Linux-audio-dev January 2005

linux-audio-dev@lists.linuxaudio.org

89 participants
86 discussions

[linux-audio-dev] Firewire support in Linux

by Jan Depner

I was interested in the thread about firewire audio device support in Linux a while back. Just the other day I noticed that Presonus is just up the road from where I live so I thought I'd drop them a line and ask about their two firewire interfaces and if they would consider releasing specs to the Linux audio community to get drivers written for them. I received two responses, both of which seem promising: > Hi Jan, > We have a Linux fan within our engineering group. I feel certain that > we > could provide driver information for the development of Linux based > FireWire connectivity. > . > . > . > Please let us know who to contact. > Kind regards, > > Jim Odom > President/CEO > PreSonus Corporation > 225-216-7887 ext. 107 and: > Mr. Depner > I'm the engineer interested in Linux that Jim Odom spoke of. I in no > way > claim to be even an experienced Linux user, I like the idea behind > it. I > can tell you that there is ongoing development of a Linux driver for > products based on the firewire solution that we employ. The name of > the > company whose solution we use is BridgeCo...you might be able to > google some > info...no promises. I'm not sure what they've released to the open > source > community at this point...but I can tell you they are working on it. > Thanks > for your interest in PreSonus and BridgeCo, and if you find out > something...let me know! Just thought I'd throw this out on the list to see what feedback I get. I hope I didn't step on anyone's toes. If you know someone who would be interested in working on drivers please pass this to them. Contact information for Presonus is available on their web page. Jan

21 years, 1 month

[linux-audio-dev] Trying to record midi with OSS

by airplays55＠yahoo.com

I am writing a HOWTO for the benefit of knoppix users who want to utilize rosegarden21 which is included on the knoppix LiveCD. Since this is for the LiveCD users, upgrading to rosegarden4 is not an option. I figured out how to get /dev/sequencer working, now sndstat tells me my first device is "MPU-401 0.0 Midi interface #1" and I am able to get rosegarden to output midi. However, I'm unable to get rosegarden21 to record any midi notes. On my midi setup screen, it lists the device as /dev/sequencer, and the record device (which I cant modify) says (null device), then if I try to record and check again, the record device says "MPU-401 0.0 Midi interface #1" but no note events got recorded into the track. I know most everybody's moved away from OSS to ALSA but does anybody remember how to test the midi input under OSS? I can't tell if my /dev/midi is working like my /dev/sequencer is. BTW I notice the notes I play on my midi keyboard do trigger the module connected to the soundcard's midi out even with no application running. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

21 years, 1 month

[linux-audio-dev] [PATCH]: better X-Station/Speedio/ReMOTE25 support

by Charles C. Bennett, Jr.

Hi All - The attached patch improves on the patch posted Friday night on the same subject. 1. a line from my copy of sound/usb/usbaudio.c didn't make the earlier patch and it's essential 2. the earlier patch was treating the output stream as USB Bulk when it should be USB Interrupt (1ms) 3. the earlier patch was not dealing with input properly causing the input side to become inoperative after the first application closes the MIDI device. As before, this has been built and tested (and tested...) on the latest Fedora Core 3 kernel from updates-released (the 741 kernel) using an X-Station with the latest X-Station firmware on a uniprocessor 32-bit intel machine. The patch should apply cleanly to any kernel tree with ALSA 1.7 and build just fine. As before I'd be glad to hear feedback from people trying this patch out with Novation Speedio or Novation ReMOTE 25 (the MIDI-only one). Thanks in Advance, ccb Next: Mixman DM-2

21 years, 1 month

[linux-audio-dev] Re: korg oasys

by Juhana Sadeharju

Hello. The old OASYS had a hardware card with five DSPs. Korg provided the patch editor for those who asked it separately. The same editor was used to make the preset patches by Korg people, if I remember correctly. It looks like I don't have the module documentation, I have only Patches and Effects doc. Perhaps it came with the patch editor. Anyone? Hey. Nobody wrote details here. Was the given url empty or what? Is it standalone keyboard with Linux inside? Is it an audio card like the old AOSYS but they now provide the patch editor running in Linux? Does it have the patch editor included? Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

21 years, 1 month

[linux-audio-dev] Lexicon USB reverb

by Steve Harris

http://namm.harmony-central.com/WNAMM05/Content/Lexicon/PR/MX200.html Seems relevant to the linux reverb plugin questions. I would imagine that the plugin sends out a packet of samples over USB, and receives a packet back, maybe with a one packet slip to reduce the plugin processing latency, at the cost of overall latency. I wonder if they'd be willing to release the USB protocol they use. Its possible they compute some of the reverb inside the plugins (eg. the ERs) as a cheap hack to hide USB latency. More likly they dont bother though. If the protocol is simple enough it could be sniffed from a windows box. No info on pricing yet. - Steve

21 years, 1 month

[linux-audio-dev] emu10k1 multichannel support

by Lee Revell

I posted this to alsa-devel but since my previous post on this list generated a lot of interest, I am just reposting it here. As promised, here's an updated patch to add real multichannel playback support (and improved multichannel capture) to the emu10k1 driver. http://www.alsa-project.org/~rlrevell/emu10k1-multichannel-v001.patch Please test it and report any problems. I am especially interested in any regressions that impact regular PCM playback (the hw:0,0 device). QuickStart: $ jackd -R -v -d alsa -P hw:0,3 -C hw:0,2 -S I tested this and it works well with 16in/16out at 128, 256, 512 frames. 32 and 64 should work too but I can't test as I'm running a stock 2.6.10 kernel for now ;-). You can check that the routing is correct by connecting a JACK client to the playback ports corresponding to the FX buses described in Documentation/Audigy-mixer.txt and Documentation/SB-Live-mixer.txt, and verifying that the output appears on that channel (the FX buses are numbered from 0 but JACK numbers clients from 1). For example (from SB-Live-mixer.txt): name='Music Playback Volume',index=0 This control is used to attenuate samples for left and right MIDI FX-bus accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. The result samples are forwarded to the front DAC PCM slots of the AC97 codec. So "alsaplayer -o jack -d alsa_pcm:playback_5,alsa_pcm:playback_6" should output to FX buses 4 and 5, which you can test by lowering the 'Music' control in alsamixer. With an SBLive, use ports 1 and 2 for the front channels, 3 and 4 for the rear channels. The Audigy uses different channels, see the above docs for more info. In addition to multichannel recording applications, this should also be useful for OpenAL implementations, which are currently restricted to using 21 sources due to the use of an extra voice per stereo PCM. This should allow up to 63 sources. This also adds some new register info including a per channel half loop interrupt that I have discovered by reverse engineering the Windows drivers. Improvements over previous versions: - Routes the 16 channels to the 16 FX buses by default. - Enables the first 16 FX capture outputs by default, required for full duplex operation at latencies lower than 512 frames. - Rewrote the voice allocator to use a more efficient round robin algorithm, eliminating the need to reserve the first 16 voices for the multichannel device. The next free voice is maintained in the card record and the search starts from there. - Use an extra voice for playback timing rather than the EFX capture interrupt. I was only ever able to get that to work at 64 frames. Also there are definite advantages to being able to use the capture and playback devices independently. - Use the newly discovered per-channel half loop interrupt source for the extra voice rather than the channel loop interrupts. For unknown reasons, this works better for multichannel playback, and does not seem to affect regular PCM playback at all. TODO: - Fix the send routing and volume controls for the multichannel device. The current (copy and paste) solution assumes either one or two voices per PCM. So the default settings work fine but changing them with the mixer is likely to have unpredictable effects. - EFX capture should capture output channels 16-32 (mostly unused now) by default, so that we only capture the sources the user has connected to the multichannel recording inputs in the DSP manager. Typically FX buses 0-15 would be connected directly to FX outputs 16-32 so the capture channels would correspond directly to the playback channels. In order for this to work the default DSP configuration has to be changed slightly. Lee

21 years, 1 month

[linux-audio-dev] Still on usx2y

by Antonio "Willy" Malara

So.. exams gone OK this semester, so i'm returning to linux-audio world :) unfortunately my tascam us122 still dont work, and the new jack backend doesn't even compile to me.. dont know what to do.. but, since someone is using this thing right now on linux, i think there's hope even for me :) willy@Zeryn:~$ uname -a Linux Zeryn 2.6.10-mm3 #1 Wed Jan 12 21:01:17 CET 2005 ppc GNU/Linux willy@Zeryn:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.8rc2 (Wed Jan 05 06:44:40 2005 UTC). willy@Zeryn:~$ jackd -d alsa -d hw:0 jackd 0.99.0 Copyright 2001-2003 Paul Davis and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK compiled with System V SHM support loading driver .. creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit control device hw:0 configuring for 48000Hz, period = 1024 frames, buffer = 2 periods Couldn't open hw:0 for 32bit samples trying 24bit instead Couldn't open hw:0 for 24bit samples trying 16bit instead Sorry. The audio interface "hw:0" doesn't support any of the hardware sample formats that JACK's alsa-driver can use. ALSA: cannot configure capture channel cannot load driver module alsa willy@Zeryn:~$ cd Samples/HistoricalBeats willy@Zeryn:~/Samples/HistoricalBeats$ aplay -v -D hw:0 amen.wav Playing WAVE 'amen.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Hardware PCM card 0 'TASCAM US-X2Y' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 4096 xfer_align : 4096 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 willy@Zeryn:~/Samples/HistoricalBeats$ willy@Zeryn:~/Pacchi/jack-0.99.10.0usx2y$ ./configure [...] willy@Zeryn:~/Pacchi/jack-0.99.10.0usx2y$ make [...] make[3]: Entering directory `/home/willy/Pacchi/jack-0.99.10.0usx2y/drivers/usx2y' if /bin/sh ../../libtool --mode=compile --tag=CC gcc -DHAVE_CONFIG_H -I. -I. -I../.. -I../../config -I../.. -I../.. -D_REENTRANT -D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -I../../config -I../.. -I../.. -D_REENTRANT -D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -MT usx2y_driver.lo -MD -MP -MF ".deps/usx2y_driver.Tpo" -c -o usx2y_driver.lo usx2y_driver.c; \ then mv -f ".deps/usx2y_driver.Tpo" ".deps/usx2y_driver.Plo"; else rm -f ".deps/usx2y_driver.Tpo"; exit 1; fi gcc -DHAVE_CONFIG_H -I. -I. -I../.. -I../../config -I../.. -I../.. -D_REENTRANT -D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -I../../config -I../.. -I../.. -D_REENTRANT -D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -MT usx2y_driver.lo -MD -MP -MF .deps/usx2y_driver.Tpo -c usx2y_driver.c -fPIC -DPIC -o .libs/usx2y_driver.o usx2y_driver.c: In function `alsa_driver_setup_io_function_pointers': usx2y_driver.c:292: error: `sample_moveswap_dS_s16' undeclared (first use in this function) usx2y_driver.c:292: error: (Each undeclared identifier is reported only once usx2y_driver.c:292: error: for each function it appears in.) usx2y_driver.c:300: error: `sample_moveswap_dS_s24' undeclared (first use in this function) usx2y_driver.c: In function `alsa_driver_configure_stream': usx2y_driver.c:334: error: `SND_PCM_FORMAT_S16BE' undeclared (first use in this function) usx2y_driver.c:334: error: initializer element is not constant usx2y_driver.c:334: error: (near initialization for `formats[3].format') usx2y_driver.c:334: error: initializer element is not constant usx2y_driver.c:334: error: (near initialization for `formats[3]') usx2y_driver.c:335: error: `SND_PCM_FORMAT_S16LE' undeclared (first use in this function) usx2y_driver.c:335: error: initializer element is not constant usx2y_driver.c:335: error: (near initialization for `formats[4].format') usx2y_driver.c:335: error: initializer element is not constant usx2y_driver.c:335: error: (near initialization for `formats[4]') make[3]: *** [usx2y_driver.lo] Error 1 make[3]: Leaving directory `/home/willy/Pacchi/jack-0.99.10.0usx2y/drivers/usx2y' [...] willy@Zeryn:~/Pacchi/jack-0.99.10.0usx2y$ if someone has some clue... :) thanx for the time spent reading this wil --- Il jazz e' lasciare che la luce brilli, lasciarla essere. -- Keith Jarrett

21 years, 1 month

[linux-audio-dev] linuxsampler crashing anyway

by Julien Claassen

Hi! I did try the following things to get current cvs to work. I used the appropriate configuration method and then edited the Synthesizer.cpp and exchanged "ASM_X86_MMX_SSE" with "CPP" as in the other synthesizer_mode functions. The sampler compiled cleanly. But when running it (with OR WITHOUT the --no-tune option) it crashes after a few seconds. It says: Zombified - calling shutdown handler. Then it keeps idling there. I have to use killall -9 linuxsampler to terminate it. CTRL-C doesn't work. Any idea. I use jack 0.99.47 (CVS from sunday. Is there anything I can do to help find this "bug?"? Some debug information, etc... 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

21 years, 1 month

[linux-audio-dev] Telephone audio and linux options

by Steven Clift

Has anyone developed any tools to capture voice conversations via a voice capable modem on linux? I am interested in using a speakerphone and phone line to webcast a meeting where I don't have a decent Internet connection. Steven Clift Steven Clift - http://publicus.net - Reply to: clift(a)publicus.net Join DoWire: http://dowire.org E-Democracy: http://e-democracy.org

21 years, 1 month

[linux-audio-dev] [ANN] liblo 0.16

by Steve Harris

Liblo, the Lite OSC library, is an implementation of the Open Sound Control [1] protocol for POSIX systems. It is written in ANSI C and released under the GNU General Public Licence. It is designed to make developing OSC applictions as easy as possible. http://plugin.org.uk/liblo/ Changes: Patch from Dave Robillard that fixes bundles of more than 4 messages. Some const char * for C++ compatibility. Added a source field to messages that represents the source from which a message was received. Useful in method handlers to determine which client sent the message. Added patch from Walco van Loon and Pix that fixes a bug in the hostname detection fallback code - Steve [1] http://www.cnmat.berkeley.edu/OpenSoundControl/

21 years, 1 month

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Linux-audio-dev January 2005