Re: [Alsa-user] Newbie question - cannot get sound to work in Ubuntu
Nigel, Thanks again. I wasn't trying to imply that you weren't helping, just asking whether people who really know what's going on inside Alsa look at the messages. Anyway, I tried your suggestion and it recreated asound.state. Exactly the same as the previous one (diff gives no changes), and alsamixer gives the same error message. By the way, the USB sound card is disconnected, so that's not what's messing things up. Best regards, Peter mailto:[EMAIL PROTECTED] www.ptoye.com - Monday, March 3, 2008, 8:04:17 PM, you wrote: On Monday 03 March 2008 18:12, Peter Toye wrote: Nigel, Thanks for this. I've looked at the modules and the only one that's missing is dialog. It seems that this is to make it easy to write scripts which put up dialogue boxes, so I wouldn't have thought that aplay would give an error message about not being able to find the default device if the module's missing. Of course, error messages and their causes aren't always correlated too well. /var/lib/alsa contains asound.state and nothing else. I've no idea asound.state is meant to contain, so I'm attaching it (assuming that this email system allows attachments) in case it's of use. It's a bit long to include in the text. Ok. Nice to see you have an asound.state. When you installed Gutsy Gibbon, this asound.state file was created, and contains the default mixer settings for alsa, and usually the sounds are muted for the sake of your ears. I use KDE, so would then open a Konsole (CLI) , and type alsamixer as user, and can unmute controls, set slider levels, and so on. Same goes I presume, if you use Gnome. Now it has been suggested to me in the past if I can't get access to alsamixer, to remove/rename the asound.state file which may be faulty. I'd suggest renaming asound.state to something like asound.state.old, then reboot, and a new asound.state will be created, then, depending on which desktop you are using, open a terminal/konsole, and type alsamixer. Does alsamixer now open? I also have a USB sound card which I originally tried attaching - got a bit further with it in that I was able to play but not record (which is what I want to use it for). But then I removed it so as not to confuse things, and reloaded Alsa to get a clean system. As a matter of interest, does anyone directly connected with the Alsa project ever reply to queries here, or is it just users? The half-sighted leading the blind as it were. Alsa developers do read the alsa-user postings, and regularly reply to users queries. I'm just a user, but try to help if I can. Just trying to help. Nigel. Best regards, Peter mailto:[EMAIL PROTECTED] www.ptoye.com - Sunday, March 2, 2008, 5:43:32 PM, you wrote: On Sunday 02 March 2008 14:53, Peter Toye wrote: James, Thanks for this. I am a member of the audio group, so problem there. The sound modules seem to be loaded OK. The relevant bit of the output from lsmod is: $ lsmod |grep snd snd_emu10k1_synth 8192 0 snd_emux_synth 35456 1 snd_emu10k1_synth snd_seq_virmidi 8064 1 snd_emux_synth snd_seq_midi_emul 7680 1 snd_emux_synth snd_emu10k1 137248 2 snd_emu10k1_synth snd_ac97_codec100644 1 snd_emu10k1 ac97_bus3200 1 snd_ac97_codec snd_pcm_oss44672 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm80388 3 snd_emu10k1,snd_ac97_codec,snd_pcm_oss snd_page_alloc 11400 2 snd_emu10k1,snd_pcm snd_util_mem5760 2 snd_emux_synth,snd_emu10k1 snd_hwdep 10244 2 snd_emux_synth,snd_emu10k1 snd_seq_dummy 4740 0 snd_seq_oss33152 0 snd_seq_midi9600 0 snd_rawmidi25728 3 snd_seq_virmidi,snd_emu10k1,snd_seq_midi snd_seq_midi_event 8448 3 snd_seq_virmidi,snd_seq_oss,snd_seq_midi snd_seq53232 9 snd_emux_synth,snd_seq_virmidi,snd_seq_midi_emul,snd_seq_dummy,snd_seq_o ss, snd_seq_midi,snd_seq_midi_event snd_timer 24324 3 snd_emu10k1,snd_pcm,snd_seq snd_seq_device 9228 8 snd_emu10k1_synth,snd_emux_synth,snd_emu10k1,snd_seq_dummy,snd_seq_oss,s nd_ seq_midi,snd_rawmidi,snd_seq snd54660 15 snd_emux_synth,snd_seq_virmidi,snd_emu10k1,snd_ac97_codec,snd_pcm_oss,sn d_m ixer_oss,snd_pcm,snd_hwdep,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd _seq _device soundcore 8800 1 snd $ So that looks OK. Peter Hi Peter. Your lsmod looks ok, and from your previous post /proc/asound/cards only shows one soundcard, so it's not like you have problems like I have, with also having a usb midi keyboard plugged in when booting. That said I know the workaround for that, so it's no problem for me. I have a bunch of distros installed, and have just booted up Kubuntu (Dapper, upgraded from Breezy). Which
Re: [Alsa-user] hda_intel no microphone
At Mon, 03 Mar 2008 22:03:13 +0100, Ferry Toth wrote: Hi Takashi, I'm sorry, please explain what is the HG version. See download page of www.alsa-project.org. You can pick up a daily snapshot tarball instead. Takashi - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] alsa-lib-1.0.16 compile error
At Mon, 3 Mar 2008 12:48:11 -0600 (CST),
Stephen Stocker wrote:
On Mon, 3 Mar 2008, Takashi Iwai wrote:
At Sat, 1 Mar 2008 03:58:16 -0600 (CST),
Stephen Stocker wrote:
Hi.
I'm running kernel 2.4.24-ck1, gcc 3.3.4. Trying to compile the
alsa-lib-1.0.16, I get the following error.
Does the patch below fix?
Takashi
diff -r 14ce0fc9a26d src/pcm/pcm_local.h
--- a/src/pcm/pcm_local.h Fri Feb 29 12:42:57 2008 +0100
+++ b/src/pcm/pcm_local.h Mon Mar 03 18:11:21 2008 +0100
@@ -944,13 +944,17 @@ typedef union snd_tmp_double {
/* get the current timestamp */
static inline void gettimestamp(snd_htimestamp_t *tstamp, int monotonic)
{
+#ifdef HAVE_CLOCK_GETTIME
if (monotonic) {
clock_gettime(CLOCK_MONOTONIC, tstamp);
} else {
+#else
struct timeval tv;
gettimeofday(tv, 0);
tstamp-tv_sec = tv.tv_sec;
tstamp-tv_nsec = tv.tv_usec * 1000L;
+#ifdef HAVE_CLOCK_GETTIME
}
+#endif
}
Hi,
I'm still getting an error, but slightly different after applying the
patch. I'm not sure how to shorten it here without losing some vital
message, so here's the complete error:
Looks like your glibc has no CLOCK_MONOTONIC definition although it
has clock_gettime() function. That's bad.
Try the patch below instead.
Takashi
diff -r 14ce0fc9a26d src/pcm/pcm_file.c
--- a/src/pcm/pcm_file.cFri Feb 29 12:42:57 2008 +0100
+++ b/src/pcm/pcm_file.cTue Mar 04 11:57:27 2008 +0100
@@ -469,7 +469,7 @@ int snd_pcm_file_open(snd_pcm_t **pcmp,
pcm-poll_fd = slave-poll_fd;
pcm-poll_events = slave-poll_events;
pcm-mmap_shadow = 1;
-#ifdef HAVE_CLOCK_GETTIME
+#if defined(HAVE_CLOCK_GETTIME) defined(CLOCK_MONOTONIC)
pcm-monotonic = clock_gettime(CLOCK_MONOTONIC, timespec) == 0;
#else
pcm-monotonic = 0;
diff -r 14ce0fc9a26d src/pcm/pcm_hw.c
--- a/src/pcm/pcm_hw.c Fri Feb 29 12:42:57 2008 +0100
+++ b/src/pcm/pcm_hw.c Tue Mar 04 11:57:27 2008 +0100
@@ -994,7 +994,7 @@ int snd_pcm_hw_open_fd(snd_pcm_t **pcmp,
if (SNDRV_PROTOCOL_INCOMPATIBLE(ver, SNDRV_PCM_VERSION_MAX))
return -SND_ERROR_INCOMPATIBLE_VERSION;
-#ifdef HAVE_CLOCK_GETTIME
+#if defined(HAVE_CLOCK_GETTIME) defined(CLOCK_MONOTONIC)
if (SNDRV_PROTOCOL_VERSION(2, 0, 9) = ver) {
struct timespec timespec;
if (clock_gettime(CLOCK_MONOTONIC, timespec) == 0) {
diff -r 14ce0fc9a26d src/pcm/pcm_local.h
--- a/src/pcm/pcm_local.h Fri Feb 29 12:42:57 2008 +0100
+++ b/src/pcm/pcm_local.h Tue Mar 04 11:57:27 2008 +0100
@@ -944,13 +944,17 @@ typedef union snd_tmp_double {
/* get the current timestamp */
static inline void gettimestamp(snd_htimestamp_t *tstamp, int monotonic)
{
+#if defined(HAVE_CLOCK_GETTIME) defined(CLOCK_MONOTONIC)
if (monotonic) {
clock_gettime(CLOCK_MONOTONIC, tstamp);
} else {
+#endif
struct timeval tv;
gettimeofday(tv, 0);
tstamp-tv_sec = tv.tv_sec;
tstamp-tv_nsec = tv.tv_usec * 1000L;
+#if defined(HAVE_CLOCK_GETTIME) defined(CLOCK_MONOTONIC)
}
+#endif
}
-
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Re: [Alsa-user] Newbie question - cannot get sound to work in Ubuntu
Anyway, I tried your suggestion and it
recreated asound.state. Exactly the same
as the previous one (diff gives no changes),
and alsamixer gives the same error message.
By the way, the USB sound card is disconnected,
so that's not what's messing things up.
Some distros do the alsactl save as part of the shutdown
sequence(/etc/rc?.d/K*). If it doesn't then once you deleted asound.state, it
should not be recreated. Not until you run alsactl save. At least that's how
it works in debian. I know this because if asound.state exists, then it
doesn't restore the mixer settings that were set and saved in some other mixer
like aumix. YMMV
I still think you're missing the /dev/'s though. Or maybe you just need the
/etc/modprobe.d/ configuration. Multiple cards can be tricky. From what I've
gathered, most of your errors are cannot open device, or device does not exist.
If you can't get past that, they you probably wont be able to do anything with
that device.
I have different cards, but this is how I set mine up. Usb audio does require
usb support. And I've been configuring mine custom-ish for a while. There's
probably an alsaconf in the alsa-utils package that sets this stuff up for you.
But I never got it to work for me back in the day. So I almost always set
things up long hand. Fortunately with usb sticks, that's just a cp or cut and
paste away.
/etc/modprobe.d/alsa_custom
#***
alias char-major-116 snd
alias char-major-14 soundcore
options snd major=116 cards_limit=4
options snd-atiixp index=0
options snd-atiixp-modemindex=1
options snd-usb-audio index=2
options snd-usb-audio index=3
alias snd-card-0 snd-atiixp
alias sound-slot-0snd-card-0
alias sound-service-0-0 snd-mixer-oss
alias sound-service-0-1 snd-seq-oss
alias sound-service-0-3 snd-pcm-oss
alias sound-service-0-8 snd-seqr-oss
alias sound-service-0-12 snd-pcm-oss
alias snd-card-1 snd-atiixp-modem
alias sound-slot-1snd-card-1
alias sound-service-1-0 snd-mixer-oss
alias sound-service-1-1 snd-seq-oss
alias sound-service-1-3 snd-pcm-oss
alias sound-service-1-8 snd-seqr-oss
alias sound-service-1-12 snd-pcm-oss
alias snd-card-2 snd-usb-audio
alias sound-slot-2snd-card-2
alias sound-service-2-0 snd-mixer-oss
alias sound-service-2-1 snd-seq-oss
alias sound-service-2-3 snd-pcm-oss
alias sound-service-2-8 snd-seqr-oss
alias sound-service-2-12 snd-pcm-oss
alias snd-card-3 snd-usb-audio
alias sound-slot-3snd-card-3
alias sound-service-3-0 snd-mixer-oss
alias sound-service-3-1 snd-seq-oss
alias sound-service-3-3 snd-pcm-oss
alias sound-service-3-8 snd-seqr-oss
alias sound-service-3-12 snd-pcm-oss
# ***
/home/user/.asoundrc
# ***
pcm.atiixp {
type hw
card 0
}
ctl.atiixp {
type hw
card 0
}
pcm.atiixp_modem {
type hw
card 1
}
ctl.atiixp_modem {
type hw
card 1
}
pcm.usb_audio2 {
type hw
card 2
}
ctl.usb_audio2 {
type hw
card 2
}
pcm.usb_audio3 {
type hw
card 3
}
ctl.usb_audio3 {
type hw
card 3
}
defaults.pcm.card 0
#
This is good enough to use the cards and have their drivers autoload when you
try to use them. If I want alsa native apps to use the non zero card, then I
mearly change the number on the defaults.pcm.card line. OSS type apps will
almost always default to 0, unless you use aoss. Not that much of an issue
since you can always change the index numbering to make whatever device be
device 0. The asoundrc defaults line is just a lot fewer lines to change, and
doesn't require restarting alsa. /etc/init.d/alsasound stop|start
Not that you have go the defaults route. Many existing apps let you specify
which card to use. By either it's hardware number or it's asoundrc label. $
mplayer -ao alsa:device=hw:0 (or device=atiixp). -D hw:0 for arecord. -d, or
-C or -P for jackd. And various gui preferences to help you utilize secondary
cards and friends.
HTH
-
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Re: [Alsa-user] RME AES-32 questions
James Courtier-Dutton wrote: John Sigler wrote: I have an RME AES-32 PCI board. http://www.rme-audio.de/en_products_hdsp_aes32.php When I boot, every channel has its volume set to 0. http://lxr.linux.no/linux/sound/pci/rme9652/hdspm.c#L3396 http://lxr.linux.no/linux/sound/pci/rme9652/hdspm.c#L965 Apparently, it's not the board's firmware, it's the device driver which mutes every channel at initialization. Does anybody know why this is done? It was done to protect one's ears. Could you expand on your explanation? Intuitively, I would have initialized every input channel and output channel to leave the signal as is (what you refer to as 0 dB). I used amixer to change the volume. Simple mixer control 'Chn',1 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 65536 Mono: 32768 [50%] What is the meaning of the level? Does 100% mean the signal is left unmodified? and 50% means the amplitude is multiplied by 0.50? http://lxr.linux.no/linux/sound/pci/rme9652/hdspm.c#L377 #define UNITY_GAIN 32768 #define MINUS_INFINITY_GAIN 0 So, volume = 32768 means leave the signal as is and volume = 0 means mute the channel. Is that correct? What would volume = 16384 and volume = 49152 mean? (What is the scale?) There is no scale there. It can have a range of value between 0 and 65536 32768 being about 50% of 65536. As a matter of fact, 32768 is exactly 50% of 65536, but I'm not sure I understand the point you're making. The driver source code seems to imply that 32768 is special in that it is named UNITY_GAIN. That would seem to indicate that this volume level corresponds to 0 dB, wouldn't it? Now that we have these dB gain levels, we could potentially set all mixer controls to 0dB, i.e. not gain and not attenuation, with only the master output control being set to a lower value so as to avoid ears breaking. I don't know where the master output control is. Do you see it in the outputs of 'amixer' or 'amixer contents' I provided in one of my previous messages? Regards. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] RME AES-32 questions
On 04/03/2008, John Sigler [EMAIL PROTECTED] wrote: James Courtier-Dutton wrote: John Sigler wrote: I have an RME AES-32 PCI board. http://www.rme-audio.de/en_products_hdsp_aes32.php When I boot, every channel has its volume set to 0. http://lxr.linux.no/linux/sound/pci/rme9652/hdspm.c#L3396 http://lxr.linux.no/linux/sound/pci/rme9652/hdspm.c#L965 Apparently, it's not the board's firmware, it's the device driver which mutes every channel at initialization. Does anybody know why this is done? It was done to protect one's ears. Could you expand on your explanation? Some time ago, there was no way to tell, from user space, if 100% or 50% was 0db, -50dB or +50dB. As a result, the safest value to use was the minimum value being either 0% or mute. That has changed with alsa 1.0.16, as dB gain information is now available. Intuitively, I would have initialized every input channel and output channel to leave the signal as is (what you refer to as 0 dB). As a matter of fact, 32768 is exactly 50% of 65536, but I'm not sure I understand the point you're making. The driver source code seems to imply that 32768 is special in that it is named UNITY_GAIN. That would seem to indicate that this volume level corresponds to 0 dB, wouldn't it? But the value of 50% does not correspond to 0 dB for all sound cards. Now that we have these dB gain levels, we could potentially set all mixer controls to 0dB, i.e. not gain and not attenuation, with only the master output control being set to a lower value so as to avoid ears breaking. I don't know where the master output control is. Do you see it in the outputs of 'amixer' or 'amixer contents' I provided in one of my previous messages? version 1.0.16 of amixer will have the dB values. You also have to have version 1.0.16 of alsa-driver and alsa-lib. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] JAVA 32 bit and a 64 bit OS ?
Hello, I have a 64bit machine with a 64bit OS. I am trying to run a java program using the 32bit java (the real time version). Mostly it works fine, I get wonderful low latency audio, but opening a midi device input crashes the program with an assertion error in rawmidi.c (line 264) I am guessing that the problem is calling a 64bit library from 32bit code? Will compiling alsa-lib with the -m32 option fix this if so how do I go about doing this so the java will use the 32bit library. Any other advice would be welcome. I could also try installing a 32bit version of ubunutu but I would prefer to find a solution that keeps the 64bit OS. Paul. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] RME AES-32 questions
At Tue, 4 Mar 2008 15:03:15 +, James Courtier-Dutton wrote: Now that we have these dB gain levels, we could potentially set all mixer controls to 0dB, i.e. not gain and not attenuation, with only the master output control being set to a lower value so as to avoid ears breaking. I don't know where the master output control is. Do you see it in the outputs of 'amixer' or 'amixer contents' I provided in one of my previous messages? version 1.0.16 of amixer will have the dB values. You also have to have version 1.0.16 of alsa-driver and alsa-lib. hdspm driver still doesn't support TLV dB information. In addition, rme9652/hdsp/hdspm drivers use control elements in a non-standard way for indirect accessing, so each mixer value isn't visible in amixer or alsactl. Takashi - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] RME AES-32 questions
James Courtier-Dutton wrote: Some time ago, there was no way to tell, from user space, if 100% or 50% was 0db, -50dB or +50dB. As a result, the safest value to use was the minimum value being either 0% or mute. That has changed with alsa 1.0.16, as dB gain information is now available. I'm not sure where user space fits in. I'm talking about the sound/pci/rme9652/hdspm.c driver which mutes every channel at initialization by calling all_in_all_mixer(hdspm, 0); cf. http://lxr.linux.no/linux/sound/pci/rme9652/hdspm.c#L3396 AFAIU, the driver knows what level corresponds to 0 db. But the value of 50% does not correspond to 0 dB for all sound cards. I'm not discussing all sound cards, I'm discussing the sound cards driven by hdspm. Is it safe to assume that, for the cards driven by hdspm, level 32768 corresponds to 0 dB? Regards. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] RME AES-32 questions
Takashi Iwai wrote: hdspm driver still doesn't support TLV dB information. In addition, rme9652/hdsp/hdspm drivers use control elements in a non-standard way for indirect accessing, so each mixer value isn't visible in amixer or alsactl. Hello Takashi, Can you tell whether the hdspm driver is able to capture raw AES frames, instead of just the PCM payload inside AES frames? Is there any sample code on how to do that? Regards. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] AC'97 0 does not respond - RESET
hello, im using debian with kernel 2.6.18 and alsa 1.0.16 and when i start the pc i have that message : AC'97 0 does not respond - RESET i need to run alsaconf after every reboot... i have a SIS soundcard and always runs under linux - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] RME AES-32 questions
Takashi Iwai wrote: At Tue, 4 Mar 2008 15:03:15 +, James Courtier-Dutton wrote: Now that we have these dB gain levels, we could potentially set all mixer controls to 0dB, i.e. not gain and not attenuation, with only the master output control being set to a lower value so as to avoid ears breaking. I don't know where the master output control is. Do you see it in the outputs of 'amixer' or 'amixer contents' I provided in one of my previous messages? version 1.0.16 of amixer will have the dB values. You also have to have version 1.0.16 of alsa-driver and alsa-lib. hdspm driver still doesn't support TLV dB information. In addition, rme9652/hdsp/hdspm drivers use control elements in a non-standard way for indirect accessing, so each mixer value isn't visible in amixer or alsactl. So, in that case mapping values from 0 to 65536 to gain values is not yet done for you. You will have to find some other method to convert them to gain values, and as the driver does not document the mapping, you probably have to use trial and error or look at a datasheet if they exist. James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Newbie question - cannot get sound to work in Ubuntu
James, Thanks. My comments are below. Best regards, Peter mailto:[EMAIL PROTECTED] www.ptoye.com - Sunday, March 2, 2008, 11:52:57 PM, you wrote: I'm a bit confused about the rest of your comment. The only file called devfs is a directory which has two subdirectories, neither of which seems to have anything interesting in it (one is empty). And I can't find a file called snddevices anywhere, but might have mistyped (I'm not currently on the Linux machine and can't get to it to check up). In the alsa-driver package (alsa-project.org) there's a script named snddevices. It's in the root path of that tarball. Basically it creates the devices your drivers/applications need to access to do sound. Many distros set this up for you if you install the needed packages. But assuming a more basic LFS approach, you need to set them up yourself. You seem to be missing those devices. Hence the snd_ctl_open error. Not sure what you mean by LFS. Don't forget I'm a Linux newbie. ls -al /dev/* | grep -i audio | wc -l 23 I get 25 here. Would a list help? None of them is called default. lsmod | grep -i snd | wc -l 21 I get 23 here, which I think I posted in an earlier mail. That's what mine lists. Various /dev/ devices. /dev/mixer* /dev/sequencer* /dev/dsp* /dev/audio* If devfs or udev didn't create these for you, then you're left with the old mknod methods. Which the script alsa-driver/snddevices uses to create the devices. pgrep udev pgrep devfs (if you don't get a pid number, then it's not ) udev is running, devfs isn't. Are they both needed? I can't even find a man page for devfs. There's a file /etc/devfs which just has two directories in it: conf.d and devices.d. conf.d has one file: nvidia-kernel-ufc which doesn't look useful, and devices.d is empty. Otherwise you may just need to: apt-get install alsa alsa-base (and various other alsa* packages.) I did this earlier to clean out anything left by my attaching the USB card. (I have a nasty feeling that the mail that I mentioned this in is awaiting the attention of the mods, as I attached the asound.state file which is a bit big.) Can't see much point in doing it again! Unless you can. Or run the snddevices script. find / -iname '*snddevice*' Not on my machine. This of course may be the problem. HTH As you can see, a bit. But it's a bit uphill. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Newbie question - cannot get sound to work in Ubuntu
On Tuesday 04 March 2008 11:40, Peter Toye wrote: Nigel, Thanks again. I wasn't trying to imply that you weren't helping, just asking whether people who really know what's going on inside Alsa look at the messages. Anyway, I tried your suggestion and it recreated asound.state. Exactly the same as the previous one (diff gives no changes), and alsamixer gives the same error message. By the way, the USB sound card is disconnected, so that's not what's messing things up. Best regards, Peter Hi Peter. I've spent most of the afternoon (as I have nothing better to do) googling your problem, and there are a few things you can try. First though, do you have a live cd of Gutsy Gibbon, or Knoppix that you can bootup with, and see if you have the sounds working? Now moving on to my googling stuff. 1: Your audigy live card (circa 2000/1) is presumably a plugin PCI card, and not an onboard one. Yes? I ask because I wondered if you were using a Dell machine, and know that Dell have audigy live cards on some of their mobo's, which use the snd-emu10k1x driver, rather than the snd-emu10k1 one. 2: If your audigy live card is a plugin PCI one, do you also have an onboard soundcard on your machine? Again, I ask because I saw a reference to snd-via82xx in /etc/modprobe.d in one of your replies. Have a look in your BIOS, to see if you can disable the onboard card if it exists. I have an onboard card on the machine that has the audigy2 soundblaster, but physically disabled it with jumpers on the mobo. Would you provide the output of lspci -v. Just the stuff for soundcards will be enough. It was also suggested to reinstall the libasound5 package, and someone got their sound working after doing that. I looked on my Dapper install which has libasound2. Don't try to remove it, then reinstall it, as it want's to remove half the OS. In Synaptic, just do a reinstall of libasound5, which doesn't mess with any other installed packages. I think that alsaconf is being deprecated, and personally havn't had to run it for ages, but if it is available on Gutsy Gibbon, it may be worth running it, and setting up your soundcard again, as below. (something someone else suggested while googling) sudo alsaconf Do you have pulseaudio installed on Gutsy Gibbon? I don't think it's installed as default, but is default on my F8 install, and caused sound problems for me. If it is installed, to disable it, you only have to remove the package alsa-plugins-pulseaudio. With it enabled, it disables alsamixer, but saying that, when trying to start alsamixer, the errors specifically mention pulseaudio, and not like your errors. Alsamixer when opened shows the default card (card0), but have a look at the manpages for alsamixer. I now open these in a webbrowser as man:alsamixer. To start alsamixer for other cards you have the -c option. For example, alsamixer -c0 brings up the default cards mixer settings. alsamixer -c1 brings up the mixer settings for card 1, and so on. That's enough for now. Quite why you're having such problems with an emu10k1 based card I don't know. I've used my audigy2 soundblaster since Fedora core1 (2003), and apart from having to set options for my usb midi keyboard in /etc/modprobe.conf have had no problems. All a bit puzzling. Be nice to see your problem resolved though. Nigel. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] alsa-lib-1.0.16 compile error
On Tue, 4 Mar 2008, Takashi Iwai wrote: Hi, I'm still getting an error, but slightly different after applying the patch. I'm not sure how to shorten it here without losing some vital message, so here's the complete error: Looks like your glibc has no CLOCK_MONOTONIC definition although it has clock_gettime() function. That's bad. Try the patch below instead. Takashi This fixes it, and alsa-lib now compiles again with no problems. I did check, and my glibc is 2.3.2, from Slackware 10. I probably should have included that in my original post, but all is working well now. :) Thanks a lot, and take care, Steve diff -r 14ce0fc9a26d src/pcm/pcm_file.c --- a/src/pcm/pcm_file.c Fri Feb 29 12:42:57 2008 +0100 +++ b/src/pcm/pcm_file.c Tue Mar 04 11:57:27 2008 +0100 snip - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] About my sound card...
Hello, I am a PARDUS LINUX user. I upgraded latest 1.0.16 alsa-driver. But, my sound card seems still unsupported by alsa. In the alsa-project web page ıt is said that support for this model is under development. After upgrade, I wrote in console #alsaconf but still it comes the same error message that my sound card is not supported. The question is when will it be supported. In year or years? Does anyone have an idea about this matter? Thank you. -- Server ACİM - Besteci (Composer) - Üniversite Öğretim Üyesi(University-College Academic Staff) - PARDUS kullanıcısı ve katkıcısı (PARDUS Linux user contributor) http://www.serveracim.net attachment: ESI-MAYA44-PCI.png- This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] About my sound card...
Hi! WHICH is your soundcard? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] About my sound card...
On Tue, Mar 4, 2008 at 6:07 PM, Julien Claassen [EMAIL PROTECTED] wrote: Hi! WHICH is your soundcard? Kindest regards Julien Maya44 PCI. It was described in a PNG attachment clipped from an ALSA web screenshot ;-) Here's the latest, from a recent alsa ML posting: Can't wait till maya44 support will be added to the official alsa tree. It will help with the development of the new driver. I don't think there will be much users until the driver is merged to kernel tree because distros don't use always alsa-driver tree. Anyway, the Maya44 support is still pending. We need some cleanups before merting Maya44 support code. I know it is being actively worked on. I would guess a few weeks. Lee - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] RME AES-32 questions
At Tue, 04 Mar 2008 17:57:15 +, James Courtier-Dutton wrote: Takashi Iwai wrote: At Tue, 4 Mar 2008 15:03:15 +, James Courtier-Dutton wrote: Now that we have these dB gain levels, we could potentially set all mixer controls to 0dB, i.e. not gain and not attenuation, with only the master output control being set to a lower value so as to avoid ears breaking. I don't know where the master output control is. Do you see it in the outputs of 'amixer' or 'amixer contents' I provided in one of my previous messages? version 1.0.16 of amixer will have the dB values. You also have to have version 1.0.16 of alsa-driver and alsa-lib. hdspm driver still doesn't support TLV dB information. In addition, rme9652/hdsp/hdspm drivers use control elements in a non-standard way for indirect accessing, so each mixer value isn't visible in amixer or alsactl. So, in that case mapping values from 0 to 65536 to gain values is not yet done for you. You will have to find some other method to convert them to gain values, and as the driver does not document the mapping, you probably have to use trial and error or look at a datasheet if they exist. 32768 = unity gain is found in hdspm.txt. But the actual dB level to be applied for other values isn't mentioned there. Takashi - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
