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[linux-audio-dev] How to use the alsa sequencer?

by Stephen Cameron

Hi, I'm trying to figure out how to use the ALSA sequencer with my app. (to date, I've been just using raw midi). What I'm supposed to use for the "name" parameter for snd_seq_open is not exactly clear. I've been using "hw:0,0" and it returns zero, but I'm not sure it's right (see aconnect -lio output below.); That's my soundcard, however, I'm really trying to send to a softsynth. [scameron@zuul ~]$ amidi -l Device Name hw:0,0 Audigy MPU-401 (UART) hw:0,1 Audigy MPU-401 #2 hw:0,2 Emu10k1 Synth MIDI (16 subdevices) hw:0,3 Emu10k1 Synth MIDI (16 subdevices) [scameron@zuul ~]$ I have code to open and set up an alsa client port that looks like this: struct midi_handle *midi_open_alsa(unsigned char *name) { int rc; struct midi_handle_alsa *mh; unsigned char clientname[255], portname[255]; sprintf(clientname, "Gneutronica (%d)", getpid()); mh = (struct midi_handle_alsa *) malloc(sizeof(*mh)); if (mh == NULL) return NULL; rc = snd_seq_open(&mh->seqp, name, SND_SEQ_OPEN_OUTPUT, 0); if (rc < 0) { printf("snd_seq_open returns %d\n", rc); free(mh); return NULL; } rc = snd_seq_set_client_name(mh->seqp, clientname); if (rc < 0) printf("snd_seq_set_client_name failed \n"); sprintf(portname, "Gneutronica output (%d:%d)", getpid(), 0); mh->outputport = snd_seq_create_simple_port(mh->seqp, portname, SND_SEQ_PORT_CAP_READ|SND_SEQ_PORT_CAP_SUBS_READ, SND_SEQ_PORT_TYPE_MIDI_GENERIC); if (mh->outputport < 0) printf("snd_seq_create_simple_port failed\n"); return (struct midi_handle *) mh; } struct midi_handle_alsa is just this: struct midi_handle_alsa { snd_seq_t *seqp; /* alsa sequencer port */ int outputport; }; This *appears* to work... When I run my app, I can see this: [root@zuul ~]# aconnect -loi client 0: 'System' [type=kernel] 0 'Timer ' 1 'Announce ' Connecting To: 15:0 client 14: 'Midi Through' [type=kernel] 0 'Midi Through Port-0' client 16: 'Audigy MPU-401 #2' [type=kernel] 0 'Audigy MPU-401 (UART)' 32 'Audigy MPU-401 #2' client 17: 'Emu10k1 WaveTable' [type=kernel] 0 'Emu10k1 Port 0 ' 1 'Emu10k1 Port 1 ' 2 'Emu10k1 Port 2 ' 3 'Emu10k1 Port 3 ' client 128: 'FLUID Synth (6096)' [type=user] 0 'Synth input port (6096:0)' client 129: 'Gneutronica (14693)' [type=user] 0 'Gneutronica output (14693:0)' There's my app, client 129, and I can do this: [root@zuul ~]# aconnect 129:0 128:0 [root@zuul ~]# aconnect -loi client 0: 'System' [type=kernel] 0 'Timer ' 1 'Announce ' Connecting To: 15:0 client 14: 'Midi Through' [type=kernel] 0 'Midi Through Port-0' client 16: 'Audigy MPU-401 #2' [type=kernel] 0 'Audigy MPU-401 (UART)' 32 'Audigy MPU-401 #2' client 17: 'Emu10k1 WaveTable' [type=kernel] 0 'Emu10k1 Port 0 ' 1 'Emu10k1 Port 1 ' 2 'Emu10k1 Port 2 ' 3 'Emu10k1 Port 3 ' client 128: 'FLUID Synth (6096)' [type=user] 0 'Synth input port (6096:0)' Connected From: 129:0 client 129: 'Gneutronica (14693)' [type=user] 0 'Gneutronica output (14693:0)' Connecting To: 128:0 [root@zuul ~]# Which, I think looks correct... the aconnect makes client 128:0 subscribe to 129:0, right? But, when I try to send events from my app, they don't seem to go through. Here's the event sending code: void midi_noteon_alsa(struct midi_handle *mh, unsigned char channel, unsigned char value, unsigned char volume) { struct midi_handle_alsa *mha = (struct midi_handle_alsa *) mh; snd_seq_event_t ev; snd_seq_ev_clear(&ev); snd_seq_ev_set_source(&ev, mha->outputport); snd_seq_ev_set_subs(&ev); snd_seq_ev_set_direct(&ev); ev.type = SND_SEQ_EVENT_NOTEON; ev.data.note.channel = channel; ev.data.note.note = value; ev.data.note.velocity = volume; ev.data.note.off_velocity = 0; ev.data.note.duration = 100; /* it's drums... there is no note off. */ printf("Sending event to port %d, note=%d, vel=%d, pid=%d\n", mha->outputport, ev.data.note.note, ev.data.note.velocity, getpid()); snd_seq_event_output(mha->seqp, &ev); return; } I get output from my app that looks like: Bass Drum Sending event to port 0, note=35, vel=100, pid=14693 Bass Drum Sending event to port 0, note=36, vel=100, pid=14693 Rim Stick Sending event to port 0, note=37, vel=100, pid=14693 Tom Hi Sending event to port 0, note=38, vel=100, pid=14693 Hand Clap Sending event to port 0, note=39, vel=100, pid=14693 (The channel is zero.) If I use raw midi (write to a file descriptor hooked to /dev/snd/midiC2D0 and use snd-virmidi acconnected to Fluidsynth... it works. I'm obviously missing something, but it's not obvious to me what it is... Any ideas? Thanks, -- steve __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

19 years, 3 months

[linux-audio-dev] Re: Linux Audio driver problem on TI OMAP 2430

by chandrashekhar singh

Hi Nishanth, Hope you are doing well. We have an issue with the Linux kernel 2.6 for OMAP 2430 which is available on TI site for download. The Stereo effect does not work. Would you advise us some thing, why it does work? We are planning to test our surround and other effect modules on this target but due to stereo does not work, we can't test our midi engine with effect module. Please advise us if any to come out of this problem. Best Regards chandrashekhar ----- Original Message ----- From: "Menon, Nishanth" <x0nishan(a)ti.com> To: "chandrashekhar singh" <cp_singh(a)faith.co.jp> Sent: Wednesday, June 07, 2006 11:01 PM Subject: RE: Linux Audio driver problem on TI OMAP 2430 Chandrashekar, If ur using MV kernel, then there are some interesting things u can play with :).. u can now choose to switch b/w the onboard speakers and the headset speakers using IOCTLs (don't remember offhand what it was). There is one interesting factor I had reported to the SDP team, recording can be done from headset mic (the 2.5 mm) only if I plug in a headset into the 3.5mm jack.. just thought I'd warn u on this.. minor irritant, but better know than being surprised.. Regards, Nishanth Menon > -----Original Message----- > From: chandrashekhar singh [mailto:cp_singh@faith.co.jp] > Sent: Tuesday, June 06, 2006 10:18 PM > To: Menon, Nishanth > Cc: Khasim, Syed > Subject: Re: Linux Audio driver problem on TI OMAP 2430 > > Hi Nishanth, > > Thanks for your prompt response which saved our alot of time > now we have tested our application on revised version of Board > component and it seems working perfectly. > > Thanking you all. > ======================== > \ chandrashekhar singh > \ cp_singh(a)faith.co.jp > / Faith Inc. Tokyo > / www.faith.co.jp > ======================== > ----- Original Message ----- > From: "Menon, Nishanth" <x0nishan(a)ti.com> > To: "Khasim, Syed" <x0khasim(a)ti.com>; "chandrashekhar singh" > <cp_singh(a)faith.co.jp> > Sent: Wednesday, May 31, 2006 9:52 PM > Subject: RE: Linux Audio driver problem on TI OMAP 2430 > > > Hi Chandrashekar, > Please look for modification to the following board revision numbers: > * T2 Companion Board, Revision 750-2012-001 (E). > * Connectivity Board, Revision 750-2003-002 (D). > * Enhanced LCD/Audio Board, Revision 750-2038-001 (C). > Regards, > Nishanth Menon > > -----Original Message----- > > From: linux-omap-open-source-bounces(a)linux.omap.com > [mailto:linux-omap- > > open-source-bounces(a)linux.omap.com] On Behalf Of Khasim, Syed > > Sent: Wednesday, May 31, 2006 7:29 AM > > To: chandrashekhar singh; linux-omap-open-source(a)linux.omap.com > > Subject: RE: Linux Audio driver problem on TI OMAP 2430 > > > > Hi Chandra shekhar, > > > > Which version of 2430 SDP is this and are you working with TWL4030 > (T2) > > audio codec? > > > > There was a noise issue with T2 audio codec and got resolved later. > This > > needs some hardware modifications. You might have to upgrade your > board. > > Please contact your vendor. > > > > Regards, > > Khasim > > > > -----Original Message----- > > From: linux-omap-open-source-bounces+x0khasim=ti.com(a)linux.omap.com > > [mailto:linux-omap-open-source-bounces+x0khasim=ti.com@linux.omap.com] > > On Behalf Of chandrashekhar singh > > Sent: Wednesday, May 31, 2006 6:07 AM > > To: linux-omap-open-source(a)linux.omap.com > > Subject: Linux Audio driver problem on TI OMAP 2430 > > > > Dear All, > > We are working on a porting on porting of a audio application on > > TI OMAP 2430. > > > > We have got running Monta-vista linux kernel 2.4 on this target board > > and ported our application, while playing we noticed that there is > > some noise generated by audio driver in back ground. > > > > For testing purpose we have downloaded Genaral Linux kernel 2.6 for > > OMAP 2430 from TI site and tested our appliction on this. > > > > The problem remain same on both linux. > > > > If any body had experience this problem and have got some solution > > please help me out. > > > > > > Thanks in advance. > > chandrashekhar > > _______________________________________________ > > Linux-omap-open-source mailing list > > Linux-omap-open-source(a)linux.omap.com > > http://linux.omap.com/mailman/listinfo/linux-omap-open-source > > _______________________________________________ > > Linux-omap-open-source mailing list > > Linux-omap-open-source(a)linux.omap.com > > http://linux.omap.com/mailman/listinfo/linux-omap-open-source

19 years, 3 months

Re: [linux-audio-dev] Re: How to use the alsa sequencer?

by Pedro Lopez-Cabanillas

On Sunday, 18 June 2006, Stephen Cameron wrote: > > Hi, > > > > I'm trying to figure out how to use the ALSA sequencer > > with my app. (to date, I've been just using raw midi). [..] > But if I take out this line > > snd_seq_ev_set_dest(&ev, 128, 0); > > And replace it with: > > snd_seq_ev_set_dest(&ev, SND_SEQ_EVENT_PORT_SUBSCRIBED, 0); > > And try to use aconnect to connect things together, it doesn't > seem to work. > > So, there's some piece of the puzzle regarding how aconnect > works that I'm missing. snd_seq_ev_set_subs(&ev); I guess you may want to generate the ALSA library documentation from the sources with doxygen. It is also available online: http://www.alsa-project.org/alsa-doc/alsa-lib/seq.html Regards, Pedro

19 years, 3 months

[linux-audio-dev] Re: Writing a winner takes it all gain filter.

by Bill Schottstaedt

Why do you say sndlib is "poorly maintained"? I did not get any bug reports that I remember. Also, it can be used with LADSPA, but I prefer a higher level approach.

19 years, 3 months

[linux-audio-dev] Writing a winner takes it all gain filter.

by Alex Polite

Hi there. Thanks to everyone who responded to the request about high level languages for audio work. I made a wikipage to order your input and my findings. http://tinyurl.com/p4zqo It definitely looks like Faust is the most serious contender for writing LADSPA plugins. It seems to be actively maintained and well documented. It claims to be able to generate different types of plugins/applications from the same code. I'll print the tutorial and read it over the weekend. My first project will be a winner-takes-it-all-gain filter that takes n number of inputs and lowers the gain on all but the loudest signal. I want to use this on recordings of conversations where each speaker has a separate microphone. First I tried sidechain ducking which didn't really work for me. Then I tried expanding each channel, so as to mute it when it fell under the threshold. That works pretty well but it's not perfect. This winner-takes-it-all thingy should be dead simple to implement and I expect it work pretty well. alex -- Alex Polite http://flosspick.org - finding the right open source

19 years, 3 months

Re: [linux-audio-dev] Writing a winner takes it all gain filter.

by Kjetil S. Matheussen

"Alex Polite": >Hi there. > >Thanks to everyone who responded to the request about high level >languages for audio work. >I made a wikipage to order your input and my findings. > >http://tinyurl.com/p4zqo > >It definitely looks like Faust is the most serious contender for >writing LADSPA plugins. It seems to be actively maintained and well >documented. It claims to be able to generate different types of >plugins/applications from the same code. I'll print the tutorial and >read it over the weekend. > I suggest that you rewrite the comment about snd. Writing "lispish, yuck" doesn't give you much credit as someone worth listening to. Also, for many other really good alternatives, like csound and supercollider, you have just written "naa". I suggest that you do some more research as well...

19 years, 3 months

Re: [linux-audio-dev] Writing a winner takes it all gain filter.

by Kjetil S. Matheussen

Steve Harris: >Hum. It's maybe not tactfuly expressed, but the s-expression syntax has a >number of objectors with informed positions. > >It is near one end of a broad spectrum of languages so inevitably not to >everyones taste. Sure. Syntax can be more compact without s-expressions, and the syntax can also be more formed towards specific purposes without s-expressions as well, like smalltalk that use {...} instead of (lambda ()...), and C that use {...} instead of (begin ...), and things like accessing array values or setting values requires more characters with s-expression since you can't use special characters for common tasks. But the fact that you have complete control over the language in a way that a non-lisper is probably not able to understand without ever using lisp macros weights up for all those things. However, when people normally bash lisp, its probably because of the following reason: All the paranthesis looks ugly and are confusing. For example, I actually spent almost two years programming lisp before I started to like lisp very much. The paranthesis confusion dissapeared quickly, but thinking lispish was harder. Before that, I thought python was the most beautiful language of them all. (I knew about 20 programming languages at that time.)

19 years, 3 months

[linux-audio-dev] [iua-mtg] CLAM 0.91.0

by Pau Arumi

CLAM 0.91.0 Release Announcement ´Spectral transformations, annotator, packaging, and desktop integration release' We are glad to announce the 0.91.0 CLAM release which comes by the hand with Music Annotator 0.3.2, Network Editor 0.3.1 and SMSTools 0.4.1. They are available for download as source tarballs and also as binary packages for Windows, Ubuntu dapper, Debian sid and Fedora Core 5. MacOsX binaries are not available for this release, but we promise they will be back soon. This release is the first official one which incorporates the new CLAM Music Annotator featuring chord extraction. Almost 30 new spectral transformations have been incorporated into the processing repository. Some of them are already available from the NetworkEditor. Application usage has received some extra stress on this release. Applications are better integrated on Windows and Linux desktops. Step by step application tutorials are available on the clam wiki for Music Annotator [1], SMSTools [2], Network Editor and Prototyper [3]. And, all of them all provide examples to start with. Please read these and other improvements in the changelog [4] or visit our website [5] for further details. We expect as much feedback as possible from all our users. Besides the mailing list, you can likely find us at #clam channel on FreeNode (IRC network). The CLAM team [1] http://iua-share.upf.es/wikis/clam/index.php/Music_Annotator_tutorial [2] http://iua-share.upf.es/wikis/clam/index.php/SMSTools_tutorial [3] http://iua-share.upf.es/wikis/clam/index.php/Network_Editor_tutorial [4] http://clam.iua.upf.edu/ChangeLog.html [5] http://clam.iua.upf.edu

19 years, 3 months

[linux-audio-dev] Re: Writing LADSPA plugins in high level language?

by Kjetil S. Matheussen

Phil Frost: > Subject: Re: [linux-audio-dev] Writing LADSPA plugins in high level > language? > To: The Linux Audio Developers' Mailing List > <linux-audio-dev(a)music.columbia.edu> > Message-ID: <20060614132522.GA3483(a)unununium.org> > Content-Type: text/plain; charset=us-ascii > > On Wed, Jun 14, 2006 at 07:47:36AM +0200, Alex Polite wrote: >> Hi there. >> >> Is it possible to write LADSPA plugins in anything but C/C++? I prefer >> perl, ruby or python. >> >> alex > > Anything but C/C++, yes. See FAUST [1], a compiled language designed > specificly for processing audio streams. Perl, Ruby, or Python, not > really. > > [1] <http://faudiostream.sourceforge.net/> The realtime extension for snd (scheme-like language) is another: http://www.notam02.no/arkiv/doc/snd-rt/ Here is a cool alsa softsynth written in that system: http://ccrma.stanford.edu/~kjetil/220c/

19 years, 3 months

[linux-audio-dev] gettimeofday() freeze?

by Steve

Hi, Tonight I actually for once got around to working a bit on my pet audio project, and thought I'd finally try to tackle my issues with a processor-intensive audio loop. So I started writing a simple timing class (C++) to do some profiling... Unfortunately, instead I've found myself wasting my time because my computer (or at least X) keeps freezing! It seems that whenever I put a call to gettimeofday() in my audio loop (which gets called about every 40-or-so milliseconds, a latency I'd like to reduce), and run the program, everything just freezes... no mouse, keyboard, no screen updates, etc. Reboot required. (I tried using ssh from another computer but got no response... strangely I did get a ping response however.) To be more specific, the loop is actually a callback from PortAudio. With gettimeofday commented out, there is no freeze, the software runs as expected. Originally I thought it might be a priority problem, but I tried running it without a priority boost and using "nice -n 19", but still I get a system freeze. It's strange, because I was under the impression that a user process should never be able to hang my whole system if not running as root. Has anyone else had issues with this? Is there a better way to profile my code? Unfortunately it's a GUI program (using SDL for graphics) so it's difficult to run it in text mode. I might try figuring out how to comment out all the graphics stuff though, to see if it has the same effect without X running. I'm using Ubuntu Dapper, up-to-date: $ uname -a Linux hal9000 2.6.15-22-686 #1 SMP PREEMPT Sun May 7 16:37:57 UTC 2006 i686 GNU/Linux Thanks, Steve

19 years, 3 months

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Linux-audio-dev