Re: [Alsa-devel] alsa-utils from CVS
On Sun, 15 Jun 2003, Erik de Castro Lopo wrote: On Sun, 15 Jun 2003 12:27:54 +1000 Erik de Castro Lopo [EMAIL PROTECTED] wrote: Hi all, I have managed to download alsa-driver, alsa-kernel and als-lib from CVS but I can't get alsa-utils. [EMAIL PROTECTED] cvs -d:pserver:[EMAIL PROTECTED]:/cvsroot/alsa login Logging in to :pserver:[EMAIL PROTECTED]:2401/cvsroot/alsa CVS password: [EMAIL PROTECTED] cvs co alsa-utils cvs server: cannot find module `alsa-utils' - ignored cvs [checkout aborted]: cannot expand modules Has anybody else managed this? It looks like SourceForge will not accept cvs co alsa-utils but will accept: cvs -z3 -d:pserver:[EMAIL PROTECTED]:/cvsroot/alsa co alsa-utils Its probably insisting that I use compression over the net. 'cvs login' only creates repository-password entries (in ~/.cvspass), so you have to specify the full repository address for checkout again, because CVS/* files does not exist at the moment (thus cvs does not know, which repository is root). Jaroslav - Jaroslav Kysela [EMAIL PROTECTED] Linux Kernel Sound Maintainer ALSA Project, SuSE Labs --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] unwanted echo of audio input
Alsa Folks, I am using ALSA 9.0 on a Dell Inspiron 8200 which uses the Intel 82801CA-ICH3 card and CS4205 chip set. I am writing a program that simulaneaously samples and plays audio. The problem I am having is that everything that is captured by the microphone is automatically echoed to the speaker. I have tried all of the mixer settings I can think of using alsamixer and cannot make the problem go away. I have also tried to fix the problem from a c program using the OSS-style commands fd = open(dev/mixer,O_RDWR); vol = 0; ioctl(fd,MIXER_WRITE(SOUND_MIXER_IMIX),vol); but without success --- the volume value is set to 0, as it should be, but no change in what actually happens. (All of my other sound-related code uses the ALSA API). I am thinking that it is possible to avoid this echoing since, using Windows, I was able to play from the cd and record without a similar mixing of input and output on the same machine. I would really appreciate any help with this. It might not be relevant, but the program I'm working on is an automatic accompaniment system. The input is the soloist's musical sound signal and the output is the orchestra recording, warped to follow the soloist. There are some nice examples on my the web page: http://fafner.math.umass.edu/music_plus_one Christopher Raphael --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] sequencer: handling non-registered parameter numbers....
On Sunday 15 June 2003 18.16, Joern Nettingsmeier wrote:
[...]
/*{SND_SEQ_EVENT_NONREGPARAM, extra_decode_nrpn},*/
/*{SND_SEQ_EVENT_REGPARAM, extra_decode_rpn},*/
};
which makes me think it might not yet be implemented
I'm using NRPNs to control the mixer in the current development
version of Audiality, and they appear to be working just fine, and
just like one would expect.
any hints or fine manuals around ?
Do you need one? :-)
What you get is plain control, value tuples, and that's all there is
to it. (Well, it's all I *want* anyway, as I don't want to do stuff
that's outside the MIDI spec...)
Here's some code, which isn't heavily tested, but does seem to do the
job:
static void alsaseq_read(struct A_device *dev, unsigned frames)
{
int more = 1;
snd_seq_event_t *ev;
ALSASEQ_data *d = (ALSASEQ_data *) dev-driver_data;
dev-read_ms-time = aev_timer;
while(more)
{
more = snd_seq_event_input(d-seq_handle, ev);
if(more 0)
break;
switch (ev-type)
{
case SND_SEQ_EVENT_CONTROLLER:
dev-read_ms-control(
ev-data.control.channel,
ev-data.control.param,
ev-data.control.value);
break;
case SND_SEQ_EVENT_NONREGPARAM:
dev-read_ms-nrpn(ev-data.control.channel,
ev-data.control.param,
ev-data.control.value);
break;
...
case SND_SEQ_EVENT_PITCHBEND:
dev-read_ms-bend(ev-data.control.channel,
ev-data.control.value);
break;
}
snd_seq_free_event(ev);
}
}
//David Olofson - Programmer, Composer, Open Source Advocate
.- The Return of Audiality! .
| Free/Open Source Audio Engine for use in Games or Studio. |
| RT and off-line synth. Scripting. Sample accurate timing. |
`--- http://audiality.org -'
--- http://olofson.net --- http://www.reologica.se ---
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Re: [Alsa-devel] sequencer: handling non-registered parameter numbers....
hello alsa gurus ! i have bought a peavey studiomix midi controller on ebay, and it sends NRPN messages (non-registered parameter numbers). when i move a slider, it sends the slider number encoded in 98 and 99 and the value in the DATA ENTRY controllers 6 and 38. i would like to map these to ordinary midi controllers, or better yet, get nrpn support into ardour. how do i get nrpn controller values from the alsa sequencer without having to parse the individual events and put them together by hand ? ardour doesn't use the sequencer. and i don't consider the nrpn messages any different from any other controller. from libmidi++'s persepective, there are 127 controller ID's, each with a value. whatever standard mapping they may have to gain, pan, or nrpn is completely ignored. 14 bit value support is almost impossible to provide: the midi spec is just ridiculous for that. --p --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] mtpav revisited.. MTC/SMPTE
Okay.. I don't know if much work has been done with ALSA or whether MusE has improved it's handling of multiple ports but the mtpav is definitely useable for all areas except...MTC/SMPTE..It just doesn't seem to lock on to a signal properly.. (the mtpav locking on to a signal sent from the computer) It will however successfully pass a signal through to This occurs in both MusE and ardour but I'm kind of ignoring the ardour results as I can't find where the frame rate is setup.. I haven't used any other clients.. I ahve to check with MusE list to see if MTC is fully implemented I don't really use MTC, but since I've got the equipment I am happy to test it.. cheers -- Allan Klinbail [EMAIL PROTECTED] --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] mtpav revisited.. MTC/SMPTE
This occurs in both MusE and ardour but I'm kind of ignoring the ardour results as I can't find where the frame rate is setup.. I haven't used any other clients.. I ahve to check with MusE list to see if MTC is fully implemented if you mean audio frame rate, ardour doesn't set it - its up to JACK. if you mean the SMPTE frame rate, its in the session state file. there is no way to set the value from the GUI at this time. --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp patch
Paul,
I've been trying for the last day or so to get some sound out of the
card. Still no luck. The setup does work fine when I boot into Windows.
I've certainly had a few problems on this end, like getting
/etc/asound.state into a funny configuration that had both the on-board
Via chipset and the HDSP 9652 in it. That's fixed, but still no sound.
I'm running as root. I've tried both Jack and straight Alsa with
aplay and alsaplayer. Everything acts like I should be getting sound,
but I don't. The Alsa drivers appear to be loaded. Restarting Alsa looks
pretty normal.
alsamixer says everything is turned up to 30. 'M' doesn't seem to
mute or unmute and channels for this card.
Can you clarify - do I need to make any 'connections' through the
HDSP 9652 to get the alsa_pcm:playback_1/2 to be enabled and supplying
audio to my amp? If so, what commands are you using?
I'm attaching asound.state, .asoundrc and a little more info. Let me
know what else you want to see.
Thanks for any pointers you can provide.
Cheers,
Mark
Wizard root # lsmod
Module Size Used byNot tainted
snd-hdsp 32556 3
snd-rawmidi15040 0 [snd-hdsp]
snd-seq-device 4352 0 [snd-rawmidi]
snd-pcm64928 2 [snd-hdsp]
snd-timer 15876 0 [snd-pcm]
snd-hwdep 5216 0 [snd-hdsp]
snd32836 1 [snd-hdsp snd-rawmidi snd-seq-device
snd-pcm snd-timer snd-hwdep]
radeon107972 1
agpgart11920 3 (autoclean)
ide-cd 27080 0 (autoclean)
cdrom 25984 0 (autoclean) [ide-cd]
snd-page-alloc 5404 0 [snd-pcm]
snd-hammerfall-mem 1920 0 [snd-hdsp]
Wizard root #
Wizard root # cat /proc/asound/card0/hdsp
RME HDSP 9652 (Card #1)
Buffers: capture df00 playback dee0
IRQ: 17 Registers bus: 0xe880 VM: 0xe08e6000
Control register: 0x10080b3
Status register: 0x2043088
Status2 register: 0x8041
FIFO status: 0
MIDI1 Output status: 0xff00
MIDI1 Input status: 0xff5e
MIDI2 Output status: 0xff00
MIDI2 Input status: 0xff4b
Buffer Size (Latency): 128 samples (2 periods of 512 bytes)
Hardware pointer (frames): 0
Passthru: no
Line out: on
Firmware version: 1
Sample Clock Source: Internal 44.1 kHz
Preferred Sync Reference: ADAT1
AutoSync Reference: ADAT1
AutoSync Frequency: 44100
System Clock Mode: Master
System Clock Frequency: 44100
IEC958 input: Internal
IEC958 output: Coaxial only
IEC958 quality: Consumer
IEC958 emphasis: off
IEC958 NonAudio: off
IEC958 sample rate: Error flag set
ADAT1: Sync
ADAT2: No Lock
ADAT3: No Lock
SPDIF: No Lock
Word Clock: No Lock
ADAT Sync: No Lock
Wizard root #
On Fri, 2003-06-13 at 21:55, Paul Davis wrote:
this patch fixes some basic problems with the hdsp driver with respect
to the hdsp9652 card. it also cleans up a few minor issues with naming
in the driver, and slightly rationalizes initialization to involve
the minimum of special-casing for the hdsp9652.
the basic problem with the hdsp9652 was related to 8 bit versus 32 bit
offsets when addressing the mixer memory. once this was fixed,
everything worked. this driver continues to work fine on my
pci+digiface unit as well.
my apologies for this taking so long - it has taken a long time to ask
RME the right question, and quite a long time to get the
answer. once i got down to it, the fix took 5 minutes!
now we just need to solve the multiface initialization problems :(
--p
state.'' {
control.1 {
comment.access 'read write'
comment.type IEC958
iface PCM
name 'IEC958 Playback Default'
value
''
}
control.2 {
comment.access 'read write inactive'
comment.type IEC958
iface PCM
name 'IEC958 Playback PCM Stream'
value
''
}
control.3 {
comment.access read
comment.type IEC958
iface MIXER
name 'IEC958 Playback Con Mask'
value
Re: [Alsa-devel] mtpav revisited.. MTC/SMPTE
On Mon, 2003-06-16 at 05:25, Paul Davis wrote: This occurs in both MusE and ardour but I'm kind of ignoring the ardour results as I can't find where the frame rate is setup.. I haven't used any other clients.. I ahve to check with MusE list to see if MTC is fully implemented if you mean audio frame rate, ardour doesn't set it - its up to JACK. if you mean the SMPTE frame rate, its in the session state file. there is no way to set the value from the GUI at this time. Thanks Paul, I meant SMPTE frame rate. cheers Allan -- Allan Klinbail [EMAIL PROTECTED] --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp patch
On Sun, 2003-06-15 at 19:42, Paul Davis wrote: RME HDSP 9652 (Card #1) Buffers: capture df00 playback dee0 IRQ: 17 Registers bus: 0xe880 VM: 0xe08e6000 Control register: 0x10080b3 You don't have the correct version of the driver. It would print: RME Hammerfall HDSP 9652 (Card #1) Buffers: capture f700 playback f6e0 IRQ: 11 Registers bus: 0xfebb VM: 0xf88af000 Control register: 0x10080f9 Control2 register: 0x800 notice the extra Control2 register printout. Cool. Something to look for anyway 15 minutes later. Bingo! OK, so the new driver hadn't gotten moved to the right place. It seems to be there now. I'm getting sound, but it's full volume and I don't seem to be able to turn it down. (And I only have a few more minutes before my kid goes to sleep. Or tries to...) ;-) I have alsamixer up and running, and the volumes turned down to 6 and it's still screaming loud. Is this like the old driver where the mixer didn't work at all? Or have I not set the right things? The other thing I notice is I only seem to be able to set 24 values in my little script to set volumes. The driver I just replaced allowed me to set all 26. OK, so a lot of progress, but I need to be able to reduce the volume badly!!! What can I send you to see if it's my problem? Also, please answer - do I need to set routing paths through the HDSP FPGA to get the mixer working? Can you supply a script to do that if it's necessary? Thanks, Mark --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp patch
On Sun, 2003-06-15 at 20:26, Mark Knecht wrote: Cool. Something to look for anyway 15 minutes later. Bingo! OK, so the new driver hadn't gotten moved to the right place. It seems to be there now. I'm getting sound, but it's full volume and I don't seem to be able to turn it down. (And I only have a few more minutes before my kid goes to sleep. Or tries to...) ;-) I have alsamixer up and running, and the volumes turned down to 6 and it's still screaming loud. Is this like the old driver where the mixer didn't work at all? Or have I not set the right things? The other thing I notice is I only seem to be able to set 24 values in my little script to set volumes. The driver I just replaced allowed me to set all 26. OK, so a lot of progress, but I need to be able to reduce the volume badly!!! What can I send you to see if it's my problem? Also, please answer - do I need to set routing paths through the HDSP FPGA to get the mixer working? Can you supply a script to do that if it's necessary? Thanks, Mark BTW: Wizard rme9652 # cat /proc/asound/card0/hdsp RME Hammerfall HDSP 9652 (Card #1) Buffers: capture de40 playback de20 IRQ: 17 Registers bus: 0xe880 VM: 0xe4cf7000 Control register: 0x10080de Control2 register: 0x800 Status register: 0x2040008 Status2 register: 0x8061 FIFO status: 0 MIDI1 Output status: 0xff00 MIDI1 Input status: 0xff00 MIDI2 Output status: 0xff00 MIDI2 Input status: 0xff00 Buffer Size (Latency): 8192 samples (2 periods of 32768 bytes) Hardware pointer (frames): 0 Passthru: no Line out: on Firmware version: 1 Sample Clock Source: Internal 48 kHz Preferred Sync Reference: ADAT1 AutoSync Reference: ADAT1 AutoSync Frequency: 48000 System Clock Mode: Master System Clock Frequency: 48000 IEC958 input: Internal IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 NonAudio: off IEC958 sample rate: Error flag set ADAT1: Sync ADAT2: No Lock ADAT3: No Lock SPDIF: No Lock Word Clock: No Lock ADAT Sync: No Lock Wizard rme9652 # --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp patch
I have alsamixer up and running, and the volumes turned down to 6 and it's still screaming loud. Is this like the old driver where the mixer didn't work at all? Or have I not set the right things? no, the mixer works, but unfortunately it appears that i didn't test enough. the code still appears to be not quite right - there's a tricky detail that you can only write 32 bit values to the mixer, but each mixer element is 16 bits, so you always have to write the one you want to modify, plus its neighbour. looks like i don't have that quite right yet. The other thing I notice is I only seem to be able to set 24 values in my little script to set volumes. The driver I just replaced allowed me to set all 26. i'll check on that. probably just a mistake on my part about the number of channels (i tend to forget the s/pdif outs). Also, please answer - do I need to set routing paths through the HDSP FPGA to get the mixer working? no. the mixer is (dis|en)abled by flipping a control register bit. if its on, its on. what needs work is the hdsp_write_gain() function. --p --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp patch
On Sun, 2003-06-15 at 20:42, Paul Davis wrote:
I have alsamixer up and running, and the volumes turned down to 6 and
it's still screaming loud. Is this like the old driver where the mixer
didn't work at all? Or have I not set the right things?
no, the mixer works, but unfortunately it appears that i didn't test
enough. the code still appears to be not quite right - there's a
tricky detail that you can only write 32 bit values to the mixer, but
each mixer element is 16 bits, so you always have to write the one you
want to modify, plus its neighbour. looks like i don't have that
quite right yet.
OK, well I'll stay tuned for possible patches and testing whatever you
need. Thanks.
If it makes any difference in simplifying your initial testing, I have
my speakers hooked to the HDSP9652 ADAT1 port, using channels 1 2. I'm
not using any other channels in that group, or any other ADAT outputs as
of yet.
If you think some other outputs do work (in terms of modifying the
volumes, let me know and I'll switch to them.
I don't suppose there is any .asoundrc magic that could reduce the
volume automagically in Alsa itself instead of in the driver? (A JAck
volume control?) ;-)
The other thing I notice is I only seem to be able to set 24 values in
my little script to set volumes. The driver I just replaced allowed me
to set all 26.
i'll check on that. probably just a mistake on my part about the
number of channels (i tend to forget the s/pdif outs).
I figured as much. No problem right now.
Also, please answer - do I need to set routing paths through the HDSP
FPGA to get the mixer working?
no. the mixer is (dis|en)abled by flipping a control register bit. if
its on, its on. what needs work is the hdsp_write_gain() function.
--p
OK, I was just remembering Roger Williams telling my some stuff back in
January when I was first trying to get this working (when we didn't know
that the volume controls didn't work) about needing to break
connections. He sent a little script file for the Digiface, but I was
not sure if this was required, or just something he kept around for test
purposes:
#!/bin/bash
ADAT1=0 1 2 3 4 5 6 7
ADAT2=8 9 10 11 12 13 14 15
ADAT3=16 17 18 19 20 21 22 23
SPDIF=24 25
function disconnect () {
for output in $@; do
input=0
while [ $input -le 25 ]; do
echo -n .
amixer cset numid=5 $input,$output,0 /dev/null
input=$((input+1))
done
done
echo
}
echo -n Disconnecting ADAT1 outputs
disconnect $ADAT1
echo -n Disconnecting ADAT2 outputs
disconnect $ADAT2
echo -n Disconnecting ADAT3 outputs
disconnect $ADAT3
echo -n Disconnecting SPDIF output
disconnect $SPDIF
Thanks,
Mark
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Re: [Alsa-devel] hdsp patch
OK, I was just remembering Roger Williams telling my some stuff back in January when I was first trying to get this working (when we didn't know that the volume controls didn't work) about needing to break connections. He sent a little script file for the Digiface, but I was not sure if this was required, or just something he kept around for test purposes: no, this doesn't do anything except write values to the mixer, and its the function inside the driver that does this which is broken. --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp patch
On Sun, 2003-06-15 at 20:56, Paul Davis wrote: OK, I was just remembering Roger Williams telling my some stuff back in January when I was first trying to get this working (when we didn't know that the volume controls didn't work) about needing to break connections. He sent a little script file for the Digiface, but I was not sure if this was required, or just something he kept around for test purposes: no, this doesn't do anything except write values to the mixer, and its the function inside the driver that does this which is broken. Thanks for the explanation. I'll watch the list for updates. Cheers, Mark --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] ens1371/ac97 broken on my machine
Once upon a time Takashi Iwai said... At Wed, 21 May 2003 16:34:08 +1000, Cameron Hutchison wrote: The driver for the ens1371 chipset no longer works on my laptop. It used to work in version 0.9.0rc3, but in version 0.9.2 and 0.93a, it no longer works. When I try to run alsamixer, I get the error No mixer elems found i put a workaround code for ens1371 to avoid this. can you try the cvs version? Ok. I finally got around to testing this again in the 0.9.4 release, and I can report that it now works for me. Thanks very much for your quick fix, and apologies for my slow response. --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
