[Alsa-user] ice1712 error
Hello, I see my ice1712-based card giving me: ICE1712: probe of :01:08.0 failed with error -12 This is on kernel 2.6.27.8. What does this mean? lspci does show the card and it used to work OK. I didn't change the config or anything. Please let me know. Thanks, Udo -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ice1712 error
Udo van den Heuvel wrote: > I see my ice1712-based card giving me: > > ICE1712: probe of :01:08.0 failed with error -12 > > This is on kernel 2.6.27.8. > What does this mean? > lspci does show the card and it used to work OK. I didn't change the > config or anything. I am on x86_64 with Fedora 10, upgraded from 9. I build my own kernels. The card used to work OK. Any ideas? -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ice1712 error
Dominique Michel wrote: >> The card used to work OK. >> >> Any ideas? >> > > You can take a look at the alsa changelogs for your chip and compare with the > changelog for the version of alsa into your kernel (1.0.17?). Maybe that > another > alsa version (last 1.0.18a) will fix it : > http://www.alsa-project.org/main/index.php/Main_Page_News Thanks for the tip. No solution yet though. I found out the card is sometimes detected OK and sometimes isn't. Cold boot/warm boot doesn't really matter. I tried kernel 2.6.27.9 and older. Card is is fine when detected. Behaviour doesn't change when I try a different PCI slot. When the card fails: # dmesg|grep 01:09 PCI: :01:09.0 reg 10 io port: [a000, a01f] PCI: :01:09.0 reg 14 io port: [a400, a40f] PCI: :01:09.0 reg 18 io port: [a800, a80f] PCI: :01:09.0 reg 1c io port: [ac00, ac3f] pci :01:09.0: supports D2 ICE1712: probe of :01:09.0 failed with error -12 When the card is OK: PCI: :01:09.0 reg 10 io port: [a000, a01f] PCI: :01:09.0 reg 14 io port: [a400, a40f] PCI: :01:09.0 reg 18 io port: [a800, a80f] PCI: :01:09.0 reg 1c io port: [ac00, ac3f] pci :01:09.0: supports D2 ICE1712 :01:09.0: PCI INT A -> Link[APC4] -> GSI 19 (level, low) -> IRQ 19 I booted into single user mode each time and checked /proc/asound/cards after each boot and then looked at dmesg. Does this mean the card is toast? Or what should I try next? Kind regards, Udo -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] alsa on h2210 issue? (mono audio gives error)
Hello,
I am trying to make flite work on my HP Ipaq H2210.
It may be an Alsa issue, I don't know.
The short version of my troubles is at
http://wiki.navit-project.org/index.php/Navit_on_%C3%85ngstr%C3%B6m#Alsa_and_flite
.
Basically it is this:
r...@h2200:~$ cat /proc/asound/cards
0 [h2200 Audio]: h2200 Audio - h2200 Audio
iPAQ h2200 Audio [codec Philips UDA1380]
r...@h2200:~$ aplay notify.wav
Playing WAVE 'notify.wav' : Signed 16 bit Little Endian, Rate 22050 Hz,
Stereo
So it works fine.
r...@h2200:~$ cat /etc/asound.conf|grep -v ^#
pcm.dsp0 {
type plug
slave.pcm "dmix"
}
ctl.mixer0 {
type hw
card 0
}
pcm.10to20 {
type route
slave.pcm hw:0
slave.channels 2
ttable.0.0 1
ttable.0.1 1
}
That's the current config.
Then I start flite:
r...@h2200:~$ flite -t Test
ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find
an usable access for '(null)'
audio_open_alsa: failed to set number of channels to 1. Invalid argument.
When I use the 10to20 mono to stereo mapping I get:
r...@h2200:~$ aplay -D10to20 notify.wav
Playing WAVE 'notify.wav' : Signed 16 bit Little Endian, Rate 22050 Hz,
Stereo
aplay: set_params:879: Broken configuration for this PCM: no
configurations available
With flite I get a similar error when I want to make stereo from the 1
channel that flite wants.
How can I solve this? Obviously I want flite to work.
Tips are very welcome!
Kind regards,
Udo
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Re: [Alsa-user] ESI Juli@ - ice1724 fails to load
Mark Trompell wrote: > but snd-ice1724 fails like that: > ALSA sound/core/init.c:183: cannot find the slot for index 0 (range > 0-1), error: -16 > ICE1724: probe of :03:02.0 failed with error -12 I had similar. I was fixed with some config option. It has to do with the ordering of teh card sand the (other) order in which they are detected. With that option I mean one can reserver a slot for a driver. -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ESI Juli@ - ice1724 fails to load
Mark Trompell wrote: > but snd-ice1724 fails like that: > ALSA sound/core/init.c:183: cannot find the slot for index 0 (range > 0-1), error: -16 > ICE1724: probe of :03:02.0 failed with error -12 I had similar. I was fixed with some config option. It has to do with the ordering of teh card sand the (other) order in which they are detected. With that option I mean one can reserver a slot for a driver. -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ESI Juli@ - ice1724 fails to load
Mark Trompell wrote: > 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia > 01:00.1 Audio device: ATI Technologies Inc Radeon HD 3870 Audio device > 03:02.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 > [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01) > > but snd-ice1724 fails like that: > ALSA sound/core/init.c:183: cannot find the slot for index 0 (range > 0-1), error: -16 > ICE1724: probe of :03:02.0 failed with error -12 Takashi Iwai explained me: It means that the given slot is already occupied, or so. Unlikely a driver error but a user-setup problem. Check index options of all sound cards. For example, try to remove index options once, and let the system load all devices. In general, using slots option of snd module is recommended. Read ALSA-Configuration.txt for details. This worked for me. (one ice1712 and two snd-hda_intel cards) Udo -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] alsa on h2210 issue: (mono audio gives error)
Hello,
I am trying to make flite work on my HP Ipaq H2210 which is running
Angstrom Linux.
It may be an Alsa issue, I don't know. I posted this about two weeks ago
but got no response. Should I try another alsa list?
The short version of my troubles is at
http://wiki.navit-project.org/index.php/Navit_on_%C3%85ngstr%C3%B6m#Alsa_and_flite
.
Basically it is this:
r...@h2200:~$ cat /proc/asound/cards
0 [h2200 Audio]: h2200 Audio - h2200 Audio
iPAQ h2200 Audio [codec Philips UDA1380]
r...@h2200:~$ aplay notify.wav
Playing WAVE 'notify.wav' : Signed 16 bit Little Endian, Rate 22050 Hz,
Stereo
So it works fine, but not for mono.
r...@h2200:~$ cat /etc/asound.conf|grep -v ^#
pcm.dsp0 {
type plug
slave.pcm "dmix"
}
ctl.mixer0 {
type hw
card 0
}
pcm.10to20 {
type route
slave.pcm hw:0
slave.channels 2
ttable.0.0 1
ttable.0.1 1
}
That's the current config.
Then I start flite:
r...@h2200:~$ flite -t Test
ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find
an usable access for '(null)'
audio_open_alsa: failed to set number of channels to 1. Invalid argument.
When I use the 10to20 mono to stereo mapping I get:
r...@h2200:~$ aplay -D10to20 notify.wav
Playing WAVE 'notify.wav' : Signed 16 bit Little Endian, Rate 22050 Hz,
Stereo
aplay: set_params:879: Broken configuration for this PCM: no
configurations available
With flite I get a similar error when I want to make stereo from the 1
channel that flite wants.
What's happening on the ipaq?
How can I solve this? Obviously I want flite to work.
Tips are very welcome!
Kind regards,
Udo
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[Alsa-user] patchbay/router tab for envy24control changes settings while playing
Hello, On my system the settings on the patchbay/router tab of envy24control *change* while playing audio over my M-Audio audiophile2496 card. The settings for the S/PDIF output need to be on Digital Mix but change into S/PDIF Out while playing for no apparent reason. This is on a AMD x86-64 Fedora 8 system while playing audio from Audacity via Jack, etc. Can anybody explain why or maybe how to debug this? Kind regards, Udo - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] sound on H2200 ipaq for flite
Hello,
I am trying to get flite working on a HP iPAQ of the H2210 type.
System sounds work.
But when I run flite, I get:
[EMAIL PROTECTED]:/etc$ flite -t "test 1 2 3"
ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find
an usable access for '(null)'
audio_open_alsa: failed to set number of channels to 1. Invalid argument.
So the default sounds card is not default? (it is using null?)
Also there might be a mono versus stereo issue.
[EMAIL PROTECTED]:/etc$ cat asoundrc
#
# simple dmix configuration
#
pcm.dsp0 {
type plug
slave.pcm "dmix"
}
ctl.mixer0 {
type hw
card 0
}
These are the defaults contents.
A line like:
pcm.!default default:h2200 Audio
doesn't really help.
[EMAIL PROTECTED]:/etc$ cat /proc/asound/cards
0 [h2200 Audio]: h2200 Audio - h2200 Audio
iPAQ h2200 Audio [codec Philips UDA1380]
[EMAIL PROTECTED]:/etc$ aplay -L
default:CARD=h2200 Audio
h2200 Audio, h2200 Audio
Default Audio Device
null
Discard all samples (playback) or generate zero samples (capture)
Does anybody know how to make the config so that flite doesn't complain
and sound works for applications?
Kind regards,
Udo
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[Alsa-user] flite on h2200 audio...
Hello,
I've been receiving some Australian help on getting flite working on the
H2210 ipaq. (thanks for that!)
But it's not 100% yet on the ipaq:
It appears that mono audio goes to the null device.
Stereo audio goes to the default device.
flite generates mono audio.
When I use the mono to stereo convertor I get:
[EMAIL PROTECTED]:/etc$ aplay -Dplug:10to20 /usr/share/gpe-conf/activate.wav
Playing WAVE '/usr/share/gpe-conf/activate.wav' : Signed 16 bit Little
Endian, Rate 44100 Hz, Mono
ALSA lib pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable
aplay: set_params:879: Broken configuration for this PCM: no
configurations available
or:
[EMAIL PROTECTED]:/etc$ flite -t test
ALSA lib pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable
audio_open_alsa: failed to get hardware parameters from audio device.
Invalid argument
asound.conf:
#
# simple dmix configuration
#
pcm.dsp0 {
type plug
slave.pcm "dmix"
}
ctl.mixer0 {
type hw
card 0
}
pcm.!default{
type plug
slave.pcm "10to20"
}
pcm.10to20 {
type route
slave.pcm hw:0
slave.channels 2
ttable.0.0 1
ttable.0.1 1
}
What is wrong here?
The dsp0 and mixer0 blocks are Angstrom default config.
Please let us know!
Thanks.
Kind regards,
Udo
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[Alsa-user] set_params:901: Sample format non available
Hello,
I am trying to record from S/PDIF on a M-Audio DiO 2496 card.
I get arecord: set_params:901: Sample format non available.
This problem looks similar to the one mentioned at:
http://www.redhat.com/archives/fedora-list/2006-June/msg00244.html
http://www.redhat.com/archives/fedora-list/2006-June/msg00282.html
These are my cards:
# arecord -l
List of CAPTURE Hardware Devices
card 0: V8237 [VIA 8237], device 0: VIA 8237 [VIA 8237]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: V8237 [VIA 8237], device 1: VIA 8237 [VIA 8237]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: M2496 [M Audio Delta DiO 2496], device 0: ICE1712 multi [ICE1712
multi]
Subdevices: 1/1
Subdevice #0: subdevice #0
arecord -L at the bottom.
I run 2.6.18 on a Fedora Core 5 box.
I have a CD-player hooked up to the M-Audio and envy24control shows the
meters moving to the sound coming in from the CD-player. So the card is
locked to the signal from the coax cable.
arecord -Dhw:1,0 -f cd gives arecord: set_params:901: Sample format non
available
arecord -Dhw:1,0 -f S32_LE -c2 -r44100 gives 906: Channels count not
available
This stuff worked under Windoze.
How can I fix this?
Am I doing something wrong? Or is there a problem?
Udo
# arecord -L
PCM list:
hw {
@args.0 CARD
@args.1 DEV
@args.2 SUBDEV
@args.CARD {
type string
default {
@func getenv
vars {
0 ALSA_PCM_CARD
1 ALSA_CARD
}
default {
@func refer
name 'defaults.pcm.card'
}
}
}
@args.DEV {
type integer
default {
@func igetenv
vars {
0 ALSA_PCM_DEVICE
}
default {
@func refer
name 'defaults.pcm.device'
}
}
}
@args.SUBDEV {
type integer
default {
@func refer
name 'defaults.pcm.subdevice'
}
}
type hw
card $CARD
device $DEV
subdevice $SUBDEV
}
plughw {
@args.0 CARD
@args.1 DEV
@args.2 SUBDEV
@args.CARD {
type string
default {
@func getenv
vars {
0 ALSA_PCM_CARD
1 ALSA_CARD
}
default {
@func refer
name 'defaults.pcm.card'
}
}
}
@args.DEV {
type integer
default {
@func igetenv
vars {
0 ALSA_PCM_DEVICE
}
default {
@func refer
name 'defaults.pcm.device'
}
}
}
@args.SUBDEV {
type integer
default {
@func refer
name 'defaults.pcm.subdevice'
}
}
type plug
slave.pcm {
type hw
card $CARD
device $DEV
subdevice $SUBDEV
}
}
plug {
@args.0 SLAVE
@args.SLAVE {
type string
}
type plug
slave.pcm $SLAVE
}
shm {
@args.0 SOCKET
@args.1 PCM
@args.SOCKET {
type string
}
@args.PCM {
type string
}
type shm
server $SOCKET
pcm $PCM
}
tee {
@args.0 SLAVE
@args.1 FILE
@args.2 FORMAT
@args.SLAVE {
type string
}
@args.FILE {
type string
}
@args.FORMAT {
type string
default raw
}
type file
slave.pcm $SLAVE
file $FILE
format $FORMAT
}
file {
@args.0 FILE
@args.1 FORMAT
@args.FILE {
type string
}
@args.FORMAT {
type string
default raw
}
type file
slave.pcm null
file $FILE
format $FORMAT
}
null {
type null
}
cards 'cards.pcm'
front 'cards.pcm.front'
rear 'cards.pcm.rear'
center_lfe 'cards.pcm.center_lfe'
side 'cards.pcm.side'
surround40 'cards.pcm.surround40'
surround41 'cards.pcm.sur
Re: [Alsa-user] set_params:901: Sample format non available
Lee Revell wrote: > > On Sat, 2006-09-30 at 11:07 +0200, Udo van den Heuvel wrote: >> >> This stuff worked under Windoze. >> >> How can I fix this? >> >> Am I doing something wrong? > > > > Yes. Use plughw:1 rather than hw:1. Or just use default:1 for > > automatic rate, format, channel conversion and dmix. Thanks for the tips. I did some tests. The setup: CD -> coax -> DiO 2496 Envy24control: masterclock spdif in rate state locked input coax The tests: arecord -d5 -r44100 -Dplughw:1 > /data/0/test0.wav Recording WAVE '(null)' : Unsigned 8 bit, Rate 44100 Hz, Mono gives low quality audio arecord -d5 -fcd -Dplughw:1 > /data/0/test0.wav Recording WAVE '(null)' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo gives silence arecord -d5 -fcd -Ddefault:1 > /data/0/test0.wav Recording WAVE '(null)' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo silence arecord -d5 -Ddefault:1 > /data/0/test0.wav Recording WAVE '(null)' : Unsigned 8 bit, Rate 8000 Hz, Mono silence arecord -d5 -r44100 -f S16_LE -Dplughw:1 > /data/0/test0.wav Recording WAVE '(null)' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono audio, mono but OK arecord -d5 -r44100 -f S16_LE -c 2 -Dplughw:1 > /data/0/test0.wav Recording WAVE '(null)' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo silence arecord -d5 -r44100 -f S16_LE -Ddefault:1 > /data/0/test0.wav Recording WAVE '(null)' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono silence So I can not record stereo from sp/dif. What is my mistake? Udo - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys -- and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Unable to record in stereo from spdif on M-Audio DIO 2496
Hello, Following up on my post about `set_params:901: Sample format non available` I did some tests which I posted here. The outcome was that I could record, even in 16-bit, but not in stereo. Stereo would yield silence. Does anybody here have the M-Audio DIO2496 working for recording from spdif? If so, please post. Below are the tests I did. Thanks, Udo >> Yes. Use plughw:1 rather than hw:1. Or just use default:1 for >> > > automatic rate, format, channel conversion and dmix. Thanks for the tips. I did some tests. The setup: CD -> coax -> DiO 2496 Envy24control: masterclock spdif in rate state locked input coax The tests: arecord -d5 -r44100 -Dplughw:1 > /data/0/test0.wav Recording WAVE '(null)' : Unsigned 8 bit, Rate 44100 Hz, Mono gives low quality audio arecord -d5 -fcd -Dplughw:1 > /data/0/test0.wav Recording WAVE '(null)' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo gives silence arecord -d5 -fcd -Ddefault:1 > /data/0/test0.wav Recording WAVE '(null)' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo silence arecord -d5 -Ddefault:1 > /data/0/test0.wav Recording WAVE '(null)' : Unsigned 8 bit, Rate 8000 Hz, Mono silence arecord -d5 -r44100 -f S16_LE -Dplughw:1 > /data/0/test0.wav Recording WAVE '(null)' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono audio, mono but OK arecord -d5 -r44100 -f S16_LE -c 2 -Dplughw:1 > /data/0/test0.wav Recording WAVE '(null)' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo silence arecord -d5 -r44100 -f S16_LE -Ddefault:1 > /data/0/test0.wav Recording WAVE '(null)' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono silence - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys -- and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Unable to record in stereo from spdif on M-Audio DIO 2496
Udo van den Heuvel wrote: > Hello, > > Following up on my post about `set_params:901: Sample format non > available` I did some tests which I posted here. > The outcome was that I could record, even in 16-bit, but not in stereo. > Stereo would yield silence. Do the findings posted at http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg08640.html still make sense or was the ICE1712 configuration changed since then? (I see two slave.channels mentions instead of one?) Who can help make this card's SPDIF input work 100%? (stereo...) Kind regards, Udo - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys -- and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Problem with an ice1712 soundcard on recording
[EMAIL PROTECTED] wrote: > arecord -f cd x.wav > > Nothing on playback. Like: silence? I have this same problem with my DiO2496. Also ice1712. But I have to use plughw:1 to get soem sound (mono) at all. So what would be different here? - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Problem with an ice1712 soundcard on recording
[EMAIL PROTECTED] wrote: > Quoting Harald Gutmann <[EMAIL PROTECTED]>: > >> arecord -f dat file.wav sounds okay. >> seems that this is the problem. > > > Hmmm > > arecord -f dat file.wav > > and I get silence so what is different ? > > just tried > > arecord -r 48000 -f S16 > > this is fine. > > So it looks like my problem is recording stereo. I have the same issue on a DiO2496 from M-Audio. > Do I need something in .asoundrc ? Dunno. I posted some links to archived posts a efw days ago about this issue. Got no response yet. - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Problem with an ice1712 soundcard on recording
Lee Revell wrote: >> arecord -f dat x.wav is silent >> >> arecord -r 441000 x.wav is not broken but distorted more then 8 bit >> quantization. >> > > I guess you mean 44100? > > Anyway that last command will record 8 bit audio at 44100Hz which still > won't sound good. > > Try arecord -r 44100 -f S16_LE maybe add -c2 to record stereo: arecord -r 44100 -f S16_LE -c2 blah.wav Overe here mono works OK but stereo gives silence. What is wrong/missing/needs configuration to make recording (from S/PDIF in my case) work? (M-Audio DiO2496) - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Problem with an ice1712 soundcard on recording
Lee Revell wrote: > On Fri, 2006-10-13 at 18:15 +0200, Udo van den Heuvel wrote: >> Lee Revell wrote: >>>> arecord -f dat x.wav is silent [] >>> Try arecord -r 44100 -f S16_LE >> maybe add -c2 to record stereo: arecord -r 44100 -f S16_LE -c2 blah.wav >> Overe here mono works OK but stereo gives silence. >> >> What is wrong/missing/needs configuration to make recording (from S/PDIF >> in my case) work? (M-Audio DiO2496) > > No idea. Must be a driver bug. Please file a report in ALSA bug > tracker. Thank, done that, I hope the fix is easy. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2526 Udo - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Problem with an ice1712 soundcard on recording
[EMAIL PROTECTED] wrote: > Quoting Udo van den Heuvel <[EMAIL PROTECTED]>: > >> Lee Revell wrote: >>> Try arecord -r 44100 -f S16_LE >> >> maybe add -c2 to record stereo: arecord -r 44100 -f S16_LE -c2 blah.wav >> Overe here mono works OK but stereo gives silence. > > Yes same here. No updates on this issue? The bug report at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2526 did see no activity. This means the ice1712 driver is dead? - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Problem with an ice1712 soundcard on recording
[EMAIL PROTECTED] wrote: > Quoting Udo van den Heuvel <[EMAIL PROTECTED]>: > >> [EMAIL PROTECTED] wrote: >>> Quoting Udo van den Heuvel <[EMAIL PROTECTED]>: >>> >>>> Lee Revell wrote: >>>>> Try arecord -r 44100 -f S16_LE >>>> maybe add -c2 to record stereo: arecord -r 44100 -f S16_LE -c2 blah.wav >>>> Overe here mono works OK but stereo gives silence. >>> Yes same here. >> No updates on this issue? >> The bug report at >> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2526 did see no >> activity. This means the ice1712 driver is dead? > > Hi, > > I have resolved my problem but my stereo recording troubles did not > involve the SPDIF. Unfortunately my SPDIF is connected to a yammaha > sound card that is unsupported under linux so I can not test SPDIF. > > My audio inputs are on lines 7/8 of my card. > > I have played around swapping lines. > > I think if I record in mono then it records all lines ? > > If I record in stereo then it uses 1/2. Hence the change from mono to > stereo made my input dissappear. Interesting theory. How do I find out on what lines my audio comes in? > I am not sure how to make arecord record my different lines. I have > found and .asoundrc for selecting output but I am not sure about input. If the default ice1712 config is broken it needs fixing. Who can help here? - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Problem with an ice1712 soundcard on recording
[EMAIL PROTECTED] wrote: [.] > You could try > > arecord -v -f cd -D dig test.wav Thanks! Will give this a go! - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] quiet mono recording! (Problem with an ice1712 soundcard on recording)
[EMAIL PROTECTED] wrote: > Quoting Udo van den Heuvel <[EMAIL PROTECTED]>: >> Interesting theory. >> How do I find out on what lines my audio comes in? > > I think the theory is correct. > > arecord -v > aplay -v > > report the transformation table ttable > > for recording mono this seems to be default > >0 <- 0*0.083 + 1*0.083 + 2*0.083 + 3*0.083 + > 4*0.083 + 5*0.083 + 6*0.083 + 7*0.083 + 8*0.083 + > 9*0.083 + 10*0.083 + 11*0.083 > > So this would explain why some people report quiet recordings! Over here I see: # arecord -v -Dplughw:1 Recording WAVE 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0*0.083 + 1*0.083 + 2*0.083 + 3*0.083 + 4*0.083 + 5*0.083 + 6*0.08 3 + 7*0.083 + 8*0.083 + 9*0.083 + 10*0.083 + 11*0.083 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : U8 subformat: STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time: 4000 tstamp_mode : NONE period_step : 1 sleep_min: 0 avail_min: 1000 xfer_align : 1000 start_threshold : 1 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Hardware PCM card 1 'M Audio Delta DiO 2496' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S32_LE subformat: STD channels : 12 rate : 8000 exact rate : 8000 (8000/1) msbits : 24 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time: 4000 tstamp_mode : NONE period_step : 1 sleep_min: 0 avail_min: 1000 xfer_align : 1000 start_threshold : 1 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 RIFF$WAVEfmt @data > For stereo it records lines 0 and 1. > > Transformation table: > 0 <- 0 > 1 <- 1 So I need to change to 8/9 After that change it gives me: # arecord -v -f cd -D dig test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 8 1 <- 9 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat: STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 5461 period_size : 1365 period_time : 30952 tick_time: 4000 tstamp_mode : NONE period_step : 1 sleep_min: 0 avail_min: 1365 xfer_align : 1365 start_threshold : 1 stop_threshold : 5461 silence_threshold: 0 silence_size : 0 boundary : 1431568384 Slave: Hardware PCM card 1 'M Audio Delta DiO 2496' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S32_LE subformat: STD channels : 12 rate : 44100 exact rate : 44100 (44100/1) msbits : 24 buffer_size : 5461 period_size : 1365 period_time : 30952 tick_time: 4000 tstamp_mode : NONE period_step : 1 sleep_min: 0 avail_min: 1365 xfer_align : 1365 start_threshold : 1 stop_threshold : 5461 silence_threshold: 0 silence_size : 0 boundary : 1431568384 (no RIFF!!) Aside from the RIFF stuff (there was no SPDIF signal active yet) it looks a bit better. Thanks! - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ICE1724, recording with iec958 at 16 bit
Andrew Gaydenko wrote: > Hi! > > The aim is very simple: to record 44100/16/2 SPDIF stream from an external > source with > bit-to-bit accuracy. The thing is I have not found a way to force ALSA iec958 > device > to work at 16 bit depth (32 bit depth works with JACK and arecord). > > If there *is not* a way to work at 16 bit depth, is there a way to resample > recorded > samples to be sure bit-to-bit accuracy will be achieved? plughw:x does not work? Just record 32 bits and throw away the bottom 16? - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ICE1724, recording with iec958 at 16 bit
Andrew Gaydenko wrote: > I don't know wich 'x' must be for SPDIF input. 'aplay -l' out is: I have an ice1712 card with similar issues. > card 0: T71Space [Terratec Aureon 7.1-Space], device 1: IEC1724 IEC958 > [IEC1724 IEC958] > Subdevices: 1/1 > Subdevice #0: subdevice #0 This is the one that should do the trick? > I have tried plug:iec958 - capture works this way with 16 bit depth. > If I understand well, this way (plug:iec958) we simply add 32-to-16 bit > covertion. And all conersion chain is 16-32-16. Can I be sure such > conversion keeps a stream with bit-to-bit accuracy? Dunno. Also depends on the hardware and driver. As I udnerstand it, data arrives at the card via TOSlink or coax in 16-bit samples. The card accepts the data and offers a 32-bit value to the driver. The driver can, at will, drop teh lowest 16 or 8 bits by specifying the wanted bitdepth. (useful in case we have 24-bit data etc) > I can not test myself as I have not an external SPDIF stream with before-hand > known > (with bit-level accuracy) content. Then I'd just play a CD and record at 32-bit, 24-bit and 16-bit. The lower remaining bits (if present) should be all 0. Compare the recordings in a sound editor, do a hexdump, etc. Short recordings will suffice. - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ICE1724, recording with iec958 at 16 bit
Andrew Gaydenko wrote: > Udo, > > How to address to this device (card 0, device 1) in 'plughw:x' syntax? > > > Andrew > > === On Saturday 04 November 2006 12:29, you wrote: === > ... >> card 0: T71Space [Terratec Aureon 7.1-Space], device 1: IEC1724 IEC958 >> [IEC1724 IEC958] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 > > This is the one that should do the trick? arecord -Dplughw:0,1 blabla or something like that? - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
