Linux-audio-dev

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7441 discussions

[linux-audio-dev] Creating the conditions for VST-like plugins on Linux.

by Juan Linietsky

Hi ! I'm writing this mail because at this point I'm seeing that the Linux Audio Development community is still going aimlessly on how to develop powerful audio applications. As we know, one of the key factors missing nowadays is audio plugins and synth plugins. Some audio plugin needs can be covered by LADSPA, but in other cases you need a lot more complex interfaces than just a few sliders.. (like is compressor with curves, graphical equalizers with band feedback, parametrics, multiband compressors, voice autotunning, surround spatial repositioning.. and everything available of course, that would become more intuitive and useful with more feedback) Also there is the need for instrument plugins, single or multi timbral (so you dont have to load all the instances at once). DSSI provides this (except the multitimbral) but you have to write the interface separated than the core, which not only is more difficult to program, but also very limiting because in many cases you cant show with much detail what is going inside the synth, like editing a sample, draw freeform envelopes, edit the oscilator generator, etc. And better not talk about effects, which need spectrums, vus, and other kind of complex feedback. So, I'm sure that many of you have asked yourselves, why cant we have a VST-like plugin architecture for Linux? The following are excuses: -Because we need core and interface separation, this way we warranty a clean design and network transparency -Because it would force both interface and core to be memory locked, as a result of being a jack clients, forcing more physical memory to be used -Because separating interface and core makes for a clear design. And I say, ALL THIS IS LIES. Every single of such reasons are of no concern to musicans. If I just want to make music, I would not care about any of such reasons. I'd just fire up an application and make music. All that seems to me more like academia-speech than real reasons. You can believe them or not, but it's PROVEN that none of these are needed to make music. However, there is a one and single reason of why we dont have VST-Like plugins in Linux: -It is impossible to make a plugin in a widget toolkit that will be hosted by another widget toolkit. (say, gtk on qt, or fltk on fox). Why is this? The answer is simple.. X11. An X client connects to the Xserver (via unix socket/tcp/wathever) then stays blocked in that connection until it receives events (keypress/mouse/etc), which are sent to the event loop. It is then the toolkit (gtk/qt/etc) the one responsible for handling the events. This way, even if you load many toolkits at the same time, and even each opens a connection... one of them has to stay blocked in the event loop. This is the REAL reason why we cant make a VST-Like plugin API in Linux, or EVEN port opensource VST plugins and use them here. Up to now, nobody has really bothered in solving this problem, instead most just attempt finding a workaround. So what I am proposing is to SOLVE this problem.. how? -Designing a library that will act as intermediate between the toolkits and the X Connection, handling the event loops for all of them, so they can run together. -Placing this library in freedesktop.org -Encouraging toolkit makers to support this library, and porting them to it As you see, this is not a simple task, and much less something I can do alone, so I will need help!! I am very sure that at some point we will get support from other groups which will also benefit from this. So before anything, I'd like to know who is interested in this, and who would like to help, offer experience, etc. Well, hope I get any feedback Cheers! Juan

19 years, 5 months

[linux-audio-dev] Chionic 1.0.0 Final (Sampler) Released!!

by Juan Linietsky

I have decided to put together a release and free Chionic to the world. Chionic is by far THE most advanced sampler that you can find for GNU/Linux, *BSD, Etc. It is not only extremely powerful, but t has everything configurable via UI. You can obtain the baby here: http://www.reduz.com.ar/chionic/ Lengthy, Lenghty, Insane List of features: 1.2.1 General * OSS and JACK Support (sorry, no time to make ALSA) * Integrated Virtual Keyboard for testing samples and patches (automatically detecting the mode in which you are). * Spectrum display for the output sound. * Voice Counter, so you can always see how many voices is Choinic processing. * Customizable Colors for some widgets. * Customizable keybindings for many widgets * Sliders all show the respective values, and offer small button so you can input a value by yourself, with the range and all presented. * Spinboxes go to top and bottom if you right click on the arrows. * Fully Functional ``Settings'' Menu.. no need to go like.. ``chionic -help'', tweak everything from the UI. 1.2.2 Samples * No, you dont control Samples directly, you have to make instruments. * Up to 1000 samples per bank * Loading of stereo/quadrophonic/whathever samples in as many slots as possible, or just merge into a monophonic sample, all upon import * Supports every single sample format thru libsndfile * Built in extremely complete sample editor with extensible plugins. This sample editor is meant for tweaking everything sample related: 1. Loop Begin/ Loop End. 2. Tuning, Autotuner. 3. Realtime centered zoom, from full view to individual samples using a slider. 4. Play Position Display! It displays the playback pointer INSIDE the sample, for every voice being played. * Sample Looping at any point, plus ping-pong (bidirectional) looping * Samples are stored in 16 bits normalized, then converted to 32bits floating point for mixing (not really any audible quality loss, since it's CD quality). * Samples are stored in a pool and become loaded on demand, when an instrument is using them or you are editing them. They are automatically freed when not using anymore. * If you permorm modifications to a sample in realtime, it will remain in memory until you save it, to avoid any kind of mistake. * Sample management is easy, with an interface full of indicators for the usage/refcount of every single sample. * Extremely fast resampler using selectable (cubic, linear, cosine or raw) interpolation method, which uses fixed point increment sections. You can have over 512 simultaneous voices with very little CPU usage. (yup!) on a 1Gz machine.. * The resampler also performs volume ramping and filtering, so the quality is perfect. * Integrated declicker, which will smooth out samples that dont start or end at DC. The result is a clean sound no matter if the samples have wrong DC. It also declicks samples that are being killed. * Experimental Sample mip-mapping feature.. in a way, it's the best upsampling interpolation you'd ever hear... in other way, it works like shit with looped samples.. as I need to refigure out how to make them.. so It's off by default. 1.2.3 Channels/Buffers/Buses/Sends/Inserts * Integrated Buffer Routing System, for assigning inserts and sends to your mix. * Integrated set of effects and presets for insert (Chorus, Reverb, Stereo FX, Stereo Echo, Band Limiter, 6,8,10,20,30 bands graphical EQs, OA-Pitch Shifter as internal plugins, plus LADSPA is supported too). * Sends can be assigned to midi controllers. * Very Smart algorithm for detecting inactivity of mixing channel buffers, so it can turn off inserts automatically when they are not used, for both internal and LADSPA inserts. * Buffer counter, so you can see how well the previous feature works :) * PANIC!!! button, to turn off all the voices. 1.2.4 Instruments * Unlimited Banks of 128 instruments for Patches * Patches are made by using samples from the pool * When a Patch is loaded, the samples are loaded too * Adjustable Velocity Sensitivity Curve * Pitch/Pan Scale and Center Separation, to be able to make piano-like instruments.. where the bass goes to the left, and treble to the right of the pan. * Inserts PER PATCH. Besides the other inserts, you can make an effect chain with the same internal and LADSPA effects that are bind to a Patch. Also, if you modify any parameter, it will be modified in realtime in all the channels where this Patch has been assigned too. * Up to 4 ``tones'' (sub voices) per Patch, each tone has: 1.2.5 Tones * Velocity Range (from which to which velocity does the tone accept notes) * Sensibility Curve Editing * Note Scaling, Coarse/Fine Tuning, Pan, Start Delay * Filters: Resonant LPF,HPF,BPF,BRF with Frequency Tracking for Cutoff, and Volume->Cutoff Sensibility in both ways. * Widget to edit the filter shape (cutoff/resonance) by just drawing on the Frequency Domain curve. * Randomness factor in all of the above parameters, for a more natural sound. * Freeform Envelopes (just draw a polygon) with Loop and Sustain Loops (which can take any 2 different points, so you can loop with any shape you want). The Display has Auto-Zooming and guides. Envelopes also display a one or more position guides which represent the place the envelopes are being played, in realtime! * LFOs (wich can be sync-ed or not) with sine, saw, square and noise waves, and editable Delay, Rate, Depth and Phase. * Envelopes and LFO for: Volume, Pan, Pitch, Cutoff and Resonance. Use all together if you want. * Sample/Note Table, to assign to EACH note, EACH different sample in the sample pool. * Of course, helper buttons for the above :) so it doesnt take a much time to do. 1.2.6 MIDI * ALSA_seq MIDI Input driver, can open up to FOUR ports, so you achieve 64 input channels. All channels are mapped into the buffers screen, and due to the optimized nature of Chionic, you can use all 64 of them without any slowdown. * Midi Jitter Correction. This means that you dont need to run JACK at low latency in order to avoid Jitter.. even at 16k buffer sizes, the midi sent to Chionic will sound fine, with no jitter. This was achieved using a very clever and complex hack, which shouldnt be there because JACK should support midi in the first place :) * Due to the wonderful nature of Linux, you have to make sure that nothing is stressing the /proc filesystem, or else the above feaure will not work correctly! Dont run any 'top' operations and BE CAREFUL with KSysguardd. Failure to comply with this will result in the midi sensing thread to improperly timestamp incoming events.. heck if I know why. * On the bright side, and if nothing of the sort is being done, you can run jack and chionic even as normal user and it's going to work fine. **************************** Sad News. I'm over with the development of chionic, on it's first release. I have to sadly accept that as much as it works the way I want, I cant communicate my sequencer to chionic in the way I need (because Jackd is too poor to do it the proper way). So in short, anyone who is interested in taking care of chionic for me and continuing development is MORE THAN WELCOME. I will gladly offer assistance on it's development. Also, choinic and it's Core/UI are separated at code level, so if you want to make a GTK app, or port it to Windows, it's not a hard task. Just let me know, so... CHIONIC IS DEAD UNTIL SOMEONE DECIDES TO PICK IT UP AND CONTINUE DEVELOPMENT (or until Jackd matures enough.. yeah right :)

19 years, 5 months

Re: [linux-audio-dev] Other real-time options

by Kjetil Svalastog Matheussen

Jean-Marc Valin: > Hi, > > I think a part of the conversation I had with Con Kolivas may be of > interest here: > > > <jmspeex> I still don't understand why this [unprivileged real-time] > hasn't gone in the kernel a long time ago. > <----> > <con> jmspeex since lkml is so hopeless as a forum for that sort of > thing, feel free to start an online petition to show how many people > there are. 1 of 2 things will happen - 1. you'll have thousands signing > up, or 2. there are less people doing it than you think This shouldn't be a problem. There are about 2300 subscribers to the linux-audio-user mailing-list: http://www.linuxdj.com/audio/lad/subscribe.php If someone sets up this forum, and more than twice of us sign up, that should show those arrogant lklm-people that there are really _a lot_ of us, and that we are strong, and very angry. Hah!

19 years, 5 months

[linux-audio-dev] jack newbie question

by Tim Blechmann

hi all ... another jack newbie question ... sometimes jacks kicks out my application, since the subgraph times out ... basically i have two questions about that: - is there an api function, that tells jack not to kick the application or a callback that tells the application that it has been kicked out? - is it possible to query the time out so that i can adapt the callback that it finished before this time out? thanks ... tim -- mailto:TimBlechmann@gmx.de ICQ: 96771783 http://www.mokabar.tk latest mp3: kMW.mp3 http://mattin.org/mp3.html latest cd: Goh Lee Kwang & Tim Blechmann: Drone http://www.geocities.com/gohleekwangtimblechmannduo/ After one look at this planet any visitor from outer space would say "I want to see the manager." William S. Burroughs

19 years, 5 months

[linux-audio-dev] RE:HELP:porting linux audio driver to RTLinux(rtlinux core driver)

by nobin matthew

Dear friends, The Real problem is, i need dma to fill codec controller FIFO in every 600us(8 samples still in codec FIFO and codec controller is half full, at that time it will generate interrupts), this will generate a sin wave which is used to controll another device. If codec controller did not get data in this period it sents last available sample again and again, thus producing a straight line. In Normal linux (2.4 kernel) interrupts and process scheduling may cause problem. This is the reason for porting linux audio driver to RTlinux and it is not possible to access normal linux driver from RTLinux threads. Nobin Mathew __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

19 years, 5 months

[linux-audio-dev] XITEL INPORT

by Charles A. Crayne

I have recently acquired a XITEL INPORT device, to capture music from my stereo system, and although [as shown below] it seems to be supported by the FC3 kernel, I have not yet managed to find any way to access the USB data stream. Any clues on how I should proceed? arecord -l **** List of CAPTURE Hardware Devices **** card 0: Solo1 [ESS ES1938 (Solo-1)], device 0: es-1938-1946 [ESS Solo-1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 /proc/asound/card1 total 0 -r--r--r-- 1 root root 0 Mar 29 17:00 id -rw-r--r-- 1 root root 0 Mar 29 17:00 oss_mixer dr-xr-xr-x 3 root root 0 Mar 29 17:00 pcm0c dr-xr-xr-x 3 root root 0 Mar 29 17:00 pcm0p -r--r--r-- 1 root root 0 Mar 29 17:00 stream0 -r--r--r-- 1 root root 0 Mar 29 17:00 usbbus -r--r--r-- 1 root root 0 Mar 29 17:00 usbid

19 years, 5 months

[linux-audio-dev] HELP:porting linux audio driver to RTLinux(rtlinux core driver)

by nobin matthew

Dear friends, I am porting pxa audio driver to rtlinux (rtlinux pro). I am mainly using pxa-ac97.c & pxa-audio.c(eliminated sound_core.c). I will try to register to two devices /dev/dsp and /dev/mixer in that. Since RTlinux kernel api are not much documented it seems very difficult to port dma and synchronization. Is it possible to use dma in rtlinux(since it needs dynamic memory allocation). I am sending the source code. can anybody tell me how to port this. Nobin Mathew __________________________________ Do you Yahoo!? Yahoo! Personals - Better first dates. More second dates. http://personals.yahoo.com

19 years, 5 months

[linux-audio-dev] ZynAddSubFX 2.2.0 release

by Paul

Hi. I released version 2.2.0 of ZynAddSubFX News: - the VST version of ZynAddSubFX is removed from the instalation until it will be more stable (hope soon :) ) - now, the instrument banks contains over 300 high quality instruments - added "Apply" a button from OscilGen window for PADsynth - added another parameter to ADsynth that controls the amount of all detunes of voices - adaptive harmonics postprocess - improved the VU-meter and added a RMS plot - Dvorak support for Virtual Keyboard - many bugs fixed and code cleanups ZynAddSubFX is a open-source software synthesizer for linux and windows. You can download it from http://zynaddsubfx.sourceforge.net Paul __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

19 years, 5 months

[linux-audio-dev] OSSRI Conference Agenda and Registration Information - most recent

by Susan Cragin

The Open-Source Speech Recognition Initiative is holding its first annual conference, to focus on the state of speech recognition geared toward Linux and the open-source community. DATES: Thursday evening, April 28, to Saturday noon, April 30, 2005. LOCATION: The Old Salt Inn / Lamie's Tavern, 490 Lafayette Road Hampton, NH 03842 USA. http://www.oldsaltnh.com Toll Free: 1.800.805.5050 Direct: 1.603.926.0330 Fax: 1.603.929.0017 Pre-registration not required for day sessions. However, room reservations are due by April 15th to get OSSRI rate ($89) and DSL in-room. CONFERENCE SCHEDULE: THURSDAY, APRIL 28 6:00 - 8:00 PM Registration, cheeses and crackers, socializing.(Cash bar.) FRIDAY, APRIL 29 8:00 - 9:00 Continental Breakfast 9:00 - 9:15 Opening Announcements and Welcome. (Susan Cragin.) 10:00 - 11:00 Discussion of the State of Speech Recognition today, and the issues developing a Linux model. (Led by Eric Johansson.) 11:00 - 12:00 Discussion of Small-Dialogue Applications. (Led by Turner Rentz.) 12:00 - 1:00 Lunch (Included in $55 fee.) 1:00 - 2:00 Demo / Video - NASA's Uses of Speech Recognition. (John Dowding, NASA.) 2:00 - 3:00 Breakout Session - Creating a Tool That Prompts for Speech. 3:00 - 4:00 Probablistic Grammar-Based Language Models for the Open-Microphone Task. (John Dowding.) 4:00 - 5:00 Breakout Session - Quality Problems in Speech-Gathering. SATURDAY, APRIL 30 8:00 - 9:00 Continental Breakfast 9:00 - 10:00 Sphinx as an engine - Current Status and Future Developments. (Willie Walker, Sun Microsystems / Sphinx 4 Developer.) 10:00 - 11:00 Deaf-Voiced Communication Using a Phoenetic Text Display. (Tristram Metcalf.) 11:00 - 11:15 Check-out. Bring luggage to conference room. 11:15 - 12:00 Open. COST: $55 per person includes entire conference and lunch on Saturday. $15 to attend each half-day (no lunch). ABOUT THE LOCATION: Hampton, New Hampshire is located about 1 hour north of Boston, right off Route 95. It is a 5-minute drive to Hampton Beach, a summertime resort, and about 15 minutes to Portsmouth NH, a charming seacoast village with many restaurants and shops. Swimmers should bring wetsuits. Questions, please feel free to contact me directly: Susan Cragin, Clerk, OSSRI susancragin(a)earthlink.net 781-416-1987 home 781-801-6829 cell

19 years, 5 months

[linux-audio-dev] jack and alsa design issue

by Hans Fugal

I'm writing an application that will use alsa in the common case, but be jack-capable. I'm faced with the following design question: Do I wrap the jack part to emulate the read/write of alsa, or do I wrap the alsa part to emulate the callback style of jack? In other words, do I push or pull from the audio segment of the program? As I understand it, alsa can be asynchronous but it requires using SIGIO which doesn't excite me. So I'd have to create another thread that selects and fills a ringbuffer. To adapt jack, I'd have a ringbuffer which is drained when the program pulls the audio. Adapting Jack seems the easier thing to do, but what do you think? -- .O. Hans Fugal | De gustibus non disputandum est. ..O http://hans.fugal.net | Debian, vim, mutt, ruby, text, gpg OOO | WindowMaker, gaim, UTF-8, RISC, JS Bach --------------------------------------------------------------------- GnuPG Fingerprint: 6940 87C5 6610 567F 1E95 CB5E FC98 E8CD E0AA D460

19 years, 5 months

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Linux-audio-dev