Linux-audio-dev

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7516 discussions

[linux-audio-dev] a Varga Csound compile problem

by Dave Phillips

Greetings: I've been trying to solve this one for a while but I can't see the light. I'm compiling Istvan Varga's beta 4.24.2 Csound for Linux, but I'm stuck here: g++ -pipe -Wall -O2 -ffast-math -fomit-frame-pointer -finline-functions -funroll-loops -fno-strict-aliasing -fPIC -DPIC -I. -I. -I./include -I./opcodes -DWINDOWS -DSFIRCAM -DSYS5 -DLINUX -DHAVE_TERMIOS_H -DIV_VERSION -DPIPES -DMACROS -DHAVE_FLTK -D_REENTRANT -DMYFLT=float -march=i686 -falign-functions=16 -falign-loops=16 -falign-jumps=16 -frename-registers -mno-ieee-fp -mno-sse -fno-exceptions -fno-rtti -fno-unroll-loops -I/usr/local/include/FL -c widgets.cpp -o widgets.o widgets.cpp: In method `int Fl_Knob::handle (int)': widgets.cpp:230: `atan2' undeclared (first use this function) widgets.cpp:230: (Each undeclared identifier is reported only once for each function it appears in.) widgets.cpp: In method `void Fl_Knob::draw_scale (int, int, int)': widgets.cpp:289: `cos' undeclared (first use this function) widgets.cpp:290: `sin' undeclared (first use this function) widgets.cpp:310: `log10' undeclared (first use this function) widgets.cpp:278: warning: `float ca' might be used uninitialized in this function widgets.cpp:278: warning: `float sa' might be used uninitialized in this function widgets.cpp: In method `void SNAPSHOT::get (vector<ADDR_SET_VALUE, allocator<ADDR_SET_VALUE> > &)': widgets.cpp:1171: `::pow' undeclared (first use here) widgets.cpp:1172: `log' undeclared (first use this function) widgets.cpp: In function `void fl_callbackJoystick (Fl_Widget *, void *)': widgets.cpp:1759: warning: `float val' might be used uninitialized in this function make: *** [widgets.o] Error 1 [dlphilp@localhost Csound-4.24.1]$ I understand that the missing functions are math functions, but math.h is included at the start of widgets.cpp. Unfortunately widgets.cpp is about 104 KB so I didn't want to send it along with this message. If anyone wants to look at it I'll gladly send it off-list. But it seems I've run into this kind of problem before, that it's some sort of C vs. C++ thing... but I also know less than zero about C++... (and only a wee bit more about C)... TIA to anyone who can assist ! Best regards, == Dave Phillips The Book Of Linux Music & Sound at http://www.nostarch.com/lms.htm The Linux Soundapps Site at http://linux-sound.org

22 years, 6 months

[linux-audio-dev] Direct Stream Digital / Pulse Density Modulation musing/questions

by Dustin Barlow

I read an interesting article on Direct Stream Digital (DSD) / Pulse Density Modulation (PDM) entitled "A Better Mousetrap" by Brian Smithers in the May 2003 issue of Electronic Musician. Since, Brian did a good job explaining PDM/DSD in quasi-layman terms, I'll just quote snippets from his article to set the stage for my questions. <quote a="EM: A Better Mousetrap" p="96"> In DSD, the amplitude of the waveform is represented by the relative density of the ones and zeros within the bitstream; for that reason the technique is also known as Pulse Density Modulation (PDM). A full positive waveform results in all ones and, whereas a full negative waveform results in all zeros. Silence would yield a string of alternating ones and zeros (Alternating ones and zeros reflect the "neutral" density of silence.) Sony claims a resulting dynamic range of 120db for DSD. In the simplest terms, DSD compares the energy of a waveform at each sample point to the accumulated energy of previous sample periods. If the energy is higher, a value of 1 is recorded; if the energy is lower, a value of 0 is recorded. Because the bitstream represents the change (delta) in energy compared to the sum (sigma) of the previous energies, another name for this technique is Sigma-Delta Modulation (SDM). </quote> The article goes on to explain how DSD uses a sampling rate of 2.8224 MHz resulting in a frequency response that extends to 100kHz. The higher sample rate can also improve stereo/surround imaging. <quote a="EM: A Better Mousetrap" p="96"> One more justification for DSD is that the steep filters required for PCM to conform to the Nyquist theorem can themselves degrade the sound due to their sharp cutoff. In fact, it has become common for PCM analog-to-digital (A/D) converters to start with a 1-bit conversion stage and then apply a digital decimation filter to convert the bitstream to PCM format. DSD essentially bypasses the decimation filter and stores the 1-bit data directly. </quote> DSD/PDM appears to be a superiour technique for recording and playing audio material. Granted, this technology may never catch on because of all the hardware and software changes that would be required to mirror what a typical PCM based DAW currently does. But, if DSD/PDM does catch on, and DAWs start being produced, how will this effect current audio DSP techniques? How would JACK's current design be effected by a 1-bit stream format? The article mentions a program called Pyramix (Windows) which features DSD support. However, for Pyramix to do EQ, dynamics, reverb processing, and to display waveforms and vu levels, it converts DSD to a "high quality" PCM format. I suspect that if DSD/PDM has any hope of succeeding in the current DAW world, there will have to be some real-time lossless converter constructed to seemlessly bridge the formats. But that of course begs the question of why use DSD in the first place if you just are forced to "decimate" it PCM to process it? Sony is currently using DSD/PDM in their SACD format, and are positioning themselves to be the competing format to PCM based DVD-A. The fact that something like this shows up in EM means that at some level the format is being given some credence and is being watched by others then just the audiophile recording crowd. Is DSD/PDM just a flash in the pan, or does it really represent a new wave of thinking regarding storing/processing audio data? Dustin _________________________________________________________________ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus

22 years, 6 months

Re: [linux-audio-dev] soft synth as plugin

by Florian Schmidt

Hi, i came across your question while browsing the web for information on the alsa9 pcm plugin architecture.. My thought on this would be: Just make your soft synth a normal application but be sure to make it work with the alsa pcm - plugin layer of alsa. Then people can use the pcm plugin - layer to mix and route the signal. The midi stuff in alsa isn't really hard (just finding docu is :). Anyways, a good option to add would be jack support. but again the alsa-pcm-plugin layer has a jack-plugin that enables every alsa app to talk to a jack server. I don't know about the details in latency and timing, but on todays systems, the alsa layer shouldn't hit performance that bad. Anyways, i think it is crucial, that the synth works with the pcm layer so people without hardware mixing capabilities can run multiple instances of the synth(s) without the hassle of setting jack up.. Florian Schmidt -- mista.tapas(a)gmx.net sound clips - http://www.arted.biz/mista.tapas

22 years, 6 months

[linux-audio-dev] SoundFontCombi v0.005

by holborn

Hi! Sorry for the english .. :-) SoundFontCombi takes the advantages of ALSA sequencer and the wavetable devices for emulate a synthetizer. I'm not programmer, this program dont have project page, CVS, or mailing list, is in proof of concept rigth now. And only the binaries are available. Is my first program in C++, my first graphic program for *nix, and belive me the source code needs a "making up" rigth now. Maybe next month when i learned how to do a Makefile :-) SoundFontCombi was tested in Debian sid and Gentoo. Requires FLTK v1.x.x (tested on v1.1.1) and ALSA with sequencer. The binaries are available in : http://personal.telefonica.terra.es/web/soudfontcombi/sfc-v0.005.tar.gz Any commets will be apreciated. Josep

22 years, 6 months

[linux-audio-dev] A C++ question

by Nick Tsocanos

Hi I know it's not a C++ specific list, but maybe someone can shed some light on this. I have encountered this problem with all C++ compilers and I don't know exactly what it is. Sometimes I might make a class, and inherit it from another. Or, the class just has virtual functions in it ( I am planning to derive others from it). In any case, I am trying to access a virtual function in an instantiated instance at runtime, and when I do, the program crashes. I try to see if it is reaching the function (printf("Hi!") from inside or something), but it doesn't. It is like the vtable is being destroyed somehow. So, is it possible to smash the vtable of a class? I can't imagine all compilers have the same quirk, because I have seen this problem on half a dozen compilers. Usually it is in classes where I am doing heavy memory management. My guess is that the vtable can be corrupted somehow by a smash.

22 years, 6 months

[linux-audio-dev] jackd fails to start: cannot create shm segment /jack-shm-registry (No such file or directory)

by Ivica Bukvic

Hi all, any ideas on this one? I have 2 soundcards configured: sblive rme 9652 my modules.conf looks like this: #install rme9652 Hammerfall buffer first before memory gets fragmented pre-install snd-hammerfall-mem alias eth0 3c59x probeall usb-interface usb-uhci alias /dev/nvidia* nvidia # ALSA native device support alias char-major-116 snd options snd major=116 cards_limit=3 alias snd-card-0 snd-emu10k1 alias snd-card-1 snd-rme9652 alias snd-card-2 snd-usb-audio # OSS/Free compatibility # SBLIVE (Card 0) alias char-major-14 soundcore alias sound-slot-0 snd-0 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 #We'll use Hammerfall (Card 1) only as an ALSA device # USB-MIDI (Card 2) alias sound-slot-2 snd-2 alias sound-service-2-1 snd-seq-oss alias sound-service-2-8 snd-seq-oss lsmod reflects that everything is loaded ok: Module Size Used by Tainted: P isofs 28244 1 (autoclean) zlib_inflate 21156 0 (autoclean) [isofs] nls_iso8859-1 3516 1 (autoclean) snd-pcm-oss 42468 0 (autoclean) snd-mixer-oss 14296 0 (autoclean) [snd-pcm-oss] snd-seq-oss 30016 0 (autoclean) (unused) snd-seq-midi 4576 0 (autoclean) (unused) udf 92128 0 (autoclean) sr_mod 16952 2 (autoclean) floppy 56028 0 (autoclean) snd-emu10k1-synth 5116 0 (unused) snd-emux-synth 32092 0 [snd-emu10k1-synth] snd-seq-midi-emul 6652 0 [snd-emux-synth] snd-seq-virmidi 4556 0 [snd-emux-synth] snd-seq-midi-event 5800 0 [snd-seq-oss snd-seq-midi snd-seq-virmidi] snd-seq 44560 2 [snd-seq-oss snd-seq-midi snd-emux-synth snd-seq-midi-emul snd-seq-virmidi snd-seq-midi-event] snd-usb-audio 47744 0 (unused) snd-rme9652 21164 0 snd-hammerfall-mem 2848 0 [snd-rme9652] snd-emu10k1 71348 0 [snd-emu10k1-synth] snd-pcm 79680 0 [snd-pcm-oss snd-usb-audio snd-rme9652 snd-emu10k1] snd-timer 19272 0 [snd-seq snd-pcm] snd-rawmidi 18304 0 [snd-seq-midi snd-seq-virmidi snd-usb-audio snd-emu10k1] snd-page-alloc 7960 0 [snd-emu10k1 snd-pcm] snd-util-mem 2984 0 [snd-emux-synth snd-emu10k1] snd-ac97-codec 40416 0 [snd-emu10k1] snd-seq-device 5740 0 [snd-seq-oss snd-seq-midi snd-emu10k1-synth snd-emux-synth snd-seq snd-emu10k1 snd-rawmidi] snd-hwdep 6272 0 [snd-emu10k1] snd 41476 0 [snd-pcm-oss snd-mixer-oss snd-seq-oss snd-seq-midi snd-emux-synth snd-seq-virmidi snd-seq-midi-event snd-seq snd-usb-audio snd-rme9652 snd-emu10k1 snd-pcm snd-timer snd-rawmidi snd-util-mem snd-ac97-codec snd-seq-device snd-hwdep] parport_pc 25320 1 (autoclean) lp 8224 0 (autoclean) parport 35232 1 (autoclean) [parport_pc lp] agpgart 41248 3 (autoclean) nvidia 1593216 10 (autoclean) nfsd 74992 8 (autoclean) af_packet 16872 1 (autoclean) 3c59x 29936 1 (autoclean) supermount 74908 2 (autoclean) ide-scsi 11440 1 scsi_mod 105556 2 [sr_mod ide-scsi] ide-cd 33952 0 cdrom 31712 0 [sr_mod ide-cd] usb-midi 19256 0 (unused) soundcore 6500 0 [snd usb-midi] usbmouse 2936 0 (unused) keybdev 2720 0 (unused) mousedev 5332 1 hid 21028 0 (unused) input 5760 0 [usbmouse keybdev mousedev hid] usb-uhci 25932 0 (unused) usbcore 75616 1 [snd-usb-audio usb-midi usbmouse hid usb-uhci] rtc 8892 0 (autoclean) ext3 62252 3 jbd 41500 3 [ext3] my /etc/asound.conf looks like this: pcm.sblive { type hw card 0 } ctl.sblive { type hw card 0 } pcm.rme9652 { type hw card 1 } ctl.rme9652 { type hw card 1 } When I do: aplay -D sblive <filename> it works just fine. Yet, when I do jackd -d alsa -d hw:0 or jackd -d alsa -d sblive or simply jackd -d alsa I get the following _every time_: jackd 0.68.0 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 key for /tmp/jack/shm/jack-shm-registry is 0x6a06252b cannot create shm segment /jack-shm-registry (No such file or directory) cannot create engine jack main caught signal 15 I am using MDK 9.1 with low-lat kernel (otherwise vanilla), and the CVS jack (4/17/03). The rpm for MDK (0.66) also has the same issue. Any help on this one is greatly appreciated! Thank you! Sincerely, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico

22 years, 6 months

[linux-audio-dev] SoundFontCombi v0.005

by holborn

22 years, 6 months

[linux-audio-dev] How can you make Jack always available?

by Nick Tsocanos

Hello, How can I make Jackd always run? Right now I have to set it up in a console window. I'd like to learn how to run it always at startup and not have to start/stop it when I need it. Also how do you use Jack in real-time mode? In order to run it I have to be root, but if I run it as root, nobody else can see Jack unless they are also root. I wish somebody would write a Jack-HOWTO so dummies like me can figure out how to run this thing. Thanks

22 years, 6 months

[linux-audio-dev] Found out what was wrong with my LADSPA manager programmer

by Nick Tsocanos

More like a golden buddha child but hey... I got them all working, it was problemos in my plugin manager code. The plugins were fine. Are you itching to know what the problem was? I'll tell you. There is the mysterious number called NAN. I found out where NAN was coming from. I was calculating the log(number) to put in the control value The culprit number=>0. Put 0 into log, and it makes NAN. Put NAN into plugin dry/wet mix control, and plugin makes zillions of NANs. And all those NANs will make your computer run slower than an Atari 800. What's even worse is trying to convert NAN to an integer and play it on a sound card.. I guess between the massive buffer underruns and trying to synthesize all NANs was why I was hearing the most horrible noises I ever heard in my life. I have no idea what NAN becomes when you cast it to an int. Some demon posessed number I suppose. It doesn't sound like white noise either. If you want to hear the most hideous noise try it. Put log(0) into a control volume and watch what happens. Try it with about 15 plugins at once. It is super loud too. I am sure it will destroy your speakers too. Don't try it with expensive speakers. Now which controls do it? A good many of them. It's not really a design defect of the plugin programmer. I am sure, they don't expect anyone but an idiot to put NAN into the control value. I didn't do it on purpose! It was a mistake! A simple error on my part. I hope somebody else has done this by accident too.. it'll make me feel better...

22 years, 6 months

[linux-audio-dev] LADSPA Hosting

by Nick Tsocanos

I got my app to host plugins, but I am having problems figuring out how to filter plugins out that won't work in my app. I only need F/X processors. If a plugin does not process an audio stream, or is used only for synthesis, or has peculiar controls that are not defined in the hints, than I want to ignore them. Is it possible? How can I find out how a plugin is used? I am reading the API header but it's not entirely clear what properties determine how a plugin is used or how I can tell if a plugin won't work for my app. Thanks

22 years, 6 months

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Linux-audio-dev