Linux-audio-dev June 2006

linux-audio-dev@lists.linuxaudio.org

70 participants
59 discussions

[linux-audio-dev] Re: [Freebob-devel] [REQUEST] test the influence of linux1394 kernel drivers on scheduling latency

by Jonathan Woithe

Pieter Palmers wrote: > This weekend I've discovered a (serious) kernel scheduling latency issue > with the current ieee1394 kernel drivers. Before I submit something > about this to lkml, I'd like some more tests. I've been able to > reproduce this on two different machines, so I suspect that this is a > more general problem. I ran the tests quickly on my firewire development machine overnight. First I started the latency monitor and left it run for about 10 minutes while I did other things, including accessing a USB stick. I then started the ieee1394 stress tester; almost immediately the monitor noted maximum latencies of the order of 1 ms (compared to tens of microseconds when the stress tester wasn't running). The output is blow. > ./latency_histogram -p 80 -n 128 -t 1 -s 1 Histogram size: 128 bins, bin size: 1us, last bin: 128us aquiring RT scheduling with priority 80 ... getting cpu speed... 67547857.763 Hz (67.548 MHz) Sleeping for 1us between timer reads... Running, press CTRL-C to stop... (+) New maximum: 62 cycles, 0.917868 (-) New minimum: 62 cycles, 0.917868 (-) New minimum: 58 cycles, 0.858650 (+) New maximum: 298 cycles, 4.411687 (+) New maximum: 43475 cycles, 643.617747 (+) New maximum: 48239 cycles, 714.145520 (+) New maximum: 52924 cycles, 783.503752 (+) New maximum: 131949 cycles, 1953.415021 (+) New maximum: 170136 cycles, 2518.747532 # of iterations: 243335 max. difference: 170136 cycles, 2518.747532us min. difference: 58 cycles, 0.858650us avg. difference: 66 cycles, 0.989601us Histogram --------- =< 1us: 33 1us : 243285 2us : 9 3us : 1 4us : 3 5us : 2 6us : 1 7us : 1 ... >= 128us: 33 The "New maximum: 298 cycles" appeared very soon after the stress tester was started, with the ones below it following soon after. Despite what the log says, this was running a 2.0 GHz "Dothan" Centrino CPU. Kernel was 2.6.16-rt25, distro was Slackware 10.2. Both the stress tester and the monitor were run with RT privilege access. The firewire interface used has a TI OHCI chipset. I apologise that the run was particularly short and that therefore the statistics aren't particularly good, but it does seem to confirm the observations you made on your machine. The large latencies only occur when the stress tester is running. Regards jonathan

18 years, 6 months

[linux-audio-dev] Re: [Freebob-devel] [REQUEST] test the influence of linux1394 kernel drivers on scheduling latency

by Pieter Palmers

Stefan Richter wrote: > Pieter Palmers wrote: >> The problem summary is that running ieee1394 ISO traffic can cause >> scheduling latency spikes up to 1ms, even for RT threads with higher >> priority. > > Do your patches to lower CPU utilization show any influence? Not tested yet. I have a hunch that the removal of the dummy read might improve the situation. But I haven't investigated this yet. > > Do you get 1394 bus resets while ieee1394stress is running? (You could > e.g. print raw1394_get_generation(handle) when it starts and when it exits.) As far as I can tell there are no bus resets, but I'll check. Pieter

18 years, 6 months

[linux-audio-dev] [REQUEST] test the influence of linux1394 kernel drivers on scheduling latency

by Pieter Palmers

Hi all, This weekend I've discovered a (serious) kernel scheduling latency issue with the current ieee1394 kernel drivers. Before I submit something about this to lkml, I'd like some more tests. I've been able to reproduce this on two different machines, so I suspect that this is a more general problem. The problem summary is that running ieee1394 ISO traffic can cause scheduling latency spikes up to 1ms, even for RT threads with higher priority. I've written a simple test suite that can be used to reproduce this behavior. The only thing needed is a firewire host controller (no firewire devices). I'd appreciate it if some people would try the test so that I can have an overview of the problem. Of course, this applies mostly to people running an -RT patched kernel. You can find the test suite here: http://freebob.sourceforge.net/old/ieee1394-latencytest.tar.gz see the README for details. Please report the maximum latency you get and the kernel/hardware you're running. Many thanks, Pieter Palmers FreeBoB developer

18 years, 6 months

[linux-audio-dev] [ANN] jack_capture V0.2.4

by Kjetil S. Matheussen

jack_capture is a small program to capture whatever sound is going out to your speakers into a file without every having to patch jack connections, fiddle around with fileformats, or set options on the argument line. This is the program I always wanted to have for jack, but no one made. So here it is. http://ccrma.stanford.edu/~kjetil/src/ Changes 0.2.3 -> 0.2.4: *Give message to stderr during recording (not only after) if any overruns occur. *Do not delete file after recording if any overruns have occured. (stupid jackreq code #$!@$) *Increased default buffer size from 0.5M to 2M.

18 years, 6 months

[linux-audio-dev] Re: linux-audio-dev Digest, Vol 33, Issue 11

by Ben Loftis

Hi Alex. What you're trying to implement is called an automixer. "Winner Takes All" is probably not going to work as well as your original expander. It will have problems when somebody coughs, drops their books, bumps the mic, tries to interject, etc, because it will duck the current speaker. Here are some of the subtleties that you might want to consider: A "filibuster" function which guarantees that once a mic is gated on, it stays on until the speaker has gone silent for a second or three, and it can't be silenced by some other mic gating on. An "AGC" (automatic gain control) algorithm on each mic that guarantees quiet speakers and loud speakers have the same perceived volume in the recording. Duck the unused mics a few dB instead of a fully gating them, or always leave one mic open. The room ambience is important and it (kinda) proves the recording hasn't been doctored. I picked up an IED 8000-series automixer which does all these things on eBay for $20.00. Dan Dugan also makes some nice ones. So if this is a one-shot deal for a serious project, you may want to consider finding something similar that is proven to work rather than rolling your own. Just my $0.02 -Ben Loftis > > 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

18 years, 6 months

[linux-audio-dev] LV2 update

by Steve Harris

There's been little news on the LV2 front here recently, all the disucssion seems to have taken place on IRC, so a quick update: Theres now a website: http://lv2plug.in/ as of a couple od days ago, thanks to Thorsten Wilms which has links to drafts of the C header file and RDF/Turtle schema. Please read the formal specs and comment. Following discussion on the l-a-u list and hard work by a lot of people there's a logo. Please don't discuss the logo except to praise it ;) choosing one was quite involved. Various generic forms (black on white etc.) will be availble on the website in due course. TODO list: * Finalise the technical aspect of the spec, get interested parties to confirm that it meets thier needs (or at least doesn't prevent them). * Write human language specification to go with .h and .ttl explaining how to use the spec, conventions etc. * Make sure there's a reasonable set of reference implementations and examples. * Test the extension mechanisms. * Publish final spec, have tea and cakes etc. - Steve

18 years, 6 months

[linux-audio-dev] LV2 plugin porting experience

by Steve Harris

I thought it might be of interest to other plugin developers to learn what my experiences were of porting my LADSPA plugins to the LV2 draft. As some of you may know the primary source for my plugins is a wierd XML format, so porting that was fiddly, but didn't involve much manual effort, "just" coding a handful of XSLT sheets. However I ported a couple by hand to get a feel for it, and basically it comes down to deleting the constants and runAdding method from the ladspa .c file, sedding some struct names and replacing occurances of LADSPA_Data with float (except the last argument to connectPort, which is a void *). It may be possible to do it automatically with a cpp/m4 hack, or a perl script or something, but I doubt its worth the effort. Writing the turtle is a little more involved, but I'm sure some enterprising person can write a program to generate .ttl from existing LADSPA .so's, then it will be a case of adding any additional data you want to express by hand. Writing the Makefile was pretty tricky, as the plugin data and code all ends up in a plugin directory, but it wasn't too bad once I figured out how best to layout the source. I decided not to use automake/autoonf/ ibtool as I felt it caused more probems than it solved with LADSPA. The Makefile would have been less critical if I didn't have quite so many plugins; I didn't want to recurse into every plugin directory, which is the obvious thing to do. - Steve

18 years, 6 months

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

by Pedro Lopez-Cabanillas

On Monday, 19 June 2006, Stephen Cameron wrote: > > http://www.alsa-project.org/alsa-doc/alsa-lib/seq.html > > Thanks, I'd found that already... it's slightly better than no > documentation. :-) But I'll take what I can get. It's a little > rough going at first when it kind of assumes you're aware of a big > picture that, in my case, I was almost totally unaware of. As you > slowly figure it out, and the bigger picture of how all the pieces > fit together -- sequencer ports, clients, channels, and how they're > named and accessed and what's what, the docs become a little clearer. > > Helps to understand it before you read it, heh. This document is the original proposal, by the ALSA sequencer author: http://www.alsa-project.org/~frank/alsa-sequencer/index.html A bit outdated, but a very intersting reading, if you want to get the big picture. Regards, Pedro

18 years, 6 months

[linux-audio-dev] loopback with hda -intel

by yvan chenard

Hi, maybe someone can help me, I find no ways to route (loop_back) the Line-in(where I have my tv-tuner pluged in) to the headphone jack, maybe someone know how to do it or know any links on documentation on how to do it. The sound card is onboard Intel D945GTP the chipset is STAC9220 (from sigmatel) (HDA_intel compatible) I am runing ALSA 1.0.11 not in the kernel(as module). kernel 2.6.16.5. The only way to get sound from LineIn to HeadPhone right now is "cat /dev/dsp > /dev/audio", but the quality is realy bad( with this command I know that the configuration of the input and output is OK). Thank You for your time. Yvan _________________________________________________________________ Mode, recettes et détente sur Sympatico / MSN Mieux vivre http://mieuxvivre.sympatico.msn.ca/Accueil/

18 years, 6 months

[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

18 years, 6 months

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Linux-audio-dev June 2006