Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7513 discussions

by Jens M Andreasen

I suppose you've all noticed this newish class of ultra portable cheap netbooks based on the Intel Atom? Forget about that, there's apparently already a new kid on the block. The tiny Beagle Board based on Texas Instruments OMAP3530 has been available for some time, then came this week the Pandora - a pocketable computer with built in game-controller, also based on the 3530 - and caused quite a stir on Slashdot. And now today I read that Dell is going to introduce a fullsized Vista laptop that as an added bonus, alternatively can fastboot into one these tiny OMAP environments. Nokia also has a product in the pipe ... And they are all running the Linux kernel of course. What is this hype all about? The bad News -------------- * Random access memory is on chip and hardwired to 128MB. This means that Desktop Environments like Gnome is out of the equation. (I realize that some might reckon this as good news ..) The good News ---------------- Hardware OpenGL ES 2.0 and an Arm processor with support for vectors of float as well as double aside, the one thing that strikes me the most is: * It's a 2 Watt system! .. including everything and the kitchen sink. So if not really running on candle light, it is not a far cry. A pocketable solar cell should be enough to get you up on your marks again, in case you run out of juice. At these power levels, battery lifetime finally appears irelevant. The real star of the show - and why I am mentioning anything at all of all of this - is the included DSP: * The TMS320C64x+ At first I was a little disappointed that this is not a floating point DSP - we have kind of settled for floats as the standard carrier for audio here lately - but since those domains are already covered by the Arm vector co-processor, I reluctantly downloaded the 1000 page manual to see if I could figure out some of what it is all about ... TI describes their DSP as an "advanced very long instruction word (VLIW) architecture", which is another way of saying that a bunch of instructions are grouped together and excuted in parallel on seperate execution units, each on their own data .. not totally unlike the these days familiar notion of SIMD, where a single instruction is executed in parallel, also on multiple data, but in the VLIW case of course way more flexible. The execution units are eight in numbers, all slightly different from each other except for two that are specialized in memory address calculations, store and load, and do not resemble anything else. This symmetric pattern of two applies to the other execution units as well. In fact - except for a few corner cases - you could view the the whole DSP as being a dual processor, each half having its own four 32bit execution units for load/store, (dual) 16bit fixpoint madd, 32/40bit shift and 32/40bit fixpoint mul as well as their own register file. The capabilities of the units overlap somewhat and is not set in stone the way I just outlined above. Furthermore, instructions may be predicated, that is to say that depending on a condition the individual instructions may be replaced with a noop so that one (or more parts) of this 8 way instruction mayhem can relax and enjoy having arrived at a result instead of mindlessly overwriting it, while the rest do their best to clean up their act. Oh, and did I mention that each VLIW can be set up to execute in parallel, serial or anyting in between .. If the above does not sound like your favourite sunday soduko, do not dispair. For private evalution or opensource developement, there is an optimizing C-compiler available as a free download from TI themselves. Final Words ------------ Both of the boards available at low cost to developers for the time being utilizes the same audio codec which also doubles up as a decision engine for power saving state (or perhaps that should be the other way around?) Samplerates are for playback: 96K, 48K, 44.1K, 32K and similar for recording except no 96K. The bittiness is 16. There are two channels in and two out and ALSA is supposed to work out of the box. References ----------- The Pandora Wiki has a convenient link page: http://pandorawiki.org/Hardware_documentation

16 years, 11 months

Guide to Linux Sound APIs

by Lennart Poettering

Heya! At the audio microconf at the Linux plumbers conference one thing became very clear: it is very difficult for programmers to figure out which audio API to use for which purpose and which API not to use when doing audio programming on Linux. Someone needed to sit down and write up a small guide. And that's what I just finished doing. I'd thus like to draw your attention to this new (long) blog story of mine containing this guide: http://0pointer.de/blog/projects/guide-to-sound-apis I'd be very thankful for comments! Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

16 years, 11 months

Re: [LAD] Guide to Linux Sound APIs

by Victor Lazzarini

That was not my experience. I put together a pulseaudio IO module for Csound using the simple API (pulse/simple.h) in about half an hour. It seemed much simpler than any alternative. And it seemed to do everything I needed from it. Victor At 13:59 29/09/2008, you wrote: >On Sun, 28.09.08 09:38, Paul Davis (paul(a)linuxaudiosystems.com) wrote: > > > > Also, I guess it depends on how you upgrade, because my workstation is > > > 8.04, which is upgraded every year or so for 2 years and a half now, > > > and I don't have pulseaudio. One of the package I wan't to add sound > > > support for is for science mostly, and many people are still using > > > Ubuntu Dapper, Fedora 3, etc... > > > > > > So it does not look like pulseaudio is that great if you want to > > > support various linux and have very little needs for audio. > > > > As Lennart tried to make reasonably clear, the primary goal of > > PulseAudio is NOT to act as a new API, but to act as a new > > *infrastructure* that supports existing APIs transparently. > > I am sure that he would be happy if it eventually takes over the world > > and everybody writes apps using its API, but that doesn't appear to be > > the goal right now. > >The reason why I don't ask application developers at this time to >adopt the native PA API is that it is a relatively complex API since >all calls are asynchronous. It's comprehensive and not redundant, but >simply too complex for everyone but the most experienced. > >Lennart > >-- >Lennart Poettering Red Hat, Inc. >lennart [at] poettering [dot] net ICQ# 11060553 >http://0pointer.net/lennart/ GnuPG 0x1A015CC4 >_______________________________________________ >Linux-audio-dev mailing list >Linux-audio-dev(a)lists.linuxaudio.org >http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev Victor Lazzarini Music Technology Laboratory Music Department National University of Ireland, Maynooth

16 years, 11 months

Specification issues in open systems

by yikyak

The following is a cross-post of an exchange that took place on the rosegarden-devel mailing list. I'm posting it here because I think it hints at something fairly serious with the current state of open audio specifications and issues with their implementation. It is not my intent to start a flame war. I come at this with over twelve years of involvement (albeit chiefly at a hobbyist level) with audio development and production on windows system (predominantly VSTis). I've used linux since the late nineties, but these issues have become more important to me since fully ditching Windows about 18 months ago. The open source philosophy has always been that, if something were not to your tastes, you were completely free to build it, or alter something else. This is true, and one of the things greatly appreciated about the system. The key issue, though, is the ability to use that facility; sometimes it is not as available as it appears. There are a great many people who would offer much more to open audio, I feel, but issues like the following need to be addressed. all the best, Chris Williams Chris Williams <yikyak> wrote: > Chris Cannam wrote: >> Chris Williams <yikyak> wrote: >>> Hello, >>> >>> I've been developing a toy DSSI plugin, partly to become >>> more familiar with the spec and partly as a platform for >>> writing base code for use with less toy instruments. I've >>> been using Rosegarden and jack-dssi-host as test hosts. >>> >>> All was going well until I increased the number of DSSI / >>> LADSPA output audio ports in the plugin. For some reason, >>> I expected that rosegarden would create extra synth >>> channels in the audio mixer for these outputs, but this >>> didn't happen. Instead, one 'synth' audio channel was >>> maintained in the mixer for the synth, out of which all >>> outputs could seemingly be heard and controlled globally. >>> >>> Does Rosegarden mix-down all the audio outputs of a given >>> synth before feeding the signal to its own internal signal >>> path, or am I misunderstanding something / being an idiot? >>> The number of outputs shows up correctly in the synths >>> 'controls' dialog along with its ID, but this seems to be >>> the only place at which they're discernible outside of the >>> synth. (I'm using v.1.7.0 from Arch's repository). >>> >> >> No, you aren't misunderstanding anything. Rosegarden is >> very simplistic in this respect -- it mixes the number of >> audio outputs up or down to match the number of channels >> on the track, which is always either 1 or 2 depending on >> whether the stereo button in the instrument parameters is >> pushed in or not. >> >> If the aim is to accept multi-MIDI-channel input and send >> multi-channel output in order to support multiple effective >> instances on separate tracks, the "DSSI way" to do that is >> to have several actual instances (perhaps sharing data >> behind the scenes) and then call run_multiple_synths once >> for all of them instead of run_synth on each one. >> Rosegarden will do this if you have the same plugin set for >> more than one track (and it supports run_multiple_synths). >> >> Unfortunately, that mechanism is rather different from any >> other plugin format. >> > > Thanks Chris, that's just the information I was looking for. > > I was thinking more of the situation where where you have > *single*-midi-channel input but, as with some synths and > samplers, want to run the output to different banks of > effects (e.g. LADSPA plugins) depending on the specific > midi note or the range in which that note lies (output > groups). This seems "theoretically" possible under DSSI > using only run_synth() (given an idiosyncratic parsing of > the DSSI spec) but not if the host routes all output ports > through the same audio channel. At the same time, I can see > the problem from the host developers' perspective: the DSSI > spec uses LADSPA's port system and there's no good reason > for an effect's output ever to be routed to multiple 'host > audio channels'. > > The two other ways of doing it would seem to be: > > 1) Incorporate a 'channel' system and ladspa hosting system > *internal* to the instrument which would then only ever > need a stereo output to the host (the massive downside of > this being that it unnecessarily replicates complex > functionality already provided by the host); or > > 2) Use run_multiple_synths() in a hackish manner for which > it probably wasn't fully intended, as you'd be writing to > separate LADSPA handles but essentially using only one midi > event queue (this would also have the nasty side effect of > requiring multiple redundant instances simply to use their > output ports). You could probably do something equally > nasty by playing with the descriptor returning and > instantiate functions. > > Anyway, thanks again. It's not a Rosegarden-specific issue, > I know; more strangeness in the DSSI spec coupled with my > own ignorance. I've been thinking about this some more, and can't let it lie. It seems to me that there's definitely a bug or broken implementation involved somewhere. Under jack-dssi-host, if I request n outputs, I get n outputs, any one of which is routable to wherever I wish to send it using jack_connect or qjackctl. Under rosegarden, if I request n outputs, I always get either one or two, depending on the stereo settings of the host- ascribed audio channel. This is a problem, because any summing rosegarden does *could* have been done internally in the plugin. What's the point of requesting, configuring and filling >2 output ports if the client is never *effectively* going to get them? If the client wanted them bumped down to one (or two) buffers, it could -- would -- have done this itself and needn't have requested and configured >2 outputs in the first place. Now, one way of approaching this is to say, "Oh, well; it's an implementation bug in rosegarden and there's nothing that can be done". That's OK as far as it goes, but it doesn't actually solve the problem / ambiguity and it's questionable as to whether it's true. Another approach is to say that rosegarden implements the DSSI spec correctly and the bug is actually in the spec itself. This is *far more serious* -- and it's far more serious for two reasons. Firstly, rosegarden is currently the only fully-featured standalone DSSI host. How are the other hosts currently implementing it (MuSE etc.) going to behave? Are they going to implement rosegarden's behaviour, or are they going to implement jack-dssi-host's behaviour? In either case, you have a situation where plugin developers simply don't know what the host is going to do, which kind of defeats the point of a specification. At this point a cynic might say that the uptake of DSSI is such that it doesn't matter too much and that the likes of LV2 will solve this problem anyway, but this brings us to the second serious problem (which the more well-versed among you will already have realised): If this ambiguity isn't specific to rosegarden but is inherent in the spec, then *LV2 doesn't solve it either*. Part of the problem with DSSI, if this is where the problem lies, is that it encourages host developers to treat synth / sampler plugins as extensions of effects plugins for the sake of simplicity. This works to an extent, but there's a subset of functionality that they do not have in common (the ability to leverage other effects plugins and host capabilities for two). There is no output-negotiation scheme in either LV2 *or* DSSI beyond the output port request and instantiation stage, and neither spec makes clear what a plugin can expect having acquired those outputs, or, from the other side, what a host is fully expected to provide in response to a given request. The assumption goes unstated, which is why there is this problem to begin with. Perhaps an extra negotiation is required, so that both the host *and* the plugin know unambiguously the contexts in which they are operating. It's not reasonable to expect a host to grant any and all resources that a synth plugin might request (2^32 outputs, for example), but at the same time the *plugin* needs to *know* the context in which it's operating, and not have to guess it based on external knowledge of specific host implementations. Anyway, I appreciate that the rosegarden developers are not fully responsible (if at all) for this situation, but it is a major issue, I feel, when it comes to standard adoption and plugin development with open systems. I may cross-post this to the LAD list as it seems just as relevant to them as to you. Thanks for your time, Chris Williams

16 years, 11 months

Re: [LAD] Something like Processing for audio

by Stefano D'Angelo

2008/9/29 Darren Landrum <darren.landrum(a)sbcglobal.net>: > Sorry for starting this entire argument. I'm just tired of getting > nowhere with all of the same tools that everyone else seems to have no > problem with. I have a very bad habit of putting myself under a great > deal of pressure to exceed everyone's expectations of me. > > Look, I know that everything I'm asking for exists on the Linux > platform. The problem is, it doesn't all exist in one place, or under a > single language. I'm convinced at this point that starting over from > scratch with a solid design is preferable to trying to use several > disparate tools and somehow glue them all together. > > I've already played around with code here and there to try out some > different approaches to this problem, but nothing that I've bothered > keeping around. Starting tonight, I'm going to draft a very detailed > design, create a code hosting account somewhere (probably Google Code), > and get started. I will keep the list apprised of any progress with > regular emails. > > It's been pointed out to me that many people on the list seem to think > that I'm trying to get someone else to code this for me. That is not and > never was my intention, and I apologize for any miscommunication on my > part for that. I am a very slow and not very good coder, though, and it > might take a little while to see any progress. > > First things first, though. A solid design. > > -- Darren > > _______________________________________________ > Linux-audio-dev mailing list > Linux-audio-dev(a)lists.linuxaudio.org > http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev > I don't know if it is relevant to this discussion (at least in an "acceptable" amount of time) but I just wanted you to know about my attempt: NASPRO (http://naspro.atheme.org). I hope people here don't take this message as spamming, because it simply is not. The ideas here are: * to make different existing and not-yet-existing sound processing technology interoperate, both general-purpose sound processing stuff (for example plugins a la LADSPA, LV2, etc.) and special purpose stuff (for example check this: http://naspro.atheme.org/content/ndf-01-file-format-overview), in both compiled and interpreted forms. * be techonlogy neutral (support for each technology implemented in external modules). * define distinct "layers", each dealing with a specific aspect of the whole problem (one for sound processing, one for GUIs, one for session handling, etc.), so that a "DSP coder" can only work on the DSP part and have all the rest automagically implemented and working (for example, you write a LADSPA plugin or write an NDF file and you get an automatically generated GUI without writing one more line of code); * have "back bridges" when possible, so that applications with support for one of NASPRO-supported technologies gets support for all other technologies without writing a single line of code. * build dead-easy-to-use tools on top of that to make it easy for non demaning applications to support DSP stuff. * build tools on top of that to do data routing among each "sound processing component" (in other words, chain-like and/or graph-like processing) - plus, since we have those back bridges, you could also use, for example, CLAM networks (as soon as CLAM will be supported) as an alternative to these tools and have the same degree of supported technology (the same goes for gstreamer, Pd, etc). * be cross-platform (apart from Mac/Windows, alternative desktop-oriented OSes like Haiku or Syllable are getting stronger these days and could become viable to do sound processing in some near or distant future). The result will hopefully be to make it also easier to develop new technologies AND without breaking interoperability. Now, since I'm only one, and I am the only one working on this, it will take an insane amount of time probably, and getting each of these abstraction layers right is astonishingly difficult already (anyone remembering GMPI?) - at the moment I'm fighting with core level stuff and I will be doing that at least for another year or two. If you can wait, I will probably have a talk about NASPRO by the end of October and will put down some slides trying to describe the inner working of it (a lot of people complained that I wasn't clear enough on the website)... Maybe this helps :-\ Stefano

16 years, 11 months

Re: [LAD] Something like Processing for audio

by AlgoMantra

[[sorry darren, this was meant for the list. i hit the wrong button..]] On Mon, Sep 29, 2008 at 1:38 PM, AlgoMantra <algomantra(a)gmail.com> wrote: > Look, I know that everything I'm asking for exists on the Linux >> platform. The problem is, it doesn't all exist in one place, or under a >> single language. >> > > I have exactly the same crib. I could not have said it better, and I am > convinced that this is a pressing issue. Let me list out the languages I > set out to learn in sequence, and finally where I stand. > > Python - PureData - CSound - Chuck - Processing (after this I shifted to > Ubuntu from XP > and found the kind of freedom I wanted) - C/C++ (fullstop) > > ------- -.- > 1/f ))) --. > ------- ... > http://www.algomantra.com > > "I'm not your personal army. I'm MY personal army." >

16 years, 11 months

multiple ICE1712 sound cards (as one huge soundcard)

by Christian Schoenebeck

Hi! I'm trying to merge two ICE1712 based soundcards, so they appear as one big soundcard in JACK. I tried it as described on John Lapeyre's website: http://www.johnlapeyre.com/linux_audio/linux_audio.html That is, I recompiled libasound2 1.0.14-rc3 with the following patch: http://www.johnlapeyre.com/linux_audio/pcm_multi.patch And compiled JACK 0.109 accordingly and finally installed the .asoundrc file, listed here: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html However when I try to start jackd with: usr/bin/jackd -t 4999 -s -R -m -d alsa -C multi_capture -P multi_playback -p 128 -n 2 -r 44100 it fails with the following messages: jackd 0.109.2 Copyright 2001-2005 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 JACK compiled with System V SHM support. loading driver .. apparent rate = 44100 creating alsa driver ... multi_playback|multi_capture|128|2|44100|0|0|nomon| swmeter|-|32bit ALSA lib pcm_hw.c:1357:(_snd_pcm_hw_open) Invalid value for card ALSA lib pcm_hw.c:1357:(_snd_pcm_hw_open) Invalid value for card cannot load driver module alsa no message buffer overruns The target system is a Debian system and I wasn't really sure where to install the .asoundrc file, so installed it as /etc/asound.conf, /etc/alsa/asound.conf and in the home directory as .asoundrc Does anybody have an idea what could be wrong or could give me pointers how to debug it? CU Christian

16 years, 11 months

[ot] capitalising words in a shell-script

by Julien Claassen

Hello there! Is there a nice and simple awk/sed trick to just capitalise a word? I didn't come up with one so far. I could do it by hand, it's a known set of words, but it would be nicer with a real word-independent mechanism. Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de

16 years, 11 months

[ANN] bss - 0.5

by Julien Claassen

Hello everyone! You might have been wondering what I'm up to. So here we go. BSS is the "BBC7 Shell Script Suite" and package of a few shell scripts to search the bbc7 website via a simple commandline interface. You can also listen to the "listen again" service, by directly providing day and time. There's also a small audio-file converter. BBC7 is the BBC's station for comedy and drama. From the criminal, to the horrifying to the hilarious, to quiz-shows to readings of everyday and high literature. Please don't investigate the hidden bbc7-listen.sh option too much. They make their living from that and they offer a very nice service, which I hated to see going down. But of course it can occasionally be helpful to dump files to your disk. The link: http://juliencoder.de/bss-0.5.tar.bz2 Requirements: bash and the usual utilities for all of them, for the listen and convert scripts it's mplayer and ecasound and the search script requires wget. Hope you'll enjoy it! Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de

16 years, 11 months

error with ffado svn rev 1348

by porl sheean

hi there. i have just tried to run the latest ffado build (rev 1348) and keep getting an 'input in flex scanner failed' error when trying to run ffado-dbus-server. the exact error is: 02522143147: Debug (ffado-dbus-server.cpp)[ 251] main: verbose level = 6 02522143184: Debug (ffado-dbus-server.cpp)[ 252] main: Using ffado library version: libffado 2.0.900-1348 input in flex scanner failed no message buffer overruns i have completely removed any old copies of ffado and checked out the build from scratch in case it was something corrupting with an older version's files being left behind or something, but still the same error. thanks for any help porl

16 years, 11 months

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Linux-audio-dev