Re: [Alsa-user] How to submit a patch

[James Dutton]

On Wed, 17 Apr 2024 at 04:32, G S  wrote:
>
> Hello ALSA community.
>
> I have a newer Lenovo laptop that has a built-in AMD audio co-processor which 
> requires adding an entry in the lookup table for the acp6x kernel module 
> (acp6x-mach.c) so the the digital mic is properly detected.
>
> I found one of the original discussions about the Linux kernel patch which 
> introduced this fix and in that thread an AMD engineer suggested that the 
> patch be submitted to the ALSA team for review and inclusion into the Linux 
> kernel.
>
> https://bugzilla.kernel.org/show_bug.cgi?id=216270
>
> Is this the case and should I send in the patch diff to the ALSA developers 
> mailing list for review?
>
> Thanks in advance for your assistance.

Read the documentation on how to submit patches here in the Linux kernel tree:
.../linux/Documentation/process

After ensuring the patch passes all the checks, then post a patch,
with a well formatted title and description, if it exists, a link to
any relevant bugzilla page, and the "Signed-off-by:"  and then send it
to the alsa-devel mailing list.


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Re: [Alsa-user] speaker-test: no correct sound output on LFE and others speakers

[James Dutton]

On Wed, 8 May 2024 at 19:53, Franco Martelli  wrote:
>
> Hello everyone,
>
> Basically I've the same issue described here:
> https://askubuntu.com/questions/1180389/speaker-test-returns-all-6-channels-to-front-speakers
>

What does this do for you:
speaker-test -c6 -twav

It should play some voice from each speaker in turn.


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[Alsa-user] hda intel speaker not working

[James Dutton]

Hi,

There is a laptop that fails to output sound from the laptop speakers
but works with headphones.
After considerable diagnostics I have made some progress:
The User followed these steps:
1) Plug the headphones in.
2) speaker-test -c2 -twav -Dplughw:CARD=1,DEV=0
3) you should hear sound through the headphones. If not, try
speaker-test -c2 -twav
You need to find a combination of the speaker-test command that
outputs sound through the headphones.
4) alsamixer -c1
This will bring up a mixer in the terminal
You use the cursor/arrow keys to move left and right.
You use the up/down to increase/decrease the volume
You use "M" to mute/unmute.
5) increase the volume and unmute all of the following: Master,
Headphone, Speaker
You can mute/unmute the headphones to check you are controlling it correctly.
6) The aim is to see if you can hear sound from both the headphones
and the laptop speakers at the same time.

The result is one actually gets sound out of the laptop speakers at step 6.
So, this is a bug with the control of the EAPD on this laptop.
Essentially, it looks like the laptop uses the same power amp for both
headphones and speakers. So the headphone EAPD control switches the
EAPD on for both headphones and speakers, and the speaker EAPD control
does nothing.
Is there an existing QUIRK that handles this scenario, or will I need
to work on a new quirk?
Codec: Realtek ALC269VC
Address: 0
AFG Function Id: 0x1 (unsol 1)
Vendor Id: 0x10ec0269
Subsystem Id: 0x10f70300
Revision Id: 0x100202

Kind Regards

James


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Re: [Alsa-user] configuring asoundrc

[James Cloos]

>>>>> "CL" == Clemens Ladisch  writes:

CL> [my attempt] disables all automatic sample rate/format conversions.

Ah.  I based it on what I had worked out years ago for my main workstation

CL> Replace it with:

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

unfortunately that gives:

ALSA lib 
/var/tmp/portage/media-libs/alsa-lib-1.2.4/work/alsa-lib-1.2.4/src/conf.c:5084:(parse_args)
 Unknown parameter 1
ALSA lib 
/var/tmp/portage/media-libs/alsa-lib-1.2.4/work/alsa-lib-1.2.4/src/conf.c:5217:(snd_config_expand)
 Parse arguments error: No such file or directory
ALSA lib 
/var/tmp/portage/media-libs/alsa-lib-1.2.4/work/alsa-lib-1.2.4/src/pcm/pcm.c:2660:(snd_pcm_open_noupdate)
 Unknown PCM plug:hdmi:0,0
aplay: main:830: audio open error: No such file or directory

(The /var/tmp/portage/media-libs/alsa-lib-1.2.4/work/ was the build dir)

-JimC
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[Alsa-user] configuring asoundrc

[James Cloos]

i'm tryin to get audio to work on a new board.

so far nothing i've tried works.

aplay -l reports:

 List of PLAYBACK Hardware Devices 
card 0: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: Generic [HD-Audio Generic], device 7: HDMI 1 [HDMI 1]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: Generic [HD-Audio Generic], device 8: HDMI 2 [HDMI 2]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: Generic [HD-Audio Generic], device 9: HDMI 3 [HDMI 3]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: Generic_1 [HD-Audio Generic], device 0: ALC888-VD Analog [ALC888-VD 
Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: Generic_1 [HD-Audio Generic], device 1: ALC888-VD Digital [ALC888-VD 
Digital]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

my goal is to default to the first one in that list.

[time passes; looks like i found a partial solution...]

this:

pcm.!default {
type hw
card Generic
device 3
}

ctl.!default {
type hw
card Generic
}

reports:

aplay: set_params:1349: Channels count non available

but some gui apps succeed, so that does not seem fatal.

nonetheless, any ideas on how to fix that?

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

[James Conroy-Finn]

Hello everyone,

I have a problem with a Class-2 compatible USB device (RODECaster Pro) that 
won't show up as an audio input when connected to my Linux machine.

This issue affects one other person who opened the following issue on the 
Ubuntu alsa-driver bug tracker:

https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1877726

In my comment [1] I've shared the log output I see when I connect this device 
to my Linux machine:

kernel: usb 2-9.4.2: new high-speed USB device number 20 using xhci_hcd
kernel: usb 2-9.4.2: New USB device found, idVendor=19f7, idProduct=0011, 
bcdDevice= 2.10
kernel: usb 2-9.4.2: New USB device strings: Mfr=1, Product=2, 
SerialNumber=3
kernel: usb 2-9.4.2: Product: RODECaster Pro
kernel: usb 2-9.4.2: Manufacturer: RODE Microphones
kernel: usb 2-9.4.2: SerialNumber: 001A
kernel: hid-generic 0003:19F7:0011.0009: hiddev1,hidraw7: USB HID v1.10 
Device [RODE Microphones RODECaster Pro] on usb-:00:14.0-9.4.2/input0
kernel: usb 2-9.4.2: parse_audio_format_rates_v2v3(): unable to retrieve 
number of sample rates (clock 1)
kernel: usb 2-9.4.2: parse_audio_format_rates_v2v3(): unable to retrieve 
number of sample rates (clock 1)
mtp-probe[4037]: checking bus 2, device 20: 
"/sys/devices/pci:00/:00:14.0/usb2/2-9/2-9.4/2-9.4.2"
mtp-probe[4037]: bus: 2, device: 20 was not an MTP device
systemd-udevd[4043]: controlC4: Process '/usr/bin/alsactl restore 4' failed 
with exit code 99.
mtp-probe[4050]: checking bus 2, device 20: 
"/sys/devices/pci:00/:00:14.0/usb2/2-9/2-9.4/2-9.4.2"
mtp-probe[4050]: bus: 2, device: 20 was not an MTP device
pulseaudio[1460]: E: [pulseaudio] module-alsa-card.c: Failed to find a 
working profile.
pulseaudio[1460]: E: [pulseaudio] module.c: Failed to load module 
"module-alsa-card" (argument: "device_id="4" 
name="usb-RODE_Microphones_RODECaster_Pro_001A-01" 
card_name="alsa_card.usb-RODE_Microphones_RODECaster_Pro_001A-01" 
namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no 
deferred_volume=yes use_ucm=yes avoid_resampling=no 
card_properties="module-udev-detect.discovered=1""): initialization failed.

[1]: 
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1877726/comments/6

I've connected a Focusrite Scarlett 2i2 and it shows up as an input as I 
expect. This leads me to believe either the RCP is not class-2 compliant as 
advertised or that there's a bug in ALSA somewhere.

If anyone has any tips on how I might fix this error parsing audio format rates 
I'd be eternally grateful!

-- 
James Conroy-Finn


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Re: [Alsa-user] Marian Marc 2 - drivers needed

[James Madgwick]

Hello Fryziu,

Your post intrigued me and I thought I would investigate it.
This card seems to be from a very low volume manufacturer and not 
produced for some years, potentially 20 years or more.
The chipsets you mention are not audio chipsets. The Xilinx is an FPGA 
and the PLX is a PCI bridge. The card appears to be entirely proprietary 
and unique.
Unfortunately this means drivers for this card would need to upload some 
binary blob or otherwise program the FPGA for it to work.
This makes writing a driver much more difficult if not impossible, 
combined with the obscurity of this card it explains why there is no driver.


I don't think you have any change of getting it working with ALSA on 
Linux directly.


However, there is another possible solution.
The manufacturers website no longer offers any downloads for this card, 
but the web archive has an older version of the site which does: 
https://web.archive.org/web/20070315160445/http://www.marian.de:80/en/downloads
These drivers are of course only for windows. My idea (which is only 
theoretical) is that you could pass-through the PCI card to a VM running 
Windows XP.
On the VM you then run the Cygwin version of pulseaudio: 
https://www.freedesktop.org/wiki/Software/PulseAudio/FAQ/#index33h3
You can then forward audio steams from Linux to the VM which is able to 
play it on the sound card using the Windows drivers.
I couldn't much about using pulseaudio on Windows and nothing about 
using the method I have outlined above. It might work (I haven't tried 
it) and you would need modern hardware to support the pass-through.


I hope this is of some help to you.

James

On 08/09/2019 08:15, Fryziu DeMol wrote:

Audio card Marian Marc 2 24/96
(earlier sold as SekD Prodif Plus)
https://i.imgur.com/fWVE3mk.jpg
distinguishable chipsets are:
VCS10XL by XILINX SPARTAN
PCI9052 by PLX TECHNOLOGY

Is there any chance to get it working?
Does anyone have some unfinished drivers,
some clue what to do to run this card on ALSA?

Fryziu


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

[James]

On 14/06/2019 15:32, Brian J. Murrell wrote:

On Fri, 2019-06-14 at 16:02 +0200, Clemens Ladisch via Alsa-user wrote:

Brian J. Murrell wrote:

# aplay -l
aplay: device_list:272: no soundcards found...

Are the sound drivers loaded? (see the output of "lsmod")

Sorry, I did check that before my previous message but forgot to
include the output:

# lsmod | grep snd
snd_seq86016  0
snd_seq_device 16384  1 snd_seq
snd_pcm   118784  0
snd_timer  40960  2 snd_seq,snd_pcm
snd94208  4 snd_seq,snd_seq_device,snd_timer,snd_pcm
soundcore  16384  1 snd


You appear to be missing: snd-hda-intel

Try "modprobe snd-hda-intel"   and see if that helps.

It should be automatically detecting it, but the modprobe is a good test.

Kind Regards

James






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

[James]

On 22/04/18 13:14, Marc Haber wrote:

On Sun, Apr 08, 2018 at 05:20:13PM +0100, James wrote:

On 31/03/18 20:02, Marc Haber wrote:
In order to get output to the digital output, you need to use the "iec958"
device.
E.g.:
speaker-test -c2 -d iec958:CARD=CMI8738,DEV=0 --rate 48000

Note that you also appear to have a digital out of the SB card, so you need
to select the card when outputting.

That would be CARD=Device to use the USB Device, and that actually
works. --rate 44100 does not seem to work. Is it possible that the USB
device is only able to play back at 48 kHz so that the PC needs to
resample, or can this be a driver issue that the frequency is not
correctly selected?

And even after successful playback with --rate 48000 and the DAT deck
still being synced to the input, iecset -c iec958:CARD=Device,DEV=0 says
Rate: 44100.


You might also need to mess with AES0 settings. Google for that.

Google results are inconclusive, a ton of forum entries like
"AES0=number fixed it for me", but nothing resembling an explanation,
and nothing in the Documentation subdirectory in the kernel tree.

Greetings
Marc


Ok, so you are making some progress and have 48000 working.

iecset is only useful up to a point. It tells you the settings when 
idle. They get overridden when you play something.


iec958:AES0=0x6,AES1=0x82,AES2=0x0,AES3=0x2

Those are the AES settings, in this case for AC3 encoding.

The numbers change depending on the format of the output.

The default is right for 48000 rate PCM, you will need different bits 
set for 44100.


One way of finding the right string is to type "iecset", it will 
describe the bits. then typing "iecset -x" will tell you which AES0 
values match it.


Some playing with iecset gives the following:

AC3
AES0=0x6,AES1=0x82,AES2=0x0,AES3=0x2
44100
AES0=0x04,AES1=0x00,AES2=0x00,AES3=0x00
48000
AES0=0x04,AES1=0x00,AES2=0x00,AES3=0x02


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

[James]

On 31/03/18 20:02, Marc Haber wrote:

Hi,

I have been experimenting with USB audio devices recently and have
settled on an USB device which seems to be a weird OEM device labeled by
the german company CSL. The device announces itself on the USB as
"0d8c:0102 C-Media Electronics, Inc. CM106 Like Sound Device". I am
pasting the output of lsusb -v, aplay -l and aplay -L below and am
willing to deliver any additional information that may helpful here.

It is very important for me that I can use the
optical ports that the device has for both recording audio to the
computer and playing back audio from the computer. Amazon Link:
https://www.amazon.de/CSL-Soundkarte-Lautsprecher-gleichzeitige-Audiogeräte/dp/B00KXAVBQY

The device is recording audio just fine, from the analog inputs and from
the optical input. What I have not been able to is playing back through
the optical output. I have connected a Sony DTC-60ES DAT deck to the
output, which does not seem to properly sync on the output. The "digital
input" indicator on the DAT deck does not stop blinking which is an
indicator of "no signal".

I have tried playing around with alsamixer, but didn't find any fader
for the digital output, nor did I find an output switch. I was also
never able to stop the device from playing back from the analog output.
Even when I select the IEC958 digital output in pavucontrol, playback
continues from the analog output.

card 2: CMI8738 [C-Media CMI8738], device 0: CMI8738-MC6 [C-Media PCI DAC/ADC]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
card 2: CMI8738 [C-Media CMI8738], device 1: CMI8738-MC6 [C-Media PCI 2nd DAC]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
card 2: CMI8738 [C-Media CMI8738], device 2: CMI8738-MC6 [C-Media PCI IEC958]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
card 3: Device [USB Sound Device], device 0: USB Audio [USB Audio]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
[2/5014]mh@fan:~ $ sudo aplay -L
iec958:CARD=SB,DEV=0
 HDA ATI SB, ALC892 Digital
 IEC958 (S/PDIF) Digital Audio Output
iec958:CARD=CMI8738,DEV=0
 C-Media CMI8738, C-Media PCI DAC/ADC
 IEC958 (S/PDIF) Digital Audio Output
iec958:CARD=Device,DEV=0
 USB Sound Device, USB Audio
 IEC958 (S/PDIF) Digital Audio Output

Hi,

In order to get output to the digital output, you need to use the 
"iec958" device.

E.g.:
speaker-test -c2 -d iec958:CARD=CMI8738,DEV=0 --rate 48000

Note that you also appear to have a digital out of the SB card, so you 
need to select the card when outputting.


You might also need to mess with AES0 settings. Google for that.

Kind Regards

James


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Re: [Alsa-user] rate coverter after dmix

[James Shatto]

On 5/9/17, remu kelly  wrote:
>
> How this can be achieved, seeing that we can't have a plugin after dmix.

Couldn't you use snd-aloop and have a plugin AFTER dmix?  It basically
creates a loopback interface who's output channel is the input
channel.  Although I've never used it (yet).  But one way to pulse
over jack without using the jack module part of pulse.

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Re: [Alsa-user] Short pauses in playback depending on output volume

[James Shatto]

Does the playback file exist on a slow storage device?  I have a few
usb sticks that pause with cd quality wav files because of the slow
I/O of the device.  But the same file compressed to flac or mp3 on the
same storage device will play without pauses.  Or the cd quality file
on any "faster" device, even sdhc cards plays without pauses.

It could be a power management issue and hardware related.  Swapping
around the ports that things are plugged into or using a powered hub
can help, sometimes.

Try running something like nmon when you trigger the issue.  With "L"
for cpu, it can be a little more informative.  With a blue "w" when
it's waiting on something from the system.  Which might indicate
hardware type issues like bus speed or swap usage.  Also check dmesg
for indications of hardware issues (iffy connections).

- James


On 1/15/17, Fabian Keller <cont...@guitargeeksvr.com> wrote:
> Hi all,
>
> While developing an audio software based on PortAudio, I discovered a
> surprising problem related to ALSA: I'm getting short pauses in the
> audio playback (sounding like typical buffer underruns) depending on
> the audio amplitude.
>
> As a test, I have generated two wave files containing pure white
> noise. One of them with an amplitude of 0.5 the other one using a full
> range amplitude of 1.0. I'm playing both files with aplay, but I'm
> getting the same behavior with other players and also with the
> software I'm developing: The 0.5 amplitude files plays without any
> issues. But the 1.0 amplitude file plays with short breaks in the
> audio stream. I'm getting about two of these breaks per minute, but
> there does not seem to be a deterministic pattern. I would guess the
> pauses are <100 ms in duration, which is why I was debugging in the
> direction of buffer underruns for many days until I discovered this
> amplitude effect. I have xruns logging enabled, so I'm pretty sure
> this is not related to that.
>
> Do you have any idea what could be causing this?
>
> System specs:
> - Ubuntu 14.04. with the default libportaudio2 (based on the last 2014
> release)
> - Standard Intel onboard sound: PCH [HDA Intel PCH], device 0: ALC892
> Analog [ALC892 Analog]
>
> Another test: Amplitude 0.9 also has breaks, so it is not just the
> full range amplitude.
>
> Thanks,
> Fabian
>
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Re: [Alsa-user] ice1712 recording

[James Shatto]

> You can mix all 4 inputs down into one stream and then record that, but
do you really mean that you can record to 4 separate application threads
concurrently without mixing?

No, I mean you can record all four channels as input at the same time
with the same app.  Hence the -c 4 aka 4 channels.  In audacity you
just select 4 channels and press record.  Once recorded you can break
them out into 4 mono channels and save each individually (unmixed).
There's no mixing involved until you configure it to do so.  WAV files
and other audio formats often contain multiple tracks.  Individual
unmixed tracks.  They are mixed at the time of playback if configured
to do so.

By default sound is often configured for stereo with one track panned
left and one track panned right.  But it's two tracks, not mixed
(until reproduced and analog-illy mixed in air).  Baring cheap sound
devices that have bleed over between tracks with unintentional mixing.
In audacity you can separate the channels and unpan them for two true
mono tracks.  As well as a few CLI options for the same.  Sox is good
for that.

With unix-isms you can send your output to stdout and pipe to stdin of
another app.  Like tee which can then save a file and redirect the
same output to another file.  I've sometimes gone this route to output
the raw WAV and a compressed MP3 at the same time to two different
storage devices / locations.

I seem to recall an arecord option to output each channel as it's own
mono file.  Which you could tail -f on other terminals to pipe that to
other things.  Seems like --separate-channels is that option.

$ arecord --help

- James

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Re: [Alsa-user] ice1712 recording

[James Shatto]

It can record from all 4.  Although many applications only care about
left / channel 1 (defaults).  I tend to run pulseaudio over jackd
setup.  As that was the only way to have your mic be an input other
than 1 for apps like skype.  Since channel 1 and 2 are typically
stereo output.

$ man arecord

-c, --channels=#
  the number of channels.  The default is one channel. .

I suspect that adding -c 4 would overcome the issue.  The -f dat has a
default of -f S16_LE -c 2 -r 48000.  So you might want to change that
parameter as well.

$ arecord -Dhw:1 -f S24_LE -t wav -c 4 -r 96000 output.wav

I have that card, although I haven't booted that old PCI system in a
while.  What I recall of that card, that should be the highest
sampling available for it.  24 bit, 96kHz, 4 channels (input).  In
theory it has a 10 channel output mixer.  It's a nice card, a shame
most newer things don't have PCI in favor of PCIe.

You might want to simplify your .asoundrc, it's more likely to get in
the way than help these days.

#-
defaults.ctl.card 1
defaults.pcm.card 1
defaults.pcm.device 0
#-

Assuming that it didn't get index 0 in /proc/asound/cards.  But did
get 1.  In alsa speak that's equivalent to -Dhw:1,0 .  Although you
might want to omit the ,0 since that's typcially playback, not
capture, so -Dhw:1

- James


On 12/4/16, Ralf Mardorf <ralf.mard...@alice-dsl.net> wrote:
> On Sun, 4 Dec 2016 20:18:24 +, zcx wrote:
>>I have a Delta 44 sound card here that uses the ice1712 chipset.
>>
>>Am I right in thinking that although the card has 4 mono inputs, it
>>can only capture one stream at a time?  arecord seems to think so...
>
> Only one app can grab the device, if you run two instances of the same
> app, only one instance can grab the device.
>
> If several apps should be able to use the device at the same time, you
> need a workaround, e.g. dmix or e.g. a sound server, such as e.g. jackd.
>
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Re: [Alsa-user] sans-pulseaudio Firefox? was: a strange thing

[James Shatto]

I've used alsa and firefox.  By default java (in debian) is configured
for pulseaudio.

FILE: /etc/java-7-openjdk/sound.properties

But both alsa and pulseaudio configs are in there (in debian).  Just
comment out pulse and uncomment alsa.  I switch between a lot,
depending on if I am home or using my laptop as a laptop.  Which also
affects icedtea-web, the java plugin for the browser IIRC.

FILE: .asoundrc
### for pulseaudio
#ctl.pulse { type pulse fallback sysdefault }
#pcm.pulse { type pulse fallback sysdefault }
#ctl.!default { type pulse fallback sysdefault }
#pcm.!default { type pulse fallback sysdefault }
### for alsa
defaults.ctl.card 0
defaults.pcm.card 0
defaults.pcm.device 0
###---end---

Comment swap there too.  As well as comment modify/swap
.config/pulse/client.conf since I pulseaudio over the network.  The 30
band calf eq chews up a lot of the CPU so I offloaded that to another
laptop.  A little high end boost to keep the ancient speakers sounding
normal-ish.  Depending on my lazy level I'll sometimes use two users,
one configured for alsa, one for pulseaudio.

About the only issue is that adobe's flash uses pulseaudio, so if
you're still using that you'll have "issues" with flash content.  In
days of old there's a compat thing you could install and it can be
made to work.  I'm not sure of the current methodology.  But at least
aoss can be avoided in most cases now.

Most of my flash stuff these days is the freshplayer plugin and
googles chrome pepperflash plugin (in firefox).  Freshplayer from
sources in debian stable, and pepperflash extracted and manually
maneuvered.  Recently moved out of the chrome.deb (version 54+) and
put somewhere else.  But it respects the .asound the client.conf
config settings.

https://get.adobe.com/flashplayer/otherversions/

The ppapi one is the pepperflash download.  My manual method puts them
in the  ~/. settings area so I never had to be root and only that one
user gets to use it. YMMV, depending on distro.  I tend towards debian
stable from a minimal install via debootstrap.  It's faster for me on
my slow internet, and I can get extras like network drivers while
still on the network with the host linux install.  Much like an
arch-chroot install.

- James

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

[James Shatto]

As previously said pavucontrol to configure pulseaudio.

BITD the default sound card was index 0.  Which could not be
overridden by some things.  So re-indexing was the desired way to
override things.  These days most things respect the .asoundrc.  And
you can have a pretty short one to change your default index #.  (if
not using pulseaudio)

FILE: ~/.asoundrc

defaults.ctl.card 2
defaults.pcm.card 2
defaults.pcm.device 0

You really only need the defaults.pcm.card # one though.  The others
are nice for things that change mixer levels in app or for hdmi audio
out which might be device 3, not 0.  Where card # is what is listed in
/proc/asound/cards.  YMMV.

$ cat /proc/asound/cards

Many apps let you override the default by parameter as well.  Such as
-D hw:2 for aplay.  Or --ao=alsa:device=hw,2 for mpv which is a fork
of mplayer(2?).  With the above .asoundrc it's simpler with just
--ao=alsa, or completely omit the option.  Things like audacity let
you select the available card under preferences.  If you wish to use
something other than the system defaults.

- James

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Re: [Alsa-user] arecord: set_params:1239: Channels count non available

[James Courtier-Dutton]

On 17 March 2014 14:30, Roger rogerx@gmail.com wrote:
 I keep getting the following error whenever specifying -c 1 or --channels=1, 
 or
 specifying any number of channels less than two channels when using an ASUS
 Essence STX soundcard and recording using the microphone line having a TRS
 jack.  (Whether using the rear or front/case microphone jack.)

 $ arecord --device=hw:0,0 --format S16_LE --rate 44100 -c1 /tmp/test.wav

 arecord: set_params:1239: Channels count non available

This due to one of two reasons:
1) The hardware cannot do it.
2) The hardware can do it, but the device driver has not implemented
support for it yet.

Have you tried using plug devices or the default
e.g
arecord --device=plughw:0,0 --format S16_LE --rate 44100 -c1 /tmp/test.wav

This will give you 1 channel, and alsa lib will do the down-mixing 2-1 for you.

James

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Re: [Alsa-user] Volume control range

[James Courtier-Dutton]

On 6 July 2013 07:04, Paul D. DeRocco pdero...@ix.netcom.com wrote:
 A few months ago, Clemens was kind enough to explain how to set a volume
 control, given its name, using the snd_ctl_elem_value_xxx functions. By
 experimenting on my Ubuntu system, it appeared that values from 0 to 0x
 ran the master volume through its entire range, as shown by having alsamixer
 running at the same time.

 Now, I run the same code on an embedded board, and it's always maxed out, or
 off at the bottom. I used alsactl store on both systems, and the master
 volume entries on both systems are identical, specifying a range of '0 -
 64', a dbmin of -6400 and a dbmax of 0. (I assume those are really
 millibels.)

 So how does one programmatically find the range of a control? I'd prefer to
 have plain linear voltage control, but I'll take anything I can get as long
 as I know what the shape of the curve is.


While you might think that the mixer control API would be simple,
unfortunately it is not.
A browse through the alsamixer source code will show you that.
A good start is to run
amixer contents
For each control is gives:
numid=1,iface=MIXER,name='Headphone Playback Volume'
  ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0
  : values=87,87
  | dBscale-min=-65.25dB,step=0.75dB,mute=0

The hardware is written with INTEGER values.
There are 2 values (a stereo control, 1 for left, 1 for right)
The min value that is valid is 0.
The max value that is valid is 87
The step size is not defined in this example. I.e. It is 1 so you can
wright values 0,1,2,3,...85,86,87  but not 88.
Some alsa drivers (not all yet) also provide metadata that allows one
to convert from the INTEGER value to dB value.
The dB values are mostly only used to display to the user as they are
better understood by the user.
There is an alsa API to convert from integer to db, and from db to integer.
For the conversion, the STEP value is important.
For example, you can set a dB of -65.25 but you cannot set a value of -65.00
For the API, the dB values are
(integer_value_returned_from_the_API_call / 100) in order to avoid
floating point.

Does this help?

Kind Regards

James

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Re: [Alsa-user] Microphone not working 00:1b.0 Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04)

[James Pic]

Hi all,

Sorry for bumping, but it's been two months without microphone, hardening
my work processes...

Is there anything I can do to help ?

Thanks in advance for any clue you could give me


On Wed, Nov 14, 2012 at 6:01 PM, James Pic james...@gmail.com wrote:

 Problem persists with 3.6.6 ... any help please ?


 On Mon, Oct 29, 2012 at 1:26 PM, James Pic james...@gmail.com wrote:

 Hello everybody,

 Internal microphone does not work on asus zenbook ux31a. Example
 recording attached as rec.wav.

 uname -a: Linux zen 3.6.3-1-ARCH #2 SMP PREEMPT Mon Oct 22 12:55:44
 CEST 2012 i686 GNU/Linux

 Also tried on ubuntu:
 https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1070325

 What can I do ?

 Regards




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Re: [Alsa-user] Microphone not working 00:1b.0 Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04)

[James Pic]

Problem persists with 3.6.6 ... any help please ?


On Mon, Oct 29, 2012 at 1:26 PM, James Pic james...@gmail.com wrote:

 Hello everybody,

 Internal microphone does not work on asus zenbook ux31a. Example
 recording attached as rec.wav.

 uname -a: Linux zen 3.6.3-1-ARCH #2 SMP PREEMPT Mon Oct 22 12:55:44
 CEST 2012 i686 GNU/Linux

 Also tried on ubuntu:
 https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1070325

 What can I do ?

 Regards




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[Alsa-user] Has anything changed recently with card addressing?

[James Cloos]

I had been using this ~/.asoundrc:

,
| pcm.!default {
| type hw
| card HDMI
| device 3
| }
| 
| ctl.!default {
| type hw
| card HDMI
| device 3
| }
`

but now I get errors from most applications.  Removing the device lines
got alsamixer to run, but audio didn't route.

My target card, as reported by aplay -l, is:

,
| card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0]
|   Subdevices: 1/1
|   Subdevice #0: subdevice #0
`

and the mplayer syntax  -ao alsa:device=hw=1.3  worked.  (Previously
that had required device=hw=3.1 to route the audio over the hdmi.)

The only changes here have been a steady set of software upgrades; the
hardware has been constant.

The relevant /dev files for that card are:

,
| /dev/snd/controlC1
| /dev/snd/hwC1D0
| /dev/snd/pcmC1D3p
`

I run Gentoo; alsa-utils is 1.0.25 (media-sound/alsa-utils-1.0.25-r2).

-JimC
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[Alsa-user] /proc/asound/card?/eld* missing

[James Cloos]

I've confirmed that hda_eld.c is compiled and linked into snd_hda_codec_hdmi,
which is compiled into the kernel (Linus' master branch).

But the eld file does not appear in /proc/asound.

Also, the card?/codec#? file lists only:

,
| Default PCM:
| rates [0x60]: 44100 48000
| bits [0x2]: 16
| formats [0x1]: PCM
`

whereas the EDID reports:

,
|   Audio data block
| Linear Pulse Code Modulation (LPCM) (2 channel(s))
| Frequencies: 32kHz 44kHz 48kHz 88kHz 96kHz 176kHz 192kHz 
| LPCM Bit Depths: 16, 20
|   Speaker allocation data block
| Front Left+Right
`

Audio to the display works, and I don't strictly /need/ the higher bit
depth or sample frequencies, but it seems to be a related symptom.  And
32kHz support would be welcome for some tasks.

Where should I look to debug this and get the eld file to show?

-JimC
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Re: [Alsa-user] turn off system beep

[James Shatto]

You might also blacklist snd-pcsp

Or maybe purge and re-install the alsa-base stuffs.

# dpkg --purge --force-all alsa-base
# apt-get install alsa-base

Make sure those needed blacklist items are there and add them if need be.

blacklist pcspkr
blacklist snd-pcspkr
blacklist pcsp
blacklist snd-pcsp

My overkill list added to my blacklist items.  You might also check
/etc/rc*.d/ for anything that might be playing sound(s) on shutdown.
If that other thing doesn't work.

$ find /etc/rc?.d/ -name 'K*'

- James


On 8/17/11, Julien Claassen jul...@c-lab.de wrote:
 Hello Xenia!
the only idea I'd have, is to re-enable the pcspkr module and in
 alsamixer 0
 or whatever you use to setup your audio ardware -, mute it.
Otherwise the real pc-speaker as such is no ALSA device, as far as I'm
 aware. The pc-speaker module is, I believe, intended to use it as a
 low-quality playback-device.
Kind regards
Julien
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[Alsa-user] no 5.1

[James]

My computer S/PDIF on the motherboard is connected to an amplifier by a 
digital cable.
I used to get 5.1 sound but now I only get left front and right front, 
no center, no sub, no rear.
VLC only has stereo; I can't tell with Amarok but I think it only stereo.

$ speaker-test -c 6

speaker-test 1.0.24.2

Playback device is default
Stream parameters are 48000Hz, S16_LE, 6 channels
Using 16 octaves of pink noise
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 2048 to 4096
Period size range from 1024 to 1024
Using max buffer size 4096
Periods = 4
was set period_size = 1024
was set buffer_size = 4096
  0 - Front Left
  4 - Center
  1 - Front Right
  3 - Rear Right
  2 - Rear Left
  5 - LFE

more /etc/asound.conf
pcm.!default {
   type plug
## Uncomment the following to use mixed analog by default
#  slave.pcm dmix-analog
## Uncomment the following to use unmixed digital by default
#  slave.pcm digital-hw
## Uncomment the following to use mixed digital by default
   slave.pcm dmix-digital
}
...
# Alias for digital (S/PDIF) output on the Audigy (hw:0,0)
# Do not use this directly--it requires specific rate,
# channels, and format
pcm.digital-hw {
  type hw
  card 0
  device 1
}

# Control device (mixer, etc.) for the Audigy card
ctl.digital-hw {
  type hw
  card 0
}
...
# Direct software mixing plugin for digital (S/PDIF) output
# on the Audigy (hw:0,0)
# Do not use this directly--it requires specific rate,
# channels, and format
pcm.dmix-digital {
  type dmix
  ipc_key 1235
  slave {
pcm digital-hw
period_time 0
period_size 1024
buffer_size 4096
rate 48000
  }
}

$ aplay -l
 List of PLAYBACK Hardware Devices 
card 0: NVidia [HDA NVidia], device 0: ALC888 Analog [ALC888 Analog]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
card 0: NVidia [HDA NVidia], device 1: ALC888 Digital [ALC888 Digital]
   Subdevices: 0/1
   Subdevice #0: subdevice #0


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Re: [Alsa-user] plughw versus hw

[James Shatto]

plughw is probably better at sharing a device than hw would be.  And
plughw probably allows for some conversion of content.  Otherwise they
are functionally the same IMO.  Not that I'd know since I haven't
really delved that deep into things.  You might check lsof or fuser to
see if something is using the sound device and keeping you from using
hw instead of plughw.  Or just use plughw since it works.

If it's a content conversion issue, you might try to create a
converted version of the media and see if that fixes the issue when
using hw.  Sox can do a lot of conversions.  Ffmpeg as well.  Many
means to an end.  Not that it fixes the issue, but it can help to
better understand the issue.

You might also try renaming your .asoundrc to see if that frees up hw
to be used in the way that you are trying to use it.  If that works,
then there's something in your .asoundrc that's getting in the way.

- James


On 6/21/11, Pierre Habraken pierre.habra...@free.fr wrote:
 On 06/20/2011 10:06 PM, alsa-user-requ...@lists.sourceforge.net wrote:

 Date: Mon, 20 Jun 2011 22:34:46 +0400
 From: Vladimir Mosgalinmosga...@vm10124.spb.edu
 Subject: Re: [Alsa-user] plughw versus hw
 To: alsa-user@lists.sourceforge.net
 Message-ID:20110620183446.ga14...@vm10124.spb.edu
 Content-Type: text/plain; charset=us-ascii

 Hi Pierre Habraken!

   On 2011.06.20 at 19:32:28 +0200, Pierre Habraken wrote next:

 I can imagine that this is a FAQ, but I could not find a clear answer :
 which precise difference(s) distinguish(es) plughw and hw from each other
 ?
 Does plughw apply sound processing that hw does not ?

 plughw *might* apply simple sound processing if needed, mostly channels
 conversion and rate conversion if required. It doesn't have to apply
 processing.
 hw doesn't support such processing only works when operating strictly in
 mode that audio card support.

 If you have device that supports only 2 channel, 16 bit 48000 mode then
 hw device won't be able to playback 2/16/44100 stream, or mono stream
 for example; you'll get an error when you try. But plughw will accept
 such streams and do the conversion. However, if you use plughw and
 output 2/16/48000 stream then no conversion is needed and most likely
 plughw won't be doing any processing.

 Note that using both hw and plughw can lead to specific problems, so
 it's best to use default device unless you have very specific
 requirements.

 Hello Vladimir,

 Thank you for your reply.

 I just bought an Asus Xonar DX sound card, for sending 24bits/96KHz
 stereo flac files to an external DAC.
 I am using Alsa 1.0.21 on a PC running Ubuntu 10.04 with Linux kernel
 2.6.32-32.
 Running aplay, I can't use hw for reading 24/96 files:

 $ aplay -D hw:0,1 Prelude.wav
 Playing WAVE 'Prelude.wav' : Signed 24 bit Little Endian in 3bytes, Rate
 96000 Hz, Stereo
 aplay: set_params:990: Sample format non available
 Available formats:
 - S16_LE
 - S32_LE
 $

 Adding the switch -f S32_LE does not help:

 $ aplay -D hw:0,1 -f S32_LE Prelude.wav
 Warning: format is changed to S24_3LE
 Playing WAVE 'Prelude.wav' : Signed 24 bit Little Endian in 3bytes, Rate
 96000 Hz, Stereo
 aplay: set_params:990: Sample format non available
 Available formats:
 - S16_LE
 - S32_LE
 $

 If I use plughw instead of hw, it works fine:

 $ aplay -D plughw:0,1 Prelude.wav
 Playing WAVE 'Prelude.wav' : Signed 24 bit Little Endian in 3bytes, Rate
 96000 Hz, Stereo
 ^CAborted by signal Interrupt...
 $

 Does it mean that the 24bits stream has to be converted to 16bits before
 being sent to the device and then to the DAC ?

 Pierre


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Re: [Alsa-user] First post

[James Shatto]

Well many source packages default to /usr/local/

Many distros default to /usr/

And the distros IGNORE /usr/local/ unless otherwise told.  It's not a
compile thing, it's a runtime thing.  Of course you could always run
things with the full paths /usr/local/bin/alsamixer and such.  But if
you add the location in $PATH, it'll find it.  But if you use the
default /usr/local/ it might look for and load the distros version
from /usr/ first.

So I generally overwrite the distro's versions.  i.e. make install and
--prefix=/usr.  Versus building debs and installing it that way which
is the preferred way to do thing.  But mainly because I can never
recall the fakeroot debian/binary stuff to make debs off the top of my
head.  And I don't always have networking setup to google it at the
stage that I'm installing alsa.  But I don't do enough from source
stuff to really consider my setup a different distro.  Just customized
per say.  If only for the optimization of not having to look through
1,000 drivers for the 1 that is actually used.  And media players with
CPU specific optimizations are always nice.

As a side note, alsa is in the 2.6 kernel tree.  Are we on 2.8 yet?
So if you compile a recent kernel, you automagically get a recent alsa
version with it.  Or if your distro offers a recent kernel.  It's done
for you.  No need to re-invent the wheel as previously said.  But
sometimes your distro doesn't package things in a way that you want to
use them.  i.e. Timidity with sequencer support.  Jackd with sequencer
support.  Alsa with OSS emulation.  And other fine tuning type needs.
Or your distro is on such an ancient kernel, that stuff just doesn't
work at all given the lack of age of your hardware versus the copious
amounts of age in your kernel version.

- James


On 6/20/11, David Henderson dhender...@digital-pipe.com wrote:
 Thanks again for the continued help James.  I knew '--prefix' was a
 'configure' option, but thought one would use it when permanently
 installing the software to a non-standard directory on the system.
 Since this software is being compiled on a temp system and installing
 to a staging directory, wouldn't the 'DESTDIR' be a better option to use
 while compiling the software so it can be packaged and installed on the
 custom distro?

 Thanks for the tips on the kernel headers and configure parameters. :)

 Dave


 On 06/19/2011 07:06 PM, James Shatto wrote:
 --prefix is a ./configure option.

 If you're going to apply the new alsa to an existing distro kernel and
 not a custom from source one.  You'll likely need to install the
 kernel-headers package for that kernel and distro.  And may need to
 manually move the old version of alsa (or remove).  Plus that whole
 depmod thing.

 $ dpkg -l '*kernel*headers*'

 Which resolves to linux-kernel-headers in debian.  Which is a psuedo
 package for:
 linux-libc-dev
 2.6.26-26lenny3
 and of course 2.6.26-26lenny3 resolves to linux-tree-2.6.26lenny3

 so:
 # apt-get install linux-libc-dev linux-tree-2.6.26-26lenny3
 (in debian 5.0 / lenny)

 If it's a custom one, just don't make clean after making the kernel.
 It should reside in /lib/modules/`uname -r`/build/ or something like
 that.  BITD, this would just be a symlink to/from /usr/src/linux and
 was what early alsa assumed by default.

 Depending on what multimedia features you need.  You might want
 --with-sequencer=yes and --with-oss=yes and a --driver=your card
 options on your alsa-driver compile.  Without those =no might be
 assumed.  And you might compile ALL drivers which could take a really
 long time.  Less so these days, but BITD, the better part of a day it
 seemed.

 It really depends on what you want interacting with your sound card.
 Timidity and other synth like software requires the
 --with-sequencer=yes if your card doesn't have native midi abilities
 (most don't these days).  And various pulse-audio and browsers and
 other things that just need --with-oss=yes or things might not work as
 expected, if at all.  Little things that you'll find out one way or
 another as you learn your way around.

 HTH,
 - James


 On 6/19/11, David Hendersondhender...@digital-pipe.com  wrote:
 Hi James, thanks for your help too. :)  I'll provide replies in the same
 fashion given.

 A) I don't want to overwrite the Kubuntu installation files as I'm
 compiling this version of alsa for my own distro.  I would prefer to use
 Kubuntu's pre-packaged software within itself.  So since the compiled
 version of alsa will be going into /opt/staging/alsa, should I include
 --prefix=/opt/staging/alsa as the parameter to configure?

 B) I'll assume at this point, that no matter what version of the Linux
 kernel is being used, it's still required to install the alsa-driver
 package.  That being said, I'm going to run into the same problem as A
 above since the version of Kubuntu I'm using to build the custom distro
 isn't using the same kernel version.  So what configure option do I
 have to pass in order

Re: [Alsa-user] First post

[James Shatto]

If you're really into going it on your own.  There's gentoo, and
there's LFS aka linux from scratch.  Both of which impose a lot of
source compilation.  The inherent problem with sources is that you run
into maintenance issues.  i.e. If you use the same install for a long
enough time, it'll eventually become unusable due to remnants of old
versions and not enough hours in a lifetime to figure out what/where
those are and manually correct.  Ultimately you'll be doing fresh
installs long before your hardware's expiration date.

Not that I don't do regular installs myself.  But I swap out hard
drives every two years to be pro-active against that type of failure.
And I do a lot of media editing, so I probably abuse my drives more
than most.

A distro is just a good ideal.  There's configuration files that you
really can't generate by hand without a pretty hefty understanding of
what you are doing.  Distros have done all this legwork for you and
provide you with a sane default configuration file where you just need
to uncomment a line to enable something or comment it to disable it.
Lots of sanity saving things in a distro that you'll be scouring
sources to figure out on your own in LFS land.  And probably
installing a distro anyway to cp their config.

There's a lot to learn.  But really you don't need to learn that
stuff.  There's no bread and butter / money in it.  Sure you'll have a
greater understanding.  And should some do or die worst case scenario
happen you'll know how to resolve it, where most other folks wont know
where to begin.  But really most IT jobs these days are installing and
uninstalling and configuration gigs.  We don't need to write a word
processor, as one (several actually) already exist.  And some of them
aren't too shabby.

As far as build systems.  The configure + make + make install is the
OLD way.  Not all sources use that one.  There's scons, mercurial, and
various *make incarnations.  And of course distro specific ways that
are compatible with their package manager(s).  Plus the typical
development role of 1001 ways to do one thing.  Fortunately alsa is
still a bit old school.  Or unfortunately depending on your POV.

- James

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Re: [Alsa-user] First post

[James Shatto]

Ummm.  I'm not sure if I follow you.

$ make
will build the objects and stuff in the current path of your source tree.

$ make install copies the executables to the system usable locations.
/usr/bin/ /lib/modules/.  /usr/share/doc/.
(which is why you need to be root in a lot of cases to run make
install, but not to run make)

A package maintainer will likely use stuff like what's in debian/rules
debian/binary and such to build a package manager package, instead of
using make install to place the important components (results) where
they need to be.  A package manager package lets you keep track of
what got installed and where and the package provides additional
features useful for long term maintenance and/or large scale
deployment.

If you want to build a package (i.e. .deb) you'd use those tools for
that to do that.  Otherwise you have to at least mimic make install.
Which is a bit futile IMO, given that you could just run make install.
 i.e. how exactly would you create your tarball?  From a diff of an
entire backup before and after make install?  By doing everything done
by make install manually?  That's fine for relatively small things
like alsa.  But for X, KDE, ???, and other more bulky entities.  You'd
need a couple lifetimes of spare time to re-invent make install.  And
scons install and  and ... and ...

Also bear in mind that if you're building something on a system other
than the one where it will be deployed.  You will run into some
version compatibility issues.  Just a minor difference in the API
between version 1.0.24 and 1.0.25 could make things unusable.  And as
previously mentioned, alsa comes with the 2.6 kernel, so you'll have
an existing version already in place that you will need to deal with,
one way or another.  When there's multiple versions of things, at
runtime things like to load in alphabetical order or ascii order at
least.  Which generally means the that OLDer version takes priority.
So even if you install your newer version, it's probably going to be
ignored unless you remove or replace the older version.  The manual
approach to dependency hell I guess, of sorts.

Lots of little things that will keep you from succeeding.  It's
probably time better spent learning an existing package management
system IMO.  Than to create your own.  Especially if you're on your
own and not part of team.  But it's almost all open source so if you
can read the source, everything that you need to know is there in one
form or another.

- James



On 6/20/11, David Henderson dhender...@digital-pipe.com wrote:
 On 06/20/2011 11:52 AM, Pierre Lorenzon wrote:
 Hi,



 From: David Hendersondhender...@digital-pipe.com
 Subject: Re: [Alsa-user] First post
 Date: Sun, 19 Jun 2011 15:28:48 -0400

 Thanks for the reply Pierre.  I checked into the blfs book, but
 it merely says these five chapters will cover alsa and then
 gives you a basic type configure  make.  This is obviously
 not going to answer the questions below. :) Any other thoughts?

 Dave


 On 06/19/2011 11:22 PM, Pierre Lorenzon wrote:
 Hi,

 It looks like to me such questions are well answered in the
 blfs book. I personnaly think that the latter is a very good
 tool to build his own custom distro.

 Bests

 Pierre


 From: David Hendersondhender...@digital-pipe.com
 Subject: [Alsa-user] First post
 Date: Sun, 19 Jun 2011 14:41:08 -0400

 Hi everyone!  I'm currently expanding my knowledge of GNU/Linux
 to
 include building packages from scratch towards an overall goal
 of a
 custom distro.  So far, I have a nice base for a command line
 OS, but
 want to expand into the multimedia aspect.  Alsa was my first
 (only?)
 choice for the audio portion, but I'm running into problems.
 The alsa
 site is somewhat overwhelming to newbies and is easy to get
 lost.  I
 have a few questions below from which I hope I can find help.
 All
 contributions are greatly appreciated. :)

 Thanks,
 Dave


 1) Currently I have downloaded alsa-driver, alsa-lib, and
 alsa-utils
 packages.  Is there an order in which these packages need to be
 compiled
 and installed?
  This question is answered by the blfs book. First alsa-lib
  and after alsa-utils.



 2) I'm currently running the relatively new Linux kernel 2.6.33
 so do I
 need the alsa-driver package?
  No ! I am running a 2.6.32 kernel and never installed
  alsa-driver. Anyway if the sound system is something very
  exotic it might be necessary ...




 Great one less thing to compile! :)

 3) I've been able to successfully compile the alsa-lib package
 and
 install it in the custom distro.  When I try to compile the
 alsa-utils
 package, I constantly get the error:

 checking for libasound headers version= 1.0.16... not present.
 configure: error: Sufficiently new version of libasound not
 found.

 I'm actually using an existing Kubuntu installation to build
 the
 packages for my custom distro.  As a result, after I compiled
 the newer
 alsa-lib, I didn't install the package

Re: [Alsa-user] First post

[James Shatto]

This is part of the reason that I use --prefix=/usr because the
/usr/includes/ are also affected by the --prefix option (i.e.
/usr/local/includes / which is empty).  And I've never really gotten
into the changing $PATH part of things.  But there's a whole slew of
-I and -L options (with a different case / case sensitive) for gcc to
bypass / customize a lot of that.  A real PITB IMO.  But just my
opinion.  i.e. Use what is already there, not re-invent it in your
image.  And yes a bit OT at this point.

- James


On 6/20/11, David Henderson dhender...@digital-pipe.com wrote:
 I think your statement here 

 i.e. how exactly would you create your tarball?  From a diff of an
 entire backup before and after make install?

  best sums it up.  Without a staging directory to install to, you
 would have to parse the entire FS in order to find what the make
 install step did.  By using a staging directory, you still run make
 install, it just installs everything in it's retained hierarchy within
 that staging directory.  That's why I said /opt/staging/alsa/bin in
 Kubuntu (build OS) becomes /bin in the custom distro.  That's what the
 DESTDIR parameter does, it allows you to retain whatever directory
 hierarchy to use, but during the make install phase, instead of using
 / as the root, it uses whatever you include (e.g.
 DESTDIR=/opt/staging/alsa) as the value pre-pended for root.

 Honestly, at this point, we've gotten way off topic. lol  These are all
 issues for me to work out, but appreciate you guys efforts. :)
 Presently, I'm thinking that alsa-utils (as we've determined alsa-driver
 probably doesn't have to be installed) is failing to compile because
 it's looking under /... for the header files and not
 /opt/staging/alsa/...  Is there a way to make the configure script look
 into that directory for the header files during the configure phase?

 Thanks again for everyone's continued efforts in getting this matter
 resolved.

 Dave



 On 06/20/2011 04:06 PM, James Shatto wrote:
 Ummm.  I'm not sure if I follow you.

 $ make
 will build the objects and stuff in the current path of your source tree.

 $ make install copies the executables to the system usable locations.
 /usr/bin/ /lib/modules/.  /usr/share/doc/.
 (which is why you need to be root in a lot of cases to run make
 install, but not to run make)

 A package maintainer will likely use stuff like what's in debian/rules
 debian/binary and such to build a package manager package, instead of
 using make install to place the important components (results) where
 they need to be.  A package manager package lets you keep track of
 what got installed and where and the package provides additional
 features useful for long term maintenance and/or large scale
 deployment.

 If you want to build a package (i.e. .deb) you'd use those tools for
 that to do that.  Otherwise you have to at least mimic make install.
 Which is a bit futile IMO, given that you could just run make install.
   i.e. how exactly would you create your tarball?  From a diff of an
 entire backup before and after make install?  By doing everything done
 by make install manually?  That's fine for relatively small things
 like alsa.  But for X, KDE, ???, and other more bulky entities.  You'd
 need a couple lifetimes of spare time to re-invent make install.  And
 scons install and  and ... and ...

 Also bear in mind that if you're building something on a system other
 than the one where it will be deployed.  You will run into some
 version compatibility issues.  Just a minor difference in the API
 between version 1.0.24 and 1.0.25 could make things unusable.  And as
 previously mentioned, alsa comes with the 2.6 kernel, so you'll have
 an existing version already in place that you will need to deal with,
 one way or another.  When there's multiple versions of things, at
 runtime things like to load in alphabetical order or ascii order at
 least.  Which generally means the that OLDer version takes priority.
 So even if you install your newer version, it's probably going to be
 ignored unless you remove or replace the older version.  The manual
 approach to dependency hell I guess, of sorts.

 Lots of little things that will keep you from succeeding.  It's
 probably time better spent learning an existing package management
 system IMO.  Than to create your own.  Especially if you're on your
 own and not part of team.  But it's almost all open source so if you
 can read the source, everything that you need to know is there in one
 form or another.

 - James



 On 6/20/11, David Hendersondhender...@digital-pipe.com  wrote:
 On 06/20/2011 11:52 AM, Pierre Lorenzon wrote:
 Hi,



 From: David Hendersondhender...@digital-pipe.com
 Subject: Re: [Alsa-user] First post
 Date: Sun, 19 Jun 2011 15:28:48 -0400

 Thanks for the reply Pierre.  I checked into the blfs book, but
 it merely says these five chapters will cover alsa and then
 gives you a basic type configure   make.  This is obviously

Re: [Alsa-user] First post

[James Shatto]

A) If you want to overwrite your existing distro's versions, you
probably want the --prefix=/usr option on your ./configure commands.
If not, be sure to change your $PATH to look at /usr/local FIRST.

B) Compile alsa-lib first, alsa-driver second.  Most compile options
only need --prefix=/usr if you want to override the default of
/usr/local.  But alsa-driver requires extra parms depending on what
you want.  Some packages are only tool sets, so make -f Makefile?  And
use them from where you made them, or copy/move them to more common
$PATH's.

C) You might have versioning conflicts depending on what you're trying
to mix and match.  libc and other things might not work well together
unless you're running the latest and greatest of every component.  And
even that is problematic some of the time.

D) unless you have a lot of time to waste, or just need the learning,
I'd recommend going with existing distros.  There's enough of them
that one might suit your current needs.  www.distrowatch.com

HTH,
- James



On 6/19/11, David Henderson dhender...@digital-pipe.com wrote:
 Thanks for the reply Pierre.  I checked into the blfs book, but it
 merely says these five chapters will cover alsa and then gives you a
 basic type configure  make.  This is obviously not going to answer
 the questions below. :)  Any other thoughts?

 Dave


 On 06/19/2011 11:22 PM, Pierre Lorenzon wrote:
 Hi,

 It looks like to me such questions are well answered in the
 blfs book. I personnaly think that the latter is a very good
 tool to build his own custom distro.

 Bests

 Pierre


 From: David Hendersondhender...@digital-pipe.com
 Subject: [Alsa-user] First post
 Date: Sun, 19 Jun 2011 14:41:08 -0400

 Hi everyone!  I'm currently expanding my knowledge of GNU/Linux to
 include building packages from scratch towards an overall goal of a
 custom distro.  So far, I have a nice base for a command line OS, but
 want to expand into the multimedia aspect.  Alsa was my first (only?)
 choice for the audio portion, but I'm running into problems.  The alsa
 site is somewhat overwhelming to newbies and is easy to get lost.  I
 have a few questions below from which I hope I can find help.  All
 contributions are greatly appreciated. :)

 Thanks,
 Dave


 1) Currently I have downloaded alsa-driver, alsa-lib, and alsa-utils
 packages.  Is there an order in which these packages need to be compiled
 and installed?

 2) I'm currently running the relatively new Linux kernel 2.6.33 so do I
 need the alsa-driver package?

 3) I've been able to successfully compile the alsa-lib package and
 install it in the custom distro.  When I try to compile the alsa-utils
 package, I constantly get the error:

 checking for libasound headers version= 1.0.16... not present.
 configure: error: Sufficiently new version of libasound not found.

 I'm actually using an existing Kubuntu installation to build the
 packages for my custom distro.  As a result, after I compiled the newer
 alsa-lib, I didn't install the package into the Kubuntu OS, but rather a
 staging directory (/opt/staging/alsa).  I'm sure the reason this is
 failing is because it's probably looking for /usr/lib/... or some other
 default location.  How do I tell the configure script for the alsa-utils
 to look in the staging directory for the header files it needs?


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Re: [Alsa-user] First post

[James Shatto]

--prefix is a ./configure option.

If you're going to apply the new alsa to an existing distro kernel and
not a custom from source one.  You'll likely need to install the
kernel-headers package for that kernel and distro.  And may need to
manually move the old version of alsa (or remove).  Plus that whole
depmod thing.

$ dpkg -l '*kernel*headers*'

Which resolves to linux-kernel-headers in debian.  Which is a psuedo
package for:
linux-libc-dev
2.6.26-26lenny3
and of course 2.6.26-26lenny3 resolves to linux-tree-2.6.26lenny3

so:
# apt-get install linux-libc-dev linux-tree-2.6.26-26lenny3
(in debian 5.0 / lenny)

If it's a custom one, just don't make clean after making the kernel.
It should reside in /lib/modules/`uname -r`/build/ or something like
that.  BITD, this would just be a symlink to/from /usr/src/linux and
was what early alsa assumed by default.

Depending on what multimedia features you need.  You might want
--with-sequencer=yes and --with-oss=yes and a --driver=your card
options on your alsa-driver compile.  Without those =no might be
assumed.  And you might compile ALL drivers which could take a really
long time.  Less so these days, but BITD, the better part of a day it
seemed.

It really depends on what you want interacting with your sound card.
Timidity and other synth like software requires the
--with-sequencer=yes if your card doesn't have native midi abilities
(most don't these days).  And various pulse-audio and browsers and
other things that just need --with-oss=yes or things might not work as
expected, if at all.  Little things that you'll find out one way or
another as you learn your way around.

HTH,
- James


On 6/19/11, David Henderson dhender...@digital-pipe.com wrote:
 Hi James, thanks for your help too. :)  I'll provide replies in the same
 fashion given.

 A) I don't want to overwrite the Kubuntu installation files as I'm
 compiling this version of alsa for my own distro.  I would prefer to use
 Kubuntu's pre-packaged software within itself.  So since the compiled
 version of alsa will be going into /opt/staging/alsa, should I include
 --prefix=/opt/staging/alsa as the parameter to configure?

 B) I'll assume at this point, that no matter what version of the Linux
 kernel is being used, it's still required to install the alsa-driver
 package.  That being said, I'm going to run into the same problem as A
 above since the version of Kubuntu I'm using to build the custom distro
 isn't using the same kernel version.  So what configure option do I
 have to pass in order for alsa to see the source code of the custom
 distro's kernel version?

 C) So far, so good, but I'll keep that in mind. :)

 D) Thanks for the URL, but this is a project that I've wanted to do for
 the last 5-7 years and now I have the ability to do so.  Not only that,
 but knowing details at this level of building an OS can also help with
 my job - so I get a two fold benefit. :)  Otherwise, I'd definitely
 follow your advice! lol

 Thanks again for your help, I look forward to hearing back from you.

 Dave


 On 06/19/2011 04:36 PM, James Shatto wrote:
 A) If you want to overwrite your existing distro's versions, you
 probably want the --prefix=/usr option on your ./configure commands.
 If not, be sure to change your $PATH to look at /usr/local FIRST.

 B) Compile alsa-lib first, alsa-driver second.  Most compile options
 only need --prefix=/usr if you want to override the default of
 /usr/local.  But alsa-driver requires extra parms depending on what
 you want.  Some packages are only tool sets, so make -f Makefile?  And
 use them from where you made them, or copy/move them to more common
 $PATH's.

 C) You might have versioning conflicts depending on what you're trying
 to mix and match.  libc and other things might not work well together
 unless you're running the latest and greatest of every component.  And
 even that is problematic some of the time.

 D) unless you have a lot of time to waste, or just need the learning,
 I'd recommend going with existing distros.  There's enough of them
 that one might suit your current needs.  www.distrowatch.com

 HTH,
 - James



 On 6/19/11, David Hendersondhender...@digital-pipe.com  wrote:
 Thanks for the reply Pierre.  I checked into the blfs book, but it
 merely says these five chapters will cover alsa and then gives you a
 basic type configure  make.  This is obviously not going to answer
 the questions below. :)  Any other thoughts?

 Dave


 On 06/19/2011 11:22 PM, Pierre Lorenzon wrote:
 Hi,

 It looks like to me such questions are well answered in the
 blfs book. I personnaly think that the latter is a very good
 tool to build his own custom distro.

 Bests

 Pierre


 From: David Hendersondhender...@digital-pipe.com
 Subject: [Alsa-user] First post
 Date: Sun, 19 Jun 2011 14:41:08 -0400

 Hi everyone!  I'm currently expanding my knowledge of GNU/Linux to
 include building packages from scratch towards an overall goal of a
 custom distro.  So far, I have

Re: [Alsa-user] Help with ALSA on new computer

[James Shatto]

It depends on the HDMI device.  For my video card, the specification
of the sound that travels over that wire is pretty strict.  ONLY AC3,
only 44.1kHz, only stereo / 2 channels, only...  And it does work if
all criteria is met.  But I much prefer to use the analog audio
(lossless / PCM).  But a single wire does have it's uses.

I imagine that it's in your configuration somewhere to make what
you're wanting happen.  .asoundrc?  pauvcontrol?  Just be aware of the
specifications of your gear.  You have two of them technically,
sending and receiving, and they may not match, but on the lowest of
lowest common denominators.  At least change the indexing so that your
HDMI card is card 0, that way even the stupid apps try to use it,
versus some other card.

- James


On 6/16/11, Jerry Geis ge...@pagestation.com wrote:
 I got a new computer (Zotac HD-ID40) ION2

 I have installed alsa 1.0.24 on Centos 5.6 x86_64.

 Not sure why it registers 2 audio devices either.

 here is info:

 lspci | grep Audio

 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition
 Audio Controller (rev 02)
 03:00.1 Audio device: nVidia Corporation High Definition Audio Controller
 (rev a1)

 lspci -n
 00:1b.0 0403: 8086:27d8 (rev 02)
 03:00.0 0300: 10de:0a64 (rev a2)
 03:00.1 0403: 10de:0be3 (rev a1)

 lspci -v
 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition
 Audio Controller (rev 02)
   Subsystem: ZOTAC International (MCO) Ltd. Device a140
   Flags: bus master, fast devsel, latency 0, IRQ 10
   Memory at fe9fc000 (64-bit, non-prefetchable) [size=16K]
   Capabilities: [50] Power Management version 2
   Capabilities: [60] MSI: Enable- Count=1/1 Maskable- 64bit+
   Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00
   Kernel driver in use: HDA Intel
   Kernel modules: snd-hda-intel

 03:00.0 VGA compatible controller: nVidia Corporation GT218 [ION] (rev a2)
 (prog-if 00 [VGA controller])
   Subsystem: ZOTAC International (MCO) Ltd. Device 3100
   Flags: bus master, fast devsel, latency 0, IRQ 5
   Memory at fd00 (32-bit, non-prefetchable) [size=16M]
   Memory at d000 (64-bit, prefetchable) [size=256M]
   Memory at ce00 (64-bit, prefetchable) [size=32M]
   I/O ports at ec00 [size=128]
   Expansion ROM at fcf8 [disabled] [size=512K]
   Capabilities: [60] Power Management version 3
   Capabilities: [68] MSI: Enable- Count=1/1 Maskable- 64bit+
   Capabilities: [78] Express Endpoint, MSI 00
   Capabilities: [b4] Vendor Specific Information: Len=14 ?
   Kernel driver in use: nvidia
   Kernel modules: nvidiafb, nvidia

 03:00.1 Audio device: nVidia Corporation High Definition Audio Controller
 (rev a1)
   Subsystem: ZOTAC International (MCO) Ltd. Device 3100
   Flags: bus master, fast devsel, latency 0, IRQ 5
   Memory at fcf7c000 (32-bit, non-prefetchable) [size=16K]
   Capabilities: [60] Power Management version 3
   Capabilities: [68] MSI: Enable- Count=1/1 Maskable- 64bit+
   Capabilities: [78] Express Endpoint, MSI 00
   Kernel driver in use: HDA Intel
   Kernel modules: snd-hda-intel

 How can I get sound over HDMI working on this computer?
 Analog sound out works fine.
 I have edited the /etc/asound.conf to point to the device 1,3 which is the
 HDMI audio.

 Thanks,

 Jerry


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Re: [Alsa-user] [SPAM] No Sound in Debian 6

[James Shatto]

Finding out what's different with more detain than Yes and No would help.

boot into knoppix

$ lsmod | grep -i snd | sort 21 | tee alsa_knoppix.log

boot into debian

$ lsmod | grep -i snd | sort 21 | tee alsa_debian.log

save these files on a common medium (flash drive) of course.

$ diff -a -U 3 alsa_knoppix.log alsa_debian.log

And of course noting /proc/asound/cards plus you might want to modinfo
the module of importance for each, to see what differs there
(versioning / parms).  And various other things generally covered in
the alsa-info.sh script.

HTH,
- James



On 5/23/11, s.keup...@arcor.de s.keup...@arcor.de wrote:
 Hello!

 I'm afraid the problem is totally not linked to the Debian Squeeze distro,
 reaosn: there are similar problems for some chipsets, especially the
 famous
 Intel HDA Audio chipset in Ubuntu:
 ...
 Note: I'm not telling that the problem happens for all chipsets, but tons
 of
 persons should probably encouter such a problem when using laptops and
 netbooks fitted with Intel hda chipsets.

 It may not be a problem of distributions, however, everything worked with
 the
 alsa packages and configuration that come with Ubuntu, but not with those
 which come with Debian.

 At this moment I overbridge the problem on a absurd virtual way: external
 speakers, there is a vol-button you can rise up.
 External speakers are not working for me.

 @all:
 I think I've found the root of my problem: I booted the computer with a
 Knoppix
 Flash drive, with which all sound output and input was fine. While checking
 the
 /proc/asound/cards of the Knoppix system, I noticed an Sigmatel Audio Card,
 additional to the HDA Intel entry. For the same Sigmatel Chip there werde
 Channels
 displayed in Alsa mixer, which seemed to actually control my volume.
 This Sigmatel entry is missing on my Debian System.

 Does anyone know how to add this card to my alsa?

 Best Regards
 Savio

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Re: [Alsa-user] [SPAM] No Sound in Debian 6

[James Shatto]

A little overkill from my description.  And so forget that versioning
would alter the module sizes.  But alsa-info has the needed info.

Knoppix - Kernel 2.6.37 - alsa 1.0.23
Debian - Kernel 2.6.32 - alsa 1.0.24

Is that the way your debian came, or did you try to fix things
manually?  Just an odd versioning combo AFAIK.  The the older debian
kernel would have the newer alsa version.

Other notes.  Aside from that WTF modinfo null. stuff for debian.

knoppix - Codec: SigmaTel STAC9205
debian - Codec: Conexant ID 2c06

00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio
Controller (rev 02)
00:1b.0 0403: 8086:284b (rev 02)
Subsystem: 1028:01f1

So the knoppix one is the one that works?

If so:  modprobe snd-hda-intel id=SigmaTel
???

Or add snd-hda-intel.id=SigmaTel to the kernels boot line in grub (or
lilo).  Or something like that, it's been a while.  Syntax might vary.
 And it might need to be in the kernel's .config if it's not already.
Unless you compile alsa / install the old 2.4.x way.  Beyond that your
guess is as good as mine.

Probably some alias / options line you can add to
/etc/modprobe.d/*alsa* as well, or instead of those two other ways,
that functionally do the same thing (with quirks).  Hopefully
something in there rings a bell for you.  Or else the old
alsa-project.org and doc stuff might hint towards a solution.  On the
surface it looks like debian is defaulting to a conexant codec and
failing and knoppix is defaulting to a sigmatel codec and NOT failing.
 Which is the same module / driver for all intents, so something
configuration is awry.  Or I could be wrong.

- James



On 5/23/11, s.keup...@arcor.de s.keup...@arcor.de wrote:
 Hello,

 Thank you.

 These are the differences of the two logs: http://pastebin.com/nAtnhMvc

 As I am not a really experienced user, I do not know how to add and
 configure the modules properly. It would be nice if you would explain this
 in more detail.

 This is the modinfo on the three sound modules installed on Knoppix:
 http://pastebin.com/ADknBwRt
 And this is the output of the alsa-info skript you mentioned - On Knoppix:
 http://pastebin.com/fHCguzr6 , On Debian: http://pastebin.com/0rAPiyEN .

 Best Regards
 Savio

 PS: I hope the Pastebin Links are ok. I just wanted the mail not get too
 messy.
 PPS: As seen in the files, I was wrong saying the there would be a Sigmatel
 card on Knoppix in /proc/asound/cards. However a Sigmatel codec is used on
 Knoppix.


Finding out what's different with more detain than Yes and No would help.

boot into knoppix

$ lsmod | grep -i snd | sort 21 | tee alsa_knoppix.log

boot into debian

$ lsmod | grep -i snd | sort 21 | tee alsa_debian.log

save these files on a common medium (flash drive) of course.

$ diff -a -U 3 alsa_knoppix.log alsa_debian.log

And of course noting /proc/asound/cards plus you might want to modinfo
the module of importance for each, to see what differs there
(versioning / parms).  And various other things generally covered in
the alsa-info.sh script.

HTH,
- James

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Re: [Alsa-user] Spikes when recording

[James Shatto]

That seems pretty regular at 8 to 10 minute intervals.  Do you live
near a subway line?  Or other electric mass transit option?  Is the
computer on a UPS or power conditioner type supply line?  I get a
spike like that when I use a battery box to power an electret mic.  If
I turn it on after pressing record.  Beyond that, your guess is as
good as mine.

- James


On 5/20/11, Peter Hoffmann p...@peter-hoffmann.com wrote:
 Hello,

 I'm recording audio 24/7 with a delta 1010 sound card and have a
 strange problem:

 Every night at 2:30 I get spikes and some inaccurancy within some
 seconds in a one hour length recording. I've upload a screenshot to
 illustrate the problem:

 http://img88.imageshack.us/img88/4562/spikesh.png

 My .asoundrc

 pcm.capt {
 type dsnoop
 ipc_key 223456
 slave {
 pcm hw:0,0
 rate 8000
 period_time 0
 period_size 320
 channels 12
 format S32_LE
 }
 }

 pcm.c1 {
  type plug
  ttable.0.0 1
  slave.pcm capt
 }

 pcm.c2 {
  type plug
  ttable.0.1 1
  slave.pcm capt
 }

 pcm.c3 {
  type plug
  ttable.0.2 1
  slave.pcm capt
 }

 pcm.c4 {
  type plug
  ttable.0.3 1
  slave.pcm capt
 }

 pcm.c5 {
  type plug
  ttable.0.4 1
  slave.pcm capt
 }

 pcm.c6 {
  type plug
  ttable.0.5 1
  slave.pcm capt
 }

 pcm.c7 {
  type plug
  ttable.0.6 1
  slave.pcm capt
 }

 pcm.c8 {
  type plug
  ttable.0.7 1
  slave.pcm capt
 }

 I'm recording with arecord -q -f cd -t wav -d 3600 -c 1 -D c1 out.wav.

  Any hints where the problem might be?

 Kind Regards,
 Peter Hoffmann

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Re: [Alsa-user] very low-level volume in both Debian and Ubuntu

[James Shatto]

I agree that sound should just work.  And I'm still kind of surprised
that a) we have to configure it with a text file.  And b) twenty years
later, that's still the case for the most part.  Not that I think that
we should give up the command line even in part.

Alsamixer isn't intuitive, but it is semi-user friendly IMO.  It's far
from perfect.  But you launch it with alsamixer and exit with the
escape key (aka boss key).  Cursor up is up in volume, cursor down is
down in volume.  The M key for mute and unmute.  The tab key to
switch between playback and capture is not that intuitive.  It should
probably default to ALL IMO.  But it does have the typical F1 help
screen.  Although I'm not sure of the accessibility options at this
time.

There does appear to be a bug report filed on it.

https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/430937

Although that dates back to 2009 and karmic.  And the fix seems to be
to install gnome-alsamixer and turn up master F.  I still think that
alsamixer is the route to fix it and it's just a level setting.

http://git.alsa-project.org/?p=alsa-driver.git;a=blob_plain;f=utils/alsa-info.sh

$ sh alsa-info.sh

And the output (in /tmp) for that one should give you information on
your card.  Take note of the mixer part for Master.  I've used aumix
in the past.  But if you have more than one card or other things, it
can be wrong / useless more often than not.  Although it's still the
only way to tell some soundcards to record from PCM out IME.  I've
never found a way to affect that setting in any other way, in the
manner needed.  Even though I can see the effect of that change in the
output of amixer.

HTH,
- James



On 5/20/11, Y P yellowpeng...@edpnet.be wrote:
 Hello James:

 On Thu, May 19, 2011 at 09:59:01AM -0500, James Shatto wrote:
 The first step would be to see if it's even an ALSA issue.

 I had to say first of all I'm not using Flash nor graphical mixers since I'm
 a VIP - vision impaired person; sometimes I use the Orca screen reader but
 my current/daily usage of Gnu/Linux OSes is command-line. Sorry if I forgot
 to precise this before.

With flash
 video (youtube) there's a speaker icon and a slider which affects the
 volume.  I recently noticed hulu had my levels way low with such an
 icon.  With mplayer there's a softvol option which might differ from
 the levels set in alsa.

 Yes but you can't go louder than the maximum of volume-level, so the problem
 remains.

And of course alsamixer to actually set your
 levels.

 I'm not using any alsamixer, it is not user-friendly so I prefer adding
 oss-compat, libsox-fmt-all and aumix, and adjust then the volumes -v -w -W
 -s at the commandline.

 I'm afraid it is really an ALSA/Pulse problem: I just googled with the
 keywords ALSA+Pulse+netbook+very+low+volume+output+problem and Google gave
 me about 7 screens of results.

 The reasons why I believe the problem is really an ALSA+Pulse issue are:
 - one of the Google results talks about an upgrade from an Ubuntu 9.10; I've
 got a volume-problem since upgrading my Ubuntu Intrepid just a few weeks
 ago, but before the release of the newest / latest Natty (11.04); so if some
 other people encounter the problem with the same Intel hda chip in Ubuntu
 9.10 (Maverick) I had probably upgraded to the problem while I hadn't it
 before ?
 - from the Google results I see that the problem is not bound to one
 specific distribution, that explains why the problem also occurs in the
 Debian Squeeze; there is also a low level on headset, so the problem is
 really bound to sound output, not to the distribution.

 technically nothing is broke and nothing needs
 fixing.  If the levels are set and maxed out and the problem persists,
 then it could be an alsa issue.

 I will probably try to install a fresh Squeeze on my Hercules eCAFE; at this
 moment the output-volume of my 11.04 is normal, loud is loud, maxed is quite
 too loud!
 The major difference with the freshly installed Debian on my EEE where I've
 got Ubuntu Intrepid before is, that maxed the volumes are stil too low,
 impossible to stream radio and puting your machine as background-radio.

Although I'd extract the audio
 content being played and look at it in audacity to see if it's not
 just the content to verify the potential source.

 I'm not using audacity at all since that tool is graphical.
 The difference I noticed you can hear it at boot time: before the problem I
 was able to hear clearly the Ubuntu tamtam at gdm login,
 at this moment I hear the Espeak voice in Debian's gdm login very very far
 away, it's unusable !

 IMHO there is a very important crucial bug happend a few versions ago and
 that causes a volume difference of 32 dB or probably much more.

Just a power user
 here and nothing really current version wise on my end to have that
 issue or know much about it myself.  But it'd be nice to know how to
 fix it, if I do run into it.

 I will continue to surf and have a look around to fix it, but I'm

Re: [Alsa-user] vert low-level volume in both Debian and Ubuntu

[James Shatto]

The first step would be to see if it's even an ALSA issue.  With flash
video (youtube) there's a speaker icon and a slider which affects the
volume.  I recently noticed hulu had my levels way low with such an
icon.  With mplayer there's a softvol option which might differ from
the levels set in alsa.  And of course alsamixer to actually set your
levels.  Make sure those are appropriate for what you're trying to do.
 If they aren't, then technically nothing is broke and nothing needs
fixing.  If the levels are set and maxed out and the problem persists,
then it could be an alsa issue.  Although I'd extract the audio
content being played and look at it in audacity to see if it's not
just the content to verify the potential source.  Just a power user
here and nothing really current version wise on my end to have that
issue or know much about it myself.  But it'd be nice to know how to
fix it, if I do run into it.

HTH,
- James


On 5/19/11, Y P yellowpeng...@edpnet.be wrote:
 Hello,

 escuse me but I'm asking myself if the problem I encounter regarding very
 low level of volumes is due to Alsa/Pulse :
 a few weeks/maybe a month ago, I upgraded my EEE netbook's Ubuntu O S; the
 result is : no longer a normal level volume;
 last weekend I did an installation of Debian on a EEE netbook, same result :
 very low level output volume from the internal EEE speakers.
 Finally today a good friend told me he did an installation of an Ubntu on a
 PowerMac, idem same problem regarding sound : very low level.

 How can I test the origins of this problem to be sure which is the source of
 the problem ?

 I did also had a problem in using an eCAFE EC-900 as DJ due to a very
 annoying bug: plugging a minijack in it won't help to cut the output to the
 internal speakers.

 Can someone tell me if one of these problems or both are being fixed soon ?

 sound is very important, appology for all these remarks! please fix them if
 possible!

 Grtnx,

 Y)ellow P)enguin


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Re: [Alsa-user] Newbie query about interfaces

[James Shatto]

It depends on what you need.  A majority of the cheap USB interfaces
are USB 1.x and only do 2 channels.  USB 2.x only recently got an
audio standard ( 5 years) and devices that are starting to use that.
With USB 3.x already being out in the wild of sorts.  IMO, if you need
more than 2 channels, you're better off with a firewire device (for
now).  And yes, they cost a good chunk of money as many of them
include microphone preamps.  At $$$ per channel.  Unless you already
have gear than can deliver 5.1 input over optical cable, you're
probably going to have to chunk out some change.  Even at $50 a
channel, 4x channels is $200-ish.

If you have line level inputs and don't need microphone preamps, you
might have a few options.  A used Delta 44 (PCI) runs about $100 USD
on craigslist and is fairly well supported driver wise.  Although
pulse audio still kind of sucks at a default configuration for it.
And various versions of alsamixer seem to disagree with the hardware
specs more often than not.  Bit it's 4 line level inputs and 4 line
level outputs (24/96) on a budget and works fine under linux.  But it
depends on the budget.  It's 1/4 connectors ONLY.  Insert external
microphone preamp(s), and external headphone preamp(s) to use it like
most OTHER interfaces, plus cables and adapters and whatnot.  If you
don't already have a lot of that stuff.  You're going to be looking at
some $$$.  Or squiggly LLL in your case.

Not that I see how any of this is an alsa issue.  Until you have
questions on a specific device.  There's other websites with forums
that discuss various interfaces and whatnot.  I have an M-Audio Mobile
Pre (USB 1.x, 2 channels in, 2 channels out) and it works fine under
linux with alsa (usb class compliant).  I also have a Delta 44 and it
works fine, with a little extra configuration in some cases.

HTH,
- James


On 5/1/11, Graham Dicker graham.dic...@antecor.com wrote:
 Dominique Michel wrote:

 Le Fri, 29 Apr 2011 16:28:47 +0100,
 Graham Dicker graham.dic...@antecor.com a écrit :

 I have been recording for many years with a Yamaha digital 4 track
 recorder. I would now like to switch to using my Suse Linux minitower
 using Ardour. I am not sure though what kind of audio interface I
 need to buy. Ideally I would like to have similar capability to what
 I am used to with my 4 track i.e. up to four instruments being
 recorded simultaneously, each on to it's own track. I understand that
 an interface like the Yamaha Audiogram 6 will work but it's not clear
 that I will be able to route each instrument on to it's own track. Is
 that what it does? Or do I need something else?

 If the audiogram 6 is recognized by alsa, it will work. Beside ardour,
 you will need jack-audio-connection-kit (jack) and some alsa mixer like
 the alsamixer. jack have several GUI like qjackctl. with it, you can
 route the audio channels as you want to.

 Ciao,
 Dominique

 Thank you for responding. I already have all the software working using the
 motherboard audio interface and have been using it to record stuff for a few
 months. But I can only do two mono tracks or one stereo track at a time - a
 painfully slow process. I have looked around for an interface with more
 inputs and find they vary from around £80 to £3000 or more. The Audiogram 6
 is within my budget, I don't want to spend more than that. The descriptions
 on the vendor websites for this and similar units in this price bracket
 don't mention simultaneous recording on multiple tracks and as far as I can
 tell they are just a kind of mixer, and can only record one track at a time
 (on any OS). Is that true? If so, what kind of interface do I need?

 Thank you

 Graham Dicker



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Re: [Alsa-user] overrun with 'arecord' - why ?

[James Shatto]

Low latency kernel?

xruns are basically a resource issue.  Web browsers have flash and
java and javascript scripts that loop for infinity and other things
that strip you of your resources.  Basically I'd start by closing your
browsers while recording.  If you're trying to capture content from
the browsers, there's other ways to accomplish that IMO.  Without
having to sample the output from a soundcard on a soundcard.
Otherwise when browsers access sound, it's generally the old OSS way
(/dev/dsp).  A few extras like java and flash have gotten smart about
more modern ways, but not all sounds from a web browser are triggered
by those methods.  And not all versions of those things are smart
about it.

Alternatively give your audio priority consideration in
/etc/security/limits.conf.  Check /proc/asound/ for information while
recording.  You might need to tweak period size or other things.
Audio might need priority in other ways.  Which could mean changing
the nice level of the recording application, or the nice level of most
everything else.  I've gotten in the habit of running povray and
ffmpeg conversions with nice -n 19, just so I can still do other
things while they run.  Otherwise they all run at the same nice level
and do battle over who's more important.  Which is not an environment
you want your realtime recording application(s) to be in.

Basically one thing at a time.  If arecord is your recording method,
and what you're recording doesn't need a gui, you might try running
without X and see if you still have xruns.  If you must, slowly add
things back like X, until it breaks to know where your limits are.
AFAIK, arecord is a single threaded application.  So if it's not being
run on the not used CPU(s), and the one that it is on is maxed...  Not
that I'd know how or IF you can choose CPU per task for non SMP aware
applications.

- James


On 4/8/11, Sergei Steshenko steshenko_ser...@list.ru wrote:
 Hello,

 I've tried to run 'arecord' as part of simultaneous playback + capture rig
 (for acoustic measurements) and noticed overruns.

 So, even plain single 'record' occasionally produces overruns:

 arecord -D hw:0,2,0 -c 2 -r 96000 -d 6 -f S32_LE recorded.wav
 Recording WAVE 'recorded.wav' : Signed 32 bit Little Endian, Rate 96000 Hz,
 Stereo
 overrun!!! (at least 855507586.521 ms long)
 .

 So, my question is: Why ?. It's a 2.6Ghz machine with SATA disks. Two
 cores,
 web browsers are the most active tasks (nothing fancy, no sound activity on
 the side of the web browsers). Effectively one core is free.

 Any ideas ?

 The gear:

 Card: HDA NVidia
 Chip: Realtek ALC883
 .

 Thanks,
   Sergei.

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Re: [Alsa-user] overrun with 'arecord' - why ?

[James Shatto]

 No, I'm not trying to capture content from browsers; the browsers have
 no relationship to what I'm doing.

If it's running on the same computer at the same time, there is a
relationship.  i.e. Fewer resources.  An xrun is a lack of resources.
(or a bug)

 arecord -D hw:0,2,0 -c 2 -r 96000 -d 6 -f S32_LE recorded.wav

So the sox variant you're using is?

rec -s -4 -L -c 2 -r 96000 recorded.wav trim 00:00:00 00:00:06

Have you tried arecord without -D ?  And/or with -t wav


- James

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Re: [Alsa-user] overrun with 'arecord' - why ?

[James Shatto]

 Please reread my message in this thread on 'sox' - it contains the complete
 command line I've used.

So apples to oranges?  since your sox only does 4 seconds (trim 1 5)
and your arecord does 6 seconds -d 6.  Statistically that's 50% more
opportunity for failure in arecord.

-

Did omitting -D help?  did adding -t wav help, so it doesn't have to
assume stuff (or not) based only on file extension?

milliseconds are what?  1/1000 of a second.  So 855507586.521 is about
85,550 seconds.  Or roughly 1,426 minutes or roughly 23 hours and 46
minutes.  Kind of odd for a 6 second capture don't you think?  Or is
the result and the example unrelated?

- James

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Re: [Alsa-user] 50 year old male with no sound coming out of his speakers

[James Shatto]

http://ubuntuforums.org/showthread.php?t=286016

does that one help?  Appears common to need to do a card reset for some reason.

If that doesn't work, you might try the snd-hda-intel driver, versus
the snd-intel8x0 that it says you're using.  I don't know which of
those drivers go with that card.  And google hits are varied.  Just a
user as far as alsa goes.  And don't have that particular card on
anything of mine.

- James


On 4/5/11, jida...@jidanni.org jida...@jidanni.org wrote:
 Gentlemen, I cranked everything up but still not an ounce of sound.
 My ALSA information is located at
 http://www.alsa-project.org/db/?f=823f89190858a6673ec0075c004db6de86c7495b
 Yes I connected headphones to the green jack and ran speaker-test(1).
 Doing the same on a different computer one hears static, but on this
 computer -- silence.

 You know what would be really neat, if there was something in /proc that
 could show that yes, there really is something plugged into that 3.5mm
 jack socket, as there is a change in resistance ohms, showing that the
 what looks like it is soldered to the board really is.

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[Alsa-user] Troubleshooting M-Audio Delta 44 (ICE1712)

[James P. Early]

Greetings everyone,
  I'm writing to request assistance with getting my M-Audio Delta 44
(ICE1712) functioning under a fresh Gentoo installation. At first, I
thought I had an issue with jackd, because I could not get it to
start. I tried many configuration options, but typically got ALSA
poll time out messages, then jackd would die. So, I decided to
examine my ALSA installation more carefully.

  Today, I decided to see if I could just get a sound to play using
aplay (from the command line without starting X11). What I notice is
that aplay just freezes at some point while processing the input wav
file. I get no sound, and I have to Ctlr-C to stop it. Here's an
example:

===

$ aplay -vv /usr/share/sounds/alsa/Front_Center.wav
Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit
Little Endian, Rate 48000 Hz, Mono
Plug PCM: Route conversion PCM (sformat=S32_LE)
  Transformation table:
0 - 0
1 - 0
2 - 0
3 - 0
4 - 0
5 - 0
6 - 0
7 - 0
8 - 0
9 - 0
Its setup is:
  stream   : PLAYBACK
  access   : RW_INTERLEAVED
  format   : S16_LE
  subformat: STD
  channels : 1
  rate : 48000
  exact rate   : 48000 (48000/1)
  msbits   : 16
  buffer_size  : 6144
  period_size  : 1024
  period_time  : 21333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min: 1024
  period_event : 0
  start_threshold  : 6144
  stop_threshold   : 6144
  silence_threshold: 0
  silence_size : 0
  boundary : 6917529027641081856
Slave: Direct Stream Mixing PCM
Its setup is:
  stream   : PLAYBACK
  access   : MMAP_INTERLEAVED
  format   : S32_LE
  subformat: STD
  channels : 10
  rate : 48000
  exact rate   : 48000 (48000/1)
  msbits   : 24
  buffer_size  : 6144
  period_size  : 1024
  period_time  : 21333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min: 1024
  period_event : 0
  start_threshold  : 6144
  stop_threshold   : 6144
  silence_threshold: 0
  silence_size : 0
  boundary : 6917529027641081856
Hardware PCM card 0 'M Audio Delta 44' device 0 subdevice 0
Its setup is:
  stream   : PLAYBACK
  access   : MMAP_INTERLEAVED
  format   : S32_LE
  subformat: STD
  channels : 10
  rate : 48000
  exact rate   : 48000 (48000/1)
  msbits   : 24
  buffer_size  : 6553
  period_size  : 1024
  period_time  : 21333
  tstamp_mode  : ENABLE
  period_step  : 1
  avail_min: 1024
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 7378022089539715072
  silence_threshold: 0
  silence_size : 7378022089539715072
  boundary : 7378022089539715072
  appl_ptr : 0
  hw_ptr   : 0
+  | 46%^C
Aborted by signal Interrupt...

==

You'll notice in this example that it has frozen at the 46% mark.
Other files may stop at different points. All indications (aplay,
lspci, /proc/asound, alsamixer, etc.) are that the card is being
recognized, and that the associated kernel module is loaded.

I've tried a number of sites for answers including:
http://alsa.opensrc.org/TroubleShooting
http://www.alsa-project.org/main/index.php/Matrix:Module-ice1712
http://www.gentoo.org/doc/en/alsa-guide.xml

... and countless forum posts

I have run alsa-info, and the result can be seen here:
http://www.alsa-project.org/db/?f=0703a7d0bd8da7a7b4bc387e107900acd0674a7b

I would greatly appreciate hearing from anyone with information about
further testing or configuration.

Thanks,
~Jim

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Re: [Alsa-user] Troubleshooting M-Audio Delta 44 (ICE1712)

[James Shatto]

Try:

speaker-test -c 2 -D hw:0

where hw:# is the number of your card as it shows in /proc/asound/cards.

snd-ice1712 is the driver.  With a Delta 44 myself.  Pulse-audio
doesn't play nice with it, so disable that if reasonable.  Most apps I
use interface with alsa or jackd directly, so I just leave pulse audio
as it came with ubuntu.  Semi working, but mostly annoying.  Complete
with re-indexing the alsa drivers so that ice1712 is card 0 to get it
working initially.  Even if pauvcontrol says my only output option is
dummy out or HDMI out from my video card (snd-hda-intel).  But
probably not your issue, so I'll stop rambling.

- James


On 4/2/11, James P. Early earl...@gmail.com wrote:
 Greetings everyone,
   I'm writing to request assistance with getting my M-Audio Delta 44
 (ICE1712) functioning under a fresh Gentoo installation. At first, I
 thought I had an issue with jackd, because I could not get it to
 start. I tried many configuration options, but typically got ALSA
 poll time out messages, then jackd would die. So, I decided to
 examine my ALSA installation more carefully.

   Today, I decided to see if I could just get a sound to play using
 aplay (from the command line without starting X11). What I notice is
 that aplay just freezes at some point while processing the input wav
 file. I get no sound, and I have to Ctlr-C to stop it. Here's an
 example:

 ===

 $ aplay -vv /usr/share/sounds/alsa/Front_Center.wav
 Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit
 Little Endian, Rate 48000 Hz, Mono
 Plug PCM: Route conversion PCM (sformat=S32_LE)
   Transformation table:
 0 - 0
 1 - 0
 2 - 0
 3 - 0
 4 - 0
 5 - 0
 6 - 0
 7 - 0
 8 - 0
 9 - 0
 Its setup is:
   stream   : PLAYBACK
   access   : RW_INTERLEAVED
   format   : S16_LE
   subformat: STD
   channels : 1
   rate : 48000
   exact rate   : 48000 (48000/1)
   msbits   : 16
   buffer_size  : 6144
   period_size  : 1024
   period_time  : 21333
   tstamp_mode  : NONE
   period_step  : 1
   avail_min: 1024
   period_event : 0
   start_threshold  : 6144
   stop_threshold   : 6144
   silence_threshold: 0
   silence_size : 0
   boundary : 6917529027641081856
 Slave: Direct Stream Mixing PCM
 Its setup is:
   stream   : PLAYBACK
   access   : MMAP_INTERLEAVED
   format   : S32_LE
   subformat: STD
   channels : 10
   rate : 48000
   exact rate   : 48000 (48000/1)
   msbits   : 24
   buffer_size  : 6144
   period_size  : 1024
   period_time  : 21333
   tstamp_mode  : NONE
   period_step  : 1
   avail_min: 1024
   period_event : 0
   start_threshold  : 6144
   stop_threshold   : 6144
   silence_threshold: 0
   silence_size : 0
   boundary : 6917529027641081856
 Hardware PCM card 0 'M Audio Delta 44' device 0 subdevice 0
 Its setup is:
   stream   : PLAYBACK
   access   : MMAP_INTERLEAVED
   format   : S32_LE
   subformat: STD
   channels : 10
   rate : 48000
   exact rate   : 48000 (48000/1)
   msbits   : 24
   buffer_size  : 6553
   period_size  : 1024
   period_time  : 21333
   tstamp_mode  : ENABLE
   period_step  : 1
   avail_min: 1024
   period_event : 0
   start_threshold  : 1
   stop_threshold   : 7378022089539715072
   silence_threshold: 0
   silence_size : 7378022089539715072
   boundary : 7378022089539715072
   appl_ptr : 0
   hw_ptr   : 0
 +  | 46%^C
 Aborted by signal Interrupt...

 ==

 You'll notice in this example that it has frozen at the 46% mark.
 Other files may stop at different points. All indications (aplay,
 lspci, /proc/asound, alsamixer, etc.) are that the card is being
 recognized, and that the associated kernel module is loaded.

 I've tried a number of sites for answers including:
 http://alsa.opensrc.org/TroubleShooting
 http://www.alsa-project.org/main/index.php/Matrix:Module-ice1712
 http://www.gentoo.org/doc/en/alsa-guide.xml

 ... and countless forum posts

 I have run alsa-info, and the result can be seen here:
 http://www.alsa-project.org/db/?f=0703a7d0bd8da7a7b4bc387e107900acd0674a7b

 I would greatly appreciate hearing from anyone with information about
 further testing or configuration.

 Thanks,
 ~Jim

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Re: [Alsa-user] MobilePre USB support

[James Shatto]

I have the mobile pre (old one, but not the oldest one).  It just
works.  USB compliant, at least for USB 1.x standards.  i.e. 2
channels input, 16 bit, 48kHz max.  The gray one with buttons on
front, and pretty much any analog connection type known to man.
Although the line input (3.5mm) does not provide the plug in power for
cheap-ish / camcorder type mics.  And the phantom power is known to be
a bit under volt, but good enough for most mics IMO.

Worked out of the box for me.  But there is some wonkyness in Debian
Lenny(5.0) with freezes (wasn't an issue in sarge).  But then again
mine is old enough that the blue LED light comes and goes.  And I'm
running the distro supplied kernel 2.6.26.  As some of the changes in
the current version of things makes it a lot more difficult to run a
custom kernel.  At least with old school ways and a working config
from a previous install.  I may upgrade to 6.0 as soon as some of my
current projects are wrapped up.  But for $80 off of craigslist, I'm
not complaining.

- James


On 3/15/11, Jonathan Wilkes jancs...@yahoo.com wrote:
 Hi,
  I'm looking for a relatively inexpensive USB audio interface that will
 work (painlessly) on a laptop running Gnewsense, which would basically
 mean either Hardy or Squeeze.  After some looking and reading I'm thinking
 about something like

 Maudio Fast Track (one with the knobs on the top)

 Fast Track Pro

 or MobilePre (one with the knobs on the top).

 Can anyone tell me which of these works the most reliably with ALSA?  Or
 if there's some other interface in this price range that has an XLR input
 and works as well as the USB Transit seems to do?

 I basically want to record into Ardour or Pure Data, and monitor with
 headphones.

 Thank you,
 Jonathan




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Re: [Alsa-user] MobilePre USB support

[James Shatto]

AFAIK, the old old one is white-ish and looks like a fallback to the
1950's.  And AFAIK, that is the only difference.  The Fast Track Pro
is probably the more liked modern one (USB 2.x?).  Although I don't
know of it's linux status.  Should be fine IMO, if it's class
compliant.  It took a good long while, but there is a 2.0 audio
standard now.

When recording you want 24 bit IMO.  16 bit is more of a delivery
format.  24 bit gives you more dynamic range, which better suits
recording IMO.  About all I use my Mobile Pre for is laptop sound.  Or
if I need to archive an odd format like reel to reel tapes or
cassettes (judges tapes).

I have a Delta 44 (24/96) on the desktop and a Korg MR-1000 (24/192 or
DSD) for anything more serious.  Except for the DSD part, all linux
compatible as well.  Although the Korg only functions as a usb storage
device as far as a computer is concerned.

M-Audio tends to use the same ADC/DAC chips in most of their gear, so
it's a fairly safe bet IMO.  Safe-er than some other options anyway.
Although not that configured by default in things like pulse audio and
such.  But the driver(s) work, always have IMO.  Some of the mixer
stuff can be a little off.  But I'm not exactly running the latest and
greatest of everything.  Most of my hardware is sufficiently old that
I don't need to in most cases.

- James


On 3/15/11, Jonathan Wilkes jancs...@yahoo.com wrote:
 Thanks, James.  From a quick google search it looks like one can still get
 these.  But just to make sure I'm talking about the same one-- what's
 the difference between the oldest one you referred to and the one
 you've got?

 And has anyone had success with the new shiny little one with top knobs?

 Thanks,
 Jonathan

 --- On Tue, 3/15/11, James Shatto wwwshad...@gmail.com wrote:

 From: James Shatto wwwshad...@gmail.com
 Subject: Re: [Alsa-user] MobilePre USB support
 To: alsa-user@lists.sourceforge.net
 Date: Tuesday, March 15, 2011, 2:04 PM
 I have the mobile pre (old one, but
 not the oldest one).  It just
 works.  USB compliant, at least for USB 1.x
 standards.  i.e. 2
 channels input, 16 bit, 48kHz max.  The gray one with
 buttons on
 front, and pretty much any analog connection type known to
 man.
 Although the line input (3.5mm) does not provide the plug
 in power for
 cheap-ish / camcorder type mics.  And the phantom
 power is known to be
 a bit under volt, but good enough for most mics IMO.

 Worked out of the box for me.  But there is some
 wonkyness in Debian
 Lenny(5.0) with freezes (wasn't an issue in sarge).
 But then again
 mine is old enough that the blue LED light comes and
 goes.  And I'm
 running the distro supplied kernel 2.6.26.  As some of
 the changes in
 the current version of things makes it a lot more difficult
 to run a
 custom kernel.  At least with old school ways and a
 working config
 from a previous install.  I may upgrade to 6.0 as soon
 as some of my
 current projects are wrapped up.  But for $80 off of
 craigslist, I'm
 not complaining.

 - James


 On 3/15/11, Jonathan Wilkes jancs...@yahoo.com
 wrote:
  Hi,
   I'm looking for a relatively
 inexpensive USB audio interface that will
  work (painlessly) on a laptop running Gnewsense, which
 would basically
  mean either Hardy or Squeeze.  After some looking
 and reading I'm thinking
  about something like
 
  Maudio Fast Track (one with the knobs on the top)
 
  Fast Track Pro
 
  or MobilePre (one with the knobs on the top).
 
  Can anyone tell me which of these works the most
 reliably with ALSA?  Or
  if there's some other interface in this price range
 that has an XLR input
  and works as well as the USB Transit seems to do?
 
  I basically want to record into Ardour or Pure Data,
 and monitor with
  headphones.
 
  Thank you,
  Jonathan
 
 
 
 
 
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Re: [Alsa-user] Help configuring HDSP9632

[James Shatto]

In theory you don't need the dmix thing anymore.  If your applications
use ALSA natively, it will automagically mix sound from several
applications (in software).  If the applications use OSS, you can
force it to use alsa with aoss.  BITD you'd run esddsp or artsdsp -m
app to do this sort of thing.  Depending on the sound daemon of
choice you happened to be using.  These days those daemons just get in
the way, chew up resources and cause XRUNs or other woes.  If your
applications are configured to use ALSA, this should be a non-issue.
Assuming that you're running something current and not RH 5.1 from
some book or something.

In the few times that I tried to use dmix BITD, it was generally the
cause of problems, not the solution.  If your applications use OSS and
you don't launch them with aoss, then they will lock the device (per
days of old).  I'm not sure if that's addressed with oss emulation or
not.  And aoss isn't perfect as something like a browser will launch
pop ups that are NOT launched with aoss and break the very thing you
were trying to avoid.  Mostly problematic with internet gaming where
the games are pop ups.  But for most other application you can select
the audio system of choice.  alsa, oss, jackd, artsd, esd,
pulse-audio, and probably others.  Alsa, having the least overhead
IMO, if you're coming up short on system resources.

check your .asoundrc and whatever system defaults were created for you
or by you in /etc/.  I'm not sure of that locations default naming
convention as it probably varies between distros.  alsa.conf?
asound.conf?  +/- an /etc/ or /etc/alsa/ or /etc/sound/ or ???  And
various tricks of old to delete the asound.state file to force new
defaults.  Located at /var/lib/alsa/asound.state on my system.  YMMV

HTH,
- James


On 2/26/11, Bill Unruh un...@physics.ubc.ca wrote:
 On Sat, 26 Feb 2011, Friedrich Ewaldt wrote:

 Hi Matt,

 I didn't use a RME HDSP9632 for quite a long time (also I never used it
 with the dmix plugin). However, the dmesg message sounds like a clock
 source problem. All I can suggest is to check for the correct rate
 settings, e.g. compare what hdspconf is showing to the output of

 cat /proc/asound/card0/hdsp

 --fe

 Matthew Robbetts schrieb am 26.02.2011 15:18:
 Hi guys,

 I've been trying off and on for weeks now, but I can't get my RME
 HDSP9632 configured under ALSA properly.

 (Is it me or does ALSA really not make this an easy process? I can't
 find anywhere to get any feedback from the system on configuration
 errors. Hell, if you make a typo in the config file, you only find out
 because of some scary-looking output from aplay. The docs on the website
 seem to be quite old and often conflict with each other and I can't find
 any relevant man pages.)

 Anyhow, the card works out of the box, insofar as it lets one
 application play sound through it at a time. So I'm trying to do the

 OK. It works. That finishes alsa.

 common thing of configuring dmix to let multiple applications output
 sound at once. Nothing fancy, really! At least, at this point.

 That in general has nothing to do with the card or the driver of the card.
 Most cards do not allow multiple inputs to all play at once. It is software.
   It is often pulseaudio or jack could be used as well.
 .

 My /etc/asound.conf file is as follows (pieced together from tuts and
 the like):

 
 pcm.!default {
 type plug
 slave.pcm hdsp9632_dmix
 hint {
 show on
 description Default device: Plugs into hdsp9632_dmix.
 }
 }

 ctl.hdsp9632_dmix {
 type hw
 card 0
 }

 pcm.hdsp9632_dmix {
 type dmix
 ipc_perm 0660
 ipc_key 1025
 ipc_key_add_uid false
 slave {
 pcm hw:0,0
 rate 44100
 channels 2
 period_size 1024
 buffer_size 4096
 }
 bindings {
 0 0
 1 1
 }
 hint {
 show on
 description hdsp9632_dmix: The dmix plugin - plugs into 
 hdsp9632.
 }
 }
 


 Using this file, I get

 # aplay -L
 null
 Discard all samples (playback) or generate zero samples (capture)
 default
 Default device: Plugs into hdsp9632_dmix.
 hdsp9632_dmix
 hdsp9632_dmix: The dmix plugin - plugs into hdsp9632.

 which is what I hope for. But, if I try and play something with vlc, I
 get an error message and No AutoSync source for requested rate comes
 up in dmesg. The card is currently set to clock master at the same
 sample rate as the audio (44.1kHz).


 If anyone can shed any light on what I'm doing wrong (and, ideally, some
 methodology on configuring ALSA which doesn't require scrabbling around
 in the dark!), I will be grateful until the end of time.

 The alsa users documentation has long long long been its greatest shortfall.
 And noone seems to be stepping up to the plate to write the docs. One of the
 problems with the open software movement

Re: [Alsa-user] Record 8 separate Line IN Channels from M-Audio Delta 1010 Card

[James Shatto]

-f cd is a shortcut for a STEREO track.  AFAIK, the output for arecord
is ONE file, with many channels in it.
i.e. -f cd == -f S16_LE -t wav -c 2 -r 44100
and
i.e. -f cdr == -f S16_BE -t wav -c 2 -r 44100
(what it gets converted to before burning a disc)

or something like that...

$ arecord -t wav -f S16_LE -c 8 -r 48000 -D ice1712 All_8_Tracks.wav
(would that be ice1724?  Dont know, just asking.)

sndfile-deinterleave
sox
audacity
ffmpeg

and probably others to chunk out each individual track and convert
them to mono.  Plus/minus on the syntax's, it's been a while and using
gray matter only.  Aften to create 5.1 ac3 audio.  ffmpeg is limited
to creating 5.0 iirc.  And other quirks for pretty much all of the
options.

- James


On 2/21/11, Sergei Steshenko steshenko_ser...@list.ru wrote:
 On Mon, 21 Feb 2011 17:37:24 +0100
 Peter Hoffmann p...@peter-hoffmann.com wrote:

 wa

 (BTW can anyone point me to a tool to split a multi channel wav into a
 file per channel?)


 SoX (sox.sf.net), 'ecasound'; 'audacity'.

 Regards,
   Sergei.

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Re: [Alsa-user] No sound, no /proc/asound/

[James Shatto]

Well depending on HOW it was obtained.  The short answer is that the
kernel primarily installs to only TWO locations.

/lib/modules/`uname -r`/

and

/boot/

So check for the #.##.## of your kernel version in those locations.
Also note a few symlinks /boot/config / boot/system /boot/kernel that
might link to the #.##.## of your kernel.  Not to worry those are
handled at your re-install.  But there might be an initrd image in
there that could linger and not update if you do anything manual-ish
that could be a trouble maker.  As in could be formed from another
version you're not actually using, but the boot loader tries to use it
anyway.  That one is the primary difference between a custom kernel
and a distro kernel in a lot of cases.

So the basic procedure might be...

$ sudo dpkg --purge --force-all kernel-*version*
(might also be some header, image, modules, or other things for that
image depending on how the distro packages it.  Purge them all.  Make
sure only for the version in question.  And keep your OLD kernels /
ALTERNATE kernels around because you'll have to boot to them to
re-install.  And/or just to do this step.)

$ sudo rm -rf /lib/modules/linux-*version*
(tab completion is your friend)

$ sudo rm -rf /boot/*version*
(make sure you're not grabbing anything important.  As long as your
alternates don't share the same version number, you should be
safe-ish)

Perhaps a good ideal to do a full backup before these steps, just in case.

$ sudo apt-get install kernel-*version*
(plus any related packages)

It's not unheard of for a distro to botch a particular kernel for a
particular purpose.  Depending on your distro and version there of.
Most times they will be updated or replaced with the latest and
greatest at the next update.  At least a couple times a year, so 30
days to six months and your issue might automagically disappear.
Otherwise try those steps above.  Perhaps an ls *version* beforehand
to ensure that you're not grabbing anything not intended to be
grabbed.  You can also mv the stuff versus rm if you want a
recovery option, but a bit more tedious and no real need to hang on to
it for all intents with alternate options.  Potentially dangerous
commands there so be weary of fat fingers.  And backup first if you
don't trust yourself.  And backup if you DO trust yourself.

Not to clutter the issue, but sometimes /boot/ is on a different
partition / device and unmounted after boot.  In that instance you
might need to do some trickery to have it be there to uninstall from.
Basically don't assume anything, verify verify verify.  With certain
permission schemes(acl/selinux) it's entirely possible that the
process is not that simple.  But it could be.

- James



On 2/15/11, Marcin Szyniszewski mszyn...@gmail.com wrote:
 Hello,

 I checked if alsa and stuff is working on other kernels - it seems it is
 working brilliantly! Mic and sound works fine!
 So the problem would be with the latest kernel. I removed it while being on
 previous one and installed it again, but the problem is still present.
 Then I tried to do all the stuff that was suggested here again, nothing
 worked. So it looks like it's the fault of kernel but reinstalling it
 somehow doesn't work! Do you have some suggestions? Maybe it's not kernel
 after all? Or maybe there's some different way to remove the kernel
 completely and reinstall it?

 My impression that that at some earlier stage audio *was* working, so the

 current lack of working is due to something like an attempt to do something

 like 'upgrade' the kernel. If so, my recommendation is to *always* do a

 backup of your system before doing anything that might furtle things up.


 I use 'clonezilla' for this every now and then to try to protect myself

 from my own idiocy. Put the backup on a removable USB HD. But there are

 various other ways you may prefer.


 Yeah, thats a good idea. Fortunately there are previous kernels available!

 Best,
 *mszynisz*


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Re: [Alsa-user] No sound, no /proc/asound/

[James Shatto]

As I suspected, the modules aren't loaded so alsa isn't even running.
Hence your original open error(s).

How did you install alsa?  Not that I think it is your issue, but it
could be.  If you boot with lilo, you need to re-install lilo after
creating a new kernel.  Even if it's technically the same version of
your old kernel.  Although most distros default to grub these days.
So not likely.

If you compiled from source at least for some modules, you'll need to
reboot to use the new kernel and the new modules.  Not really
applicable to sound as you probably didn't change any PCIe or other
internals to gain the functionality.

In the old days if you compiled from source you could insmod
(modprobe) the modules in alsa-driver-???/modules/ until you got the
right order and all of the modules loaded.  This is representative of
the errors that you're seeing.  You can't load a certain module
because another module wasn't loaded before it.  That has those
symbols (functions) that it needs.  Which brings things full circle to
alsa isn't properly installed.

$ sudo dpkg -l '*alsa*'

Only pay attention to the ones that start alsa or alsa-.  On my debian
setup (similar to ubuntu) I have alsa, alsa-base,
alsa-firmware-loaders, alsa-headers, alsa-source, alsa-tools,
alsa-tools-gui, and alsa-utils.  On my system all of those are
installed, except alsa-firmware-loaders, alsa-headers (needed to
compile other things from source against it), and alsa-tools-gui.  IMO
you are probably missing alsa-base.  This should have entries in
/etc/modprobe.d/alsa* for autoloading your modules (without concerning
yourself about the order of insertion).  It could also be that you
haven't run depmod -a, or your distro didn't.  Which updates a sort of
list of what modules are related so they can also load when the other
is loaded.  IME, alsa is independent of this list and relies on other
things (/etc/modprobe.d/).

If you haven't solved your issue by now, I guess you're stuck with the
old school ways.  Meaning you'll likely have to create a
/etc/modprobe.d/ entry for alsa so it can auto load at boot.  Which
might look something like:

#--- START - /etc/modprobe.d/alsa_custom.conf ---#

alias   char-major-116  snd
alias   char-major-14   soundcore

options snd major=116   cards_limit=3

# duplicate this following sequence for each soundcard you have
# and bump (or omit) the index=# depending on the order / priority
# that you desire.  And adjust the first # in the sound- aliases to
# match the index number.

# your specific module NEXT LINE (and the next one)
options snd-hda-intel  index=0
alias   snd-card-0  snd-hda-intel

# this one assumes OSS emulation, you might need to
# reference alsa-project.org to find a different one if you
# opted out on that option.  --with-oss=yes ?
# (been a while)
alias   sound-slot-0snd-card-0
alias   sound-service-0-0   snd-mixer-oss
alias   sound-service-0-1   snd-seq-oss
alias   sound-service-0-3   snd-pcm-oss
alias   sound-service-0-8   snd-seq-oss
alias   sound-service-0-12  snd-pcm-oss

#--- END ---#

And 20 years after linux started, we're still configuring sound from
the command line.  Be sure to reboot OR try to use the soundcard to
get the modules to auto magically load.  They generally load at boot
because your distro will likely try to restore mixer settings.  And
therefor try to use your soundcard.  (which is or was failing for you)

- James


On 2/12/11, Marcin Szyniszewski mszyn...@gmail.com wrote:
 Thank you all for the replies! Very appreciated! :)

  $ sudo modprobe [module]
 FATAL: Error inserting snd
 (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd.ko): Unknown
 symbol in module, or unknown parameter (see dmesg)
 WARNING: Error running install command for snd
 WARNING: Error inserting snd_pcm
 (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd-pcm.ko):
 Unknown symbol in module, or unknown parameter (see dmesg)
 WARNING: Error inserting snd_hwdep
 (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd-hwdep.ko):
 Unknown symbol in module, or unknown parameter (see dmesg)
 WARNING: Error inserting snd_hda_codec
 (/lib/modules/2.6.35-25-generic/kernel/sound/pci/hda/snd-hda-codec.ko):
 Unknown symbol in module, or unknown parameter (see dmesg)
 FATAL: Error inserting snd_hda_intel
 (/lib/modules/2.6.35-25-generic/kernel/sound/pci/hda/snd-hda-intel.ko):
 Unknown symbol in module, or unknown parameter (see dmesg)

 This doesn't look good. What do you think is wrong??

 Note that * is a wildcard.  So /dev/dsp* is any devices that start
 with /dev/dsp.
 Yes, of course. I did ll in this folder and went through the whole
 list. Nothing's there.

 # modprobe snd-hda-intel
 Gives me permission errors.

 $ sudo modprobe snd-hda-intel
 Gives the result above.

 $ sudo pavucontrol
 sudo: pavucontrol: command not found

 $ lsmod | grep -i snd
 snd_page_alloc  7120  0

 $ cat /proc/asound/cards
 cat: /proc/asound/cards: No such file

Re: [Alsa-user] No sound, no /proc/asound/

[James Shatto]

 $ sudo dpkg -l '*alsa*'
 Desired=Unknown/Install/Remove/Purge/Hold
 |
 Status=Not/Inst/Conf-files/Unpacked/halF-conf/Half-inst/trig-aWait/Trig-pend
 |/ Err?=(none)/Reinst-required (Status,Err: uppercase=bad)
 ||/ Name Version
  Description
 +++---
 un  alsa none
   (no description available)
 ii  alsa-base1.0.23+dfsg-1ubuntu4
   ALSA driver configuration files
 ii  alsa-firmware-loaders1.0.23-3ubuntu1
  ALSA software loaders for specific hardware
 ii  alsa-oss 1.0.17-4
   ALSA wrapper for OSS applications
 ii  alsa-source  1.0.23+dfsg-1ubuntu4
   ALSA driver sources
 ii  alsa-tools   1.0.23-3ubuntu1
  Console based ALSA utilities for specific hardware
 ii  alsa-tools-gui   1.0.23-3ubuntu1
  GUI based ALSA utilities for specific hardware
 ii  alsa-utils   1.0.23-2ubuntu3.4
  Utilities for configuring and using ALSA
 ii  alsamixergui 0.9.0rc2-1-9
   graphical soundcard mixer for ALSA soundcard driver
 ii  bluez-alsa   4.69-0ubuntu2
  Bluetooth audio support
 ii  gnome-alsamixer  0.9.7~cvs.20060916.ds.1-2
  ALSA sound mixer for GNOME
 ii  gstreamer0.10-alsa   0.10.30-2
  GStreamer plugin for ALSA
 un  libsdl1.2debian-alsa none
   (no description available)

 Looks like there's some problem with alsa :( How to fix this?

Well there's the old school ways.  When all else fails, re-install.
Fortunately in linux that's not as dreaded as it sounds


$ sudo dpkg --purge --force-all alsa alsa-base alsa-firmware-loaders
alsa-oss alsa-source alsa-tools alsa-tools-gui alsa-utils alsamixergui

(removes the packages)


$ sudo apt-get install alsa alsa-base alsa-firmware-loaders alsa-oss
alsa-source alsa-tools alsa-tools-gui alsa-utils alsamixergui

(puts them back)


HTH,
- James

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Re: [Alsa-user] No sound, no /proc/asound/

[James Shatto]

Basically the same way.  To redo your kernel.  dpkg --purge
--force-all  apt-get install.  Just make sure that you DO NOT do it
to the kernel that you are currently running.  Which might mean
installing an older kernel as a safe recovery and boot to that before
recovering the kernel you want to run.  This week anyway.

Definitely a module mismatch.  But is it because the kernel you are
running is mangled with old modules, old initrd images, old ???
Because the boot loader isn't using the NEW kernel version?  Depmod
and any number of things depending on how you came about your current
config.  Assuming the old school insmod route can't be made to work at
all.  Which requires some functional knowledge of your system.  i.e.
lsmod (from a working version / live CD).  But requires no
configuration to load the modules, outside of the right sequence.  And
full paths if you use insmod, and not modprobe.

$ sudio find /lib/modules/`uname -r`/ -iname '*snd*.*o'

I've been assuming that you've been running a distro supplied kernel.
I guess the question should be asked, how did you come by your current
kernel?  Supplied by the distro or did you do something different?  In
either case you might want to try a distro supplied kernel.
Preferably one that differs from the version (name) that you are
currently using.  At least in terms of simple fixes.

Beyond that you might rm ~/.asoundrc and the
/etc/modprobe.d/alsa_custom.conf when you reinstall alsa.  Or at least
mv to ~/ with different names so you can easily recover them.
Otherwise it appears that you might have installed alsa from source,
and an update to the same kernel version might have overwritten in
part your changes.  The rm step to happen between dpkg --purge and
apt-get install.  For the kernel you might want to rm the
/lib/modules/2.6.35.???/ for the kernel in question, just in case
something lingered.  Between purge and install of course.  While
running a differently named kernel.

Otherwise a fresh FULL reinstall should fix your issue.  Assuming that
your card is supported in the first place, which it appears to be or
it would have never worked.  Otherwise we could troubleshoot for days
without more information about how you got to your current state of
affairs.  Not that you'd have that standard M$ answer.  I installed
AOL and now XXX doesn't work anymore...

- James



On 2/12/11, Bill Unruh un...@physics.ubc.ca wrote:
 On Sat, 12 Feb 2011, Marcin Szyniszewski wrote:

 On Sat, Feb 12, 2011 at 16:26, James Shatto wwwshad...@gmail.com wrote:

 $ sudo depmod -a
 $ sudo modprobe snd-hda-intel
 WARNING: Error inserting snd_timer
 (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd-timer.ko): Unknown
 symbol in module, or unknown parameter (see dmesg)

 This usually means that you have a module mismatch-- the modules you loaded
 are not the up to date modules for your kernel.  You may well have neglected
 to uninstall previous modules before puttin in the new ones.

 I would  remove the current kernel and then reinstall the kernel forcing it
 to reinstall everything
 (I have no idea how debian does this-- I use a rpm based system).
 None of the alsa modules are being installed so it is not surprizing you are
 getting no sound.

 WARNING: Error inserting snd_pcm
 (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd-pcm.ko): Unknown
 symbol in module, or unknown parameter (see dmesg)
 WARNING: Error inserting snd_hwdep
 (/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd-hwdep.ko): Unknown
 symbol in module, or unknown parameter (see dmesg)
 WARNING: Error inserting snd_hda_codec
 (/lib/modules/2.6.35-25-generic/kernel/sound/pci/hda/snd-hda-codec.ko):
 Unknown symbol in module, or unknown parameter (see dmesg)
 FATAL: Error inserting snd_hda_intel
 (/lib/modules/2.6.35-25-generic/kernel/sound/pci/hda/snd-hda-intel.ko):
 Unknown symbol in module, or unknown parameter (see dmesg)

 Looks like module loader is not willing to cooperate :/ Do you know what's
 going on?

 Thank you all for the replies! Please help!

 Best,
 *mszynisz*


 --
 William G. Unruh   |  Canadian Institute for| Tel: +1(604)822-3273
 PhysicsAstronomy  | Advanced Research  | Fax: +1(604)822-5324
 UBC, Vancouver,BC  |   Program in Cosmology | un...@physics.ubc.ca
 Canada V6T 1Z1 |  and Gravity   |  www.theory.physics.ubc.ca/

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Re: [Alsa-user] No sound, no /proc/asound/

[James Shatto]

Most times when I get something like that it has to do with the
/dev/'s not being present.  Could be that udev isn't running on your
box.  Or isn't configured for alsa.  It could also be something else
like snd-pcm-oss not auto loading.  And it's friends, snd-mixer-oss
snd-seq-oss.  Basically cannot open means some sort of missing
something or bad permissions.  Is the user in the audio group?  Do the
/dev/audio* and /dev/dsp* stuff exist?  In the old days we'd run
./snddevices from the alsa-driver source tree.  But that's probably
not the solution of choice these days.

# /etc/init.d/alsa-utils restart
# /etc/init.d/udev restart

# groups user
# grep -i audio /etc/group

lsmod, dmesg, and all of the other stuff that's probably covered by
that alsa-info.sh script thing.

- James


On 2/11/11, Jim Lesurf j...@audiomisc.co.uk wrote:
 In article
 AANLkTikA=hHDEy3pCsamVvgye7u9=_4pqqw_pscjb...@mail.gmail.com,
Marcin Szyniszewski mszyn...@gmail.com wrote:
  Is the file /usr/bin/alsamixer present, or /sbin/alsa ?
 
  Or the /usr/share/alsa directory?
 
  You should have these or equivalents IIUC.

 /usr/bin/alsamixer is present and gives: cannot open mixer: No such file
 or directory

 Did you issue alsamixer as the command or the full pathname? If the former,
 maybe something is wrong with your path/environment setup.

 Afraid I don't know what the problem is, so I can only suggest some ideas
 and diagnostics to check.

 I am wondering if your OS install hasn't actually loaded the modules
 correctly for your hardware.

 Try the command 'lsmod' to list the modules that are loaded. If the list is
 too long use 'lsmod | grep snd' to just list the ones that have 'snd' in
 their names.

 You can then use modinfo module name to check details of each module.

 Or modprobe (with care!) to alter what is loaded.

 Do you have another sound system like Pulse active? if so, that may be
 interfering with the direct use of ALSA.

 You could also put a simple redefinition of the ALSA default into an
 .asoundrc file and see if that can be made to work with aplay. But from
 what you have said I have doubts about that.

 You might also consider trying to install the latest version of ALSA in
 case what you have isn't suitable for your hardware or is furtled in some
 way.

 Sorry I can't be more help. But I hope the above may be useful.

 Slainte,

 Jim

 --
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Re: [Alsa-user] No sound, no /proc/asound/

[James Shatto]

Note that * is a wildcard.  So /dev/dsp* is any devices that start
with /dev/dsp.

It looks like you don't have the modules loaded.

# modprobe snd-hda-intel
$ sudo modprobe snd-hda-intel

(depending on your distro / $ is user / # is root)

It might be /etc/init.d/alsasound or other named thing depending on
your version and distro.  It might not even be in /etc/init.d/
depending on your distro.  It looks like you have pulse audio running,
so you might try the pavucontrol application.  Should be accessible
through the speaker icon in the taskbar in ubuntu.  Or just run it
from a terminal.

$ sudo pavucontrol

You appear to be installed and with permissions, but if you don't have
/dev/dsp and friends, then you don't have alsa running.  Probably
didn't load up the modules at boot.  Not completely uncommon on a new
install.  Someplace to start looking anyway.

$ lsmod | grep -i snd
$ cat /proc/asound/cards

- James


On 2/11/11, Torsten Schenk torsten.sch...@zoho.com wrote:
 I also use ubuntu (10.04) and it came to happen that the system didn't load
 the modules automatically any more. I don't know why that happened or where
 this loading is prohibited. Just try to load the module manually and see if
 that works. If so, you could also post this on a ubuntu mailing list.

   $ sudo modprobe [module]

 You need to replace [module] with the module that fits your card,
 eventually snd-hda-intel or snd-usb-audio, these are very common cards.

 Greets, Torsten

  On Fri, 11 Feb 2011 20:54:49 +0100 Marcin Szyniszewski  wrote 

 Did you issue alsamixer as the command or the full pathname? If the
 former,
 maybe something is wrong with your path/environment setup.
I used it as both. Nothing works :/

 I am wondering if your OS install hasn't actually loaded the modules
 correctly for your hardware.
Everything worked before. I tried to make my mic work and sound
stopped to work. Now nothing works :P

 Try the command 'lsmod' to list the modules that are loaded. If the list
 is
 too long use 'lsmod | grep snd' to just list the ones that have 'snd' in
 their names.
$ lsmod | grep snd
snd_page_alloc 7120 0

But I don't know what that means :P

 You can then use modinfo  to check details of each module.
 Or modprobe (with care!) to alter what is loaded.
Ok, and what modules should I check?

 Do you have another sound system like Pulse active? if so, that may be
 interfering with the direct use of ALSA.
Stopping pulse and reinstalling ALSA didn't work. :(

 You might also consider trying to install the latest version of ALSA in
 case what you have isn't suitable for your hardware or is furtled in some

 way.
I think I have the latest version.

 Sorry I can't be more help. But I hope the above may be useful.
Thanks for help :)

 Most times when I get something like that it has to do with the
 /dev/'s not being present. Could be that udev isn't running on your
 box. Or isn't configured for alsa. It could also be something else
 like snd-pcm-oss not auto loading. And it's friends, snd-mixer-oss
 snd-seq-oss. Basically cannot open means some sort of missing
 something or bad permissions. Is the user in the audio group? Do the
 /dev/audio* and /dev/dsp* stuff exist? In the old days we'd run
 ./snddevices from the alsa-driver source tree. But that's probably
 not the solution of choice these days.

/dev/audio* doesn't exist, as well as /dev/dsp*
Should I do something about that??

 # /etc/init.d/alsa-utils restart
bash: /etc/init.d/alsa-utils: No such file or directory

 # /etc/init.d/udev restart
Rather than invoking init scripts through /etc/init.d, use the service(8)
utility, e.g. service udev restart
Since the script you are attempting to invoke has been converted to an
Upstart job, you may also use the restart(8) utility, e.g. restart udev
restart: Rejected send message, 1 matched rules; type=method_call,
sender=:1.45 (uid=1000 pid=9806 comm=restart)
interface=com.ubuntu.Upstart0_6.Job member=Restart error
name=(unset) requested_reply=0 destination=com.ubuntu.Upstart
(uid=0 pid=1 comm=/sbin/init))

 # groups
mszynisz : mszynisz adm dialout fax cdrom floppy tape audio dip video
plugdev fuse netdev lpadmin admin sambashare

 # grep -i audio /etc/group
audio:x:29:pulse,mszynisz

 lsmod, dmesg, and all of the other stuff that's probably covered by
 that alsa-info.sh script thing.
My output of alsa-info.sh script is attached.

Please help, I really need my sound :(

Best,
mszynisz
--

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Pinpoint memory and threading errors before they happen.
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Re: [Alsa-user] No sound, no /proc/asound/

[James Shatto]

Except that your web browser likely defaults the OSS, not ALSA.  And
OSS emulation IS part of alsa.  Even if you have to launch an
application with aoss to use the alsa sound drivers.  It's probably
not native alsa, but it is coded as part of alsa's drivers, and
therefor part of alsa.  But yeah, check /dev/snd* for items as well.
It does vary depending on version of alsa, version of the kernel, and
other things.

$ ls -l /dev/* /dev/*/* | grep -i audio

In either case your audio group will likely be assigned to the audio
devices available to you.  By all means nitpick that I used grep -i
audio, versus awk '{ print $4   $9 }' | grep -i audio or
something.

- James


On 2/11/11, Bill Unruh un...@physics.ubc.ca wrote:
 On Fri, 11 Feb 2011, James Shatto wrote:

 Note that * is a wildcard.  So /dev/dsp* is any devices that start
 with /dev/dsp.

 It looks like you don't have the modules loaded.

 /dev/dsp and /dev/audio are the oss sound drivers, not alsa. alsa has an oss
 emulation module, which will create those but they are NOT needed for using
 alsa. What you have under alsa is a buch of entry points under /dev/snd

 Now if your program uses the oss sound system, then you must load the
 alsa-oss
 emulators as well (snd_seq_oss, snd_pcm_oss snd_mixer_oss)
 This will generate the various /dev/dsp entry points.



 # modprobe snd-hda-intel
 $ sudo modprobe snd-hda-intel

 (depending on your distro / $ is user / # is root)

 It might be /etc/init.d/alsasound or other named thing depending on
 your version and distro.  It might not even be in /etc/init.d/
 depending on your distro.  It looks like you have pulse audio running,
 so you might try the pavucontrol application.  Should be accessible
 through the speaker icon in the taskbar in ubuntu.  Or just run it
 from a terminal.

 $ sudo pavucontrol

 You appear to be installed and with permissions, but if you don't have
 /dev/dsp and friends, then you don't have alsa running.  Probably

 Totally false. /dev/dsp is NOT part of alsa.

 didn't load up the modules at boot.  Not completely uncommon on a new
 install.  Someplace to start looking anyway.

 $ lsmod | grep -i snd

 That is a good starting point.

 $ cat /proc/asound/cards

 - James


 On 2/11/11, Torsten Schenk torsten.sch...@zoho.com wrote:
 I also use ubuntu (10.04) and it came to happen that the system didn't
 load
 the modules automatically any more. I don't know why that happened or
 where
 this loading is prohibited. Just try to load the module manually and see
 if
 that works. If so, you could also post this on a ubuntu mailing list.

   $ sudo modprobe [module]

 You need to replace [module] with the module that fits your card,
 eventually snd-hda-intel or snd-usb-audio, these are very common
 cards.

 Greets, Torsten

  On Fri, 11 Feb 2011 20:54:49 +0100 Marcin Szyniszewski  wrote 

 Did you issue alsamixer as the command or the full pathname? If the
 former,
 maybe something is wrong with your path/environment setup.
 I used it as both. Nothing works :/

 I am wondering if your OS install hasn't actually loaded the modules
 correctly for your hardware.
 Everything worked before. I tried to make my mic work and sound
 stopped to work. Now nothing works :P

 Try the command 'lsmod' to list the modules that are loaded. If the
 list
 is
 too long use 'lsmod | grep snd' to just list the ones that have 'snd'
 in
 their names.
 $ lsmod | grep snd
 snd_page_alloc 7120 0

 But I don't know what that means :P

 You can then use modinfo  to check details of each module.
 Or modprobe (with care!) to alter what is loaded.
 Ok, and what modules should I check?

 Do you have another sound system like Pulse active? if so, that may be
 interfering with the direct use of ALSA.
 Stopping pulse and reinstalling ALSA didn't work. :(

 You might also consider trying to install the latest version of ALSA in
 case what you have isn't suitable for your hardware or is furtled in
 some

 way.
 I think I have the latest version.

 Sorry I can't be more help. But I hope the above may be useful.
 Thanks for help :)

 Most times when I get something like that it has to do with the
 /dev/'s not being present. Could be that udev isn't running on your
 box. Or isn't configured for alsa. It could also be something else
 like snd-pcm-oss not auto loading. And it's friends, snd-mixer-oss
 snd-seq-oss. Basically cannot open means some sort of missing
 something or bad permissions. Is the user in the audio group? Do the
 /dev/audio* and /dev/dsp* stuff exist? In the old days we'd run
 ./snddevices from the alsa-driver source tree. But that's probably
 not the solution of choice these days.

 /dev/audio* doesn't exist, as well as /dev/dsp*
 Should I do something about that??

 # /etc/init.d/alsa-utils restart
 bash: /etc/init.d/alsa-utils: No such file or directory

 # /etc/init.d/udev restart
 Rather than invoking init scripts through /etc/init.d, use the
 service(8)
 utility, e.g. service udev restart
 Since the script you

Re: [Alsa-user] [alsa-devel] Creative Sound Blaster Audigy SE Mic problem

[James Courtier-Dutton]

On 24 November 2010 13:38, Grega Fajdiga gregor.fajd...@guest.arnes.si wrote:
 Hello,

 I am using Ubuntu 10.10 with a Creative Sound Blaster Audigy SE.

 The snd_ca0106 module is loaded.

What does this show?
cat /proc/asound/cards

It will tell me if the driver is recognised or not.

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Re: [Alsa-user] Attempting to redirect (and understand) ALSA

[James Le Cuirot]

On Wed, 17 Nov 2010 18:32:01 +1100 (EST)
Howard Lowndes lan...@lannet.com.au wrote:

 I can get sound by doing:
   arecord -D plughw:2,0 | aplay

I have a SAA7134 hybrid too and when I used to do analogue stuff, I
didn't use the internal connector but I can't remember why. If you can
figure that out, it's probably the best way. Otherwise try setting a
lower buffer/period time/size on aplay. You can use -v to see what
settings it is using now and work from there.

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Re: [Alsa-user] Basic S/PDIF Recording

[James Le Cuirot]

On Mon, 25 Oct 2010 09:50:19 -0400
Paul Braman brama...@gmail.com wrote:

 I'll assume I want to read in blocks of 1536 bytes-at-a-time as long
 as ALSA is properly synchronizing to the S/PDIF frame and giving me
 aligned blocks. Is this an assumption I can make?

Probably not. I recently thought I would be clever and decode my
external AC3 source in software without buying an expensive home
theatre system but I found out the hard way that this generally isn't
possible. You can get away with a little jitter when dealing with PCM
but AC3 just can't tolerate it cleanly enough. You end up with short
bursts of noise every couple of seconds. If you want to try it for
yourself, it's as simple as...

arecord -Dspdif -f dat -t raw | spdifextract | ac3dec -6

spdifextract is a small program you can get from here...
http://forums.gentoo.org/viewtopic-p-4472816.html#4472816

I spoke to the original author of the code and he never managed to work
around this problem either. Simply adjusting the buffer or period size
doesn't help because the data has already been lost by this point.

Apparently a PLL is needed to synchronise the clock frequency but I
haven't been able to determine whether any sound cards out there have
these at all. I've heard of some Creative cards having on-board AC3
decoders but I think this may have simply been for DVDs being played on
the machine itself, not for external sources.

James

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Re: [Alsa-user] Basic S/PDIF Recording

[James Le Cuirot]

On Mon, 25 Oct 2010 15:18:03 +0100
James Le Cuirot ch...@aura-online.co.uk wrote:

 Apparently a PLL is needed to synchronise the clock frequency but I
 haven't been able to determine whether any sound cards out there have
 these at all. I've heard of some Creative cards having on-board AC3
 decoders but I think this may have simply been for DVDs being played
 on the machine itself, not for external sources.

Actually now I think about it, the guy in that Gentoo thread says he
got it working with a Terratec Aureon 5.1 USB MkII. One of those is
significantly cheaper than a home theatre system. I'll try to contact
him as I want to double check whether he really got it working. Those
posts were from years ago though.

James

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Re: [Alsa-user] Basic S/PDIF Recording

[James Le Cuirot]

On Mon, 25 Oct 2010 10:45:10 -0400
Paul Braman brama...@gmail.com wrote:

 Interesting, but I see a couple of idiological problems with this
 approach.
 
 dat implies reading 32-bit frames at a rate of 48KHz. That's all
 fine and good but an S/PDIF bitstream is going to be pumping data
 faster than that rate. Assuming the embedded audio is 48KHz, 16-bit,
 stereo then each frame of raw 32-bit PCM is encoded within a 64-bit
 S/PDIF frame. It's as if arecord is told to set up to read at only
 half the speed it needs to be reading. I'd suspect xrun conditions to
 cause the noise.

I'm certainly no expert but I don't think it works quite like that.
Taken from http://ac3filter.net/guides/ac3filter_spdif...

Since compressed data is transmitted in place of PCM data, the bitrate
of the compressed stream must exactly match uncompressed stereo 16-bit
PCM bitrate. As a rule, compressed stream (even a multi-channel one)
having a lower bitrate, compressed stream must be padded with zeros to
match PCM bitrate.

It goes on to say that DTS can be converted to use 14 bits instead of
16 to lessen the harsh noise you get when compressed data is mistaken
for PCM.

When you try to record from S/PDIF with arecord, it only allows S16_LE
and S32_LE and I'm pretty sure I tried both.

James

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Re: [Alsa-user] output from one application to input of another

[James Le Cuirot]

On Sat, 9 Oct 2010 19:18:38 -0400
Michael Di Domenico mdidomeni...@gmail.com wrote:

 Is there a way to take the output from the first application and
 redirect it into a /dev device for input into this second program?

This may not be the answer you're looking for but I'm pretty sure
PulseAudio can do this. There are probably other solutions but I'm not
that familiar with OSS.

James

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Re: [Alsa-user] Alsa and Jack

[James Le Cuirot]

On Tue, 5 Oct 2010 22:37:06 +0800
Samuel Kidman samkid...@gmail.com wrote:

 I'm having issues getting audio programs that don't use jack to make
 sounds while i'm running jack. as soon i quit jack everything works
 again. Applications that use jack work fine when jack is running but
 i can't get them to make a sound without jack. Ultimately I would
 like  to have everything working with jack and making sounds all at
 the same time.

What distribution are you using and do you have the
file /usr/lib/alsa-lib/libasound_module_pcm_jack.so?

James

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Re: [Alsa-user] Basic PCM Recording

[James Le Cuirot]

On Tue, 5 Oct 2010 09:30:37 -0400
Paul Braman brama...@gmail.com wrote:

 The problem is that there is no good way to discover or set these
 things that seems official. There are semi-official suggestions of
 set to maximum buffer and divide into four periods or buffer about
 a second and divide into about 8 periods. It all just tastes too
 wishy-washy for me. But, I'm only one person.

I'm not in your position but if I was, I would certainly want something
more concrete too. I once discussed these issues with the developer of
Twinkle, a SIP client that uses ALSA. He ended up having to provide
configuration options in the GUI for stuff that the end user really
shouldn't need to know about.

James

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Re: [Alsa-user] [alsa-devel] Creative Sound Blaster X-Fi Titanium PCIe support

[James Le Cuirot]

On Fri, 01 Oct 2010 19:40:50 +0200
Thor Kristoffersen tho...@gmail.com wrote:

 Ok, I have the following requirements to a soundcard:
  - PCI Express
  - 96kHz/24-bit
  - Optical SPDIF I/O
  - Works correctly in ALSA
 
 Do you know of any card that fulfils these requirements?  (Preferably
 less than EUR250.)

Sounds like the ASUS Xonar D2X could work for you. It doesn't have
hardware mixing though.

James

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Re: [Alsa-user] Strange alsa behaviour

[James Courtier-Dutton]

On 2 September 2010 11:12, Cyril Russo stage.nexvis...@laposte.net wrote:

  Hi,

  I've an issue with my new Creative Audigy sound card.
 I'm using a Debian Squeeze (with official 2.6.32-5-amd64 kernel) system.
 I've done this step to ensure I'm using the latest version:
 sudo module-assistant auto-install alsa (which installed the driver from
 alsa-driver 1.0.23's package)

 The sound card is correctly detected and it's working, but I've an
 issue, in that each channel appears as a different device.
 So in all the software using Alsa I have to select a device and it
 outputs on a single stereo channel for this particular device.

 For example, this command lists:
 # aplay -l
  List of PLAYBACK Hardware Devices 
 card 0: CA0106 [CA0106], device 0: ca0106 [CA0106]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
 card 0: CA0106 [CA0106], device 1: ca0106 [CA0106]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
 card 0: CA0106 [CA0106], device 2: ca0106 [CA0106]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
 card 0: CA0106 [CA0106], device 3: ca0106 [CA0106]
   Subdevices: 1/1
   Subdevice #0: subdevice #0


It is 4 stereo channels because that is what the hardware is.
The hardware has an option to output 8 channels in one stream, but the
xruns are atrocious in that mode as the hardware buffer used is too
small. (I think it has 1ms of buffer or something un-useably small
like that.
I therefore only present the 4 stereo channels option in the driver.

Kind Regards

James

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[Alsa-user] Softvol plugin - is there a way to have Volume Control AND Mute Switch?

[James Huk]

Hello everybody.

This is my first post in here - hopefully this is the right place to
ask this...

I would like to have a software volume control (using softvol plugin),
which can be muted, I know that, if I add resolution 2 at the end of
softvol configuration, I will get the mute/unmute switch - but this
will eliminate the possibility to control volume level other then to
turn it On or Off, what I would like, is to have both controls -
so that I could control volume level AND would be able to mute/unmute
it at demand (same way like the Master control works) - is there a
way to do this?

Thanks in advance for the info.

Best regards.

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Re: [Alsa-user] How to downmix 5.1 to stereo?

[James Courtier-Dutton]

On 4 July 2010 16:51, Manuel Reimer manuel.s...@nurfuerspam.de wrote:
 James Courtier-Dutton wrote:
 Why would you need to downmix 5.1 to 2.0 ?
 Most Linux applications do the downmix for you.
 I.e. You tell it how many speakers you have, and it outputs the sound to 
 them.
 e.g. The xine media player

 Gentoo uses Totem and I couldn't find the relevant setting for the audio
 channels.


In totem.

Edit-Preferences-Audio
Audio Output type: Set it to Stereo for 2.0

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Re: [Alsa-user] no sound, audigy 2zs

[James Courtier-Dutton]

On 11 June 2010 18:31, Kristoffer Gustafsson k...@dreamwld.com wrote:
 Hi.
 Now I've gotten the audigy 2zs as first soundcard.
 I edited the /etc/modprobe.d/asound/alsa-base.conf file.
 there i set option snd-hda-intel index=-2
 And alsactl init says that audigy is the card I'm using now, but no sound at
 all from the card.
 It works to play music, no errors when using mplayer, but no sound at all is
 heard.
 Is there a bug or so in the driver for debian squeeze, or have I missed
 something.
 I have set all controls in alsa mixer to 100%

If you can use the command line, try the following:

speaker-test -D plug:front:0 -c2 -twav

You should here a voice from the speakers.

You might have connected the speakers into the wrong socket. The
correct socket is the Green one.
One option is to set the speaker-test program running, and then try
the speakers in each socket until one works.

Kind Regards

James

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Re: [Alsa-user] ftp.alsa-project.org down?

[James Shatto]

It doesn't appear to be NAT.  At least not anything that I have
control over.  Same error(s) on the router box with or without
firewall.  FTP to my other ISP's base web space works fine.

$ curl ftp.alsa-project.org/pub/driver/alsa-driver-1.0.23.tar.bz2
curl: (56) FTP response reading failed

$ wget ftp.alsa-project.org/pub/driver/alsa-driver-1.0.23.tar.bz2
Error in server response, closing control connection.

$ ftp
open ftp.alsa-project.org
421 Service not available, remote server has closed connection

and so on and so on.

traceroute to alsa0.alsa-project.org (212.20.107.51), 30 hops max, 40
byte packets
 1  192.168.2.1 (192.168.2.1)  1.566 ms  1.975 ms  2.980 ms
...
 5  user45.embarqnow.net (64.45.249.45)  21.211 ms  22.044 ms  22.377 ms
 6  ge-6-14.car2.Houston1.Level3.net (4.78.10.17)  30.828 ms  18.662
ms  17.253 ms
 7  ae-2-5.bar2.Houston1.Level3.net (4.69.132.238)  19.686 ms  24.716
ms  25.090 ms
 8  ae-7-7.ebr1.Atlanta2.Level3.net (4.69.137.142)  49.183 ms  49.600
ms  48.246 ms
 9  ae-63-60.ebr3.Atlanta2.Level3.net (4.69.138.4)  47.535 ms  47.883
ms  47.829 ms
10  ae-2-2.ebr1.Washington1.Level3.net (4.69.132.86)  57.109 ms
55.129 ms  57.400 ms
11  ae-61-61.csw1.Washington1.Level3.net (4.69.134.130)  54.671 ms
ae-91-91.csw4.Washington1.Level3.net (4.69.134.142)  45.316 ms  44.817
ms
12  ae-82-82.ebr2.Washington1.Level3.net (4.69.134.153)  50.233 ms
49.554 ms ae-72-72.ebr2.Washington1.Level3.net (4.69.134.149)  50.113
ms
13  ae-44-44.ebr2.Frankfurt1.Level3.net (4.69.137.61)  139.208 ms
ae-43-43.ebr2.Frankfurt1.Level3.net (4.69.137.57)  134.281 ms
ae-44-44.ebr2.Frankfurt1.Level3.net (4.69.137.61)  166.472 ms
14  ae-5-5.car2.Prague1.Level3.net (4.69.135.50)  179.133 ms  179.015
ms  179.089 ms
15  ae-11-11.car1.Prague1.Level3.net (4.69.135.41)  175.002 ms
175.047 ms  175.025 ms
16  212.162.8.14 (212.162.8.14)  156.870 ms  181.953 ms  182.332 ms
17  perexsoft.customer.vol.cz (212.20.107.218)  164.298 ms  184.698 ms
 159.328 ms
18  * * *
19  * * *
20  * * *
21  * * *
22  * * *
23  * * *
24  * * *
25  * * *
26  * * *
27  * * *
28  * * *
29  * * *
30  * * *

- James


On 5/30/10, Jaroslav Kysela pe...@perex.cz wrote:
 On Sat, 29 May 2010, James Shatto wrote:

 My debian distro comes with a 2.6.26-2-686 kernel.  Which has version
 1.0.17 of alsa.  I was hoping to just install the 1.0.23 version from
 alsa-project.org.  But the links to download the sources don't appear
 to work.  Is the ftp site down?  Is there some other way to get these
 sources without extracting them from another more recent kernel?

 wget -c ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.23.tar.bz2

 --2010-05-29 16:09:25--
 ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.23.tar.bz2
   = `alsa-driver-1.0.23.tar.bz2'
 Resolving ftp.alsa-project.org... 212.20.107.51
 Connecting to ftp.alsa-project.org|212.20.107.51|:21... connected.
 Logging in as anonymous ...
 Error in server response, closing control connection.
 Retrying.

 The command works for me. It seems like a local issue in your network
 (perhaps a broken NAT gateway)?

   Jaroslav

 -
 Jaroslav Kysela pe...@perex.cz
 Linux Kernel Sound Maintainer
 ALSA Project, Red Hat, Inc.



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[Alsa-user] ftp.alsa-project.org down?

[James Shatto]

My debian distro comes with a 2.6.26-2-686 kernel.  Which has version
1.0.17 of alsa.  I was hoping to just install the 1.0.23 version from
alsa-project.org.  But the links to download the sources don't appear
to work.  Is the ftp site down?  Is there some other way to get these
sources without extracting them from another more recent kernel?

wget -c ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.23.tar.bz2

--2010-05-29 16:09:25--
ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.23.tar.bz2
   = `alsa-driver-1.0.23.tar.bz2'
Resolving ftp.alsa-project.org... 212.20.107.51
Connecting to ftp.alsa-project.org|212.20.107.51|:21... connected.
Logging in as anonymous ...
Error in server response, closing control connection.
Retrying.

I'm interested in doing this because jackd requires 1.0.18 or better
version(s) of alsa for alsa support (oss might actually work) if
compiled from sources.  And my current version of mplayer wont compile
with jack support against my current version of jackd.  aka dependency
hell in source mode.

Thanks,
- James

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Re: [Alsa-user] WAV offset with ALSA

[James Courtier-Dutton]

2010/1/20 Dean Montgomery dmo...@sd73.bc.ca:
 Why would the WAV be offset using ALSA but not OSS?

 See attached picture:
 * top = alsa
 * bottom = oss

 http://dean.sd73.bc.ca/mod/resource/view.php?id=21


It might be a bug in the driver. You do not give any information
regarding which sound card you have.
The audio on the ALSA seems to look to be much louder than the OSS one.
Do you have the volume controls turned up?

What are you measuring this on. There should be components in the
output that remove any offset.
This might instead be a mic problem.

Kind Regards

James

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Re: [Alsa-user] Microphone + AudioOut to HDMI

[James Shatto]

On Tue, 5 Jan 2010 01:07:21 +0300
An St vit@gmail.com wrote:

 Hello!
 
 Please help. I can't get working microphone at HDMI output.

HDMI audio normally has some sort of limit in place.  For my ATI HD4550 video 
card, the audio has to be transmitted in an AC3 codec(5.1 surround).  AKA 
compressed, it will not work with PCM audio which might be where your making a 
connection is giving you trouble.  That capability might depend on your 
graphics card, but that's the quirk of mine.  Assuming that your HDMI audio is 
provided via a graphics card.  

Maybe alsa can handle the AC3 conversion transparently / internally, or NOT.  
It's seems a bit destined for problems IMO, so I just avoid the issue with an 
RCA cable from a dedicated soundcard.  Fortunately my HDTV has a channel with 
HDMI/DVI input and RCA audio input so running that machine on a 42 display is 
possible.  

If my graphics card does handle PCM audio over HDMI, it's probably limited to 2 
channels, 48kHz, 16 bit, and all that jazz.  Maybe even 44.1kHz.  I really 
haven't checked the specs that recently on it.  But I only have one receiving 
device for HDMI audio (HDTV) so it's not a priority to explore for me.  Which 
is kind of ironic since the audio device registers and an hda-intel device.  
AKA high definition audio.  But the limits are listed in the manual.  Not that 
I've looked at it in the past year+.

HTH,
James

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Re: [Alsa-user] Redirecting the output audio to the microphone input

[James Shatto]

On Wed, 9 Dec 2009 09:23:44 -0200
Kazuo Teramoto kaz@gmail.com wrote:

 On Wed, Dec 9, 2009 at 5:16 AM, James Shatto shado...@earthlink.net wrote:
  You can set the record device to PCM (aumix term, never been able to find
  the equivalent alsamixer way).  Although you'll likely need to adjust your
  volume levels to get a good level (which might be below audible levels).
 
 I cant find the setting to set the record device to PCM can you give
 me a amixer command line for it?

Like I said aumix seems to be the only command line one I've found with that 
option/feature.  Not that I've tried to do it other ways.

$ aumix -q
$ aumix -v R
$ aumix -q

The options for -p and -w seem to set it to mic for microphone input.  In the 
gui mode it's the square with the green center square, click the one for master 
volume (first one) to set it to the pcm output as your default recording 
channel.  It should turn/be red in the gui, when it is.  There might be an 
amixer option for that.  Alsamixer seems to only let me toggle mic capture ON, 
but NOT off, which undoes the change aumix made.  I mainly use this feature to 
make a WAV of festival output for text to speech.

Seems like aumix says Capture to on and Mic to off (relative to capture), 
when aumix changes things.  With alsamixer I seem to only be able to set Mic to 
ON.  Bear in mind that this is completely dependent on your hardware supporting 
this feature.


 
  Roughly %50 volume on my laptop.  And it requires a soundcard that supports
  that.  Otherwise use a cable to connection line out to line in.  Which could
  be on that machine or another one.
 
 This is not a solution in my case, I don't have a line in jack connection.
 
A somewhat related idea is how I can read audio 
from a file and pipe it to microphone input.

  If you have a file, you don't really need to, outside of some sort of
  realtime performance setup with effects.  But if you have a file, just how
  real time is it?  Sox, ffmpeg, audacity, and a few other applications can
  convert file formats to other formats.  Or play them back, you don't really
  need to record them, if they're already in file format.  And if your
  soundcard isn't full duplex, you might already be getting some bleed through
  accidentally.
 
 I need what  I asked for, because I like to emulate a microphone. I
 like to play sounds in a program that only accept mic input, but cant
 take files an input e.g. Skype and other voice programs, with games
 (Counter Strike Source), etc. I not doing this to convert files (if Í
 needed to convert I had searched for a converting solution, I'm not
 that stupid =] )

Then don't call them files.  Use sources or other more appropriate terms.

 
 -- 
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 Marie Sklodowska Curie.

For the resulting changes after $ aumix -v R

$ amixer get Mic
Simple mixer control 'Mic',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch 
cswitch-exclusive
  Capture exclusive group: 0
  Playback channels: Mono
  Capture channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono: Playback 27 [87%] [6.00dB] [off]
  Front Left: Capture [off]
  Front Right: Capture [off]

$ amixer get Capture
Simple mixer control 'Capture',0
  Capabilities: cvolume cswitch cswitch-joined
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 15
  Front Left: Capture 8 [53%] [12.00dB] [on]
  Front Right: Capture 8 [53%] [12.00dB] [on]

I haven't been able to get this result by anything other than aumix.  And I'm 
not familiar with the amixer equivalent.  But here's what it's like BEFORE I 
change the capture device AWAY from Mic with aumix.

$ amixer get Mic
Simple mixer control 'Mic',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch 
cswitch-exclusive
  Capture exclusive group: 0
  Playback channels: Mono
  Capture channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono: Playback 27 [87%] [6.00dB] [off]
  Front Left: Capture [on]
  Front Right: Capture [on]

$ amixer get Capture
Simple mixer control 'Capture',0
  Capabilities: cvolume cswitch cswitch-joined
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 15
  Front Left: Capture 8 [53%] [12.00dB] [on]
  Front Right: Capture 8 [53%] [12.00dB] [on]

Your hardware may vary.  My hardware is an ATI IXP SB400 on my compaq presario 
laptop.  1002:4370

Other options might be to use jackd and qjackctl to make associations, or some 
form of pulse audio.  There's many means to an end.  Bear in mind that piping 
line out to mic in, WILL result in feedback if there's any sort of playthrough 
/ relation between the two channels.

- James

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Re: [Alsa-user] Redirecting the output audio to the microphone input

[James Shatto]

On Tue, 8 Dec 2009 21:26:14 -0200
Kazuo Teramoto kaz@gmail.com wrote:

 Hello.
 
 I like to redirect the sound I hear in the speakers to microphone, so
 it can be recorded with e.g. arecord.

You can set the record device to PCM (aumix term, never been able to find the 
equivalent alsamixer way).  Although you'll likely need to adjust your volume 
levels to get a good level (which might be below audible levels).  Roughly %50 
volume on my laptop.  And it requires a soundcard that supports that.  
Otherwise use a cable to connection line out to line in.  Which could be on 
that machine or another one.

 A somewhat related idea is how I can read audio from a file and pipe
 it to microphone input.

If you have a file, you don't really need to, outside of some sort of realtime 
performance setup with effects.  But if you have a file, just how real time is 
it?  Sox, ffmpeg, audacity, and a few other applications can convert file 
formats to other formats.  Or play them back, you don't really need to record 
them, if they're already in file format.  And if your soundcard isn't full 
duplex, you might already be getting some bleed through accidentally.

 I think alsa can do this with some sort of combination of plugin file,
 dsnoop and some asoundrc-fu but I cant get all the concepts to create
 a solution by myself.
 
 Someone can help me please?
 
 Thanks,
 Kazuo Teramoto

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Re: [Alsa-user] USB soundcard advice

[James Shatto]

On Mon, 2 Nov 2009 20:42:48 +0100
Y.A. Bolawy bol...@gmail.com wrote:

 Hi all,
 
 I'd like some advice on a USB soundcard. The reason for getting one is
 that I'd like to have good quality sound on all the computers I use or
 will use. The quality should be good enough to allow speech
 recognition. Of course, that is possible to some extend with any
 soundcard, but if the quality is low it has a big impact.
 
 Unfortunately, the better the quality of the cards, the less standard
 compliance they seem to be. At least that seems to be the underlying
 message of everything I've read so far.

USB Audio is a standard.  As long as the box says class compliant, it should 
work out of the box in linux.  Only one caveat though as it wont default to 
card 0 and be your default card.  Since you probably have a motherboard with 
onboard sound.  Configure accordingly.  I have a USB M-Audio Mobile Pre and it 
works fine.  Although web browsers don't seem to use it properly even though I 
have it configured to card 0.  I've never had a problem recording from it 
though.  Not really the best audio option, but loads better than most stock 
soundcards.

HTH,
James

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Re: [Alsa-user] Multiple cards, alphanumeric names?

[James Shatto]

On Tue, 03 Nov 2009 18:25:24 -0600
Jonathan E. Brickman j...@joshuacorps.org wrote:

 OK.  I now find myself happily educated in card names (HD2 in my 
 case), devices as being items on cards (HD2,0 et cetera), and 
 subdevices whose names appear to be used in rather different 
 locations.  My next question:  What if I had two cards of this type?  Do 
 I have to use numeric names, or is there an alphanumeric rule built in 
 somewhere which gives me HD2(0) or some such?
 
 J.E.B.

I think that you're getting grub device names confused with alsa names.  
Normally you can address them by hw:0 or hw:1 or hw:2.  Basically hw:0,1 for 
capture device and hw:0,0 for playback.  You can give more meaningful names in 
your .asoundrc configuration.  But generally NOT HD#, that's a grub thing for 
Hard Drive.  Although most alsa apps reference them by -c # where the # matches 
their designation in /proc/asound/cards.  Many apps that use alsa use something 
like -D hw:2 or -ao alsa:device=hw:2 and that is assuming that you don't want 
to just use the defaults.

HTH,
- James

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Re: [Alsa-user] Very frustrating ALSA MIDI issue

[James Gadsby]

These are my soundcards:

 aplay -l
 List of PLAYBACK Hardware Devices 
card 0: NVidia [HDA NVidia], device 0: Cirrus Analog [Cirrus Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: NVidia [HDA NVidia], device 1: Cirrus Digital [Cirrus Digital]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

Additionally, doing a modprobe of snd-usb-audio returns:

FATAL: Module snd_usb_audio not found.




 On Fri, Aug 28, 2009 at 9:44 AM, Clemens Ladisch 
 cladi...@googlemail.comwrote:

 James Gadsby wrote:
  Simply put, MIDI devices are not detected in the audio programs I use,
  despite being detected at the USB level (as seen via lsusb). I know
 these
  devices work with other PCs,

 Linux?

  Here is my lsmod output

 There is no snd-usb-audio.  It should be loaded automatically for any
 supported device.  Which device are you trying to use?


 Best regards,
 Clemens



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Re: [Alsa-user] audio loopback in linux

[James Shatto]

On Fri, 28 Aug 2009 10:38:16 +0200 (CEST)
Julien Claassen jul...@c-lab.de wrote:

 Hi!
I'm not sure, if alsa does it, still. But you can do it with jackd (Jack 
 Audio Connection Kit). It's a low latecny audio server and a lot of Linux 
 Audio software support it. You can find packages in your distro.
Then you simply do:
 jack_connect system:capture_1 system:playback_1
 jack_connect system:capture_2 system:playback_2
Or install some GUI connection tool, e.g. qjackctl.
Hope that helps
  Julien
 

jack + qjackctl does this (or the CLI alternative).  If your hardware supports 
it, it'll be listed in alsamixer.  My delta 44 is anyway.  On that card it 
shows as  H/W   H/W 1   H/W 2   H/W 3  and you just change it from PCM 
Out to HW In 0 or whatever source you want.  Bear in mind that if input is a 
mic and the speaker is loud enough you'll get feedback.  And if you're doing it 
for some sort of TV Capture card that audio has less latency than video, so 
you'll hear them talk before their lips move (just slightly) which can/will 
drive you nuts.

HTH,
James

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[Alsa-user] Very frustrating ALSA MIDI issue

[James Gadsby]

Hello,

I've been spending quite a long time attempting to get MIDI support for my
music applications with ALSA. I need to say that all these MIDI devices
function fine and as they should do on other hardware. The hardware I'm
trying to work with is a mid-2009 MacBook Pro, running on Ubuntu 9.04. Now,
being a MacBook, it requires some small fixes and tweaks to fully work under
Linux, the most important bit for me being sound support. In fact, sound has
only worked at all on the latest MacBooks since early August, by compiling,
building and installing alsa-utils from an unstable (now stable) snapshot.
Luckily, every aspect of sound through ALSA now works wonderfully, and it's
only a matter of solving the very last issue, but a big one (as I use Linux
as an amateur music production system).

Simply put, MIDI devices are not detected in the audio programs I use,
despite being detected at the USB level (as seen via lsusb). I know these
devices work with other PCs, but not with my MacBook.

When I compiled ALSA, the ./configure was as follows:

./configure --enable-dynamic-minors   --with-cards=hda-intel
--with-sequencer=yes

followed by the usual make and sudo make install. As you can see, alsa is
compiled with with-sequencer and should surely have MIDI support?

I started trying to add midi-related modules with modprobe, such as
snd_seq_device. Here is my lsmod output as I type, and yet with no support
for MIDI:

lsmod
Module  Size  Used by
nls_iso8859_1  12032  1
isofs  39844  1
nls_cp437  13696  1
vfat   18816  1
udf87716  0
fat58272  1 vfat
crc_itu_t  10112  1 udf
binfmt_misc16776  1
ppdev  15620  0
bridge 56212  0
stp10500  1 bridge
bnep   20224  2
uinput 15616  2
video  25360  0
output 11008  1 video
joydev 18368  0
applesmc   29616  0
led_class  12036  1 applesmc
input_polldev  11912  1 applesmc
coretemp   13952  0
lp 17156  0
parport42220  2 ppdev,lp
snd_hda_codec_cirrus20224  1
snd_hda_intel  36576  3
snd_hda_codec  89216  2 snd_hda_codec_cirrus,snd_hda_intel
snd_hwdep  15364  1 snd_hda_codec
snd_pcm_oss46208  0
snd_mixer_oss  22912  1 snd_pcm_oss
snd_pcm84100  3 snd_hda_intel,snd_hda_codec,snd_pcm_oss
snd_seq_oss36352  0
snd_seq_midi_event 15232  1 snd_seq_oss
snd_seq57648  4 snd_seq_oss,snd_seq_midi_event
uvcvideo   63368  0
snd_timer  29192  2 snd_pcm,snd_seq
snd_seq_device 15372  2 snd_seq_oss,snd_seq
ieee80211_crypt_tkip16896  0
compat_ioctl32  9344  1 uvcvideo
pcspkr 10496  0
appleir12416  0
snd67748  17
snd_hda_codec_cirrus,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_seq,snd_timer,snd_seq_device
soundcore  15200  1 snd
snd_page_alloc 17032  2 snd_hda_intel,snd_pcm
nvidia   8950276  38
shpchp 40212  0
wl   1281364  0
btusb  19608  2
usb_storage99648  1
videodev   41600  1 uvcvideo
v4l1_compat21764  2 uvcvideo,videodev
hid_apple  14336  0
bcm597416512  0
ieee80211_crypt13444  2 ieee80211_crypt_tkip,wl
agpgart42696  1 nvidia
mbp_nvidia_bl  12176  0
usbhid 42336  0
ohci1394   38576  0
ieee1394   94660  1 ohci1394
forcedeth  61712  0
fbcon  46112  0
tileblit   10752  1 fbcon
font   16384  1 fbcon
bitblit13824  1 fbcon
softcursor  9984  1 bitblit

I have no idea where to go from here, as everything should work. But it
doesn't. I use LMMS for music production.

Thanks for any help relating to this, it's been a very frustrating issue.
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Re: [Alsa-user] hda-intel ad1998b troubles (front channels only)

[James Le Cuirot]

On Thu, 26 Feb 2009 14:26:33 + (UTC)
shy_reclusive_alsa_user ymail_u...@ymail.com wrote:

 speaker-test --channels 8 (blah, blah...)

I've only posted to this list once but I have a suggestion for you.
What is this blah blah stuff? ;) You need to specify the device name
when doing surround. For 8 channels, I guess you want -Dsurround71.

Regards,
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[Alsa-user] Prevent boom/pop when loading emu10k1?

[James Le Cuirot]

Hi guys,

I have an Audigy 1 (Platinum) and I was wondering if there was any way
to prevent the annoying boom/pop sound when loading the module. I
notice that the default mixer levels aren't all muted - Master is but
some of the others are set to 100%. Maybe this is the cause? I don't
think this happened with Windows but I don't have it installed anymore
so I couldn't swear to it.

Cheers,
James

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Re: [Alsa-user] broken digital sound again

[James]

James wrote:
 Clemens Ladisch wrote:
   
 James wrote:
   
 
 $ speaker-test -D iec958 -c 6
 ...
 Channels count (6) not available for playbacks: Invalid argument
 
   
 S/PDIF does not support uncompressed surround sound; you have to play
 stereo data (use -c 2) or AC-3/DTS-compressed data.


 Best regards,
 Clemens
   
 
 Speaker-test used to output to all my speakers.
 -c 2 DOES output to my left and right front speakers but I can't get
 anything out of the others in my 5.1

 $ aplay -D iec958 -c 2 /storage/music/goodbye3.wav
 Playing WAVE '/storage/music/goodbye3.wav' : Signed 16 bit Little
 Endian, Rate 22050 Hz, Mono
 aplay: set_params:966: Channels count non available

Grrr, Amarok just randomly started working.
Still no KDE system sounds but I care more about digital Amarok.

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Re: [Alsa-user] broken digital sound again

[James]

Clemens Ladisch wrote:
 James wrote:
   
 $ speaker-test -D iec958 -c 6
 ...
 Channels count (6) not available for playbacks: Invalid argument
 

 S/PDIF does not support uncompressed surround sound; you have to play
 stereo data (use -c 2) or AC-3/DTS-compressed data.


 Best regards,
 Clemens
   
Speaker-test used to output to all my speakers.
-c 2 DOES output to my left and right front speakers but I can't get
anything out of the others in my 5.1

$ aplay -D iec958 -c 2 /storage/music/goodbye3.wav
Playing WAVE '/storage/music/goodbye3.wav' : Signed 16 bit Little
Endian, Rate 22050 Hz, Mono
aplay: set_params:966: Channels count non available

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[Alsa-user] broken digital sound again

[James]

My digital sound broke again :-(
I like Gentoo because it updates alot but I hate it because it breaks my
digital sound alot. :-(
I am using the 2.6.28.2 kernel and alsa 1.0.19

$ aplay -l
 List of PLAYBACK Hardware Devices 
card 0: NVidia [HDA NVidia], device 0: ALC888 Analog [ALC888 Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: NVidia [HDA NVidia], device 1: ALC888 Digital [ALC888 Digital]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

$ aplay -L
default:CARD=NVidia
HDA NVidia, ALC888 Analog
Default Audio Device
front:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
Front speakers
surround40:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Digital
IEC958 (S/PDIF) Digital Audio Output
null
Discard all samples (playback) or generate zero samples (capture)

$ speaker-test -D iec958 -c 6

speaker-test 1.0.19

Playback device is iec958
Stream parameters are 48000Hz, S16_LE, 6 channels
Using 16 octaves of pink noise
Channels count (6) not available for playbacks: Invalid argument
Setting of hwparams failed: Invalid argument


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Re: [Alsa-user] emu 0404 usb problems

[James Courtier-Dutton]

Clemens Ladisch wrote:
 Aleksander Kamenik wrote:
 Since the upgrade from Fedora 9 to Fedora 10, the EMU 0404 USB card does 
 not work any more.
 
 Only in PulseAudio?
 
 The alsa version changed form .17 to 19.
 
 There have been no relevant changes in either the ALSA driver or
 alsa-lib.
 
 The gist of the problem ass told by Lennart:
 quote
 Ah, that's interesting. Apparently your sound card does not support
 integral number of periods.
 
 It should; the USB driver accepts nearly anything.
 
 Please run PA with the environment variable LIBASOUND_DEBUG set to 1.
 Alsa-lib will then output some information about the error on stderr.
 
 Also, under which project should this be filed?
 
 alsa-lib
 
 
 Best regards,
 Clemens
 


I do not think that this is a bug in alsa generally.
The driver for the 0404 usb is not finished.
I think that this is related to sample rate differences.
The driver lets one send samples at any rate 44.1, 48 etc.
The 0404 USB internally only works at a fixed but configurable rate via
a vendor specific processing unit control.
I.e. if the 0404 USB is internally set to 44.1 and one sends 48 to it
things go wrong.

Kind Regards

James

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[Alsa-user] default to digital

[James]

how to I set alsa to use -Dplug:spdif as the default?

$ aplay -L
default:CARD=NVidia
HDA NVidia, ALC888 Analog
Default Audio Device
front:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
Front speakers
surround40:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Digital
IEC958 (S/PDIF) Digital Audio Output
null
Discard all samples (playback) or generate zero samples (capture)

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Re: [Alsa-user] maybe this is why I can't get digital output

[James]

David McCloskey wrote:
 That's just a warning.  Usually it will be working after that warning.

 ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.pdf

 Dave

 On Sun, Jan 11, 2009 at 11:48 PM, James bjloc...@lockie.ca wrote:
   
 hda_codec: Unknown model for ALC883, trying auto-probe from BIOS...
 

Maybe I need some sort of dmix setup so I can play system sounds through
SPDIF.

I recompiled the kernel (2.6.28) with alsa debugging.

Cannot find slave Headphone Playback Volume, skipped
Cannot find slave Speaker Playback Volume, skipped
Cannot find slave Mono Playback Volume, skipped
Cannot find slave Line-Out Playback Volume, skipped
Cannot find slave Speaker Playback Switch, skipped
Cannot find slave Mono Playback Switch, skipped
ALSA device list:
  #0: HDA NVidia at 0xf9df8000 irq 20

azx_pcm_prepare: bufsize=0x1, format=0x31
hda_codec_cleanup_stream: NID=0x6
hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x31
hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x31
hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x31
hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x31
azx_pcm_prepare: bufsize=0x1, format=0x21
hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x21
hda_codec_cleanup_stream: NID=0x8
hda_codec_cleanup_stream: NID=0x2
hda_codec_cleanup_stream: NID=0x3
hda_codec_cleanup_stream: NID=0x4
hda_codec_cleanup_stream: NID=0x5
azx_pcm_prepare: bufsize=0x1, format=0x31
hda_codec_cleanup_stream: NID=0x6
hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x31
hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x31
hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x31
hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x31
azx_pcm_prepare: bufsize=0x1, format=0x21
hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x21
hda_codec_cleanup_stream: NID=0x8
hda_codec_cleanup_stream: NID=0x2
hda_codec_cleanup_stream: NID=0x3
hda_codec_cleanup_stream: NID=0x4
hda_codec_cleanup_stream: NID=0x5
azx_pcm_prepare: bufsize=0x1, format=0x31
hda_codec_cleanup_stream: NID=0x6
hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x31
hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x31
hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x31
hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x31
azx_pcm_prepare: bufsize=0x1, format=0x21
hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x21



It's annoying everything dmesg shows is all interleaved.


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[Alsa-user] maybe this is why I can't get digital output

[James]

hda_codec: Unknown model for ALC883, trying auto-probe from BIOS...


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[Alsa-user] no digital aplay -L but digital aplay -l

[James]

I can't digital output for analog works fine.
Should aplay -L show a digital output?

$ aplay -L
default:CARD=NVidia
  HDA NVidia, ALC888 Analog
  Default Audio Device
front:CARD=NVidia,DEV=0
  HDA NVidia, ALC888 Analog
  Front speakers
surround40:CARD=NVidia,DEV=0
  HDA NVidia, ALC888 Analog
  4.0 Surround output to Front and Rear speakers
surround41:CARD=NVidia,DEV=0
  HDA NVidia, ALC888 Analog
  4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=NVidia,DEV=0
  HDA NVidia, ALC888 Analog
  5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=NVidia,DEV=0
  HDA NVidia, ALC888 Analog
  5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=NVidia,DEV=0
  HDA NVidia, ALC888 Analog
  7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
null
  Discard all samples (playback) or generate zero samples (capture)

$ aplay -l
 List of PLAYBACK Hardware Devices 
card 0: NVidia [HDA NVidia], device 0: ALC888 Analog [ALC888 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: NVidia [HDA NVidia], device 1: ALC888 Digital [ALC888 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0


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Re: [Alsa-user] digital output with hda_intel

[James]

On 12/28/08 23:04, James wrote:
 Something happened to my ALSA.
 It plays sound through the regular speakers but not through the digital
 out of the motherboard.
 I can play DVDs through the digital out so I know the output should work.
 It used to work.

 ┌──[AlsaMixer v1.0.18 (Press Escape to
 quit)]──┐
 │ Card: HDA NVidia │
 │ Chip: Realtek ALC888 │
 │ View: [Playback] Capture All │
 │ Item: IEC958 Default PCM

 I *think* I need to change the IEC958 Default to digital.

$ aplay -L
default:CARD=NVidia
 HDA NVidia, ALC888 Analog
 Default Audio Device
front:CARD=NVidia,DEV=0
 HDA NVidia, ALC888 Analog
 Front speakers
surround40:CARD=NVidia,DEV=0
 HDA NVidia, ALC888 Analog
 4.0 Surround output to Front and Rear speakers
surround41:CARD=NVidia,DEV=0
 HDA NVidia, ALC888 Analog
 4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=NVidia,DEV=0
 HDA NVidia, ALC888 Analog
 5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=NVidia,DEV=0
 HDA NVidia, ALC888 Analog
 5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=NVidia,DEV=0
 HDA NVidia, ALC888 Analog
 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
null
 Discard all samples (playback) or generate zero samples (capture)

$ aplay -l
 List of PLAYBACK Hardware Devices 
card 0: NVidia [HDA NVidia], device 0: ALC888 Analog [ALC888 Analog]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
card 0: NVidia [HDA NVidia], device 1: ALC888 Digital [ALC888 Digital]
   Subdevices: 1/1
   Subdevice #0: subdevice #0


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[Alsa-user] digital output with hda_intel

[James]

Something happened to my ALSA.
It plays sound through the regular speakers but not through the digital 
out of the motherboard.
I can play DVDs through the digital out so I know the output should work.
It used to work.

┌──[AlsaMixer v1.0.18 (Press Escape to 
quit)]──┐
│ Card: HDA NVidia │
│ Chip: Realtek ALC888 │
│ View: [Playback] Capture All │
│ Item: IEC958 Default PCM

I *think* I need to change the IEC958 Default to digital.

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Re: [Alsa-user] Digital bit perfect ouptut with ALSA

[James Courtier-Dutton]

Paulo Moura Guedes wrote:
 For my case where I connect to the Benchmark DAC1 via USB (which supports 
 24bit 96khz), does my sound card have any influence in the process?
 
No.

If the sound card does 24bit 96khz, and the original sound file is 24bit
96khz, ALSA will not touch/modify the samples. If the original sound
file is 24bit 44.1khz, and the usb sound device can only do 24bit 96khz,
ALSA will have to resample them to get 44.1khz into the 96khz pipe.

Kind Regards

James

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Re: [Alsa-user] Digital bit perfect ouptut with ALSA

[James Courtier-Dutton]

Paulo Moura Guedes wrote:
 I'm trying to get bit perfect output out of my linux box, but I can't find 
 much 
 info on the web. I'm using ALSA.
 
 Some questions:
 
 - does Linux/ALSA features dynamic sample rates?
 - is it possible to set the bit-depth? (in my case to 24 bit)
 - what other variables do i have to consider in order to get bit transparent 
 ouput, 
 i.e., no resampling at all?
 - is it possible to check if ouput is bit-perfect?
 
 Thanks in advance,
 Paulo
 

All the answers are yes.

If the hardware supports a particular rate, alsa will use it, on a per
sound file basis.
I.e. first sound file might be 24bit 48khz, alsa will output to the
hardware using this.
If the second sound file is 24bit 44.1khz, alsa will output to the
hardware using that.
If the hardware does not support a method to get bit perfect output to
the hardware, i.e hardware does 48khz only, but sound file is 44.1khz,
alsa will use a low quality resample method to at least get some sound.



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Re: [Alsa-user] Audigy2 recording level much too low - an (one and half) year later

[James Courtier-Dutton]

Zbigniew Baniewski wrote:
 Hallo,
 
 Over a year ago I've reported the problem with very low microphone input
 sensitivity, while recording using Audigy2, what is making this card unusable
 for VoIP.
 
 At that time, I've received a tip (thanks again), to fix it by introducing
 a line:
 
 snd_ac97_write_cache(emu-ac97, AC97_REC_GAIN, 0x0f0f);
 
 ...directly after a line:
 
 snd_ac97_write_cache(emu-ac97, AC97_REC_SEL, 0x);
 
 ...in sound/pci/emu10k1/emumixer.c
 
 Yes, this works OK. So why not introduce this line for steady, in every ALSA
 release? The problem is, that while there are binaries for kernel available,
 one has to patch, then compile every new kernel version again and again.
 
 Couldn't be possible to insert this line into emumixer.c for steady?
 I suppose, that every Audigy2 user will be grateful for making this. Even,
 if it can be treated as temporary hack, there could be placed a remark
 in the code, like f.e.: temporary fix, will be changed in the future. I'm
 patching those kernels over a year - and this temporary fix doesn't seem
 to spoil anything.
 
 Keeping the broken micro-input handling - and not doing with this anything
 since years - doesn't seem to have much sense. So why not fix it at least
 such way, when it _just works_, and it's tested (at least by me)?

It is a levels problem.
Mic captures at 24bits.
The DSP handles 24bits fine.
When the sound is passed from the DSP to the CPU only the top 16bits of
the 24bit value are passed. This accounts for the low capture levels.
The snd_ac97_write_cache(emu-ac97, AC97_REC_GAIN, 0x0f0f); reduces
the analogue capture headroom so is not ideal.
I think a better fix would be to add DSP code to adjust the 24bit value
into 16bits.
The current 24bit to 16bit conversion introduces about 48 dB loss.




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[Alsa-user] Bluetooth headset woes

[James Dominy]

Hi all,

I have successfully managed to output sound to my bluetooth headset using
mplayer and aplay, hooray! Two things elude me still though...

First, I cannot get audio capture to work via the headset, and second, I don't
know how to tell various applications like skype/exaile/audacious etc... to use
the headset. I can do it with mplayer, via the command line, but the device
doesn't show up as an option in the gui version.

So I suppose a directed set of questions is in order:

1) What do I have to do to get audio capture working. Is there something special
I need to add to my .asoundrc in addition to the following

pcm.bluetooth {
type bluetooth
device 00:18:E4:19:35:11
}


2) I'm guessing the way to go to get exaile, audacious etc... to 'detect' the
bluetooth headset as a device would be to create a virtual device in my
.asoundrc which wraps the bluetooth device, but I have no idea how to do that.
Any suggestions?

Thanks,
James

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Re: [Alsa-user] Emu 1212M on Debian Lenny

[James Courtier-Dutton]

Dominique Pautrel wrote:
 I'm nearly sure (but need to be prouved) that 1212M and 1820M use the
 same base : E-MU 1010 PCI card, with I/O ADAT and I/O SPDIF, so 10
 inputs and 10 outputs, plus firewire port and a plug who look like an
 RJ45 port.
 
 The 1212M has a daughter board, E-MU 0202 I/O daughter card : 2 analog
 inputs + 2 analog outputs.
 
 The 1820M has an additional Rack, but I've never seen it so I can't say
 nothing about it.
 
 On this page http://www.emu.com/products/welcome.asp?category=505 you
 see the 1010 PCI card on 3 products.
 
 What I'm sure is that these products are from the same family.
 
 Perhaps if you test it you could tell more ... I'm sorry I can't say
 more about it
 
 Regards
 Dom.
 
 
 Le dimanche 26 octobre 2008 à 17:48 +0100, Bo Forslund a écrit :
 Is 1212M the same as the 1820M.

 I just wonder. It might be useful for me to follow this thread.

 Regards
 Bo

 

Dom is correct, the base PCI card is what I call the 1010.
The 1212 adds another 0202 card that goes in another PCI slot for analog
in/outs.
The 1820 adds an external audio dock that gives more analog in/outs.

The extra bit of information is that their are a number of different
versions of the 1010 and audio dock, each needing their own unique firmware.

James

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Re: [Alsa-user] M-Audio Delta 66 Not working and cannot load modules after compile

[James Shatto]

 (/lib/modules/2.6.24-19-generic/ubuntu/sound/alsa-driver/acore/seq/snd-seq-device.ko):
  

there's your problem

Alsa from source will likely install to:
/lib/modules/`uname -r`/kernel/sound/

Which means you likely have two versions:
/lib/modules/`uname -r`/ubuntu/sound/

find /lib/modules/`uname -r`/ -iname '*snd-*.*o'

depmod -ae will pick up both version.  You probably need to remove one version 
to get it to work.  Then rerun depmod.

I'm not sure if soundcore.ko is alsa, or the kernel.  In days of old it was 
part of OSS, or so I thought.  It might just be easier to do a custom kernel.  
With alsa compiled over it.  That way there's not multiple versions / 
locations.  As the path assumptions would match.

HTH,
James

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Re: [Alsa-user] M-Audio Delta 66 Not working and cannot load modules after compile

[James Shatto]

On Mon, 29 Sep 2008 23:22:27 -0500
John Beavers [EMAIL PROTECTED] wrote:

 Hello all,
 
 I have a  few problems.  My main problem is that I have installed an 
 M-audio Delta 66, and it will output no sound, and does not recognize 
 input from sound sources, either.  But before we get to that, I have a 
 more pressing issue.  I tried compiling the latest build of Alsa, and 
 when I get to the step of inserting the driver, it gives me all sorts of 
 errors:
 
 sudo modprobe snd-ice1712
 FATAL: Error inserting snd 
 (/lib/modules/2.6.24-19-rt/ubuntu/sound/alsa-driver/acore/snd.ko): 
 Unknown symbol in module, or unknown parameter (see dmesg)
 WARNING: Error running install command for snd
 WARNING: Error inserting snd_seq_device 
 (/lib/modules/2.6.24-19-rt/ubuntu/sound/alsa-driver/acore/seq/snd-seq-device.ko):
  
 Unknown symbol in module, or unknown parameter (see dmesg)

I have a Delta 44 and it works fine.  The inputs are picky in that they need to 
be fed with a line level source.  Which for me means using a microphone preamp. 
 Even on some sources that may not need one in other circumstances / other 
cards.

That being said, the unknown symbol is common.  If insmod-ing without deps, you 
can only do this in a specific order.  If you boot using lilo and didn't rerun 
lilo to install the newer kernel, then it may be having version conflicts.  If 
you didn't run depmod -ae after installing the newer alsa you might also have 
trouble.  

snd-ice1712 should be the right module(s).  For me I modify the 
/etc/modules.conf configuration or modprobe.d / modutil.d modern equivalents.  
While disabling distro supplied defaults.  In my case it looks something like 
this.

# /etc/modprobe.d/alsa_custom

alias   char-major-116  snd
alias   char-major-14   soundcore

options snd major=116   cards_limit=1
options snd-ice1712 index=0

alias   snd-card-0  snd-ice1712
alias   sound-slot-0snd-card-0
alias   sound-service-0-0   snd-mixer-oss
alias   sound-service-0-1   snd-seq-oss
alias   sound-service-0-3   snd-pcm-oss
alias   sound-service-0-8   snd-seq-oss
alias   sound-service-0-12  snd-pcm-oss

# END

With this configuration, anytime you try to use the sound device the modules 
are automatically loaded.  Also when you modprobe snd-ice1712, it picks up the 
dependants in whatever order they were supposed to be used in.  

Reindexing it to 0 makes it the default soundcard.  There's other ways to do 
that, but this simplifies things for OSS type apps.  Like festival / mozilla / 
.

There are other issues if you're not using udev and/or didn't run the 
snddevices script to create the /dev devices (not to be run if you ARE using 
udev).  But that doesn't appear to be your issue.  And other ways to implement 
the above custom configuration with alsaconf and other utilities.  I just never 
got them to work for me back in the day, and never adapted to letting current 
tools try to do it for me.

- James

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Re: [Alsa-user] ALSA vs M-Audio fast track ULTRA

[James Shatto]

 I need to make multitrack recordings; I' m looking for a sound card 
 usb2  model
 of  at least 4/6/8 balanced inputs, XLR with phantom power to 48V and 
 audio resolution 24-bit/96kHz
 and with many analog audio outputs maybe XLR balanced, SPDIF in / out 
 and MIDI in / out / trough.

For those specs you need a firewire device.  USB has limited bandwidth.  The 
best I've come across that work, are 2x 16 bit 48kHz input with simultaneous 2x 
16 bit 48kHz output.  The best I've seen is 2x 24 bit 96kHz, and the reviews on 
them are not great.  Buggy, not full duplex at that rate, and other driver-ish 
issues.  Even in windows.  The USB bus is very limited and at a very minimum 
has latency issues if you want to multitrack.  Even at 16 bit 48 kHz.

If PCI is a possibility, then an Echo Layla 3G might be to your liking.  But I 
don't know about it's linux support status.  I just don't know if you're gonna 
find a device like that, that works in either windows or linux.  Unless it's a 
firewire device like a Presonus Firepod / Firebox / FP10 / Whatever the 
marketing name of the year is.  Or a PCI device.  Go PCI or Firewire, you'll 
have many more options, and not as many headaches.

HTH

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Re: [Alsa-user] ALSA vs M-Audio fast track ULTRA

[James Shatto]

 Hello everyone! I now resigned to not being able to use
 the m-audio fast track ULTRA usb soundcard with my LINUX-DAWs,
 someone can recommend another card usb I can afford
 multitrack audio recordings of quality, which is working with Linux?

As said before, my M-Audio Mobile Pre seems class compliant.  And otherwise 
works.  My Delta 44 (pci) works too.  The Delta worked for several months 
before they finally came up with Windows Vista drivers.  But it doesn't sound 
like you want another M-Audio.  So probably the Lexicon Omega / Alpha type 
cards might work for you.  What type of inputs are you needing?  And how many 
inputs?  TRS / XLR / 3.5mm stereo?  There's several out there depending on your 
needs.  What's your budget?  

If you need a lot more inputs at higher rates, then firewire might be better 
suited.  

freebob.sf.net

HTH,
- James

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