Re: [Alsa-devel] Re: VIA8233/8235 testers wanted
Hi, At Wed, 15 Jan 2003 00:48:46 +0300, Anton Worshevsky wrote: > > Hello , > > > if someone has VIA8233, VIA8233A, VIA8233C or VIA8235 chipset, could > > you help the testing of the new driver? > > the new driver code is found at > > > > http://www.alsa-project.org/~iwai/via82xx.c > > i has VIA8235 and ALC650 codec. After installing new driver from cvs from > 20030113, there is following problem: > > In 5.1-channel dvd playback with xine using surround51 device, > i has swapped Rear and Center/lfe channels. > RL <-> Center, RR <-> LFE > i fixed this with following patch for via82xx.c > > 812,813c812,815 > < case 5: slots = (1<<0) | (2<<4) | (5<<8) | (3<<12) | (4<<16); break; > < case 6: slots = (1<<0) | (2<<4) | (5<<8) | (6<<12) | (3<<16) | (4<<20); >break; > --- > > //case 5: slots = (1<<0) | (2<<4) | (5<<8) | (3<<12) | (4<<16); break; > > //case 6: slots = (1<<0) | (2<<4) | (5<<8) | (6<<12) | (3<<16) | (4<<20); >break; > > case 5: slots = (1<<0) | (2<<4) | (3<<8) | (4<<12) | (5<<16); break; > > case 6: slots = (1<<0) | (2<<4) | (3<<8) | (4<<12) | (5<<16) | (6<<20); >break; > > please note that 5ch mode is untested. > I'm not sure that it is alsa problem, xine' may be. hmm, it could be due to the setting of ALC650. how is the status of "Exchange Center/LFE" mixer switch? > It will be nice to have a small prog for channel position testing in different > playback modes, because now i know no 5.1ch source other then video player. > In general multichannel playback has good quality now (with rc6 sound was > cracking). > > > the new driver can (hopefully) play multiple streams simultaneously on > > VIA8233, VIA8233C and VIA8235, but NOT on VIA8233A. > > It's worked, but second stream plays only left channel. > > Almost all OSS application such as games (UT, tux-racer and other) now producing > cracking and skipping sound. Only XMMS OSS-plugin plays good. OSS-sound was good > with rc6. does the patch below change something? ciao, Takashi via-frag-fix.dif Description: Binary data
Re: [Alsa-devel] Anonymous CVS broken?!?
At Tue, 14 Jan 2003 22:58:44 +0100, Christian Esken wrote: > > Hello, > > I treid to download latest CVS version, but failed logging in (using empty password >as > described on the web page). > > > chris@bjork:~/CVS/alsa> cvs -d >':pserver:[EMAIL PROTECTED]:/cvsroot/alsa' login > Logging in to :pserver:[EMAIL PROTECTED]:2401/cvsroot/alsa > CVS password: > cvs [login aborted]: connect to cvs.alsa-project.org(66.35.250.207):2401 failed: >Connection refused > > > Am I doing sth. wrong here?!? no, it seems like a problem of sourceforge... you can try CVS snapshot instead. Takashi --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] terratec dmx xfire 1024 ac3
At Wed, 15 Jan 2003 11:23:58 +0100, Guido Bakker wrote: > > On Wednesday 15 January 2003 10:08, you wrote: > > Hi, > > > > At Wed, 15 Jan 2003 09:04:21 +0100, > > > > Guido Bakker wrote: > > > hi, > > > > > > i'm trying to use the optical out of my terratec dmx xfire 1024 with an > > > ac3 stream. > > > > > > root@thebox:~/alsa/alsa-tools-0.9.0rc6/ac3dec# ./ac3dec -C test.ac3 > > > Using PCM device 'iec958:AES0=0x2,AES1=0x82,AES2=0x0,AES3=0x2' > > > ALSA lib pcm.c:1719:(snd_pcm_open_conf) Invalid type for PCM > > > iec958:AES0=0x2,AES1=0x82,AES2=0x0,AES3=0x2 definition (id: iec958, > > > value: cards.pcm.iec958) > > > snd_pcm_open: Invalid argument > > > Output open failed > > > > > > i'm getting this error, any ideas? > > > > > > i'm running alsa rc6. > > > > try the cvs version. > > should have been already fixed. > > I'm now stuck to 2ch ac3 only. > > Forced audio codec: hwac3 > Opening audio decoder: [hwac3] AC3 pass-through SP/DIF > No accelerated IMDCT transform found > AUDIO: 48000 Hz, 2 ch, 16 bit (0x400), ratio: 56000->192000 (448.0 kbit) > Selected audio codec: [hwac3] afm:hwac3 (AC3 through SPDIF) > > alsa-init: testing and bugreports are welcome. > alsa-init: requested format: 48000 Hz, 2 channels, AC3 > alsa-init: soundcard set to iec958:AES0=0x2,AES1=0x82,AES2=0x0,AES3=0x2 > alsa9: 48000 Hz/2 channels/4 bpf/32768 bytes buffer/Signed 16 bit Little > Endian > AO: [alsa9] 48000Hz 2ch AC3 > Building audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/8bit... > [format] Sample format big endian AC3 not yet supported ^^^ i guess the file is corrupt or you forgot byte-swapping. Takashi --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] trident problem
At 15 Jan 2003 13:01:06 -0800, Alex Romosan wrote: > > the latest cvs update (from a few days ago, since sourceforge seems to > have problems with anonymous cvs) broke the trident driver. when i try > to play a sound file i get the following: > > kernel: ALSA ../../alsa-kernel/pci/trident/trident_memory.c:192: bad MAGIC >(0xa55a5a5a) > > any ideas? this must have been already fixed on the latest cvs... Takashi --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] terratec dmx xfire 1024 ac3
> > > > i'm trying to use the optical out of my terratec dmx xfire 1024 with
> > > > an ac3 stream.
> > > >
> > > > root@thebox:~/alsa/alsa-tools-0.9.0rc6/ac3dec# ./ac3dec -C test.ac3
> > > > Using PCM device 'iec958:AES0=0x2,AES1=0x82,AES2=0x0,AES3=0x2'
> > > > ALSA lib pcm.c:1719:(snd_pcm_open_conf) Invalid type for PCM
> > > > iec958:AES0=0x2,AES1=0x82,AES2=0x0,AES3=0x2 definition (id: iec958,
> > > > value: cards.pcm.iec958)
> > > > snd_pcm_open: Invalid argument
> > > > Output open failed
> > > >
> > > > i'm getting this error, any ideas?
> > > >
> > > > i'm running alsa rc6.
> > >
> > > try the cvs version.
> > > should have been already fixed.
> >
> > I'm now stuck to 2ch ac3 only.
> >
> > Forced audio codec: hwac3
> > Opening audio decoder: [hwac3] AC3 pass-through SP/DIF
> > No accelerated IMDCT transform found
> > AUDIO: 48000 Hz, 2 ch, 16 bit (0x400), ratio: 56000->192000 (448.0 kbit)
> > Selected audio codec: [hwac3] afm:hwac3 (AC3 through SPDIF)
> >
> > alsa-init: testing and bugreports are welcome.
> > alsa-init: requested format: 48000 Hz, 2 channels, AC3
> > alsa-init: soundcard set to iec958:AES0=0x2,AES1=0x82,AES2=0x0,AES3=0x2
> > alsa9: 48000 Hz/2 channels/4 bpf/32768 bytes buffer/Signed 16 bit Little
> > Endian
> > AO: [alsa9] 48000Hz 2ch AC3
> > Building audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/8bit...
> > [format] Sample format big endian AC3 not yet supported
>
>^^^
>
> i guess the file is corrupt or you forgot byte-swapping.
When i use the same file on a sb live 1024, it works fine.
--
Guido Bakker
Interne Automatisering
Sogeti Nederland B.V.
T: +31 (0)20 660 66 00 + nakiesnummer 0281
e-mail: [EMAIL PROTECTED]
printk("??? No FDIV bug? Lucky you...\n");
2.2.16 /usr/src/linux/include/asm-i386/bugs.h
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Re: [Alsa-devel] terratec dmx xfire 1024 ac3
At Thu, 16 Jan 2003 10:55:37 +0100, Guido Bakker wrote: > > > > > > i'm trying to use the optical out of my terratec dmx xfire 1024 with > > > > > an ac3 stream. > > > > > > > > > > root@thebox:~/alsa/alsa-tools-0.9.0rc6/ac3dec# ./ac3dec -C test.ac3 > > > > > Using PCM device 'iec958:AES0=0x2,AES1=0x82,AES2=0x0,AES3=0x2' > > > > > ALSA lib pcm.c:1719:(snd_pcm_open_conf) Invalid type for PCM > > > > > iec958:AES0=0x2,AES1=0x82,AES2=0x0,AES3=0x2 definition (id: iec958, > > > > > value: cards.pcm.iec958) > > > > > snd_pcm_open: Invalid argument > > > > > Output open failed > > > > > > > > > > i'm getting this error, any ideas? > > > > > > > > > > i'm running alsa rc6. > > > > > > > > try the cvs version. > > > > should have been already fixed. > > > > > > I'm now stuck to 2ch ac3 only. > > > > > > Forced audio codec: hwac3 > > > Opening audio decoder: [hwac3] AC3 pass-through SP/DIF > > > No accelerated IMDCT transform found > > > AUDIO: 48000 Hz, 2 ch, 16 bit (0x400), ratio: 56000->192000 (448.0 kbit) > > > Selected audio codec: [hwac3] afm:hwac3 (AC3 through SPDIF) > > > > > > alsa-init: testing and bugreports are welcome. > > > alsa-init: requested format: 48000 Hz, 2 channels, AC3 > > > alsa-init: soundcard set to iec958:AES0=0x2,AES1=0x82,AES2=0x0,AES3=0x2 > > > alsa9: 48000 Hz/2 channels/4 bpf/32768 bytes buffer/Signed 16 bit Little > > > Endian > > > AO: [alsa9] 48000Hz 2ch AC3 > > > Building audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/8bit... > > > [format] Sample format big endian AC3 not yet supported > > > >^^^ > > > > i guess the file is corrupt or you forgot byte-swapping. > > When i use the same file on a sb live 1024, it works fine. what about running ac3dec without -C option (i.e. analog two channels)? it must work correctly regardless of spdif support. Takashi --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] cs64xx distortion (GameTheater XP)
At Thu, 16 Jan 2003 00:14:10 +0100, Christian Esken wrote: > > Hi, > > I wonder what the status of the cs64xx distortion problem is. > I tested the 2003-01-14.tar.bz2 snapshot with my GameThaeter 7.1 XP > and it still shows those heavy distortions on analog output. > > Digital out is OK most of the time, but sometimes fails (as described in >cs62xx_lib.c). > I thought you might be interested in the syslog: > > PCI: Found IRQ 7 for device 00:0a.0 > ALSA ../../alsa-kernel/pci/cs46xx/cs46xx_lib.c:3855: hack for Hercules Game Theatre >XP enabled > ALSA ../../alsa-kernel/pci/cs46xx/cs46xx_lib.c:3866: Crystal EAPD support forced on. well, it looks like you passed external_amp=1 option to the module, which you don't need usually. anyway, try the attached patch (and without external_amp=1 option). it will clean up the amplifier controls. Takashi cs46xx-amp.dif Description: Binary data
Re: [Alsa-devel] General Guidelines? (MIDI)
On Thu, Jan 16, 2003 at 08:39:23AM +0100, Clemens Ladisch wrote: >Brian Victor wrote: >> Monitoring /proc/asound/seq/queues shows the events being queued, but as >> soon as the queue starts, all 400 notes leave the queue instantly; none >> are played. >> >> snd_seq_ev_schedule_tick(&ev, m_queue, 0, tick); >What is the queue's tempo? 400 ticks may not be much. [before starting] owned by client: 128 lock status: Locked queued time events : 0 queued tick events : 400 timer state: Stopped timer PPQ : 128 current tempo : 1200 current BPM: 5 current time : 0.0 s current tick : 0 [after starting] queue 0: [Queue-0] owned by client: 128 lock status: Locked queued time events : 0 queued tick events : 0 timer state: Running timer PPQ : 128 current tempo : 1200 current BPM: 5 current time : 1.14000 s current tick : 12 (Which begs another question: what does PPQ stand for? Something per quarter?) >> snd_seq_ev_set_subs(&ev); >This will send the note events to the subscribers of your program's port. >Are there any? Yes. I have a sleep in my program that allows me to run from another terminal: aconnect 64:0 128:1; aconnect 128:0 64:0 (Yes, there's an input port, too.) -- Brian --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] General Guidelines? (MIDI)
Brian Victor wrote: >On Thu, Jan 16, 2003 at 08:39:23AM +0100, Clemens Ladisch wrote: >>Brian Victor wrote: >>> Monitoring /proc/asound/seq/queues shows the events being queued, but as >>> soon as the queue starts, all 400 notes leave the queue instantly; none >>> are played. afaik, you should use snd_seq_event_output() instead of snd_seq_event_output_direct() -- the latter is supposed to by-pass the queue for instant transmission. you may want to look at alsa-lib/test/playmidi1.c for another reference. >(Which begs another question: what does PPQ stand for? Something per >quarter?) ticks/quarter (aka 'parts per quarter'). hth, tim --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] General Guidelines? (MIDI)
On Thu, Jan 16, 2003 at 04:32:05PM +0100, Tim Goetze wrote: >afaik, you should use snd_seq_event_output() instead of >snd_seq_event_output_direct() -- the latter is supposed >to by-pass the queue for instant transmission. I've tried both, actually. output() does not show any queued events in /proc/asound/seq/queues; output_direct() does. miniArp.c uses output_direct, and seems to work. Neither produces any output in my test program. I've removed all attachments to wxWindows, so anyone should be able to compile my test program. So I don't have to keep sending snippits out of context, the code is here: http://www.personal.psu.edu/users/b/h/bhv1/nowx.cc or http://www.personal.psu.edu/users/b/h/bhv1/nowx.cc.html (syntax colored) (g++ nowx.cc -o nowx -lasound) >you may want to look at alsa-lib/test/playmidi1.c for >another reference. That could be useful, but at a glance I don't see anything important in there that I'm not doing. This is really perplexing me. >>(Which begs another question: what does PPQ stand for? Something per >>quarter?) >ticks/quarter (aka 'parts per quarter'). I figured it was something like that. Thanks. -- Brian --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] General Guidelines? (MIDI)
At Thu, 16 Jan 2003 16:32:05 +0100 (CET), Tim Goetze wrote: > > Brian Victor wrote: > > >On Thu, Jan 16, 2003 at 08:39:23AM +0100, Clemens Ladisch wrote: > >>Brian Victor wrote: > >>> Monitoring /proc/asound/seq/queues shows the events being queued, but as > >>> soon as the queue starts, all 400 notes leave the queue instantly; none > >>> are played. > > afaik, you should use snd_seq_event_output() instead of > snd_seq_event_output_direct() -- the latter is supposed > to by-pass the queue for instant transmission. well, this is not exact: snd_seq_event_output_direct() sends the event without "output buffer" on the user-space. if the event record has a proper queue value (i.e. event.queue != SND_SEQ_QUEUE_DIRECT), it is scheduled on the specified queue. for by-passing the scheduling on the queue, you need to mark the queue via snd_seq_ev_set_direct(). snd_seq_event_output() will put the event onto the output buffer. the buffered events won't be sent and stay on the buffer until either the buffer becomes full or snd_seq_drain_output() is called explicitly. this i/o-buffer was introduced to reduce the amounts of read/write, ioctls. this could be implemented more sofisticatedly as a high-level library... Takashi --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: trident problem
At Thu, 16 Jan 2003 10:06:14 -0800, Alex Romosan wrote: > > Takashi Iwai <[EMAIL PROTECTED]> writes: > > > At 15 Jan 2003 13:01:06 -0800, > > Alex Romosan wrote: > >> > >> the latest cvs update (from a few days ago, since sourceforge seems to > >> have problems with anonymous cvs) broke the trident driver. when i try > >> to play a sound file i get the following: > >> > >> kernel: ALSA ../../alsa-kernel/pci/trident/trident_memory.c:192: bad MAGIC >(0xa55a5a5a) > >> > >> any ideas? > > > > this must have been already fixed on the latest cvs... > > i downloaded the latest cvs snapshot (2003-01-14) but the problem is > still there. i got it. sorry, my typos. the patch attached. Takashi trident-fix2.dif Description: Binary data
RE: [Alsa-devel] Sound Blaster 4.1 problem (ens1371)
Hi, The simulation approach still has a lot of bugs... so I tried another approach. I was reading the datasheets of the ENS-1371 chip, and I noticed that almost all configuration is done using IO space registers. So I wrote a little tool to read out the contents of these registers in Win98, along with the sample rate converter memory and the CODEC registers. I started to play around with the settings of the SB and found out the following: * When switching from/to 4speaker mode register BASE+04h is altered as follows: v B404: 24280EC00010 0100 0010 1000 1110 1100 ^ This bit (bit 26) gets set when in 4speaker mode. (note: the format of the line is "IO addr: HEX result \t binary formatted result") * I also noticed that the windows volume control "Master" alters codec registers 0x02 (master volume) and 0x38 (surround out volume) simultaniously and keeps them at the same value. * There seem to be differences in settings of the samplerate convertor between windows and ALSA. Do we have any docs on this SRC? I'm now going to try put these things in a little patch, and hopefully *1 is the only thing that's important. Pieter > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED]]On Behalf Of Jaroslav > Kysela > Sent: dinsdag 14 januari 2003 20:20 > To: Pieter Palmers > Cc: [EMAIL PROTECTED] > Subject: Re: [Alsa-devel] Sound Blaster 4.1 problem (ens1371) > > > On Tue, 14 Jan 2003, Pieter Palmers wrote: > > > Anyway, the statement I wanted to make is: maybe I can use the > same approach > > to try spying on the ens1371. I already experienced that you > can obtain a lot > > of info fast if you know what you're looking for. > > > > Should you have any suggestions on how I should build/enhance > the 'bridge', > > they are very welcome. I'm new to linux driver design, and > kernel level code. > > Thanks for explaining. Perhaps, you can log all accesses to i/o ports for > ens1371 hardware from the windows drivers? It might be helpful - we can > decode the init sequence. > > Jaroslav > > - > Jaroslav Kysela <[EMAIL PROTECTED]> > Linux Kernel Sound Maintainer > ALSA Project, SuSE Labs > > > > --- > This SF.NET email is sponsored by: Take your first step towards giving > your online business a competitive advantage. Test-drive a Thawte SSL > certificate - our easy online guide will show you how. Click here to get > started: http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0027en > ___ > Alsa-devel mailing list > [EMAIL PROTECTED] > https://lists.sourceforge.net/lists/listinfo/alsa-devel --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] cs64xx distortion (GameTheater XP)
Takashi Iwai wrote: >Christian Esken wrote: >> >> I wonder what the status of the cs64xx distortion problem is. >> I tested the 2003-01-14.tar.bz2 snapshot with my GameThaeter 7.1 XP >> and it still shows those heavy distortions on analog output. > >well, it looks like you passed external_amp=1 option to the module, >which you don't need usually. > >anyway, try the attached patch (and without external_amp=1 option). >it will clean up the amplifier controls. Tried it. Unfortunately the patch does not help. Probably the distortion is slightly less, but still the soundcard is basically unusable. I removed the external_amp=1 and reloaded the driver - and I sure that actually the new driver got loaded. Chris --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] ens1731 patch to enable rear output
Hi, It seems that my previous assumption was correct. I added a patch for the ens1370.c driver that enables the rear output. It seems to be a mirror of the front output, but the 'surround' slider in alsamixer controls volume separately from the output. By the way: the CVS server at sourceforge still refuses connections. So I applied this to the latest CVS version I had locally. Pieter - 1947,1950d1946 < #ifdef CHIP1371 < /* enable the rear outputs */ < ensoniq->cssr |= (1 << 26); < #endif -- --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] General Guidelines? (MIDI)
On Thursday 16 January 2003 21:08, Brian Victor wrote:
> I've removed all attachments to wxWindows, so anyone should be able to
> compile my test program. So I don't have to keep sending snippits out
> of context, the code is here:
>
> http://www.personal.psu.edu/users/b/h/bhv1/nowx.cc or
> http://www.personal.psu.edu/users/b/h/bhv1/nowx.cc.html (syntax colored)
> (g++ nowx.cc -o nowx -lasound)
In short: start the queue before sending events, add snd_seq_drain_output at
end of SetTempo method.
I wrote some example programs in plain C while learning ALSA API (player,
recorder, metronome, monitor...)
http://perso.wanadoo.es/plcl/alsautil.tar.bz2
More examples, for Kylix and FreePascal:
http://perso.wanadoo.es/plcl/
HTH
--- nowx.cc.old Thu Jan 16 22:11:17 2003
+++ nowx.cc Thu Jan 16 22:08:59 2003
@@ -64,14 +64,14 @@
wxMidiOutput outport(sequencer);
wxMidiQueue queue(sequencer);
int x;
- for (x = 0; x < 400; ++x)
- {
-queue.NoteOn(outport, x, 127, x);
- }
cout << "Queue will start in five seconds" << endl;
sleep(5);
cout << "Starting queue" << endl;
queue.StartQueue();
+ for (x = 0; x < 400; ++x)
+ {
+queue.NoteOn(outport, x , 127, x);
+ }
while (1) {
sleep(1);
}
@@ -181,6 +181,7 @@
snd_seq_queue_tempo_set_ppq(tpo, 128);
snd_seq_set_queue_tempo(m_drv.GetAlsaSeq(), m_queue, tpo);
snd_seq_queue_tempo_free(tpo);
+ snd_seq_drain_output(m_drv.GetAlsaSeq());
}
void wxMidiQueue::StartQueue()
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Re: [Alsa-devel] cs64xx distortion (GameTheater XP)
Christian Esken wrote: Takashi Iwai wrote: Christian Esken wrote: I wonder what the status of the cs64xx distortion problem is. I tested the 2003-01-14.tar.bz2 snapshot with my GameThaeter 7.1 XP and it still shows those heavy distortions on analog output. well, it looks like you passed external_amp=1 option to the module, which you don't need usually. anyway, try the attached patch (and without external_amp=1 option). it will clean up the amplifier controls. Tried it. Unfortunately the patch does not help. Probably the distortion is slightly less, but still the soundcard is basically unusable. I removed the external_amp=1 and reloaded the driver - and I sure that actually the new driver got loaded. Does the sound quality improve if you lower the volume of PCM and/or master? If so, does reloading the cs46xx module (or restarting alsa) help? Try /etc/init.d/alsa force-restart as root or modprobe -r snd-cs46xx modprobe snd-cs46xx as root. Chris 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] cs64xx distortion (GameTheater XP)
I wonder what the status of the cs64xx distortion problem is. I tested the 2003-01-14.tar.bz2 snapshot with my GameThaeter 7.1 XP My opinion: the GameTheater XP it's a cheap soundcard with a beautiful blue box. Usually cheap soundcard got a cheap design as result of a tiny bugdet, so just dont expect to much from the analog part's. Actually I got a Hercules GameTheater XP 6.1, when volume is over about a ~80% the output is sometimes distorcionated, but I believe that's something we cant do anything about in the driver. Limit some volume control's in the driver dont feel like a coherent solution. The card is still usable, I really dont expect to better quality on analog output with this soundcard. ALSA ../alsa-kernel/core/pcm_lib.c:123: Unexpected hw_pointer value (stream = 0, delta: -512, max jitter = 8192): wrong interrupt acknowledge? ALSA ../alsa-kernel/core/pcm_lib.c:123: Unexpected hw_pointer value (stream = 0, delta: -512, max jitter = 8192): wrong interrupt acknowledge? ALSA ../alsa-kernel/core/pcm_lib.c:123: Unexpected hw_pointer value (stream = 0, delta: -512, max jitter = 8192): wrong interrupt acknowledge? Feel like theese messages should not be there, could be a bug in cs46xx driver. I am wondering about the status, as there is still this comment in cs64xx_lib.c: --- /* CD mute change ? */ /* Benny: this hack dont seems to make any sense to me, at least on the Game Theater XP, Turning of the amplifier just make the PCM sound very distorcionated. is this really needed */ --- Should only matter if you unmute the analog CD input, and with the new DSP code it dont matter at all. /Benny --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] General Guidelines? (MIDI)
On Thu, Jan 16, 2003 at 10:27:35PM +0100, Pedro Lopez-Cabanillas wrote: >In short: start the queue before sending events, add snd_seq_drain_output at >end of SetTempo method. ...and then there was light! Er.. sound! Now that it's working, I'm left again with conceptual questions. I was under the impression that I could put events in a queue, start it, stop it, and start it from the beginning again. Is that not the case? The events seem to leave the queue once sent. Likewise, I was under the impression that queues handled sequencing events by timestamp if they arrived in non-sequential order. Is that so? Do I just need to stay ahead of the queue for this to work? Does that mean, I can't reliably send the queue all events for one voice, followed by all events for another voice? Thank you, Pedro, for showing me how to fix my code. That was a huge help. -- Brian --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] FIX: soundblaster rear channels (ens1371)
Finally I figured out how to enable the rear channel separately from the front on the creative labs CT5880 (= modified ens1371 as it seems). I'll describe the settings to enable this, but I'm not going to submit a patch, as I have noticed that I messed up my ens1370.c too much, and I can't get a clean version because CVS doesn't work. It's a minor change, so I assume it can be done by the maintainers. The whole thing is in three bits in the BASE+0x04 register (ES_REG_STATUS in the driver). These bits are bit27, bit26 and bit24 There are three modes of operation: 1) No rear output: bit27=bit26=bit24=0 2) rear output is a mirror of the main output, but controlled by the 'surround' slider of the mixer. I assume the ens1371 mixes the two 'devices' and sends the mix to both MAIN and SURROUND DAC. This mode is selected by setting bit27=bit24=0 and bit26=1 3) independant rear (surround) and front output. Using the current driver, this has the strange side-effect that HW:0,0 becomes the rear output and HW:0,1 becomes the front. So HW:0,0 is controlled by the 'surround' mixer control, and HW:1,0 is controlled by the PCM and Master mixer controls. To select this mode set bit27=bit24=1 and bit26=0. It seems that bit27=1 and bit24=bit26=0 is identical, but the windows driver clearly does the first, so why not? It works... front only, no rear:bit27=0 bit26=0 bit24=0 rear mirrors front: bit27=0 bit26=1 bit24=0 front & rear independant: bit27=1 bit26=0 bit24=1 (x?) I patched my driver by inserting the following code around line 1947: (I included two lines of overlap to make the location easier to find) = outb(ensoniq->uartc = 0x00, ES_REG(ensoniq, UART_CONTROL)); outb(0x00, ES_REG(ensoniq, UART_RES)); #ifdef CHIP1371 /* enable the rear outputs This seems to work ensoniq->cssr |= (0 << 27) | (1 << 24); but the windows driver does this, so let's also do it */ ensoniq->cssr |= (1 << 27) | (1 << 24); /* Use this for mirror mode ensoniq->cssr |= (1 << 26);*/ #endif outl(ensoniq->cssr, ES_REG(ensoniq, STATUS)); #if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE) = Maybe there is a better place to put this? I don't know... I put it in the snd_ensoniq_create() function because I always want 4ch output, and I don't see the use of the other modes in an ALSA enviroment. Regards, Pieter PS: I'm also developing a driver for my Maxisound ISIS, where do I look for information on ALSA/linux driver developement? Does anyone have a 'template' ALSA driver? Does ALSA support non-DMA audio transfer? I believe the ISIS uses this kind of transfers, but I don't know for sure yet. I know it's stupid design not to use DMA, but there is nothing to do about it I guess. --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] cs64xx distortion (GameTheater XP)
Benny Sjostrand <[EMAIL PROTECTED]> wrote: > My opinion: the GameTheater XP it's a cheap soundcard with a beautiful > blue box. Usually cheap soundcard got a cheap design as result of a > tiny bugdet, so just dont expect to much from the analog part's. Constantly saying that it's a cheap soundcard isn't really a solution though. While it might not be a card suited for an audiophile it's certainly better than several other cards out there. And the Windows drivers don't have any problems like this. ;) My GTXP works well enough though, though sound does get disorted/etc when using more than ~80% volume as well. I hardly see that as a problem though. -- Richard Olsson E: [EMAIL PROTECTED] W: http://www.nyo-box.net/ --- 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-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
