[Alsa-user] BCD2000
Can someone please write a driver for the BCD2000 and BCD3000 please.. I am a DJ and would like to use this device in the linux version of my DJ Software Thanks..- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] please test the beta CMI8788 driver
ATT TynTyn wrote: Under Fedora 7 with the newest kernel, I get ~ 0.25 samples every ~ 0.75 sec. on the main output. That is, 0.25 s of sound at normal pitch, then 0.75 s of silence? Is it different with 2/4/6 channels (the -c parameter)? Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Mixer problem with cmipci
Jan-Benedict Glaw wrote: On Thu, 2007-08-30 23:43:26 +0200, Jan-Benedict Glaw [EMAIL PROTECTED] wrote: Just out of interest, I ordered another, similar card (Ultron Octosound 7.1, lspci: C-Media Electronics Inc CM8738 (rev 10), /proc/asound/cards: C-Media PCI CMI8738-MC8 (model 68)) I have the same chip here. [...] After the FLAC ended, I fired up speaker-test again (and of course switched off Four Channel Mode.) This time, it's different than before: front center/LFE, front left/right and side left/right are okay, but rear right/left are highly distorted. I can reproduce this. Register 0x04 ~ Before: 0x07 CM_ASFC_SHIFT | 0x00 CM_DSFC_SHIFT | CM_BREQ After:0x07 CM_ASFC_SHIFT | 0x07 CM_DSFC_SHIFT | CM_BREQ (DAC Sample Frequency changed) 0 is the power-on default of 5512 Hz, 7 is 48 kHz. DAC and ADC aren't quite the correct names, both DMA channels can be used for either playback and capture, and the driver uses one for stereo and the other for multichannel playback. Does the FLAC file have a sample rate of 44.1 kHz or 48 kHz? Register 0x08 ~ Before: CM_CHB3D5C | CM_SPDIF_SELECT1 | 0x03 CM_CH1FMT_MASK | 0x00 CM_CH0FMT_SHIFT After:CM_CHB3D5C | CM_SPDIF_SELECT1 | 0x03 CM_CH1FMT_MASK | 0x03 CM_CH0FMT_SHIFT (Ch0 Format changed) 0 is 8-bit mono, 3 is 16-bit stereo (or more than two channels). Register 0x18 ~ Before: CM_TXVX | CM_FM_EN | CM_AC3EN2 | CM_VIDWPPRT | CM_ENCENTER | CM_FLINKOFF After:CM_TXVX | CM_FM_EN | CM_AC3EN2 | CM_VIDWPPRT | CM_SPDF_AC97 | CM_ENCENTER | CM_FLINKOFF (SPDIF/Out changed from 44.1kHz to 48kHz) My guess is that the change to SPDIF/Out frequency and DAC Sample Frequency are annoying at best, but irrelevant here. Am I probably right that the changed ch0 format is the cause of the distortion? I think it's the sample rate. It seems I can 'cure' the distortion by playing a 48 kHz file before running speaker-test. Can you confirm this? The chip has two sample rate registers (ASFC and DSFC), and these are internally mapped to three destinations (front+side+center DACs, rear DAC, capture ADC) depending on the state of a bunch of other bits. I guess the driver doesn't handle these bits correctly so that the rear sample rate is still the old value (although the two xSFC registers appear to have the correct value). Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] XLR3 interfaces?
James Shatto wrote: Have a look at the Edirol UA-25. It is definitely supported. Thanks, that looks like Exactly what I'm looking for. Out of curiousity, since I've never used anything like these(yet), what else does it do? Regular sound card stuff, or just a recording interface. It does recording and playback. Like many other USB devices, it probably has few or no software-controllable mixer controls. What do you mean with regular sound cars stuff? Onboard midi, or just an interface? It has two MIDI jacks, input and output. What is the difference between onboard midi and interface? Multiple channel output with hardware mixing? USB devices don't do hardware mixing; sending all those streams over the bus would take more resources than mixing them in software. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] problems with dmix, a52 and upmix
Michael KAufmann wrote: Is there any way to make dmix work with a52 as slave? No. As it's currently designed, dmix requires a hw device as slave because it needs access to the sound card's DMA buffer and interrupt for timing purposes. Regards, Clemens - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Mixer problem with cmipci
On Tue, 2007-09-04 08:56:22 +0200, Clemens Ladisch [EMAIL PROTECTED] wrote: Jan-Benedict Glaw wrote: On Thu, 2007-08-30 23:43:26 +0200, Jan-Benedict Glaw [EMAIL PROTECTED] wrote: [...] After the FLAC ended, I fired up speaker-test again (and of course switched off Four Channel Mode.) This time, it's different than before: front center/LFE, front left/right and side left/right are okay, but rear right/left are highly distorted. I can reproduce this. Register 0x04 ~ Before: 0x07 CM_ASFC_SHIFT | 0x00 CM_DSFC_SHIFT | CM_BREQ After: 0x07 CM_ASFC_SHIFT | 0x07 CM_DSFC_SHIFT | CM_BREQ (DAC Sample Frequency changed) 0 is the power-on default of 5512 Hz, 7 is 48 kHz. DAC and ADC aren't quite the correct names, both DMA channels can be used for either playback and capture, and the driver uses one for stereo and the other for multichannel playback. Does the FLAC file have a sample rate of 44.1 kHz or 48 kHz? 44.1kHz: $ sndfile-info 04_my_immortal.flac Version : libsndfile-1.0.17 File : 04_my_immortal.flac Length : 28262646 Sample Rate : 44100 Frames : 11651808 Channels: 2 Format : 0x00170002 Sections: 1 Seekable: TRUE Duration: 00:04:24.213 Signal Max : 31624 (-0.31 dB) Register 0x18 ~ Before: CM_TXVX | CM_FM_EN | CM_AC3EN2 | CM_VIDWPPRT | CM_ENCENTER | CM_FLINKOFF After: CM_TXVX | CM_FM_EN | CM_AC3EN2 | CM_VIDWPPRT | CM_SPDF_AC97 | CM_ENCENTER | CM_FLINKOFF (SPDIF/Out changed from 44.1kHz to 48kHz) My guess is that the change to SPDIF/Out frequency and DAC Sample Frequency are annoying at best, but irrelevant here. Am I probably right that the changed ch0 format is the cause of the distortion? I think it's the sample rate. It seems I can 'cure' the distortion by playing a 48 kHz file before running speaker-test. Can you confirm this? After alsaplayer'ing one of the 48kHz files that speaker-test uses with `-t wav', speaker-test also works as expected. Confirmed. MfG, JBG -- Jan-Benedict Glaw [EMAIL PROTECTED] +49-172-7608481 Signature of: Alles sollte so einfach wie möglich gemacht sein. the second : Aber nicht einfacher. (Einstein) signature.asc Description: Digital signature - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Problems with 32 bit apps on 64 bit OS using dmix and snd-ioctl32
I have a problem using 32 bit (i386) playback applications on a 64 bit (x86_64) CentOS4/RHEL4 based kernel (based on kernel 2.6.9). I've installed alsa-driver, alsa-lib and alsa-utils 1.0.13 (newer versions of alsa-driver don't build with this kernel) If I try to run a 32 bit copy of aplay on a 64 bit machine that doesn't support hardware mixing (e.g. HDA NVidia) - i.e. uses dmix by default, I get: aplay: pcm_write:1268: write error: Invalid argument and /var/log/messages reports: Sep 4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(3) cmd(40045402){00} arg(ae88) on /dev/snd/timer Sep 4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(5) cmd(c008551a){00} arg(080678a0) on /dev/snd/controlC0 Sep 4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(5) cmd(c008551b){00} arg(080678a0) on /dev/snd/controlC0 Sep 4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(3) cmd(54a0){00} arg(0320) on /dev/snd/timer I've done some searching to find a solution - and found out that in more recent kernels, snd-ioctl32 has been obsoleted - and if I do use a more modern kernel, then using a 32 bit version of aplay works OK on a newer 64 bit kernel. However, for other (not sound related) reasons, I need to run the RHEL4/CentOS4 2.6.9 kernel ... I had a look at the alsa-driver code, and acore/ioctl32/timer32_new.c contains: #if 0 /* ** FIXME ** * The following four entries are disabled because they conflict * with the TCOC* definitions. * Unfortunately, the current ioctl32 wrapper uses a single * hash table for all devices. Once when the wrapper is fixed * with the table based on devices, they'll be back again. */ MAP_COMPAT(SNDRV_TIMER_IOCTL_START), MAP_COMPAT(SNDRV_TIMER_IOCTL_STOP), MAP_COMPAT(SNDRV_TIMER_IOCTL_CONTINUE), MAP_COMPAT(SNDRV_TIMER_IOCTL_PAUSE), #endif If I change the '#if 0' to '#if 1', recompile and reload with this new snd-ioctl32 module, then the 32 bit aplay works without an error ... although /var/log/messages still reports: Sep 4 17:05:26 wstemp-b kernel: ioctl32(aplay:28926): Unknown cmd fd(3) cmd(40045402){00} arg(adb8) on /dev/snd/timer Sep 4 17:05:26 wstemp-b kernel: ioctl32(aplay:28926): Unknown cmd fd(5) cmd(c008551a){00} arg(080678a0) on /dev/snd/controlC0 Sep 4 17:05:26 wstemp-b kernel: ioctl32(aplay:28926): Unknown cmd fd(5) cmd(c008551b){00} arg(080678a0) on /dev/snd/controlC0 ... but at least I get sound out! So, what is the danger of using these 'disabled' ioctls? I can't find any reference to 'TCOC* definitions' - what are these ??? Thanks James Pearson - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Serial Midi via snd-serial-u16550 not working
hi, my friend has some problem to get snd-serial working, so i posting his message to this list: === First of all hi everybody. This is my very first post in this forum. I'm totally new in producing music under linux but after I have configured programs like Ardour, Rosegarden (in sync via Jack audio connection kit) and some other programs I want to chance from Windows (Nuendo/Cubase/Reason) to Linux because the programs were running excellent and have a big potential because of the function to rewire all the devices by using JACK. But befor telling you long stories now to my real problem: Since a few days I try to get my Technics keyboard SX-KN5000 working by using midi via the serial port. I want to do this because first I have no midi port slot for my onboard soundcard of my A8N-SLI Premium motherboard and second the serial port has a much faster speed and so latencies were near zero even in big midi-projects. A I've read MANY tutorials the last days and searched in many different forums to get this problem solved. After I had first used the 'UART16550 serial MIDI driver' as built-in kernel module but the midi device did not appear in the list of my devices Code: # aconnect -o client 20: 'MPU-401 UART' [type=kernel] 0 'MPU-401 UART MIDI' and in many how-tos I've read about using this driver as a loadable module I've changed this. OK, now I'm at the point where I don't know how to go on. I have read tutorials like this ones http://alsa.opensrc.org/index.php/Serial#Simple_serial_converter http://www.geocities.jp/midi_organ_net/alsa/ http://forums.gentoo.org/viewtopic-t-27919.html?sid=7abc10e4e733063cdc749811f114382c but no one doesn't really help me. My serial port is com1 (/dev/ttyS0). Almost all discussions about configuring this driver include that you first have to make sure, that the serial port isn't in use. So I do a Code: # setserial /dev/ttyS0 uart none setserial without the other parameters says Code: # setserial /dev/ttyS0 /dev/ttyS0, UART: unknown, Port: 0x03f8, IRQ: 4 So I know that my port is 0x03f8 and the irq is 4. Now I do a modprobe to load the driver but I only get an error: Code: # modprobe -v snd-serial-u16550 port=0x3F8 irq=4 install { /bin/setserial /dev/ttyS0 uart none; } ; /sbin/modprobe --first-time --ignore-install snd-serial-u16550 insmod /lib/modules/2.6.22-gentoo-r2/kernel/sound/drivers/snd-serial-u16550.ko snd_port=0x3f8 snd_irq=4 snd_speed=38400 FATAL: Error inserting snd_serial_u16550 (/lib/modules/2.6.22-gentoo-r2/kernel/sound/drivers/snd-serial-u16550.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for snd_serial_u16550 In my /var/log/messages it says Code: snd_serial_u16550: Unknown parameter `snd_port' I don't understand this. I haven't specified any snd_port in the modprobe command. I first thought, that I have to, but running modprobe with snd_port (and e.g. snd_irq) doesn't help. Even if I call it without any parameters the same error message appears. Maybe someone has the same error as I or knows any help. Maybe it's just a config error somewhere. At the beginning I experimented with configuring some configfiles which you see below. But I don't know if they will be ever used by my sound system. (Should I mention that all other sound drivers are all built-in kernel modules) /etc/modules.d/alsa Code: # Alsa kernel modules' configuration file. # ALSA portion alias char-major-116 snd # OSS/Free portion alias char-major-14 soundcore ## ## IMPORTANT: ## You need to customise this section for your specific sound card(s) ## and then run `update-modules' command. ## Read alsa-driver's INSTALL file in /usr/share/doc for more info. ## ## ALSA portion ## alias snd-card-0 snd-interwave ## alias snd-card-1 snd-ens1371 ## OSS/Free portion ## alias sound-slot-0 snd-card-0 ## alias sound-slot-1 snd-card-1 ## pre-install snd-serial-u16550 /bin/setserial /dev/ttyS0 uart none alias snd-card-1 snd-serial-u16550 options snd-serial-u16550 port=0x3f8 irq=4 speed=38400 # OSS/Free portion - card #1 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-seq-oss alias sound-service-0-12 snd-pcm-oss ## OSS/Free portion - card #2 serial midi alias sound-service-1-1 snd-seq-oss alias sound-service-1-8 snd-seq-oss alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss # Set this to the correct number of cards. options snd cards_limit=2 /etc/modprobe.d/sound Code: options snd-serial-u16550 port=0x3f8 irq=4 The rest of my soundsystem works fine. I would be thankful for any help I could get. Is something missing? Please tell me. Here the last thing which could be helpful: Code: # uname -a Linux marcus 2.6.22-gentoo-r2 #1 SMP Thu Aug 30 18:37:44 CEST 2007 i686 AMD Athlon(tm) 64 X2 Dual Core Processor 4400+ AuthenticAMD GNU/Linux Thanks Marcus ===
Re: [Alsa-user] Front Output Jack problems with AD1986A/M5461
Have you checked all of your mixer settings? The front one might be PCM2 or Headphone. And might only be accessible in alsamixer. Nope, the only things I have listed in alsamixer under playback are: Master, PCM, Line, CD, Mic, Mic Boos, IEC958, and Aux. All are unmuted and set at full. Any other ideas? - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user