Linux-audio-dev February 2003

linux-audio-dev@lists.linuxaudio.org

97 participants
92 discussions

[linux-audio-dev] Call for participation: MMA GMPI working group

by RonKuper＠Cakewalk.com

The Generalized Music Plug-In Interface (GMPI) working group of the MIDI Manufacturer's Association (MMA) is seeking the input of music and audio software developers, to help define the technical requirements of GMPI. The objective of the GMPI working group is to create a unified cross-platform music plug-in interface. This new interface is hoped to provide an alternative choice to the multitude of plug-in interfaces that exist today. Among the many benefits of standardization are increased choice for customers, lower cost for music plug-in vendors and a secure future for valuable market-enabling technology. Like MIDI, GMPI will be license free and royalty free. Phase 1 of the GMPI working group's effort is to determine what is required of GMPI: What sorts of capabilities are needed to support existing products and customers? What are the emerging new directions that must be addressed? Phase 1 is open to any music software developer and is not limited to MMA members. It will last a minimum of three months, to be extended if deemed necessary by the MMA. Discussions will be held on an email reflector, with possible meetings at major industry gatherings such as AES, NAMM and Musik Messe. Following the collection of requirements in Phase 1, the members of the MMA will meet to discuss and evaluate proposals, in accordance with existing MMA procedures for developing standards. There will be one or more periods for public comment prior to adoption by MMA members. If you are a developer with a serious interest in the design of this specification, and are not currently a member of the MMA, we urge you to consider joining. Fees are not prohibitively high even for a small commercial developer. Your fees will pay for administration, legal fees and marketing. Please visit http://www.midi.org for more information about membership. To participate, please email gmpi-request(a)freelists.org with the word "subscribe" in the subject line. Please also provide your name, company name (if any) and a brief description of your personal or corporate domain of interest. We look forward to hearing from you. Sincerely, Ron Kuper GMPI Working Group Chair

21 years

[linux-audio-dev] Bristol Synth

by Anthony

Hi, I've been playing a lot with bristol synth and really love it. So much so that I've been trying to 'Jackify' it. Actually, I'm pretty much done, but can't figure out the internal audio format. Its interleaved floats I think, but not normalised to [-1,1]. If any of the developers are here could you help me out? I can hear noise, but I need to tune the maths. TIA. --ant

21 years, 4 months

Re: [linux-audio-dev] Re: image problem [was Re: [Alsa-devel] help for a levelmeter]

by Patrick Shirkey

>> That said, I think Patrick is right to start thinking about this now. Thanks. >I think he's completely right - I'm not sure about this bank account >thing but I do think that now is the time to be demoing, talking up and >generally approaching people and companies about Linux music software. >I wrote us up (and mentioned a few other apps) in the latest edition of >Linux User - John at mstation.org has been very kind so far as well. >Now is the right time to be talking to people and getting the >"products" out there. If it works - why not tell people about it? The reason I believe we need to have various bank accounts are because we cannot afford to waste money on excessive service charges and not everyone has access to credit cards. If we have the accounts in the right countries then people can just donate cash. From a professional perspective we need to show our prospective clients that we have sound financial thinking. It's mostly a subconscious need that consumers have. They want to know that the money they are investing is being given to people/companies/organisations who use it. Most people don't really care how it is used although we have the moral justification on our side too. This is from the Sound on Sound advertising package. "The main target market of Sound On Sound is the professional and semi-professional musician who is the kind of person that will have the spending ability to purchase a large range of products from synthesizers to samplers, mixing desks to microphones, multitracks to monitors, effects to expanders and computer hardware and software. They are not time wasters who do not know their profession - they are serious and mature individuals working with a reasonable budget." If we want to appeal to this audience we need to prove to them that they are investing in professional audio. We need to wine them and dine them (metaphorically). If they look into our commmunity and say these are just amateur geeks who have made some interesting things happen it won't work. If we take the intiative and lead them into our world they will come at it from the perspective that we are professionals who have created a very credible concept that we are proud of and want them to enjoy using. They will ask "What kind of cash have we invested" and if we come back with "Ahh, well we don't actually have a scope on the financial side of our open community." They are just going to look around for a while and leave. If we can show them that not only are we mathematics and logics wizards but that we also have solid business sense then they are going to stick around and see what we have to offer. A lot of them will probably invest just to test the waters or to keep up with the play. I want to see an advertising campaign happen that will educate and encourage the mass of potential user to take the step. I also want to make sure that we have covered our asses when they finally walk in through the doors. It's a choice between being amatuer enthusiasts or professionals. If we come across as professionals people won't give a toss about geekyness. -- Patrick Shirkey - Boost Hardware Ltd. For the discerning hardware connoisseur Http://www.boosthardware.com Http://www.djcj.org - The Linux Audio Users guide ======================================== "Um...symbol_get and symbol_put... They're kindof like does anyone remember like get_symbol and put_symbol I think we used to have..." - Rusty Russell in his talk on the module subsystem

21 years, 5 months

[linux-audio-dev] XAP spec - early scribbles

by Tim Hockin

All - here are some early scribbles of a XAP spec. I took the doc that I wrote as an overview and started formalizing it. Please read it over and pooint out the places where it is missing stuff? The Table of Contents needs a lot more meat. Once I have that, I can write a lot or a little on each subject. Again - EARLY SCRIBBLES :) That said, the ToC, content, organization, structure, and spelling are all up for abuse. Have at it The XAP Audio Plugin API Specification $Id: xap_overview.txt,v 1.1 2003/01/15 10:55:50 thockin Exp $ 0.0 Meta 0.1 Guilty Parties 0.2 Goals 0.3 Terminology 0.4 Conventions 1.0 Overview (introduce ideas, no details) 1.1 Controls 1.2 Events 1.3 Channels 1.4 Ports 2.0 The Descriptor 2.1 Meta-Data 2.2 State Changes 2.2.1 Create 2.2.2 Destroy 2.2.3 Activate 2.2.4 Deactivate 2.3 Channels 2.4 Port Setup 2.5 Controls and EventQueues 2.6 Processing Audio 2.7 Errorcodes Host struct - Threading Control struct - RT - we have SYS controls Port struct Events - ramping -EVQ and Events Channels Ports Instruments and Voices We have Voice Controls Special controls - sys controls - MIDI controls Tempo and Meter Sequencer Controls Macros 0.1 Guilty Parties XAP is a combined effort of many people on the linux-audio-dev (linux-audio-dev(a)music.columbia.edu) mailing list. The discussion is open, and anyone interested is welcome to join in. 0.2 Goals The main goal of this project is to provide an API that is full-featured enough to be the primary plugin system for audio creation and playback applications, while remaining as simple, lightweight, and self-contained as possible. The focus of this API is flexibility tempered with simplicity. 0.3 Terminology In order to read this document without having your head spin, you should probably understand the following terms, first. * Plugin: A chunk of code, loaded or not, that implements this API (e.g. a .so file or a running instance). * Host The program responsible for loading and controlling Plugins. * Instrument/Source: An instance of a Plugin that supports the instrument API and is used to generate audio signals. Many Instruments will implement audio output but not input, though they may support both and be used an an Effect, too. * Effect: An instance of a Plugin that supports both audio input and output. * Output/Sink: An instance of a Plugin that can act as a terminator for a chain of Plugins. Many Outputs will will support audio input but not output, though they may support both and be used as an Effect, too. * Voice: A playing sound within an Instrument. Instruments may have multiple Voices, or only one Voice. A Voice may be silent but still active. * Event: A time-stamped notification of some change of something. * Control: A knob, button, slider, or virtual thing that modifies behavior of the Plugin. Controls can be master (e.g. master volume), per-Channel (e.g. channel pressure) or per-Voice (e.g. aftertouch). * Port: An audio input or output. * EventQueue A control input or output. Plugins may internally have as many EventQueues as they deem necessary. The Host will ask the Plugin for the EventQueue for each Control. FIXME: what is the full list of things that have a queue? Controls, Plugin(master), each Channel? * VVID Virtual Voice ID. Part of a system that allows sequencers and the like to control synth Voices without having detailed knowledge of how the synth manages Voices. * Channel A grouping of Controls and Ports, similar to MIDI Channels. * Tick The unit of musical time, used with the tempo and meter interfaces. The unit is decided by the maintainer of the timeline, in order to keep musical time calculations exact as far as possible. * Cue Point A virtual marker on the musical time line, marking a position to which the plugin should be able to jump at any time, without delay. This is used by hard disk recorders, and other plugins that may need to perform time consuming and/or nondeterministic processing as a result of timeline jumps. 0.4 Conventions //FIXME: datatypes (XAP_foo) and documentation style 1.0 Overview XAP Plugins live in shared object files. A shared object file holds one or more plugin descriptors, accessed by index. Each descriptor holds all the information about a single plugin - it's identification, meta-data, capabilities, controls, and access methods. The Plugin descriptors are retrieved by an exported function in each shared object. XAP plugins are always in one of two states - ACTIVE or IDLE. IDLE Plugins are not capable of processing audio, and must be activated. Plugins will spend most of their time in the ACTIVE state. After loading Plugins, the Host instantiates them, establishes Port and EventQueue connections, and activates them. Once ACTIVE, a Plugin expects to be run repeatedly on small blocks of data. When audio processing is done, the Host can deactivate the Plugin. XAP is designed to be used in realtime scenarios. XAP plugins specify their realtime capabilities, and Hosts can allow or disallow operations based on that information. All XAP audio data is processed in 32-bit floating point form. Values are normalized between -1.0 and 1.0, with 0.0 being silence. 1.1 Controls XAP uses the idea of Controls as abstract carriers of Plugin parameters and other information. Controls can represent things like knobs and buttons, but they can also represent things like filenames, MIDI aftertouch or channel pressure. Not only do they represent audio parameters, but they can represent chunks of system information, such as tempo or transport position. Like audio hardware, knobs and other controls can be global to a Plugin. However, XAP also allows Instrument Plugins to provide per-Voice controls. Controls come in a few datatype flavors, and can have min/max limits and default values, as well as hints to the host about what they are, semantically. Hosts can use the hints to automatically connect things, where appropriate. Controls get their data via Events. Events can immediately set the value of a Control, or they can establish a ramp - a target and duration for the Plugin to change the control more smoothly. 1.2 Events Almost everything that happens during the ACTIVE state is communicated via Events. The Host can send Events to Plugins, Plugins can send Events to the Host, and Plugins can send Events to other Plugins (if they are so connected). All Events are timestamped. That means that any Control change, or any other Event is sample-accurate. XAP hosts have a running timer which counts sample-frames - this is what the timestamp is based on. Events are passed to Plugins on EventQueues. This allows a Plugin to receive any number of Events with a minimal per-Event overhead. 1.3 Channels //FIXME: 1.4 Ports This is an audio API, so it wouldn't be complete without some mechanism to transport audio Data. This is a Port. Each Port carries a single stream of mono audio data. Plugins may allow the Host to disable Ports. If a Port is disabled, the Plugin will not read from or write to it. If a Port is not disabled, it must be connected by the host to a valid buffer. 2.0 The Plugin Descriptor The Plugin descriptor is a static(*) data structure provided by the Plugin to describe what it can do. Descriptors are retrieved by calling the xap_descriptor() function of a XAP shared object. This function is called repeatedly with an index parameter, starting at 0 and incrementing by one on each call. The function returns the Plugin descriptor for each index, up to the number of Plugins in the shared object file, at which time it returns NULL. Plugin indices are always sequential, and once a NULL is returned, the host can assume there are no more Plugin descriptors to be queried. (*) The descriptor for wrapper Plugins may change upon loading of a wrapped Plugin. This is a special case, and the Host must be aware of it. //FIXME: how? 2.1 Meta-Data The Plugin descriptor provides several fields for Plugin meta-data. The fields are as follows: id_code: the vendor and product encoding api_code: the XAP API version code of this Plugin ver_code: the Plugin version code - used to identify Plugin data flags: Plugin-global flags label: a short, unique Plugin identifier name: the user-friendly Plugin name string version: the version string author: the author string copyright: the copyright string license: the license string url: the URL string notes: notes about the Plugin 2.2 State Changes As mentioned above, Plugins exist in one of two states - ACTIVE and IDLE (well, three if you count non-existance as a state). The Plugin descriptor holds the methods to create and destroy instances of Plugins, and to change their state. 2.2.1 Create A Plugin is instantiated via it's descriptor's create() method. This method receives two key pieces of information, which the Plugin will use throughout it's lifetime. The first is a pointer to the Host structure (see below), and the second is the Host sample rate. If the Host wants to change either of these, all Plugins must be re-created. This is where the Plugin's internal structures can be allocated and initialized. There is no required set of supported sample rates, but Plugins should support the common sample rates (44100, 48000, 96000) to be generally useful. If the Plugin does not support the specified sample rate, this method should fail. Hosts should always check that all Plugins support the desired sample rate. Once created, the Plugin instance is in the IDLE state. 2.2.2 Destroy Plugins are destructed via the descriptor's destroy() method. Plugins can only be destroyed when in the IDLE state. All Plugin-allocated resources must be released during this method. After this method is invoked, the Plugin handle is no longer valid. This function can not fail. 2.2.3 Activate From the IDLE state, a Plugin can be changed to the ACTIVE state via the activate() method. Passed to this method are two arguments which are valid for the duration of the ACTIVE state - quality level and realtime state. The quality level is an integer between 1 and 10, with 1 being lowest quality (fastest) and 10 being highest quality (slowest). Plugins may ignore this value, or may provide less than 10 discrete quality levels. The realtime state is a boolean value which is boolean TRUE if the Plugin is in a realtime processing net or boolean FALSE if it is not realtime (offline). This method can only be called from the IDLE state. Once ACTIVE, a Plugin may process audio. 2.2.4 Deactivate From the ACTIVE state, a Plugin can be changed to the IDLE state via the deactivate() method. This method can only be called from the ACTIVE state. 2.3 Channels //FIXME: 2.4 Port Setup //FIXME: kinda needs Channels Once a Plugin is loaded, the Host must connect the audio Ports. All Ports in a Plugin must be connected or disabled. The Plugin descriptor provides a connect_port() method, which the host must call to connect a buffer pointer to a Port. Once connected, a Port remains connected to the specified buffer until the Host disables it or connects it to a different buffer. All Plugins are assumed to not use the same input buffers as output buffers, unless the Plugin flags indicate that it safely handles in-place processing. Plugins may allow the Host to disable Ports, rather than connect them. The Plugin descriptor provides an optional disable_port() method. If this method is provided, and it returns successfully, the Host can ignore this port. Once disabled, a Port remains disabled until the Host connects it, at which point it becomes enabled. 2.5 Controls and EventQueues //FIXME: kinda needs Channels In order to be of any use, a Plugin must provide Controls. Control changes are delivered via Events. Events are passed on EventQueues. Each Control has an associated EventQueue, on which Events for that Control are delivered. In addition, there is an EventQueue for each Channel and a master Queue for the Plugin. The Plugin can internally use the same EventQueue for multiple targets. The Host queries the Plugin for EventQueues via the get_input_queue() method. In order to allow sharing of an EventQueue, the get_input_queue() method also returns a cookie, which is stored in each Event as it is delivered. This allows the plugin to use a simpler EventQueue scheme internally, while still being able to sort incoming Events. Controls may also output Events. The Host will set up output Controls with the set_output_queue() method. 2.6 Processing Audio A Plugin which has been activated and properly set up, may be called upon to process audio. This is done through the descriptor's run() method. This method gets the current sample-frame timestamp as an argument. In this method the Plugin is expected to examine and handle all new Events, and to read from or write to it's Ports. This method may only be called from the ACTIVE state. 2.7 Errorcodes All this stuff needs to be integrated somewhere... Tempo and Meter ---- XAP uses Controls for transmitting tempo and meter information. If a Plugin defines a TEMPO control, it can expect to receive tempo Events on that control. The Host must define some unit of musical-time measurement (Tick), which represents the smallest granularity the host wants to work with. This is the basis for tempo and meter. The host publishes the current count of Ticks/Beat via the host struct. Control: TEMPO Type: double Units: ticks/sec Range: [-inf, inf] Events: Hosts must send a TEMPO Event at Plugin init and when tempo changes. Control: METER Type: double Units: ticks/measure Range: [0.0, inf] Events: Hosts must send a METER Event at Plugin init and when meter changes. Hosts should send a METER Event periodically, such as every measure or once per second. Control: METERBASE Type: double Units: beats/whole-note Range: [1.0, inf] Events: Hosts must send a METERBASE Event at Plugin init and when meter changes. This mechanism gives Plugins the ability to be aware of tempo and meter changes, without forcing information into plugins that don't care. A Plugin can sync to various timeline events, easily. The Host struct also provides a mechanism to query the timestamp of the next Beat or Bar, and to convert timestamps into the following time formats: * Ticks * Seconds * SMPTE frames Sequencer Control ---- XAP plugins may be aware of certain sequencer events, such as transport changes, positional jumps., and loop-points. These data are received on Controls. Control: POSITION Type: double Units: ticks Range: [0.0, inf] Events: Hosts must send a POSITION Event at Plugin init, when transport starts, and when transport jumps. Hosts should send a POSITION Event periodically, such as every beat, every measure or once per second. Control: TRANSPORT Type: bool Units: on/off Events: Hosts must send a TRANSPORT Event at Plugin init and when transport state changes. Control: CUEPOINT Type: double Units: ticks Range: [0.0, inf] Events: Hosts must send a CUEPOINT Event when Cuepoints are added, changed, or removed. Control: SPEED Type: double Units: scalar Range: [-inf, inf] Events: Hosts must send a SPEED Event at Plugin init and when play speed changes. Instruments and Voices ---- XAP instruments can be either voiced or non-voiced. Non-voiced instruments are essentially always on, and their output is controlled purely by controls, such as the gate control of a modular synth or the hand-distance of a theremin. Non-voiced instruments are monophonic per-channel. Voiced controls are more structured. They must handle Virtual Voice IDs (VVIDs). VVIDs are unsigned 32 bit integers, which are allocated by the host and passed to instruments via the XAP_EV_VVID_* events. Instruments may use VVIDs as a direct index into the host structure's VVID table, which is an array of unsigned 32 bit integers. The data in the host VVID table is for use by the instrument. VVIDs must be allocated before use and de-allocated before re-use. Instruments can maintain an internal mapping between VVIDs and actual voices, which allows them to handle voice allocation in a purely abstracted manner from the host. Once allocated, the host can send events for a VVID until the VVID is de-allocated. A VVID has two states - active and inactive. After allocation, the VVID is inactive. Control events received while inactive can be assumed to set the control state for the activation event. A VVID is activated via the VOICE control, which all instruments must provide. Once in the active state, the instrument may produce sound for the voice. A VVID is deactivated via the VOICE control, which puts the VVID in the inactive state. It is important to note that just because a VVID has received a VOICE OFF event, it is not necessarily silent. It may have a long release phase, which is dependant on the instrument. Control: VOICE Type: bool Units: on/off Events: Hosts send VOICE Events any time after a VVID has been allocated but before it has been deallocated. Some instruments will choose to only examine some controls at activation time (such as velocity for MIDI-like instruments) or at deactivation time (such as release velocity). These controls are referred to as latched. Setting an init-latched control after activation may or may not set the value, and may or may not have any effect - this is instrument dependant. A VVID may be re-used. That is to say, the host can leave a VVID allocated after deactivation, and re-activate it later. This can be used for emulation of MIDI synths, where the VVID is akin to the MIDI pitch. A VVID may be deallocated before the associated voice is done playing. The instrument should continue to play the voice, as no VOICE OFF event has been received. Likewise, a voice may end before it has received a VOICE OFF (such as a drum hit). The VVID can change to the inactive state and track control changes until another VOICE ON is received. Because VVIDs and voices have no exact correlation, instrument plugins have a great deal of control over their voice operations. An Instrument can be mono, and cut the voice and restart for each new VOICE ON, or it man be massively polyphonic. The host always alerts the plugin to the state of a VVID, and to changes of that state. Once a VVID has been deallocated, it can be re-used. Plugins can assume that VVIDs are at least global per-Plugin. The host will not activate the same VVID in different Channels at the same time.

21 years, 7 months

[linux-audio-dev] question re: hammerfall cards

by Thomas Vander Stichele

Hey, I just got a new box and two hammerfall 9636 cards for a project at work. The motherboard is a KT4 Ultra that has some onboard audio chip I don't care about but can't turn off either in the bios. I've put in the first hammerfall card, got alsa 0.9.0 rc7 from the freshrpms site (it's a redhat box), and started configuring things. I've got the feeling I'm missing some things, so here are some simple questions. a) It looks like the Hammerfall driver doesn't have a mixer interface, is this correct ? Here's an ls in the relevant dir: [root@framboos asound]# ls card0/ id pcm0c pcm0p rme9652 b) It looks like the onboard audio chip is controlled by an OSS driver, it doesn't show up in the alsa drivers either, which is fine by me, since I'm not going to use it. Is there any problem with OSS modules being loaded at the same time as ALSA modules ? c) I bought the card so I could record optical S/PDIF. The manual says I need to tell the card that I want the ADAT1 Input source to be Optical S/PDIF. Right now I get, in /proc/asound/card0/rme9652: [root@framboos card0]# cat rme9652 RME Digi9636 (Rev 1.5) (Card #1) Buffers: capture cdc00000 playback cda00000 IRQ: 5 Registers bus: 0xde000000 VM: 0xd088a000 Control register: 44008 Latency: 1024 samples (2 periods of 4096 bytes) Hardware pointer (frames): 0 Passthru: no Clock mode: autosync Pref. sync source: ADAT1 ADAT1 Input source: ADAT1 optical IEC958 input: Coaxial IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 Dolby: off IEC958 sample rate: error flag set ADAT Sample rate: 44100Hz ADAT1: No Lock ADAT2: No Lock ADAT3: No Lock Timecode signal: no Punch Status: 1: off 2: off 3: off 4: off 5: off 6: off 7: off 8: off 9: off 10: off 11: off 12: off 13: off 14: off 15: off 16: off 17: off 18: off Which I read as the Input source being ADAT optical instead of S/PDIF. How do I set it to S/PDIF ? d) the card came with a sub-D-connector that connects to the card's 15 pin port, which branches off two RCA jacks. I don't suppose these RCA jacks provide an analogue output by any chance on which I can monitor for sound ? I'm sure I'll bug you with more questions later on, but these are the most pressing ones at this point. Any help is greatly appreciated. Thomas -- The Dave/Dina Project : future TV today ! - http://davedina.apestaart.org/ <-*- thomas (dot) apestaart (dot) org -*-> Oh, baby, give me one more chance <-*- thomas (at) apestaart (dot) org -*-> URGent, the best radio on the Internet - 24/7 ! - http://urgent.rug.ac.be/

21 years, 7 months

[linux-audio-dev] POSIX clocks now in linux 2.5

by Bob Ham

>From the 2.5.63 changelog: o POSIX clocks & timers Phat :) -- Bob Ham <rah(a)bash.sh>

21 years, 7 months

[linux-audio-dev] Re: [linux-audio-user] Re: New LADSPA Hints (Was: [ANNOUNCE] tapiir-0.7.1)

by Mike Rawes

Think this needs to go to LAD too... --- Steve Harris <S.W.Harris(a)ecs.soton.ac.uk> wrote: > I think there was one more too, but I can't think what it was... I remember one about sidechains or something. I don't know what a sidechain is, though, I just recall seeing it somewhere :) > I agree AUDIO_RATE_CONTROL should be renamed. > > There was a suggestion on ardour-dev that a hint to say whether control outs > were supposed to be informative or a source of control data might help, > but I'm not sure about it. Not sure what 'informative' means here... what information do we get if we ignore the control data on the output? > Does someone want to reword these in a more meaningful way? If not I'l do > it, then you'l be sorry ;). I'll have a go :) (The LADSPA_IS_* things will need to be added too) /* Hint MOMENTARY indicates that that a control should behave like a momentary switch, such as a reset or sync control. LADSPA_HINT_MOMENTARY may only be used in combination with LADSPA_HINT_TOGGLED. */ #define LADSPA_HINT_MOMENTARY 0x40 /* Hint RANDOMISABLE indicates that it's meaningful to randomise the port if the user hits a button. This is useful for the steps of control sequencers, reverbs, and just about anything that's complex. A control with this hint should not result in anything too suprising happening to the user (eg. sudden +100dB gain would be unpleasant). */ #define LADSPA_HINT_RANDOMISABLE 0x80 /* Plugin Ports: Plugins have `ports' that are inputs or outputs for audio or data. Ports can communicate arrays of LADSPA_Data (for audio or continuous control inputs/outputs) or single LADSPA_Data values (for control input/outputs). This information is encapsulated in the LADSPA_PortDescriptor type which is assembled by ORing individual properties together. Note that a port must be an input or an output port but not both and that a port must be one of either control, audio or continuous. */ [...] /* Property LADSPA_PORT_CONTINUOUS_CONTROL indicates that the port is a control port with data supplied at audio rate. */ #define LADSPA_PORT_CONTINUOUS_CONTROL 0x10 - Mike > > - Steve > > > On Wed, 15 Jan 2003 18:13:35 +0000 > > Steve Harris <S.W.Harris(a)ecs.soton.ac.uk> wrote: > > > > > There have been a few suggestions recently, I'l try to summarise them > > > for comment. > > > > > > MOMENTARY. A hint to suggest that a control should behave like a > > > momentary switch, eg. on for as long as the user holds down the > > > key/mouse button/whatever. Useful for reset or sync controls for > > > example. Would be useful in the DJ flanger. Only applies to TOGGLED > > > controls. > > > > > > AUDIO_RATE_CONTROL. Hints than an audio control should/could be > > > controlled by a high time res. slider or control data, but shouldn't > > > be connected to the next audio signal by default. I can't think of any > > > simple examples off hand, but combined with MOMENTARY it could be used > > > for sample accurate tempo tapping. > > > > > > RANDOMISABLE. Hints that its useful/meaningful to randomise the port > > > if the user hits a button. This is useful for the steps of control > > > sequencers, reverbs, and just about anything that's complex. Allows > > > you to specify which controls can be randomised without anything too > > > supprising happening to the user (eg. sudden +100dB gain would be > > > unpleasent). __________________________________________________ Do You Yahoo!? Everything you'll ever need on one web page from News and Sport to Email and Music Charts http://uk.my.yahoo.com

21 years, 7 months

[linux-audio-dev] Help wanted with audio app design

by Arthur Peters

I have been working on an audio processing/synthesis application that should have the power of Reason and Buzz plus the ability to be used effectivly in live proformance. Some of you may have read my earlier emails on this subject. I call it Voltage. The main features that I want but are not availiable in other software are: - Linux and Open-Source - the ability to route and process sequence data (MIDI like data). This would allow very powerful chordization and arpegiation. - the ability to create sequence loops in a flexible way (like Reason Matrix except without the limitations). I'd also like to be able to record/modify them live without stoping the playback. - the ability to control things in complex ways using scripting of some sort (maybe as a machine that is scripted to create certain sequence events) What I am looking for is someone to help me (an inexperienced software designer) design this program. That person don't nesseserily need to help code the program. I've desided I need help after multiple failed attempts. I think my current ideas are closer to a working design than past ones, but I don't want to code it then find out doesn't work (like I did last time round). If you are willing to help or have any tips or books I should read, etc. please email me. Thanks much. -Arthur -- Arthur Peters <amp(a)singingwizard.org>

21 years, 7 months

Re: [linux-audio-dev] Fwd: CSL Motivation (fwd)

by Stefan Westerfeld

Hi! On Tue, Feb 25, 2003 at 07:48:11PM +0200, Kai Vehmanen wrote: > Date: Tue, 25 Feb 2003 12:20:22 -0500 > From: Paul Davis <paul(a)linuxaudiosystems.com> > Reply-To: linux-audio-dev(a)music.columbia.edu > To: linux-audio-dev(a)music.columbia.edu > Subject: Re: [linux-audio-dev] Fwd: CSL Motivation > > >There are discussions on kde-multimedia about > >the future of Linux/Unix multimedia (especially sound). > >This is one of the most interesting messages. > > CSL is proposed primarily as a wrapper layer around existing APIs. as > such, it seems to me to have no particular merits over PortAudio, > which has the distinct advantages of (1) existing CSL also "exists". > (2) working on many platforms already and You're right about that one: CSL is not as complete as PortAudio w.r.t to portability. > (3) using well-developed abstractions. I do not believe that something is ever a "well-developed abstraction" by itself. Something is always a "well-developed abstraction" from something, designed to achieve a certain purpose. From the PortAudio homepage: | PortAudio is intended to promote the exchange of audio synthesis software | between developers on different platforms, and was recently selected as the | audio component of a larger PortMusic project that includes MIDI and sound | file support. This clearly states the purpose: if you want to write audio synthesis software, then you should use PortAudio. Then, I assume, the abstraction is well- developed. However, it does not state: "... is intended to play sound samples with sound servers easily. Or: ... is intended to port existing applications easily. Or: ... is intended to let the application choose its programming model freely." No. PortAudio makes a lot of choices for the software developer, and thus provides an easy abstraction. This will mean, however, that actually porting software to PortAudio will probably be hard (compared to CSL), whereas writing new software for PortAudio might be convenient, _if_ the software falls in the scope of what the abstraction was made for. > CSL was > written as if PortAudio doesn't exist. I don't know if this a NIH > attitude, or something else, but I see little reason not use consider > PortAudio as *the* CSL, and by corollary, little reason to develop Yet > Another Wrapper API. Well, I gave you some. The paper gives some more. Basically, CSL is intended for porting _most_ free software, whereas PortAudio is intended for portable synthesis software. I think PortAudio would benefit in _supporting_ CSL, rather than aRts for instance, because CSL is more generic ; once new sound servers (like MAS) come up, you need not patch PortAudio all the time, but just one place: a CSL driver. The same is valid for other meta-frameworks like SDL. > the only reason i was happy writing JACK was > precisely because its not another wrapper API - it specifically > removes 90% of the API present in ALSA, OSS and other similar HAL-type > APIs. I am glad you did write JACK, although back then I thought it was just another try to redo aRts (and we had some heated discussions back then), because some people seem to like it. If some people will like CSL, why not? If you added CSL support to JACK right now, you would never need to bother with any of the "sound server guys" like me again, because you could always say: "support CSL in your sound server thing, and then JACK will support your sound server". On the other hand, if you added JACK support to CSL, you could also mix the output of all of these "sound servers" into JACK, without endangering your latency properties. Cu... Stefan -- -* Stefan Westerfeld, stefan(a)space.twc.de (PGP!), Hamburg/Germany KDE Developer, project infos at http://space.twc.de/~stefan/kde *-

21 years, 7 months

[linux-audio-dev] LAD meeting - Linux Sound Night

by Frank Barknecht

Hallo, with the LAD meeting getting closer, I'm getting a bit curious about, what the plans are for the open "Linux Sound Night" on 15.3.? Will we hear some of you guys perform and Paul records it? ciao -- Frank Barknecht _ ______footils.org__

21 years, 7 months

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Linux-audio-dev February 2003