Linux-audio-dev February 2003

linux-audio-dev@lists.linuxaudio.org

97 participants
92 discussions

[linux-audio-dev] anyone going to PyCon?

by Paul Winkler

I know there are a few pythonistas around these parts... Are any of you going to PyCon? I am! Would love to meet other linux-audio-dev regulars and lurkers. http://www.python.org/pycon/ -- Paul Winkler http://www.slinkp.com Look! Up in the sky! It's THE FEMINIST! (random hero from isometric.spaceninja.com)

21 years, 10 months

[linux-audio-dev] [ANN] JACK Rack 1.3.0

by Bob Ham

Hi all, New stuff. Good. * you can now use plugins with unequal numbers of input and output ports. any excess ports get exposed as extra jack ports with names like "amplif_1-1_i2", of the form "<shortened plugin name>_<rack position>-<plugin copy>_{i,o}<port>" * wet/dry controls for the output of each plugin * logarithmic controls * much more resilient file loader http://pkl.net/~node/jack-rack.html Bob -- Bob Ham <rah(a)bash.sh>

21 years, 10 months

Re: [linux-audio-dev] PTAF link and comments

by Laurent de Soras [Ohm Force]

Hello everybody, I'm one of the PTAF authors and I've just subscribed to this list, seeing the standard is discussed here. I will try to answer your questions and to collect feedback. I apologize for this long mail. > * Three states; created, initialized and activated. > This may be useful if plugins have to be instantiated > for hosts to get info from them. Why not just provide > metadata through the factory API? I prefered to keep the factory API quite simple. In most cases, the factory would produce only one kind of plug-in. Detaching the plug-in capabilities from the plug-in itself may become a pain. With factory-provided metadata and only two states, the trivial solution is to instanciate internally the plug-in to get its properties. If it brings it directly in the Initialized states, it will likely increase the overhead. There is also an advantage when the plug-in is a wrapper, as the factory can just reflect a directory content, possibly filtered and checked. Lastly, plug-in properties may depends on contextual data, like OS or host versions, date, CPU, environment variable, etc. - things sometimes useful for copy protections. Hmm that makes me think about the issue of dual recursion in property checking (host-plug-host-plug etc) => infinite loops. > * Assuming that audio and sequencer stuff is in different > threads, and even have plugins deal with the sync > sounds like a *very* bad idea to me. I think there is a misunderstanding here. The "Sequencer" thread is not a thread sending notes or events or whatever related to the sequence. It is the generic thread for the host, being not the GUI thread nor the audio thread. The word "Sequencer" is badly choosen, someone already told me that too. If anyone has a better idea for its name... Of course I'm 100% for having audio and sequencing in the same thread. That's why there is only one opcode to process audio and sequence data at once. However... some VST hosts actually dissocitate them, allowing process() function to overlap with setParameter() or ProcessEvents(), which is 1) confusing 2) source of bugs, hard to track 3) difficult to synchronize. > * GUI code in the same binaries is not even possible on > some platforms. (At least not with standard toolkits.) I'm presonally not familiar with Linux GUI toolkits (I'm confused with Gnome, KDE, X, Berlin, etc, sorry for my ignorance), is there a problem with them ? In this case wouldn't be possible to launch the GUI module from the plug-in ? What would you recommend ? .Agreed, this is not ideal, ohm force are a host based processing shop, so haven't thought about these issues. The TDM people will obviously have different ideas. Yes we have tought about it. TDM is not a problem here because Digidesign is absolutely opposed to wrappers or non-Digi-controlled standards, it is part of the SDK license agreement :). UAD cards are a more interesting case because their plug-ins show as VSTs. It seems that the current code architecture isn't an issue for them. Actually this is mostly a design choice. If the plug-in must assumes that its GUI and DSP codes may be run on different locations (on two computers for example), both API and plug-ins should be designed to take that into account, especially instanciation and communication between both parts. Current design is oriented toward a single object, making the API simpler and allowing flexible and hidden communication between GUI and plug- in core. Anyway the idea is indubitably interesting. > * Using tokens for control arbitage sounds pointless. Why > not just pipe events through the host? That turns GUI/DSP > interaction into perfectly normal control routing. Several APIs (MAS for example, and recent VST extensions) use a similar system, it works like a charm. I don't understant why just sending events to the host would work. If several "parameter clients" act at the same time, how would you prevent the parameter to jump continuously ? One solution is to make the host define implicit priority levels for these clients, for example 1 = automation, 2 = remote controls, 3 = main GUI. This is good enough for parameter changes arriving simultaneously, but is not appropriate for "interleaved" changes. Another solution is to use tokens. It can acts like the first solution, but also allows the host to know that user is holding his fingers on a knob, maintaining it at a fixed position, because the plug-in doesn't have to send any movement information. This is essential when recording automations in "touch" mode. > * Why use C++ if you're actually writing C? If it won't > compile on a C compiler, it's *not* C. It was for the structured naming facilities offered by C++ (namespaces). It makes possible to use short names when using the right scope and to be sure that these names wouldn't collide with other libraries. I agree that it's not essential and everything could be written in C. > * I think the VST style dispatcher idea is silly Performance of the calls is not a critical issue here. In the audio thread, there is roughly only one call per audio block. And final plug-in or host developer will likely use an wrapper, adding more overhead. But it's the easiest way to : 1) Ensure the bidirectional compatibility across API versions 2) Track every calls when debugging. By having a single entry point you can monitor call order, concurence, called functions, etc. The system is simple and can be wrapped to your solution at a very low cost (just bound checking and jump in table) > * Is using exceptions internally in plugins safe and portable? It depends on your compiler. To specify, the use of exception is allowed within a call only if it has no side effect for the caller, whatever it is. I think most of the C++ compilers are compliant with this rule; it seems to be the case for all compilers I've worked with. The sentence about exception came after many plug-in developer questions on mailing lists: "can I use safely exceptions in my plug-in ?" The answer generally given was "Yes, but restricted to certain conditions, I don't know them that much so I wouldn't recommend their use". In the PTAF API, we wanted to clarify this point. And no, C++ is not required for PTAF plug-in, because it defines a communication protocol at the lowest possible level. C/C++ syntax is just an easy and common way to write it. > * UTF-8, rather than ASCII or UNICODE. Actually it IS Unicode, packed in UTF-8. This is the best of both worlds, because strings restricted to the ASCII or partial Latin-1 (character sets are stored in UTF-8 exactly like in "unpacked" form. Alternatively, "exotic" characters may be represented using multi-byte sequences. > * Hosts assume all plugins to be in-place broken. Why? > * No mix output mode; only replace. More overhead... There is several reasons for these specifications : 1) Copy/mix/etc overhead on plug-in pins is *very* light on today's GHz computers. Think of computers in 5, 10 years... This would have been an issue if the plug-ins were designed to be building blocks for modular synth, requiring hundreds of them per effect or instruments. However this is not the goal of PTAF which is inteded to host mid- to coarse-grained plug-ins. A modular synth API would be completly different. 2) This is the most important reason: programmer failure. Having several similar functions differing only by += vs = are massively prone to bugs, and it has been confirmed in commercial products released by major companies. Implementing only one function makes things clear and doesn't requires malicious copy/paste or semi-automatic code generation. 3) Why no in-place processing ? Same reasons as above, especially when using block-processing. More, allowing "in-place" modes can lead to weird configurations like crossed-channels, without considering the multi-pins / multi-channels configurations with different numbers of inputs and outputs. Also, leaving intact the input buffers is useful to make internal bypass or dry mix. 4) One function with one functionning mode is easier to test. When developing a plug-in, your test hosts probably won't use every function in every mode, so there are bug which cannot be detected. In concrete terms, this is a real problem with VST. I think it's a good idea that an API specification takes care about programming errors when performance is not affected much. Indeed let's face it, none of us can avoid bugs when coding even simple programs. Final user will attach value to reliable software, fore sure. > * Buffers 16 byte aligned. Sufficient? I don't know. We could extend it to 64 ? For what I've read about 64-bit architecture of incoming CPUs, 128-bit registers are still the widers. Enforcing 16-byte alignment would ensure that SIMD instructions can be used, and with maximum efficiency. SSE/SSE2 still works when data is not aligned to 16 but seems slower. However Altivec produce bus error and this instruction set is the only way to get decent performance with PPCs. So for me, 16 byte is a minimum. It could be increased, but to which size ? It requires a visionary here ;) > * Events are used for making connections. (But there's > also a function call for it...) This is the PTAF features which embarass me... Having two ways to control one "parameter" is awful in my opinion. I think about replacing all these redundant functions by just one, callable only in Initialized state. It would be kinda similar to the audio processing opcode, but would only receive events, which wouldn't be dated. > * Tail size. (Can be unknown!) Yes. Think about feedback delay lines with resonant filters or waveshapers, etc. It may be only a few ms or an infinite time. The tail size is not essential because plug-in can tell if it has generated sound or not at each processing. It is rather a hint for the host. > * "Clear buffers" call. Does not kill active notes... I can't remember exactly why I wrote that ??? A fair requirement would be "can be called only when there isn't active notes any more". > * Ramping API seems awkward... What would you propose ? I designed it to let plug-in developpers bypass the ramp correctly. Because there are chances that ramps would be completly ignored by many of them. Indeed it is often preferable to smooth the transitions according to internal plug-in rules. > * It is not specified whether ramping stops automatically > at the end value, although one would assume it should, > considering the design of the interface. Hmmm... I start to see your point. You mean that it is impossible for the plug-in to know if the block ramp is part of a bigger ramp, extending over several blocks ? > * Note IDs are just "random" values chosen by the sender, > which means synths must hash and/or search... Yes. It is easy to implement and still fast as long as the polyphony is not too high, but I agree, there is probably a more convenient solution. You were talking about VVIDs... what is the concept ? > * Hz is not a good unit for pitch... Where have you read that pitch was expressed in Hz ? Pitch unit is semi-tone, relative or absolute, depending on the context, but always logarithmic (compared to a Hz scale). > * Why both pitch and transpose? Final note pitch is made of two parameters: Base Pitch and Transpose. Both are from Pitch family (semi-tones). Base Pitch is a convenient way to express both note frequency or drum hit id or other way to identify a note within the timbral palette of an instrument. This solutions refers to a debate on the vstplugins mailing-list some months ago (I think it was this ML, correct me if I'm wrong). If you want only a drum number, just round Base Pitch to the nearest integer, and do what you want with the fractional part. If you want to synthesize a piano with its "weird" tuning, remap the Base Pitch to this tuning to get the real pitch in semi-tones, possibly using a table and interpolation. If you want to make an instrument imposing a specific micro-tuning, remap Base Pitch as you want, by stretching or discretization. These are examples of the Base Pitch, giving backward compatibility with MIDI note Ids. Now you can use the Transpose information for pitch bend, here on a per-note basis (unlike MIDI where you had to isolate notes into channels to pitch-bend them independently). Thus final pitch can be calculated very easily, and log-sized wavetables switching/crossfading remains fast. I think we can cover most cases by dissociating both informations. > * Why [0, 2] ranges for Velocity and Pressure? As Steve stated, 1.0 is for medium, default intensity, the middle of the parameter course as specified in MIDI specs. But I don't get why it is "wrong" ? IMHO it isn't more wrong than the 0..64..127 MIDI scales. There is just a constant factor between both representation. > * TransportJump: sample pos, beat, bar. (Why not just ticks?) But musically speaking, what is tick, and to what is it relative ? If you want to locate a position in a piece, you need two informations: the absolute date and the musical date, if relevant. Bar has a strong rhythmic value and refering to it is important. Beat is the musical position within the bar, measured ... in beats. Tick trick is not needed since the beat value is fractional (~infinite accuracy). > * Parameter sets for note default params? Performance > hack - is it really worth it? Good question. > * Why have normalized parameter values at all? > (Actual parameter values are [0, 1], like VST, but > then there are calls to convert back and forth.) Normalized parameter value is intended to show something like the potentiometer course, in order to have significant sound variations along the whole range, along with constant control accuracy. This is generally what the user wants, unless being masochist. Who wants to control directly IIR coefficients ? (ok this example is a bit extreme but you get my point). Also automation curves make sense, they are not concentrated in 10% of the available space any more. So parameter remapping is probably unavoidable, being done by the host or by the plug-in. Let the plug-in do it, it generally knows better what to do :) Making the parameters available under a "natural" form is intended to facilitate numerical display (48% of the knob course is not a relevent information for a gain display, but -13.5 dB is) and copy/paste operations, as well as parameter exchange between plug-ins. > * The "save state chunk" call seems cool, but what's > the point, really? This is not mandatory at all, there is a lot of plug-in which can live without. It is intended to store the plug-in instance state when you save the host document. Let's take a simple example: how would you make a sampler without it ? You need to store somewhere sample information (data themselves or pathnames). It is not possible to do it in a global storage, like a file pointed by an environment variable, Windows registry, Mac preference files or whatever. Actually the content is up to you. GUI data (current working directory, tab index, skin selection, etc), control data (LFO phases, random generator states, MIDI mappings, parameters which cannot be represented with the classic way...), audio data (samples, delay line contents even)... Everything you find useful for user, avoiding him repetitive operations each times he loads a project. > * Just realized that plugin GUIs have to disconnect > or otherwise "detach" their outputs, so automation > knows when to override the recorded data and not. > Just grabbing a knob and hold it still counts as > automation override, even if the host can't see any > events coming... (PTAF uses tokens and assumes that > GUI and DSP parts have some connections "behind the > scenes", rather than implementing GUIs as out-of- > thread or out-of-process plugins.) Token is managed by the host and requested explicitly by the plug-in GUI. There is no need of hidden communication here. Or I miss something ? -- Laurent ==================================+======================== Laurent de Soras | Ohm Force DSP developer & Software designer | Digital Audio Software mailto:laurent@ohmforce.com | http://www.ohmforce.com ==================================+========================

21 years, 10 months

[linux-audio-dev] JACK 0.50 released

by Taybin Rutkin

Jack Audio Connection Kit 0.50.0 Released The Jack team is pleased to announce the release of version 0.50.0 of the Jack low-latency audio server. Jack allows applications to share data and audio devices in synchronous operation, and has already seen a year of hard testing and refinement. The API has stabilized for the foreseeable future, although backwards compatibility is not guaranteed. More information on Jack is available at the group's web site, [1]http://jackit.sourceforge.net/. Source packages for Jack 0.50.0 are available [2]here. What's new: * Audio block sizes are fixed during runtime so clients can have more efficient algorithms. * No partial blocks will be delivered. Again for efficient client algorithms. * Thread scheduling hidden from clients for better portability. * Cleanly compiles with gcc-3.3. * Works on 64-bit platforms Work is ongoing to improve transport control. Developers and users interested in Jack should sign up to [4]jackit-devel, our mailing list. References 1. http://jackit.sourceforge.net/ 2. http://jackit.sourceforge.net/releases/current/ 3. http://jackit.sourceforge.net/apps/ 4. http://jackit.sourceforge.net/lists/

21 years, 10 months

[linux-audio-dev] XAP stuff still pending

by Tim Hockin

Items we haven't agreed to a solution on (or that I missed the solution for :) 1) Do plugins load the default value for all their controls or do we have to send an event for data they already know? 2) Do we need a run_adding() form of run()? If so what are the semanitcs and how to specify the run_adding_gain?

21 years, 10 months

[linux-audio-dev] [ANN] ALSA Patch Bay 0.5.1 (bug fix)

by Bob Ham

Hi, There was a problem compiling the gtkmm ui in 0.5.0 with the latest sigc++.. oops :) Fixed now. Trivial patch attached, and a new tarball and everything up on the website. http://pkl.net/~node/alsa-patch-bay.html Bob -- Bob Ham <rah(a)bash.sh>

21 years, 10 months

[linux-audio-dev] [ANN] ALSA Patch Bay 0.5.0

by Bob Ham

Hi all, Now supports jack. It needs the patchbay-facilitating jack functions that are only in cvs atm, tho; there's been no tarball release with them yet. And yes, it's still called "ALSA" patchbay :) * added jack driver, also installing a jack-patch-bay link to alsa-patch-bay and a jack-patch-bay.desktop * added client subscriptions, similar to qjackconnect; press a read port twice http://pkl.net/~node/alsa-patch-bay.html Bob -- Bob Ham <rah(a)bash.sh>

21 years, 10 months

[linux-audio-dev] [ANN] AlsaPlayer 0.99.74

by Andy Lo A Foe

Hello! AlsaPlayer 0.99.74 is out! This release includes a ton of new features and many bug fixes. Most notable are preliminary Icecast/Shoutcast streaming support (mp3 and ogg sources), CDDB lookup for CD tracks, a new xosd interface plygin, expanded libalsaplayer API, FLAC http streaming, command line player control and much more. Work is also progressing on a new gtk2 based interface (to be released in a couple of weeks). Get your copy at: http://www.alsaplayer.org ftp://ftp.alsa-project.org/pub/people/andy/alsaplayer-0.99.74.tar.bz2 http://prdownloads.sourceforge.net/alsaplayer/alsaplayer-0.99.74.tar.bz2 Please report bugs, comments, etc. to the alsaplayer-devel mailing list. Enjoy! Andy @ alsaplayer.org

21 years, 10 months

[linux-audio-dev] ZynAddSubFX 1.0.7

by Nasca Paul

Hi. I relased ZynAddSubFX 1.0.7 . News: - some settings (like samplerate) are set at runtime (by comand line) - added Distorsion effect - added controllers, and NRPNs for changing all effects parameters by midi - bugs removed and other improovements See at http://sourceforge.net/projects/zynaddsubfx or http://zynaddsubfx.sourceforge.net. __________________________________________________ Do you Yahoo!? Yahoo! Mail Plus - Powerful. Affordable. Sign up now. http://mailplus.yahoo.com

21 years, 10 months

RE: [linux-audio-dev] newest audio server for Linux (yep, yet ano ther)

by STEFFL, ERIK (SBCSI)

> -----Original Message----- > From: David O'Toole [mailto:dto@gnu.org] > > > Probably. I think it was Bjarne Stroustrup who said > > > something along the > > > lines of "Every use of a define is an instance of a programmer not > > > programming correctly." But I was just wondering if it was some > > > portability thing or something. > > > > I think the basic idea is not to use preprocessor for > programming... > > Did Bjarne say that before or after he invented templates? :-) :-) I have exactly the same problem with templates, it pretends to be dynamic while it's just statically generated (=similar to preprocessor, which I guess is your point) erik

21 years, 10 months

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Linux-audio-dev February 2003