Re: [Alsa-user] Continuous Recording
Clemens Ladisch schrieb am 06.03.2015 um 08:34: Alexandre Ratchov wrote: On Thu, Mar 05, 2015 at 09:39:17AM -0800, Robin _ wrote: does anyone know of a readymade open source program for 24/7/365 logging? If JACK programs are an option: How about Rotter? http://www.aelius.com/njh/rotter/ I think it was designed exactly for that purpose. It's lightweight and was running stable when I tested it (ages ago). Best, Friedrich AFAICS, any program recording raw samples could do the job (arecord, sox, etc...) The problem is how to always keep available space on the filesystem. The simpler option would to record to a unix pipe and then store data into 1-day long files without closing the pipe (for instance with the dd utility). Then, from time to time move old files away to free space. arecord automatically creates a new file when the old file becomes too big (2 GB for .wav) or exceeds the length specified with the --max-file-time option. Regards, Clemens -- Dive into the World of Parallel Programming The Go Parallel Website, sponsored by Intel and developed in partnership with Slashdot Media, is your hub for all things parallel software development, from weekly thought leadership blogs to news, videos, case studies, tutorials and more. Take a look and join the conversation now. http://goparallel.sourceforge.net/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- Dive into the World of Parallel Programming The Go Parallel Website, sponsored by Intel and developed in partnership with Slashdot Media, is your hub for all things parallel software development, from weekly thought leadership blogs to news, videos, case studies, tutorials and more. Take a look and join the conversation now. http://goparallel.sourceforge.net/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Capturing stereo signals into files
Hi Vincent, Vincent Gulinao schrieb am 02.05.2013 19:19: Hi everyone, I have a task to setup a system that will continuously capture multiple stereo signals using a MADI audio card (RME HDSPe MADI) and write them into files (perhaps in 1 hour chunks). Few checks I've learned while googling on the topic looks like that my linux setup seem to recognize the card at least. I'm a total noob in ALSA and audio concepts in general. I've read many docs and forums (e.g. arecord, asoundrc, pulseaudio, jack, etc.) here and there. But while I wait all those info settle in my mind, can somebody pitch here some high-level hints on what things I need to know, or what sort of questions should I be asking, to accomplish this task and make sure I'm on the right direction? Thanks in advance! There's an application targeted at recording audio continuously in chunks of 1 hour length called 'rotter': http://www.aelius.com/njh/rotter/ I didn't use this application for quite some time, so I don't know whether this program still does what you are looking for. But at the time I used the program it just worked: lightweight and solid. rotter requires that you run the jack audio server (you would let jack run all the time). For sure there are other (better?) options that I don't know of, but your goal (continuously in 1 hour chunks) seems to be exactly what rotter was built for... -- Introducing AppDynamics Lite, a free troubleshooting tool for Java/.NET Get 100% visibility into your production application - at no cost. Code-level diagnostics for performance bottlenecks with 2% overhead Download for free and get started troubleshooting in minutes. http://p.sf.net/sfu/appdyn_d2d_ap1 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Help configuring HDSP9632
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
common thing of configuring dmix to let multiple applications output
sound at once. Nothing fancy, really! At least, at this point.
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.
Thanks a lot,
Matt
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Re: [Alsa-user] rme Hammerfall
Felix Pfeifer wrote:
Hi list,
is there a possibility to use 2 Hammerfall cards at the same time?
I would like to cascade them and use 48 adat channels for 1 application.
I heard that some people do that, but i guess they work on windows.
greets
Felix
Hi Felix,
this is definitely possible under linux and widely used. Just copy the
configuration below into your ~/.asoundrc and tell your app to use the
device named adat48ch. Make sure that both cards are running on
perfectly in sync. A wordclock cable is probably the best solution.
The configuration below is written for a system with onboard sound
(hw:0) and two hammerfalls (hw:1 and hw:2). The channels 48 and 49 (i.e.
the 49th and 50th channel) are the spdif connection of the 1st hammerfall.
have fun!
pcm.adat48ch {
type multi;
slaves.a.pcm hw:1,0;
slaves.a.channels 26;
slaves.b.pcm hw:2,0;
slaves.b.channels 26;
bindings.0.slave a;
bindings.0.channel 0;
bindings.1.slave a;
bindings.1.channel 1;
bindings.2.slave a;
bindings.2.channel 2;
bindings.3.slave a;
bindings.3.channel 3;
bindings.4.slave a;
bindings.4.channel 4;
bindings.5.slave a;
bindings.5.channel 5;
bindings.6.slave a;
bindings.6.channel 6;
bindings.7.slave a;
bindings.7.channel 7;
bindings.8.slave a;
bindings.8.channel 8;
bindings.9.slave a;
bindings.9.channel 9;
bindings.10.slave a;
bindings.10.channel 10;
bindings.11.slave a;
bindings.11.channel 11;
bindings.12.slave a;
bindings.12.channel 12;
bindings.13.slave a;
bindings.13.channel 13;
bindings.14.slave a;
bindings.14.channel 14;
bindings.15.slave a;
bindings.15.channel 15;
bindings.16.slave a;
bindings.16.channel 16;
bindings.17.slave a;
bindings.17.channel 17;
bindings.18.slave a;
bindings.18.channel 18;
bindings.19.slave a;
bindings.19.channel 19;
bindings.20.slave a;
bindings.20.channel 20;
bindings.21.slave a;
bindings.21.channel 21;
bindings.22.slave a;
bindings.22.channel 22;
bindings.23.slave a;
bindings.23.channel 23;
bindings.24.slave b;
bindings.24.channel 0;
bindings.25.slave b;
bindings.25.channel 1;
bindings.26.slave b;
bindings.26.channel 2;
bindings.27.slave b;
bindings.27.channel 3;
bindings.28.slave b;
bindings.28.channel 4;
bindings.29.slave b;
bindings.29.channel 5;
bindings.30.slave b;
bindings.30.channel 6;
bindings.31.slave b;
bindings.31.channel 7;
bindings.32.slave b;
bindings.32.channel 8;
bindings.33.slave b;
bindings.33.channel 9;
bindings.34.slave b;
bindings.34.channel 10;
bindings.35.slave b;
bindings.35.channel 11;
bindings.36.slave b;
bindings.36.channel 12;
bindings.37.slave b;
bindings.37.channel 13;
bindings.38.slave b;
bindings.38.channel 14;
bindings.39.slave b;
bindings.39.channel 15;
bindings.40.slave b;
bindings.40.channel 16;
bindings.41.slave b;
bindings.41.channel 17;
bindings.42.slave b;
bindings.42.channel 18;
bindings.43.slave b;
bindings.43.channel 19;
bindings.44.slave b;
bindings.44.channel 20;
bindings.45.slave b;
bindings.45.channel 21;
bindings.46.slave b;
bindings.46.channel 22;
bindings.47.slave b;
bindings.47.channel 23;
bindings.48.slave a;
bindings.48.channel 24;
bindings.49.slave a;
bindings.49.channel 25;
}
ctl.adat48ch {
type hw;
card 1;
}
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Re: [Alsa-user] Fwd: KnoppMyth/LinHES Linux - Analog audio out to ear phones works but not to TV. Ubuntu 9-i386 LiveCD everything works fine!
Matt Snow wrote: On Fri, Jul 10, 2009 at 2:20 AM, Clemens Ladischcladi...@googlemail.com wrote: Matt Snow wrote: On Thu, Jul 9, 2009 at 12:05 AM, Clemens Ladischcladi...@googlemail.com wrote: Matt Snow wrote: After upgrading to KnoppMyth/LinHES R6(ArchLinux), the analog out plays audio perfectly through head phones or powered speakers, but the 3.5mm stereo-to-RCA cable going to RCA input on the TV doesn't produce any sound at all. Isn't that the same jack on the card? Or where have you connected the headphones, the speakers and the TV? there are several analog jacks on the card for different channels. i'm using the front speaker jack, but have tried all jacks with the TV and ear phones. So the headphones or speakers in the front jack work, and the to-RCA cable in the *same* jack does not? This is obviously a broken cable (or TV). I didn't read the whole thread, but - just an idea: Couldn't it also be some jack-detection of the soundcard? In case it uses the impedance of the connected device for the plug-in dectection, it might not notice the TV if its inputs have a higher impedance than your phones or powered speakers. You might think that, but its fine. I take the exact same hdmi+dvi adaptor, and the rca-to-stero cable, plug in to either my macbook pro, or macbook with mini displayport adaptor to hdmi and I get audio and video. Put the plugs right back to the HTPC, get nothing. leave them there, boot a i386 Ubuntu 9.x Jaunty liveCD and Audio out of the TV works. Perhaps the plug-in detection code has changed (if there is/was any used for this card). TV and cables are fine, and the hardware is all fine as well. It's obviously a kernel/driver issue. i'm still working on compiling a new kernel but haven't had the time to do further troubleshooting. Best regards, Clemens ..Matt --fe -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] RME HDSP and Suse 10.1, will NOT work...
Tim Barker schrieb:
...
The card appears to be recognised by the system, the red light on the front
of the breakout box does extinguish on startup as expected (and even
extinguishes when inserting the card while the system is running, which
surprised me).
hotplug should work. this is ok.
I've used the modules.d/sound script for the hdsp as
suggested in several emails and websites. The problem is the card doesn't
appear in Jack, Qjackctl or many other things.
How do you check? have a look in /proc/asound/cards if your hdsp is
mentioned there.
Does hdspmixer work?
If everything seems ok, write the following in your .asoundrc (create
the file if it doesn't exist in your home directory):
pcm.multiface_test {
type plug
ttable.0.0 1
ttable.1.1 1
slave.pcm {
type hw
card 0
}
}
and then try
aplay -D multiface_test SOME_WAVE_FILE.WAV
(you have to change the card number 'card 0' if the hdsp is not the
first card in your system. you can get the card number from
/proc/asound/cards)
If aplay runs fine, you can try to find out how to get sound out of the
multiface = matrix mixer settings, see here:
http://alsa-project.org/alsa-doc/doc-php/template.php?company=RMEcard=Hammerfall+DSP+9652.chip=FPGAmodule=hdsp
and/or
use hdspmixer
Whilst hdspconf works fine
and even under a shell recognising the card and responding, hdsploader
does not (refer listing #1 below), I get the same error message I received
before.
You don't need to use the hdsploader by hand. The firmware gets loaded
automitically.
I have read that I need to use Version 10 of the pcmcia card
firmware (I'm running version 11) but have no method of moving back to the
old firmware (despite what everybody says there appears to be no software on
the RME site that allows one to go backwards and I have all of the old
loader programmes on my harddrive anyway). Are there other commands
required when loading the module (hdsp.ko)?
isn't the module name 'snd-hdsp.ko'?
where do they go? What other
commands are needed to get the box to work?
I don't have any experience regarding the firmware version problem as
I'm only using 'older' hdsp cards.
Interestingly the listing from dmesg shows the card being recognised by the
system...
pccard: CardBus card inserted into slot 0
PCI: Enabling device :03:00.0 (0080 - 0082)
ACPI: PCI Interrupt :03:00.0[A] - Link [C0B8] - GSI 11 (level, low) -
IRQ 11
ALSA sound/pci/rme9652/hdsp.c:771: Hammerfall-DSP: wait for FIFO status = 0
failed after 30 iterations
ALSA sound/pci/rme9652/hdsp.c:650: Hammerfall-DSP: loading firmware
ALSA sound/pci/rme9652/hdsp.c:683: Hammerfall-DSP: finished firmware loading
...although clearly something there isn't working (failed after 30
iterations) and it does seem to to be loading the firmware. And further
down in the listing...
ALSA sound/pci/rme9652/hdsp.c:4948: Hammerfall-DSP: Firmware already
present, initializing card.
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[Alsa-user] SiS748/963L, CMI9739, pcm not working
Hi, I've collected some additional information to my previous mail CMI9739 pcm not working: attached you find the output of proc/asound/card0/pcm0p/sub0/status when running aplay. Before running aplay, the status is 'closed'. When running aplay, the only number that changes over time is 'tstamp'. 'hw_ptr' does not change. I first tried with alsa 0.9.something as shipped with mandrake 9.2, then with a new compiled alsa1.0.0rc1. The behaviour is the same in both cases. I also tried the modules.conf delivered by mandrake: above snd-intel8x0 snd-pcm-oss alias sound-slot-0 snd-intel8x0 and the modules.conf build by the alsaconf script. This results in exactly the same behaviour: Modules etc. seem to be loaded correctly, mixer is working, but pcm is not processing (no interrupts visible with e.g. xosview) regardless of using alsa, mmap, oss-compatilbility layer... Is there any additional information I should provide for debugging? TIA fe state: RUNNING trigger_time: 1070397701.434501000 tstamp : 1070397750.192659000 delay : 7194 avail : 4806 avail_max : 0 - hw_ptr : 4804 appl_ptr: 11998
[Alsa-user] CMI9739 pcm not working
Hi, I've set up a new PC using Mandrake 9.2. The onboard sound chip was correctly detected and installed. The modules are loaded, the sound-devices exist, /proc/asound/ looks ok, alsamixer is working fine, but... ... there is no sound output from PCM. Playing back a CD through analoug soundcard input works, I can e.g. control CD volume in alsamixer. I've tested play, aplay, xmms/oss, xmms/alsa. xmms hangs when I hit the play button. Seems like the soundcard doesn't start processing. This is a ASRock K7S8XE Mainboard with onboard sound which works under win2k correctly. The codec is shown as C-Media Electronics CMI9739 in alsamixer (which is the correct name). The only thing that doesn't look good is /proc/interrups (see attachment). The SiS SI7012 should be the sound chip, so I assume the counter shouldn't be 0. I've attached the output of the aadebug script, maybe it's of some help. What should I try next? Thanks in advance! CPU0 0: 634299 XT-PIC timer 1: 17509 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 0 XT-PIC usb-ohci, ehci_hcd, SiS SI7012 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi 11: 104699 XT-PIC usb-ohci, HiSax 14: 71444 XT-PIC ide0 15: 19226 XT-PIC ide1 NMI: 0 LOC: 634259 ERR: 93 MIS: 0 ALSA Audio Debug v0.0.4 - Mo Dez 1 22:58:53 CET 2003 Kernel Linux localhost 2.4.22-10mdk #1 Thu Sep 18 12:30:58 CEST 2003 i686 unknown unknown GNU/Linux Loaded Modules snd-seq-oss30464 0 (unused) snd-seq-midi-event 6016 0 [snd-seq-oss] snd-seq42576 2 [snd-seq-oss snd-seq-midi-event] snd-pcm-oss42756 0 snd-mixer-oss 14288 0 [snd-pcm-oss] snd-intel8x0 21896 1 snd-ac97-codec 52620 0 [snd-intel8x0] snd-pcm78980 1 [snd-pcm-oss snd-intel8x0] snd-timer 18500 0 [snd-seq snd-pcm] snd-mpu401-uart 4624 0 [snd-intel8x0] snd-rawmidi17856 0 [snd-mpu401-uart] snd-seq-device 5788 0 [snd-seq-oss snd-seq snd-rawmidi] snd-page-alloc 9300 0 [snd-intel8x0 snd-pcm] snd44612 0 [snd-seq-oss snd-seq-midi-event snd-seq snd-pcm-oss snd-mixer-oss snd-intel8x0 snd-ac97-codec snd-pcm snd-timer snd-mpu401-uart snd-rawmidi snd-seq-device] Modules Conf -- above snd-intel8x0 snd-pcm-oss alias sound-slot-0 snd-intel8x0 Proc Asound --- Advanced Linux Sound Architecture Driver Version 1.0.0rc1. Compiled on Dec 1 2003 for kernel 2.4.22-10mdk with versioned symbols. 0 [SI7012 ]: ICH - SiS SI7012 SiS SI7012 at 0xdc00, irq 5 0: [0- 0]: ctl 25: [0- 1]: digital audio capture 16: [0- 0]: digital audio playback 24: [0- 0]: digital audio capture 1: : sequencer 33: : timer 00-00: Intel ICH : SiS SI7012 : playback 1 : capture 1 00-01: Intel ICH - MIC ADC : SiS SI7012 - MIC ADC : capture 1 Dev Snd --- controlC0 pcmC0D0c pcmC0D0p pcmC0D1c seq timer CPU --- model name : AMD Athlon(tm) XP 2400+ cpu MHz : 2000.103 RAM --- MemTotal: 255780 kB SwapTotal: 0 kB Hardware -- 00:00.0 Host bridge: Silicon Integrated Systems [SiS] 746 Host (rev 10) 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] Sound Controller (rev a0)
Re: [Alsa-user] CS46xx spdif output + ogle question
Hi, see below... Henri wrote: sorry, but i'm still stuck. mplayer doesn't work w/ -ao alsa9 aplay works w/ -D hw:0,0 and hw:0,2 alsa-init: soundcard set to spdif alsa-init: unable to set periodsize: Invalid argument Could not open/initialize audio device - no sound. Audio: no sound Video: no video how about: try -ao alsa9 -abs [1234]. 1 through 4 sets different buffer/periodsizes. (don't know where I found this, maybe in the manpages or some mailing list). I don't use SPDIF output, but get the same error for the analog outputs of my cs46xx-based xfire. The following command works for me for 4channel analog out: mplayer -ao alsa9:surround40 -abs 1 -channels 4 HTH --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Re: cs46xx - alsa-0.9.x freezes the system
Hmm, I also had some lockups with the nvidia drivers. Download newest nvidia driver + GLX module. Check for the correct driver and GLX version (must match you kernel). I solved my problem by seeting the AGP mode to Option NvAgp 0. This works for my TNT2 M64 card, don't know if there are similar setting for yours. Romulus Pfeiffer wrote: I already wrote to Lance that I also have got a GeForce2-MX400 and that there must be the coherence. But I am not using a screensaver. I also get lockups with several 0.9rc x, 0.9.1 and 0.9.2 Today I have set with xset # xset s 0 0 dpms 0 0 0 and I played music over 3 hours with 0.9.2 without lockups but I could not believe that this is a solution. Maybe there is world wide another person with the same problem. --- This SF.net email is sponsored by: ValueWeb: Dedicated Hosting for just $79/mo with 500 GB of bandwidth! No other company gives more support or power for your dedicated server http://click.atdmt.com/AFF/go/sdnxxaff00300020aff/direct/01/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user --- This SF.net email is sponsored by: ValueWeb: Dedicated Hosting for just $79/mo with 500 GB of bandwidth! No other company gives more support or power for your dedicated server http://click.atdmt.com/AFF/go/sdnxxaff00300020aff/direct/01/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] rme9652/hammerfall playback problems
Ede Wolf wrote:
Hello,
I have a problem with 0.9.1 and the hammerfall card (true for 0.9_rcx, too): I am only able to play back mono files. I have seen this issue has occurred before, but without any solution, so maybe there is some more experience outside there by now.
aplay -D foo.wav just works, if test.wav is mono, otherwise the card remains silent (though seems to play).
Hi,
I'd write something like the following into my .asoundrc
pcm.two_channel {
type plug
ttable.0.0 1
ttable.1.1 1
slave.pcm {
type hw
card 0
}
route_policy copy
}
pcm.koax {
type plug
ttable.0.24 1
ttable.1.25 1
slave.pcm {
type hw
card 0
}
route_policy copy
}
If you now start aplay
aplay -D two_channel some_stereo_file.wav
your card should playback through the first two channels (that's the
first two channels of ADAT out 1).
And you can playback via SPDIF out using
aplay -D koax the_same_or_some_other_stereo_file.wav
If this doesn't work there could be a problem with aplay and the WAVs
used. Perhaps using another player or WAVs from another source would
help. Just an idea...
good luck,
fe
P.S. Have you tried selecting the subdevices, e.g.
aplay -D hw:0,0,0 foo.wav
aplay -D hw:0,0,1 foo.wav ...
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Re: [Alsa-user] Hercules Gametheater XP (cs46xx)
Hi, IIRC, the phase reversal problem (causes lack of bass) was fixed with the patch from 2002-12-09. On 2002-12-10 a patch was released with some corrections that worked my DMX XFire, i.e. I would try 2002-12-09.tar.bz2 from cvs plus the patch from 2002-12-10. An additional patch (regarding playback of mono files) was released 2003-01-05. (Don't know if you need this) HTH fe Johan Groth wrote: Sylvain Petreolle wrote: --- Bradach, Jack [EMAIL PROTECTED] a écrit : I'm still needing to know if anyone's gotten a Hercules Gametheater XP to work using ALSA so that it doesn't sound awful, and if so how. What about ANY of the hercules cards? See the soundcard matrix : http://www.alsa-project.org/alsa-doc/ and follow the instructions. Running a Gamesurround Fortissimo III here. Something has happend to the driver. I had a cvs version that worked well but I can't find it anymore. rc6 is missing bass and rc7 produces very distored sound. I will check in the mail archive if I can find the patch to rc6 to make it work again. Cheers, Johan --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Game Theater XP, no bass. saving mixer settings forshutdown restart
Nathan Allworth wrote: 2 things here, i finally got this thing working, using rpms. my first problem is there is no bass like there is when im in windows, it sounds good, just no bass, the only thing i've tested this in is xmms, so maybe thats my problem. but i've tried fooling with the alsamixer app with no success, along with changing out put devices in the xmms alsa plugin. nothing seems to work. nothing is plugged into center, subwoofer, or ac3 coax/optical. Try using cvs version instead of 0.9.0rc6. But cvs version may cause some problems with SPDIF (digital) input. (At least with my cs46xx based Terratec DMX XFire) and my second thing is whenever i restart my mixer settings get lost settings get lost, how can i make it save this??? this should be done by the /etc/init.d/alsa and the /usr/sbin/alsactl scripts. Game Theater XP 6.1, speakers that _do_ have bass when used with other OSes on the same machine.(4 speakers w/subwoofer which runs thru rca cables to 4 speaker output, the subwoofer does _not_ run thru to the subwoofer plugin, and its _not_ made too.) Redhat 8.0, alsa .9 rc6 installed via rpms, on the athlon arch. thanks for your help and time. fe --- This SF.NET email is sponsored by: Thawte.com Understand how to protect your customers personal information by implementing SSL on your Apache Web Server. Click here to get our FREE Thawte Apache Guide: http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0029en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
RE: [Alsa-user] CROM (getting kmod cdrom to talk with alsa)
Hi, I don't know anything about the ibm 600e but perhaps there is no audio cable connecting cdrom and soundcard in it. This is needed if your CDROM should do the DA conversion (CDROM acting as audio CD player). Some windows versions are grabbing the CD in real time via IDE and deliver the PCM stream to the soundcard which does the DAC then. Under linux, there should be a plugin for xmms to do the same. You can also use alsaplayer (take the newest version which allows to select tracks). I recently re-installed the ALSA package and can't seem to get my cdrom to produce audio. Last install required no additional effort for the cdrom device. Now, RealPlayer produces audio ok...sort of, but no CDrom. Perhaps I need to give some command to kmod before launching the Gnome cd-player. ? The mailing list makes mention of this but I did not locate the manner of accomplishing. Please advise.. *** more info*** I have ibm 600e thinkpad with cs-4236 card my /etc/modules.conf looks a bit like this: -- +++ GMX - Mail, Messaging more http://www.gmx.net +++ NEU: Mit GMX ins Internet. Rund um die Uhr für 1 ct/ Min. surfen! --- This SF.NET email is sponsored by: FREE SSL Guide from Thawte are you planning your Web Server Security? Click here to get a FREE Thawte SSL guide and find the answers to all your SSL security issues. http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0026en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA setup ... redux
Hi, I am also using a cs46xx based card (Terratec XFire 1024) and haven't seen such error messages with any alsa version I've installed so far. I've installed the alsa rpms of several Mandrake distros as well as several tarballs from the alsa project page. I've also never seen this errors in any mail of a cs46xx user to alsa-users or alsa-devel list. I don't know if your problem is caused by using debian, the alsa version you've installed, your soundcard (which one? maybe the codec isn't supported(? just a wild guess) ) or some other reason I don't know about. (You can start alsamixer to see which codec(s) is/are used on your card. E.G. my XFire has a CS4294 rev 5 on it) Perhaps benny (currently improving the cs46xx driver -- which is not easy because of missing information from cirrus) or someone else of the developers can help (perhaps a mail to developers list could help). cg wrote: Hello all, I posted an inquiry a few days ago - without response. It was about ALSA ../../alsa-kernel/pci/cs46xx/cs46xx_lib.c:464: cs46xx: failure waiting for FIFO command to complete ALSA ../../alsa-kernel/pci/cs46xx/cs46xx_lib.c:163: AC'97 read problem (ACSTS_VSTS), codec_index 1, reg = 0x0 errors on a Debian installation. --- This SF.NET email is sponsored by: FREE SSL Guide from Thawte are you planning your Web Server Security? Click here to get a FREE Thawte SSL guide and find the answers to all your SSL security issues. http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0026en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Terratec Sixpack 5.1 with cs46xx
Hi, that's a bug in rc6 version of cs46xx driver. The lack of bass is caused by phase reversal of one channel, i.e. wrong sign of the values of one channel. Therefore correlating sound (low frequency components are stronger correlated because of the long wavelengths) in left and right channel will cancel out more or less when using loudspeakers and the stereophonic impression is also wrong. Up to now all reports of this problem stated that this error occurs only with analog out, not SPDIF -- but perhaps other cs46xx based cards were used. The cvs version should fix this problem for analog out, perhaps also for digital out. But with cvs version SPDIF input doesn't work as it should and did with rc6 version (at least for my Terratec XFire). If the cvs version doesn't fix the bass problem then benny should know what to do :-) fe Hannes Ebner schrieb: Hi everybody, I recently tried to get the SPDIF/out working on my card -- I'm using Debian Woody 3.0 with Kernel 2.4.19. The Soundcard identifies itself as Sound Fusion CS46xx with a Cirrus Logic CS4297A rev 4-chip. After upgrading from alsa-source 0.9+0beta12-3/stable to alsa-source 0.9.0rc6-3/testing an recompiling the modules I got SPDIF working -- but: now I have really flat sound without Bass (I don't know if other characteristics are also missing, the only that I heard so far was the missing Bass). It's the same with Analog-Out as well as with Digital-Out -- what could I do to solve the problem? All worked fine when I was using 0.9beta12... Do I need a special configuration in /etc/alsa? Regards, Hannes --- This SF.NET email is sponsored by: SourceForge Enterprise Edition + IBM + LinuxWorld = Something 2 See! http://www.vasoftware.com ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] device names
Hi, regarding the XFire initialization: do you reload the cs46xx module automatically and -if so- how? I have to do su -c '/etc/init.d/alsa restart' manually after rebooting and would like to have this done automatically by some change in an init script or my modules.conf but don't know how to do this best. At the moment the soundcard section of my modules.conf consists of only one line which loads alsa+oss modules: above snd-cs46xx snd-pcm-oss regards fe Richard Stevens wrote: Hi, I guess I won't try it then. Hopefully this gets resolved at some stage. I don't mind another alsa workaround anyways since there is already one. I have to reload the alsa drivers for my Terratec DMX XFire 1024 after the initial load at boottime. Otherwise I only get heavily distorted sound. After reloading it's ok. I started a thread with a question about that a while ago. AFAIR the terratec Cards doesn't get initialized properly the first time. The second time it works. So I just create the missing devices after the reload. It could be worse ;) CU and thanks again, Richard --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Capturing 0s to buffer. Help Needed
Hi Pete, start alsamixer in a shell and set the input source (hit space at the correct item). You should also set the capture flag for the 'ADC' item (which is the analog-digital converter) and for the 'capture' item. When I'm recording I have the 'ADC' volume set to 100 and change the 'capture' volume as I like. That should do what you want. fe Pete Barnard wrote: Hello. I have a problem capturing sound from my sound card. I am just capturing a buffer full of Zeros. Have tried putting my audio input into both line-in and mic inputs of my sound card but just get zeros. I have installed ALSA drivers, libraries and utils 0.9.0rc6 and am using Red Hat Linux 7.2. My sound card is a Cirrus Logic CS46xx. Have run alsamixer and everything is now unmuted. --- This SF.NET email is sponsored by: Geek Gift Procrastinating? Get the perfect geek gift now! Before the Holidays pass you by. T H I N K G E E K . C O M http://www.thinkgeek.com/sf/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-devel] Re: [Alsa-user] fm801 driver status?
Hi! Hopefully, the following information is of some help. Takashi Iwai wrote: to be sure, could you elaborate the symptom again and the detail of your system (kernel version, applied patches, ALSA version)? kernel: 2.4.19-16mdkcustom, compiled with sound and without alsa. I compiled from the kernel source delivered with mandrake 9.0 and did not apply any patches. alsa: cvs snapshot 2002-12-09.tar.bz2 with your patch (that fixes the i/o port error in /var/log/messages) applied system: duron 850, 128 MB PC 133 NMC (the name changed to enmic, I think) board, KT-133 chipset with via686a. Award bios 01/10/2001-8363-686A-6A6LMNM9C-00 especially, please check the following: - loading the module and unloading it immediately works? yep. /proc/asound builds up immediately, lsmod shows snd, ... modules immediately after modprobing - the port and irq number are correct, i.e. match with the resources listed on lspci? info from /proc/asound/cards: FM801-AS at 0xec00, irq 10 lspci output: 00:0b.0 Multimedia audio controller: Fortemedia, Inc Xwave QS3000A [FM801] (rev a0) Subsystem: Fortemedia, Inc: Unknown device 1319 Flags: bus master, medium devsel, latency 40, IRQ 10 I/O ports at ec00 [size=128] (IRQ 10, I/O ec00 also shown by window?) - playback (via aplay) works? does the system hang up at the start, during the playback or at the stop? playback works. I even hear the correct sound when I plug in headphones :-) The system locks when stopping playback. (If only I had infinite audio files...) - during the playback, does the count in /proc/interrupts increases? (check with another terminal) the count at '10' (I assume this is irq 10) increases during playback by approx. 10-15 per second. a typical reason of such a hang up is either the wrong spinlocks, unexpected infinite loops in the driver code, or the interrupt storms. if it's the interrupt storm, then it can be related with the hardware, often motherboard chipset, or BIOS version. or, could it be because of ACPI? did you apply it? I unselected ACPI before compiling the kernel. ACPI is switched off in the bios. (I have only APM running as power management) ciao, Takashi Is there anything else I could do? cheers, fe --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Newbie vs. Turtle Beach Santa Cruz info
Hi! The new cs46xx driver can talk to (up to) 2 codecs on one card now . You can see duplicate controls because your card uses 2 codecs (CS4297A and CS4294) for multichannel playback/(recording?). Therefore, e.g. one master controls your front output, one master controls your surround outputs and so on. But I don't know exactly which controls correspond to which channels. btw: The new driver also does hardware mixing (no need for sound daemons like arts,.. any more. But that feature is available since rc2, I think), SPDIF output, (theoretically) SPDIF input... hope that helps fe P.S. perhaps you should try using a newer version (cvs) because there was a bug with some versions including rc6: For some reason one channel was phase reversed (i.e. wrong sign of the sample values) which causes a wrong stereophonic impression. Furthermore sound will lack of bass when using loudspeakers because of this. Lance A. Brown wrote: Greetings, I just upgraded to ALSA 0.9.0rc6 on my system at home with a Turtle Beach Santa Cruz (Sound Fusion CS46xx, Cirrus Logic CS4297A rev 4,Cirrus Logic CS4294 rev 5) and now have a whole LOT more controls in alsamixer. I checked the TB site but didn't find anything useful about learning how to handle all this. Is there someplace in the ALSA docs that explains what all these new controls are/do? Is there someplace else I can look? In particular, I'm seeing duplicates of a lot of the controls. Master Volume and Master Volume 1, for example. Thanks, --[Lance] --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-devel] Re: [Alsa-user] fm801 driver status?
Takashi Iwai wrote: you can check this also via lspci command. in the output of lspci in your last mail, the i/o ports and irq are missing. please check again? here we go... 00:0b.0 Multimedia audio controller: Fortemedia, Inc Xwave QS3000A [FM801] (rev a0) Subsystem: Fortemedia, Inc: Unknown device 1319 Flags: bus master, medium devsel, latency 40, IRQ 10 I/O ports at ec00 [size=128] Capabilities: [dc] Power Management version 1 00:0b.1 Input device controller: Fortemedia, Inc Xwave QS3000A [FM801] (rev a0) Subsystem: Fortemedia, Inc: Unknown device 1319 Flags: bus master, medium devsel, latency 40, IRQ 11 Capabilities: [dc] Power Management version 1 the first entry seems to be the audio functionality, the second one must be the game port. IRQ 10 is shared with USB controller, IRQ 11 is shared with VGA controller. The soundcard is working correctly under win98 without the need to change any hardware or bios settings. win98 reports: fm801 pci audio, IRQ 10, I/O: EC00-EC7F plus fm801 pci audio, IRQ 11 and 'no conflicts'. I.e. win98 shows up the same IRQs used as lspci, that should be correct, then. But lspci doesn't name any memory regions used by this card. Do you have any idea what I could try? (I'll test some other bios settings, mounting the card in another slot when I find some spare time). what is the bios configuration? pnp os is yes? then set it to no. is ACPI enabled on kernel? pnp was set to 'yes' in bios. I set it to 'no', but the system still locks after loading alsa driver. To be more precisely: I did modprobe snd-fm801, started aplay, stopped aplay, started arecord and after some seconds the system hang. But the system hang just some seconds after modprobing without using the audio device in a try before. Here's the /var/log/messages output before(after) the system hangs: Dec 12 19:13:27 localhost kernel: PCI: Found IRQ 10 for device 00:0b.0 Dec 12 19:13:27 localhost kernel: PCI: Sharing IRQ 10 with 00:07.2 Dec 12 19:13:27 localhost kernel: PCI: Sharing IRQ 10 with 00:07.3 Dec 12 19:13:27 localhost kernel: PCI: Found IRQ 11 for device 00:0b.1 Dec 12 19:13:27 localhost kernel: ALSA ../alsa-kernel/pci/fm801.c:922: unable to grab region 0x0-0x7f Dec 12 19:14:56 localhost syslogd 1.4.1: restart. ACPI is not enabled in kernel (but APM is, which should not be a problem, I think). Any further hints? I just don't want to crash my filesystems too often. Hopefully, ext3 does its job. Ahh! Perhaps I should remount read-only next time :-) fe --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-devel] Re: [Alsa-user] fm801 driver status?
Takashi Iwai schrieb: [...] 00:0b.0 Multimedia audio controller: Fortemedia, Inc Xwave QS3000A [FM801] (rev a0) Subsystem: Fortemedia, Inc: Unknown device 1319 Flags: bus master, medium devsel, latency 40, IRQ 10 I/O ports at ec00 [size=128] Capabilities: [dc] Power Management version 1 00:0b.1 Input device controller: Fortemedia, Inc Xwave QS3000A [FM801] (rev a0) Subsystem: Fortemedia, Inc: Unknown device 1319 Flags: bus master, medium devsel, latency 40, IRQ 11 Capabilities: [dc] Power Management version 1 i need the pci vendor/device id for them. could you give me the output of lspci -n for two devices above? the audio chip should be 0x1319:0x0801. lspci -n output: 00:0b.0 Class 0401: 1319:0801 (rev a0) 00:0b.1 Class 0980: 1319:0801 (rev a0) and that's part of the output of 'lspcidrake -v': snd-fm801 : Fortemedia, Inc|Xwave QS3000A [FM801] [MULTIMEDIA_AUDIO] (vendor:1319 device:0801 subv:1319 subd:1319) snd-fm801 : Fortemedia, Inc|Xwave QS3000A [FM801] [INPUT_OTHER] (vendor:1319 device:0801 subv:1319 subd:1319) does this information help or do you need any other outputs? thanks! (The fm801 chip on this card is one of the older models without SPDIF input. I looked this up some time ago because I saw the possibility to get an digital input with this card using an soldering iron and some el. parts for a few cents) fe --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Re: fm801 driver status?
Adam Jones wrote: On 12-Dec-02, Takashi Iwai wrote: I've got a fm801 based sound card which locks my computer completely some seconds after loading the alsa driver module. it looks like a hardware problem rather than the driver itself. as lspci shows, your fm801 card has no resources at all. Out of interest, have you made sure to turn off Plug and Play OS in the BIOS? Thanks for your hint. pnp os was set to 'yes', I set it to 'no', but the problem persists ... --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-devel] Re: [Alsa-user] fm801 driver status?
Takashi Iwai wrote: lspci -n output: 00:0b.0 Class 0401: 1319:0801 (rev a0) 00:0b.1 Class 0980: 1319:0801 (rev a0) thanks, could you try the attached patch? at least, the weird messages for allocation of invalid i/o ports should disappear. not sure whether this cures the lock-up problem, though. Takashi done. The error in /var/log/messages doesn't appear any longer (I only get these 'sharing IRQ ...' messages). Thanks! But the system still locks up when stopping playback :-( fe --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] fm801 driver status?
Hi! I've got a fm801 based sound card which locks my computer completely some seconds after loading the alsa driver module. I compiled drivers with debug=detect -- please see my mail(s) from 12/09/2002 on alsa userlist for more info. What else could I do? Meanwhile I found several reports describing the same problem. Therefore my question(s): Should it be possible to run a fm801 soundcard using the 0.9.0rc6 or cvs driver? Anyone who has installed such a card successfully? Any ideas what I'm doing wrong? (I have installed some other soundcards using alsa09 without problems.) Thanks a lot in advance! fe --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-devel] Re: [Alsa-user] fm801 driver status?
Hi, many thanks for your quick reply! Here's some more info: Mandrake Control Center recognices the card as follows (partly in german, sorry, but perhaps that's no problem in ERL/NUE ;-) ): Hersteller: Fortemedia, Inc Bus: PCI Bus: 1319:801:1319:1319 Standort auf den Bus: 0:b:0 Beschreibung: Xwave QS3000A [FM801] Modul: snd-fm801 Medienklasse: MULTIMEDIA_AUDIO I had no alsa installed while running hardware detection. The soundcard is working correctly under win98 without the need to change any hardware or bios settings. win98 reports: fm801 pci audio, IRQ 10, I/O: EC00-EC7F plus fm801 pci audio, IRQ 11 and 'no conflicts'. I.e. win98 shows up the same IRQs used as lspci, that should be correct, then. But lspci doesn't name any memory regions used by this card. Do you have any idea what I could try? (I'll test some other bios settings, mounting the card in another slot when I find some spare time). TIA fe Takashi Iwai schrieb: At Thu, 12 Dec 2002 15:45:29 +0100, Friedrich Ewaldt wrote: Hi! I've got a fm801 based sound card which locks my computer completely some seconds after loading the alsa driver module. I compiled drivers with debug=detect -- please see my mail(s) from 12/09/2002 on alsa userlist for more info. What else could I do? Meanwhile I found several reports describing the same problem. Therefore my question(s): Should it be possible to run a fm801 soundcard using the 0.9.0rc6 or cvs driver? Anyone who has installed such a card successfully? Any ideas what I'm doing wrong? (I have installed some other soundcards using alsa09 without problems.) it looks like a hardware problem rather than the driver itself. as lspci shows, your fm801 card has no resources at all. something wrong with either the soundcard itself or the combination with the mobo... can you verify the hardware? Takashi --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] fm801 locks system
Hi, because the alsa drivers shipped with Mandrake 9.0 (RPMs, version 0.9.0rc2) locked my system hard when using my fm801 soundcard, I recompiled the kernel with sound support and without alsa, compiled alsa-dirvers, -lib and utils version 0.9.0rc6 (I have the correct kernel sources of the running kernel installed) and installed them. But the problem persitst: The system locks hard (num lock doesn't respond to keypresses, no magic sysreq. keys working...) some seconds after modprobing the snd-fm801 module. I recompiled alsa-drivers with 'debug=detect' switch and that's what I find as last messages in my /var/log/messages before the system hangs : Dec 8 20:37:59 localhost kernel: PCI: Found IRQ 10 for device 00:0b.0 Dec 8 20:37:59 localhost kernel: PCI: Sharing IRQ 10 with 00:07.2 Dec 8 20:37:59 localhost kernel: PCI: Sharing IRQ 10 with 00:07.3 Dec 8 20:37:59 localhost kernel: PCI: Found IRQ 11 for device 00:0b.1 Dec 8 20:37:59 localhost kernel: ALSA ../alsa-kernel/pci/fm801.c:922: unable to grab region 0x0-0x7f If this helps, here's my lspci output: 00:00.0 Host bridge: VIA Technologies, Inc. VT8363/8365 [KT133/KM133] (rev 02) Flags: bus master, medium devsel, latency 8 Memory at d000 (32-bit, prefetchable) [size=64M] Capabilities: [a0] AGP version 2.0 Capabilities: [c0] Power Management version 2 00:01.0 PCI bridge: VIA Technologies, Inc. VT8363/8365 [KT133/KM133 AGP] (prog-if 00 [Normal decode]) Flags: bus master, 66Mhz, medium devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 Memory behind bridge: d400-d6ff Prefetchable memory behind bridge: d700-d7ff Capabilities: [80] Power Management version 2 00:07.0 ISA bridge: VIA Technologies, Inc. VT82C686 [Apollo Super South] (rev 22) Subsystem: VIA Technologies, Inc. VT82C686/A PCI to ISA Bridge Flags: bus master, stepping, medium devsel, latency 0 00:07.1 IDE interface: VIA Technologies, Inc. Bus Master IDE (rev 10) (prog-if 8a [Master SecP PriP]) Subsystem: VIA Technologies, Inc. Bus Master IDE Flags: bus master, medium devsel, latency 32 I/O ports at e000 [size=16] Capabilities: [c0] Power Management version 2 00:07.2 USB Controller: VIA Technologies, Inc. USB (rev 10) (prog-if 00 [UHCI]) Subsystem: VIA Technologies, Inc. (Wrong ID) USB Controller Flags: bus master, medium devsel, latency 40, IRQ 10 I/O ports at e400 [size=32] Capabilities: [80] Power Management version 2 00:07.3 USB Controller: VIA Technologies, Inc. USB (rev 10) (prog-if 00 [UHCI]) Subsystem: VIA Technologies, Inc. (Wrong ID) USB Controller Flags: bus master, medium devsel, latency 40, IRQ 10 I/O ports at e800 [size=32] Capabilities: [80] Power Management version 2 00:07.4 Bridge: VIA Technologies, Inc. VT82C686 [Apollo Super ACPI] (rev 30) Subsystem: VIA Technologies, Inc. VT82C686 [Apollo Super ACPI] Flags: medium devsel Capabilities: [68] Power Management version 2 00:0b.0 Multimedia audio controller: Fortemedia, Inc Xwave QS3000A [FM801] (rev a0) Subsystem: Fortemedia, Inc: Unknown device 1319 Flags: bus master, medium devsel, latency 40, IRQ 10 I/O ports at ec00 [size=128] Capabilities: [dc] Power Management version 1 00:0b.1 Input device controller: Fortemedia, Inc Xwave QS3000A [FM801] (rev a0) Subsystem: Fortemedia, Inc: Unknown device 1319 Flags: bus master, medium devsel, latency 40, IRQ 11 Capabilities: [dc] Power Management version 1 01:00.0 VGA compatible controller: Matrox Graphics, Inc. MGA G200 AGP (rev 03) (prog-if 00 [VGA]) Subsystem: Matrox Graphics, Inc. Millennium G200 AGP Flags: bus master, medium devsel, latency 40, IRQ 11 Memory at d700 (32-bit, prefetchable) [size=16M] Memory at d400 (32-bit, non-prefetchable) [size=16K] Memory at d500 (32-bit, non-prefetchable) [size=8M] Expansion ROM at unassigned [disabled] [size=64K] Capabilities: [dc] Power Management version 1 Capabilities: [f0] AGP version 1.0 *** (end of lspci output) Any ideas what's the reason for my problems? Maybe it has something to do with IRQs (IRQ 10 an IRQ 11 shared with other devices)? If it is an IRQ problem, what could I do to solve it? Or - is this a known problem with FM801? Many thanks in advance for your help! fe --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] fm801 locks system, additional information
I forgot to mention the following information: The system has an enmic/NMC board (KT133 chipset, via686a) with duron processor. The onboard soundcard (vi82xx) is switched off in the bios. (onboard sound works with oss driver without problems, if I switch it on) If you need any further information, just let me know ... --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Terratec DMX XFire 1024 cs46xx module
Hi Benny, I changed from Mandrake's AlsaRPMs(=0.9.0rc2) to tarballs (0.9.0rc6) now. I noticed some (not-)changes: * the DAC control slider now works for left right channel (it worked only for one channel before). Thanks! * the DSP hang problem when switching the SPDIF controls is gone! Great! * the DC offset/popping noise/distortion problem remains (I have to reload the module after rebooting to eliminate it. What could have changed while loading the module the second time -- or perhaps while unloading the module before??) * I tried what you've written below to get 4 channel out working -- but with no success: aplay -D hw:0,1 or -D hw:0,2 plays the wav file, but I don't hear anything from my first or second output. If I playback to -D hw:0,0, I can hear the same sound from both outputs. (I.e. the driver behaves like before. But I'm absolutely sure that I have installed the new cs46xx module after compiling) I don't know enough about the internals of the driver, but do I have to change some other lines to put the codec in the extended mode? Should I notice any change in alsamixer if I get two independent stereo outputs (e.g. a second pcm slider; I don't see changes)? And now for something (not) completely different :^) The SPDIF input seems to get out of sync after unmuting or bit errors (or ...?) sometimes. You have to mute/unmute the SPDIFin to get undistorted sound. So I can't use the linux driver to record from my digital radio (48kHz spdif output). As this doesn't happen with the win98 driver, I compared the SPDIF input behaviour of my soundcard under windows and linux: * under linux you can hear the sound immediately after switching on the SPDIF input (unmuting) * under windows you'll hear the sound after some seconds. In this time, the driver seems to find the sample rate and to sync to the SPDIF stream. (Do you know whether the win or linux driver does some error checking/fixing for SPDIF input?) Does there exist some DSP code to perform these tasks (obtain sample rate and sync to input signal)? In the 'cwcspud.dat' file of the win98 wdm driver, there is written something about a SPDIF Monitor task (uses cwcsndct.osp code). *another problem with SPDIF input is the fact that you can record the PCM output only to get the SPDIF stream. (Bad, if any program/shell causes pcm output while recording ...). The behaviour of the old win98 driver shipped with the card was quite similar. With the newer wdm driver from terratec, you can select the SPDIF input as the only input to be recorded. Perhaps the newer win98 driver contains better DSP code you could use for your driver? I don't know if it really uses other DSP code or if the driver is written in another way somehow. If I could provide some additional information needed for solving the DC offset problem, or test some source modifications to enable 4channel output -- just let me know... fe Benny Sjostrand wrote: [...] Btw, do you have any idea how to get 4 indep. channels working with this one codec? Finnally something that I believe I got a answer -;) In theory the aproach should be: [...] 1 - AC97 CODEC configuration: Front channel left and right goes through slot 3 and 4 Ext. left and right goes through slot 5 and 11 Case 1, only one AC97 CODEC, 4 channel output is not supported yet by the current driver ... Now, in dsp_spos.c example: /* create the CODEC output task */ codec_out_scb = cs46xx_dsp_create_codec_out_scb(chip,CodecOutSCB_I,0x0010,0x, MASTERMIX_SCB_ADDR, CODECOUT_SCB_ADDR,timing_master_scb, SCB_ON_PARENT_SUBLIST_SCB); creates a task in DSP that will output samples on slot 3 and 4 from the following mapping: AC 97 primary and secondary links output slot 3 0x000 - 0x00F AC 97 primary and secondary links output slot 4 0x010 - 0x01F AC 97 primary and secondary links output slot 5 0x020 - 0x02F AC 97 primary and secondary links output slot 6 0x030 - 0x03F AC 97 primary and secondary links output slot 7 0x040 - 0x04F AC 97 primary and secondary links output slot 8 0x050 - 0x05F AC 97 primary and secondary links output slot 9 0x060 - 0x06F AC 97 primary and secondary links output slot 10 0x070 - 0x07F AC 97 primary and secondary links output slot 11 0x080 - 0x08F AC 97 primary and secondary links output slot 12 0x090 - 0x09F 0x is the address to the left-channel slot and 0x0010 is the offset to get to the right-channel slot, finally, to have 4 channel on the Terratec the following change (in pure theory should work) in dsp_spos.c sec_codec_out_scb = cs46xx_dsp_create_codec_out_scb(chip,CodecOutSCB_II,0x0010,0x0040, REAR_MIXER_SCB_ADDR, SEC_CODECOUT_SCB_ADDR,codec_in_scb, SCB_ON_PARENT_NEXT_SCB); (which currenlty is configured to output sample on slot 7 and 8 on extended CODEC) change to:
Re: [Alsa-user] Terratec DMX XFire 1024 cs46xx module
Hi Richard, hi Benny(perhaps you have some idea what's going on ;-)) Richard Stevens wrote: [...] The problem is massively distorted sound. I get the driver loaded and there is some sound coming out of the card. But it's distorted. If I set very low levels for Master, PCM and DAC in alsamixer the distortion is not that bad but I have to set my amplifier to a really high level to even hear something. As soon as I set the levels higher distortion gets worse and worse until at some stage the right channel disappears and only very distorted sound comes out of the left channel. I thought about a broken card but it works perfectly with windows. same problem for me and my DMX XFire 1024 :-(. I think the distortion sounds like a DC offset introduced by the PCM channel. Do you also get popping noise when changing the PCM volume? It just sounds like changing the supply voltage of a running amplifier for me. Perhaps I can measure the DC level before the output capacitors or --perhaps even better-- at the input stage of the output amps? (But I don't know if the DAC used on this card can output DC voltage and if the LM386 amps are AC or DC coupled. I'll have to find a screwdriver and some time first.) Not only the PCM sound is distorted, but everything that goes through the mixer, I think. If you lower the PCM volume and playback e.g. a CD through the CD input of the card, then the quality is OK. This seems to be a problem with the XFire only, as none of the other cs46xx users has reported this problem. Perhaps it has something to do with the audio codec (the XFire uses one codec for 4 channels, other cs46xx use two codecs for this)? Or some GPIO? I have this problem with Mandrake 8.1 and 9.0. The problem also existed with the old DSP code (before Benny started hacking). Typing '/etc/init.d/alsa restart' (or 'force-restart' ??? I don't know exactly right now), solves the distortion problem for me every time. (But then the rights for the audio devices are gone for users, so I have to run apps as root or use the ./sndconfig script.) So it must be some initialization problem. Would it help to compare some /proc/asound/ outputs before/after restarting alsa? If so, which should I look at? Benny, thanks a lot for your new DSP code. Hardware mixing works great! I didn't test variable period sizes up to now because I'm using Mandrake RPMs at the moment. So, patching isn't possible right now, but I'll change back to tarballs or cvs soon :-). Btw, do you have any idea how to get 4 indep. channels working with this one codec? regards, fritz e. --- This sf.net email is sponsored by: To learn the basics of securing your web site with SSL, click here to get a FREE TRIAL of a Thawte Server Certificate: http://www.gothawte.com/rd524.html ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Terratec DMX XFire 1024 cs46xx module
Hi, I attached the outputs of 'cat /proc/asound/card0/*' and 'cat /proc/asound/card0/dsp/*' before and after reloading the snd-cs46xx module. After reloading the distortion and popping noises are gone. Does this help finding an error? Richard Stevens wrote: When they get loaded the second time after unloading it pops once more. confirmed If you reload them after that, that popping sound won't come again. So once the card is in a usable state, the pops are gone also. I have to correct one thing, too. Reloading the alsa drivers once solves the problem always now. I only have trouble reloading them if sound was in use before. Then sometimes some modules are marked in use even though I killed all apps that used them. Probably those were the cases where reloading didn't work as a workaround. seems so ... One other thing, there are two channels named IEC 958 in my alsa mixer. When I unmute the second one the card starts to generate a high frequency beep that can only be stopped by rebooting. No idea what that is ;) That's a problem with IEC 958 (=SPDIF) input (or was it output? btw: SPDIF in works without problem for 48kHz. It's quite easy to build your own optical SPDIF input for the XFire). Unmuting while playing back PCM causes the DSP to hang. I thought this issue was solved some time ago. In my Mandrake RPMs (they are alsa09rc2) the problem is not solved. Try a newer alsa version or don't touch this control while playing back PCM :-) or reload the module afterwards ... (But then the rights for the audio devices are gone for users, so I have to run apps as root or use the ./sndconfig script.) So it must be some Is Mandrake using devfs? Sorry, this problem occured with Mandrake 8.1 and tarballs only. I just tried to do '/etc/init.d/alsa restart' and realized there is no alsa script anymore! Now I used modprobe -r snd-cs46xx modprobe snd-cs46xx and only had to restore the mixer settings afterwards. regards, fritz e. proc_asound_outputs.tar.gz Description: application/gzip
Re: [Alsa-user] Terratec DMX XFire 1024 cs46xx module
Hi! Benny Sjostrand wrote: same problem for me and my DMX XFire 1024 :-(. I think the distortion sounds like a DC offset introduced by the PCM channel. Do you also get popping noise when changing the PCM volume? It just sounds like changing the supply voltage of a running amplifier for me. Perhaps I can measure the DC level before the output capacitors or --perhaps even better-- at the input stage of the output amps? (But I don't know if the DAC used on this card can output DC voltage and if the LM386 amps are AC or DC coupled. I'll have to find a screwdriver and some time first.) done. The DC level before the coupling capacitors at the inputs of the amps changes from 2.26V down to 0.03V when rising both PCM and master to maximum volume. The higher master and PCM level, the smaller the DC level and the more distorted the sound gets. When setting master to max and rising PCM level, in the end (at 0.03V) you don't hear any sound. The right channel reaches this state one step of the PCM mixer before the left channel. (There was some mail about slightly different levels right/left on this list, I think. Perhaps that's the reason for this slightly different behaviour of the channels?) After reloading the snd-cs46xx module, the DC level of both channels stays at 2.26V regardless of the mixer settings and the sound doesn't get distorted. So, what could be the difference after reloading the module? (Did you get my /proc/asound/... dumps?) Not only the PCM sound is distorted, but everything that goes through the mixer, I think. If you lower the PCM volume and playback e.g. a CD through the CD input of the card, then the quality is OK. This seems to be a problem with the XFire only, as none of the other cs46xx users has reported this problem. Perhaps it has something to do with the audio codec (the XFire uses one codec for 4 channels, other cs46xx use two codecs for this)? Or some GPIO? Is the sound from that comes from the SPDIF interface OK ? If I could answer this question, I would be happy. Then I would have an audio component with SPDIF input :-) (Not true! I've got this XFire soundcard...) Just would try to understand if the problem is digital, DSP related issue, or something analog, CODEC related ... Most soundcard got some kind of amplifier, that possibly is controlled by some GPIO, like the case of Santa Cruz, and Hercules GTXP. If Terratec wont release and technical specifications about the card, then you can try to play with the EPGIO hack in the cs46xx driver (compile alsa defining the CONFIG_SND_CS46XX_DEBUG_GPIO option) and try find out if any of the EGPIO0-7 can solve the problem. /Benny Before updating to Mandrake 9.0 I played around with the GPIO controls but didn't notice any change/improvement. At the moment I'm using the mdk-rpms, so I can't tests this in detail until I change back to cvs or tarball. Sorry! regards fritz e. --- This sf.net email is sponsored by: To learn the basics of securing your web site with SSL, click here to get a FREE TRIAL of a Thawte Server Certificate: http://www.gothawte.com/rd524.html ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: Re: [Alsa-user] cs46xx s/pdif: front/rear reversed?
Hi! You need to unmute the optical out. It shows up as a device with no volume controls or anything, all you can do is mute/unmute it. How do I do that? It's been a while since I've fiddled around with the settings... Would IEC 958 Output be the SPDIF output? How would I unmute it, as well as make it stereo (or enable more channels...)? Yes, IEC 958 Output is the SPDIF output. And you can only switch it on/off by unmuting/muting it. So a mono control is enough. I don't know how to do this using amixer, but you could try alsamixer, the console based 'graphical' ALSA mixer tool. Hitting space you can select your input source and hitting m you can mute/unmute channels. And there are some other keys... :) Have a look at man alsamixer. -- GMX - Die Kommunikationsplattform im Internet. http://www.gmx.net --- This sf.net email is sponsored by: OSDN - Tired of that same old cell phone? Get a new here for FREE! https://www.inphonic.com/r.asp?r=sourceforge1refcode1=vs3390 ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: Re: [Alsa-user] cs46xx s/pdif: front/rear reversed?
Hi! Just out of interest... Is S/PDIF support for that card recent? (i.e. in CVS) I inquired a couple of months ago, and was told that there wasn't S/PDIF support... I have exactly the same card. Yes, S/P-DIF ouput support is quite new. S/P-DIF input is even newer! (There are some other new features for cs46xx cards) You'll have to decomment two lines in ./alsa/alsa-driver/pci/cs46xx/Makefile to activate these new features: # Hack for new dsp flag EXTRA_CFLAGS = -DCONFIG_SND_CS46XX_NEW_DSP=1 CONFIG_SND_CS46XX_NEW_DSP=y fritz_e -- GMX - Die Kommunikationsplattform im Internet. http://www.gmx.net --- This sf.net email is sponsored by: OSDN - Tired of that same old cell phone? Get a new here for FREE! https://www.inphonic.com/r.asp?r=sourceforge1refcode1=vs3390 ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] how do i connect ALSA outputs to ALSA inputs?
Hi all, I asked this question several weeks before but did not succeed in solving my problem :-( is there a possibility to redirect audio output from one application (that outputs to some pcm device that I've defined in my .asoundrc) to a second pcm device so that I can record the sound from the second pcm device? (The output I'd like to record is 5 or 6 channel pcm.) I've got application1 that outputs to an alsa device and application2 that records from another alsa device. (If this helps: application1 is Anders Torger's BruteFIR -- I want to process several channels and use the output as input of my second application ...) Is there a possibility to connect output and input not electrically by a cable but by some alias/plugin/device/... definition in the .asoundrc file? (I've tried some time using aserver and the shm plugin, but no success...) Or -- any other ideas/solutions? Frans Ketelaars told me I could use JACK, but I didn't get my problem solved. I don't know if it's needed but http://jackit.sourceforge.net/ seems to offer all you want :) HTH, -Frans Can someone please tell me how I can solve my problem using JACK or how I can connect ALSA outputs to inputs any other way? I think this should be no big deal, but up to now I did not find any solution ... many thanks in advance, fritz_e -- GMX - Die Kommunikationsplattform im Internet. http://www.gmx.net --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] using copy plugin = file descriptor in bad state error
Hi,
I tried to get the copy plugin running using the following lines in my
.asoundrc:
pcm.cp_test {
type copy # Copy PCM
slave cp_slave # Slave name
}
pcm.pb {# playback device for copy plugin test
type hw
card 0
}
pcm_slave.cp_slave {
pcm pb
}
I can use the cp_test device for playback with aplay, i.e.
aplay -D cp_test some_wav_file.wav
works.
But using the cp_test device as output with BruteFIR(0.99), I get the
following error:
ALSA lib pcm_hw.c:384:(snd_pcm_hw_start) SNDRV_PCM_IOCTL_START failed: File
descriptor in bad state
Could not start audio: File descriptor in bad state.
Failed to start I/O module, aborting.
Using the pb device with BruteFIR *does* work (and of course with aplay,
too).
Is this an Alsa-, copyPlugin-, asoundrc- or BruteFIR-problem?
Here's some data about the redhat linux box used:
Advanced Linux Sound Architecture Driver Version 0.9.0rc1.
Compiled on Jun 14 2002 for kernel 2.4.9-13smp (SMP) with versioned symbols.
0 [card0 ]: ES1371 - Ensoniq AudioPCI
Ensoniq AudioPCI ES1371 at 0xc400, irq 5
1 [card1 ]: RME9652 - RME Digi9652 (Rev 1.5)
RME Digi9652 (Rev 1.5) at 0xda00, irq 11
Using card1 instead of card0, I get the same error.
However, I still don't see how the copy plugin could help me transferring
the data from an ALSA output to an ALSA input. Maybe the cp_slave gets the
audio data, but can I use this slave pcm as an ALSA input device for another
application?
Any ideas?
fritz_e
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[Alsa-user] problem with type file device (save audio to disk)
Hi all!
I've got some problems in writing a 5.1 audio stream from DVD (or a mpeg2
ripped from a DVD) to harddisk as one (or six) WAV file(s).
I tried to use a program called a52dec for this, but the audio files I get
seem to consist of 2 channels only (2ch downmix) when using the -owav option.
Using the -owavdolby option results in a currupt WAV file (at least, I
cannot read it in several programs that can read multichannel WAV).
So, my next idea was to define an alsa file device to capture the audio
output of XINE. In XINE I'm using the alsa PCM device surround5, as described in
my .asoundrc (see below).
I've set the sampling rate to 48kHz because I think this is the correct bit
rate of videoDVD audio tracks.
So, I get 6 audio channels written in one binary file (interleaved 16bit
intergers).
The only problem is:
There are bit errors in the audio stream (approx. every 0.3 seconds). The
5.1 audio output I get from my soundcard is ok!
Switching on/off resampling in XINE or specifying other sampling rates in
.asoundrc doen't help.
Perhaps it's a problem with the 5.1 audio decoder of XINE (it's liba52, I
think). I've noticed rising the CPU load up to 80%-100% (from 30%) when
switching on the 5.1 output in XINE.
This behaviour does not change when I use the PCM device called 'hammerfall'
(no writing to disk). Playing 2 channel sound only does not cause this high
CPU load.
So my questions are:
* Is the high CPU load a XINE problem?
* Did I define the alsa PCM device(s) correctly?
* Can I use a52dec to solve my problem? How do I have to use it?
* Is there another possibility to extract six descrete audio channels from a
mpeg2 file (from DVD) and save them to disk?
Any ideas?
Thanks for any answers,
f.ewaldt
.asoundrc:
pcm.hammerfall {
type plug
slave {
pcm{
type hw
card 1}
rate 48000
}
route_policy copy
ttable.0.1 0.1
ttable.1.2 0.1
ttable.2.3 0.1
ttable.3.4 0.1
ttable.4.5 0.1
ttable.5.6 0.1
}
pcm.surround5 {
type file
slave.pcm hammerfall
file /mnt/somedirectory/dvd/audio.raw
}
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