[linux-audio-dev] [ANN] New releases

New releases and updates at Aliki-0.0.3 Impulse response measurement. - Many bug fixes, should be a bit more stable now... - Added flexible export options. - Manual updated. Jace-0.0.4Low-weight convolution engine for JACK and ALSA. - Now i

Re: [linux-audio-dev] promoting LAC 2007

On Tue, Mar 27, 2007 at 10:27:11PM +0200, Frank Barknecht wrote: > Linux didn't stay an "amateuer" platform in other areas, why should > free software not be professionally used in the audio world as well? There is no good reason why free software shouldn't be used, but not everything required in

Re: [linux-audio-dev] promoting LAC 2007

On Tue, Mar 27, 2007 at 04:17:16PM +0200, Frank Barknecht wrote: > Maybe one day there will be a Linux version of Live, but it's > not something I particularily look forward to, as I wouldn't > use it anyways unless it gets opensource'd. There are probably many of us thinking the same way. But

Re: [linux-audio-dev] 2-channel stereo compatible ambisonics...

On Thu, Mar 15, 2007 at 02:00:46PM +0100, Joern Nettingsmeier wrote: > unless i'm very much mistaken, UHJ-encoded material can be played back > on any stereo rig without problems (some even report better stereo width > and precision). Correct. > the only drawback i heard about is that UHJ-enc

Re: [linux-audio-dev] Getting out of the software game

On Wed, Mar 14, 2007 at 02:26:35PM -0400, Lee Revell wrote: > The interface does not change that fast. Indeed it doesn't, and that is quite normal - after so many years it should be quite clear to both kernel and driver developers what constitutes a good interface. One more reason to define and f

Re: [linux-audio-dev] Getting out of the software game

On Wed, Mar 14, 2007 at 06:32:14PM +0100, Christian Schoenebeck wrote: > I think most of the people on this list know these kind of issues. And I > totally agree that this is an argument to avoid using binary drivers, but > it's definitely NOT a sufficient argument to completely reject a BDI. I

Re: [linux-audio-dev] Getting out of the software game

On Wed, Mar 14, 2007 at 11:16:46AM -0400, Lee Revell wrote: > With binary drivers kernel debugging requires the cooperation of the > vendor in the best case, and lots of guesswork and reverse engineering > in the worst case. I'd say _driver_ debugging requires the cooperation of the vendor. You c

Re: [linux-audio-dev] LFO Phaser LADSPA plugin in details

On Mon, Mar 12, 2007 at 12:53:28AM +0300, Andrew Gaydenko wrote: > Will anybody find a minute ot two to explain me how does the plugin > work - I mean a user POV rather technical realization details. (Assuming you mean my plugin from the MCP package) This is an emulation of an analog phase delay

[linux-audio-dev] new releases

New releases of JAAA and JAPA and the libraries they depend on are now available at jaaa-0.4.1: bugfixes. japa-0.2.0: bugfixes, white and pink noise generators now built-in. clalsadrv-1.2.1: bugfixes. This version should now work correctly w

Re: [linux-audio-dev] Linux Audio Conference 2007 - program online!

Hi Stefano, > >> I still don't know if I can get there... > >> I got really disappointed to see that there's no easy way to get to > >> Berlin via train from Turin, I'm considering bus, and I need someone > >> else to come with me too... :-( You could try to get in contact with Daniele Torelli <[

Re: [linux-audio-dev] 2007..

On Sun, Dec 31, 2006 at 01:10:54PM +0100, Alberto Botti wrote: > Il giorno dom, 31/12/2006 alle 02.25 +0100, Christoph Eckert ha scritto: > > > Congrats for the new job! Hope you'll enjoy yourself in Italy. :) See > > > you in Berlin. > > > > of even better in Italy - will there be a LAD-party?!?

[linux-audio-dev] 2007..

Hello all, First of all, my best wishes for 2007 to all Linux Audio Developers ! 2007 will be a special year for me. As some of you already know, I said goodbey at Alcatel Space two months ago, and starting 8 Jan 2007 I'll be working at LAE - Laboratorio di Acustica ed Elettroacustica -

Re: [linux-audio-dev] Laptop mic-input sound quality.

On Mon, Dec 04, 2006 at 04:05:37PM -0500, Ivica Ico Bukvic wrote: > Because PCMCIA != internal soundcard (even if it is physically inserted into > the laptop's body). Namely, it is enclosed in its own metal casing, not to > mention PCMCIA enclosure both of which allow for a much better R/F shieldi

[linux-audio-dev] Website moving - please update your links

Hello all, Since I'm preparing to move abroad, I've transferred my website to a host that's independent of my local ISP. The new site is www.kokkinizita.net/linuxaudio 'Kokkini Zita' means 'red zeta' - the Greek letter, and all the 'i' are pronounced as English 'ee', not 'y'. It's also my 'pro

Re: [linux-audio-dev] Yet Another Scrolling Scope

On Wed, Nov 22, 2006 at 08:14:11PM +0100, Dominic Sacré wrote: > Got one problem though: high CPU load (I'm not kidding). At small window > sizes everything's fine, but when it exceeds a certain size (a little > less than fullscreen on my machine), CPU load immediately goes up from > ~5% to

[linux-audio-dev] Yet Another Scrolling Scope

A first release of Yass is now avaiable at . Yass is a 'scrolling scope' jack client. It has been hanging around in prototype form for some time. This is a first beta release. Main features: - up to 32 channels, - variable scrolling speed, - configurable trace colors, - automatic gain c

Re: [linux-audio-dev] snd_pcm_poll_descriptors_revents() question

On Wed, Nov 08, 2006 at 05:58:40PM +0100, Clemens Ladisch wrote: > > In that case, how can one test if *all* pollfd for a given > > pcm are ready ? > > You cannot. The state of the file descriptors is not necessarily > related to the state of the PCM device (which is why this function > exists).

[linux-audio-dev] snd_pcm_poll_descriptors_revents() question

On Wed, Nov 08, 2006 at 02:24:30PM +0100, Fons Adriaensen wrote: > Is snd_pcm_poll_descriptors_revents() more than an > accessor ? If it is, the name is a quite misleading. To answer my own question, it seems that it *is* more than an accessor. The docs leave one thing unclear. Does thi

Re: [linux-audio-dev] OSS will be back (was Re: alsa, oss , efficiency?)

On Wed, Nov 08, 2006 at 02:12:09PM +0100, Clemens Ladisch wrote: > Fons Adriaensen wrote: > > Basically what happens comes down to (ignoring error > > checking and timeouts): > > > > - The set of pollfd is poll()'ed until all are ready. > > Applications

Re: [linux-audio-dev] OSS will be back (was Re: alsa, oss , efficiency?)

On Wed, Nov 08, 2006 at 12:20:42AM +, James Courtier-Dutton wrote: > The current jackd skips a step in the processing of the poll events. Looking at the code it seems already quite elaborate. Basically what happens comes down to (ignoring error checking and timeouts): - The set of pollfd is

Re: [Jackit-devel] [linux-audio-dev] Re: Multiplexing 4 channels on SPDIF

On Mon, Oct 30, 2006 at 01:30:53PM -0500, Lee Revell wrote: > On Mon, 2006-10-30 at 18:52 +0100, Fons Adriaensen wrote: > > The real question is how to fit this into the existing architecture: > > > > - hardware presents itself as 2 * 96 kHz > > - user wants to s

Re: [linux-audio-dev] OSS will be back

On Fri, Nov 03, 2006 at 04:39:18PM +, Simon Jenkins wrote: > On Fri, 2006-11-03 at 08:53 +0100, Fons Adriaensen wrote: > > [...] > > I'd say that the essential feature of JACK is not that it is a > > callback based system, but that it presents and expects audio >

Re: [linux-audio-dev] OSS will be back

On Fri, Nov 03, 2006 at 06:37:13PM +1100, Loki Davison wrote: > On 11/3/06, Jens M Andreasen <[EMAIL PROTECTED]> wrote: > >On Fri, 2006-11-03 at 13:42 +1100, Loki Davison wrote: > > > >> mmm. I think they are missing the point about ALSA vs OSS api here. It > >> doesn't matter. The only one who sho

Re: [linux-audio-dev] Re: Alpha release of Aliki

On Wed, Nov 01, 2006 at 09:48:37AM -0600, Ben Loftis wrote: > Can you please tell us more about Aliki? > > I am using Denis Sbragion's DRC program to generate an impulse response > file, and his suite of graphing tools to generate various views of the > measurement. > > What is the output of Al

[linux-audio-dev] Alpha release of Aliki

For the brave: an alpha release of Aliki (Room Impulse Response Measurement) is now available on: http://users.skynet.be/solaris/linuxaudio along with a manual that should get you started. This is basically the code used at the LAC2006 workshop, cleaned up a but. As said, ALPHA, incomplete, pr

Re: [Jackit-devel] [linux-audio-dev] Re: Multiplexing 4 channels on SPDIF

On Mon, Oct 30, 2006 at 02:55:33PM -0500, Paul Davis wrote: > On Mon, 2006-10-30 at 18:52 +0100, Fons Adriaensen wrote: > > > > > - hardware presents itself as 2 * 96 kHz > > - user wants to see a device with 4 * 48 kHz. > > interestingly, ADAT devices do the

Re: [Jackit-devel] [linux-audio-dev] Re: Multiplexing 4 channels on SPDIF

On Mon, Oct 30, 2006 at 11:41:36AM -0500, Lee Revell wrote: > On Mon, 2006-10-30 at 17:35 +0100, Alfons Adriaensen wrote: > > > What they want to do is trick hardware 2-ch recorders into accepting > > the 4-ch signal. So what is in reality 4 channels at 48 kHz will be > > presented as 2 channels

[linux-audio-dev] Multiplexing 4 channels on SPDIF

Hello all, Core Sound willsoon be offering a tetrahedral (Ambisonic) microphone at a very reasonable price. They are also working on a combined preamp + AD converter unit for this mic. This will be able to multiplex the 4 channels over a single SPDIF link, b

Re: [linux-audio-dev] best option for audiovisual synchrony

On Fri, Oct 20, 2006 at 10:44:48PM +0200, Tim Goetze wrote: > Back in the 80s, the humble Commodore 64 could be readily programmed > to fire an interrupt on vertical sync. Have 20 years of progress > really deprived us of this fine feature, or is it just missing from X? Same on the humble BBC

Re: [linux-audio-dev] best option for audiovisual synchrony

On Fri, Oct 20, 2006 at 03:51:09PM +0100, Dave Griffiths wrote: > > The tricky bit is of course getting a flash that's totally synchronous > > with the beep. Absolute synchrony is not achievable without dedicated > > hardware, but we need to get an approximation that's within the few ms > > range.

Re: [linux-audio-dev] Paper on dynamic range compression

On Wed, Oct 18, 2006 at 12:14:45PM +0100, John Rigg wrote: > The fact remains that a lot of high end professional users consider many > of the free software plugins to be "nearly unusable" (see Ben Loftis' > earlier post in this thread). This isn't intended as a criticism of the > developers, just

Re: [linux-audio-dev] Paper on dynamic range compression

On Wed, Oct 18, 2006 at 06:50:39PM +1000, Erik de Castro Lopo wrote: > ... > Its very easy to get carried away with trying to reach some sort > of audio perfection. Things like upsampling in order to apply > compression is over-engineering. I'd agree. This is a non-problem for a well-designed an

Re: [linux-audio-dev] Paper on dynamic range compression

On Tue, Oct 17, 2006 at 06:50:04PM +0200, David Olofson wrote: > On Tuesday 17 October 2006 16:43, Fons Adriaensen wrote: > > > It could be. OTOH, most DACs today would upsample and filter before > > the real conversion takes place, and could allow for this. But maybe > &

Re: [linux-audio-dev] Paper on dynamic range compression

On Tue, Oct 17, 2006 at 09:59:10AM -0400, Paul Davis wrote: > On Tue, 2006-10-17 at 11:56 +0200, Fons Adriaensen wrote: > > > 'THE SAMPLES ARE NOT THE SIGNAL'. The real peak level of a > > signal when converted to the analog domain can be several > > d

Re: [linux-audio-dev] Paper on dynamic range compression

On Mon, Oct 16, 2006 at 07:48:35PM +0100, Dan Mills wrote: > The gain control signal has energy right the way out > to the band limit (and probably aliased around it), > never mind what happens when that hits the multiplier! The question is: how much of this HF energy is there ? There shouldn't

Re: [linux-audio-dev] Paper on dynamic range compression

On Thu, Oct 05, 2006 at 05:12:20PM +0100, Steve Harris wrote: > The SC* plugins do the same as TAP (calculate the gain every 4 samples), > but I interpolate the gain values between each computation. The > attch/deacay times were slow enough in my testing that it was OK to do > that. It should be

Re: [linux-audio-dev] very nice looking HW

On Wed, Sep 06, 2006 at 06:55:22PM +0200, Florian Schmidt wrote: > On Wednesday 06 September 2006 16:30, nescivi wrote: > > On Wednesday 06 September 2006 16:25, Alfons Adriaensen wrote: > > > Any chance of this ever being supported in Linux ? > > > > > > http://www.marian.de/en/products/ucon_cx

Re: [linux-audio-dev] I'd like my brain back (idiot developer)

On Thu, Aug 31, 2006 at 11:13:52AM -0400, Gene Heskett wrote: > Looking at the subject line prompts me to ask if you have it (your brain) > backed up on tape somewhere? Most of us have claimed that at one point or > another... To help Paul find back his brain, we should all burn candles, chant

Re: [linux-audio-dev] job offer... [Fwd: Algorithm Development Manager (Full-Time)]

On Sat, Aug 26, 2006 at 12:06:16AM +0200, David Olofson wrote: > What really annoys me about these is that they're usually written to > give the impression of a personal message from someone who would know > what you're doing and what kind of social network you have. Sometimes > you can't reall

[linux-audio-dev] Kokkini Zita

(Since this was rejected on LAA ('no reason given'), I'll answer the question here.) On Mon, Aug 14, 2006 at 09:20:36PM +0400, Andrew Gaydenko wrote: > What do "Kokini Zita" words mean? I have thought, Fons Adriaensen is the > author > of the listed apps :-)

Re: [linux-audio-dev] Re: Akai's MPC4000 Sampler/Workstation Open Source Project

On Thu, Jul 27, 2006 at 08:46:27PM +0200, Jay Vaughan wrote: > > > > There are public-domain RTOSes available that are suitable for this > > > > task. To those, you can add drivers for USB and FAT32. Without an > > > > RTOS to give you hard real-time scheduling, you have no chance to > > > > achie

Re: [linux-audio-dev] light C++ set for WAV

On Thu, Jul 27, 2006 at 06:58:32AM +1000, Erik de Castro Lopo wrote: > Yes, that was my idea. So if the sndfile.hh has: > > class Sndile > { >int method (/* params */) ; > } > > int method (/* params */) > { > /* whatever */ > } > > do I need to add

[linux-audio-dev] Improved decoder plugins for Linux

Hello all, I just updated the small set of Ambisionics related LADSPA plugins available at . The three decoders now feature optional phase aligned shelf filters and independent control for LF and HF gain of the velocity components (from in-phase over max rE to max rV). The distance compensation

Re: [linux-audio-dev] light C++ set for WAV

On Thu, Jul 13, 2006 at 09:56:58PM +0400, Andrew Gaydenko wrote: > I mean some minimal C++ class set like: WavFile, WavHeader, WavFrame with > few apparent methods (open/close, read/write frame(s)). Libsndsfile is plain C, but will do what you want without any fuss. You could write a WAV specif

Re: [linux-audio-dev] Converting a 24bit sample to 16bit

On Sat, Jul 08, 2006 at 01:34:44PM +0100, James Courtier-Dutton wrote: > Is there a standard way of converting a 24bit sample to 16bit? > I ask because I think that in different scenarios, one would want a > different result. > 1) scale a 24bit value to a 16bit by simple multiplication by a fracti

[linux-audio-dev] [OT] No-one was ever fired for having hired FA

Hello all, I finally took the big step and handed in my resignation at my employer, Alcatel Alenia Space. After 3 years of CAD, 3 years of real-time kernels, and 11 years in space telecoms, I want to return to my first love which is audio, acoustics, and music. My activities in LAD have certainly

Re: [linux-audio-dev] fst, VST 2.0, kontakt

On Sun, Jul 02, 2006 at 10:49:54AM +0200, Thorsten Wilms wrote: > To me your behaviour of accusing Dave of plain lying is not > acceptable. You seem to implicate dishonesty. I someone states as a fact and without any qualification a certain interpretation of a text, while knowing very well that

Re: [linux-audio-dev] fst, VST 2.0, kontakt

On Sat, Jul 01, 2006 at 04:09:42PM -0400, Dave Robillard wrote: > Whether or not you agree with the licensing practise, calling it "open > source" is as misleading as calling MS shared source "open source". > Defend the license/exception if you want, but don't intentionally > mislead people about

Re: [linux-audio-dev] Envelopes

On Sat, Jul 01, 2006 at 08:22:16AM +0200, Jens M Andreasen wrote: > Linear attack sounds OK. Given the exponential way we perceive volume, > this *is* the desired function. That's the rationale for having exponential volume controls. In the case of a release profile, it's rather because many real

Re: [linux-audio-dev] Re: LADSPA Extension for Extra GUI Data

On Sun, Jun 25, 2006 at 06:57:47PM +0100, Steve Harris wrote: > I agree that describing it as volts is a bit odd, but it instantly makes > people like me feel at home. There's not reason why a digital modular neds > units for its modulation sources. It's just real numbers. I never mentioned 'volt

Re: [linux-audio-dev] IR FFT smoothing

On Sat, Jun 24, 2006 at 07:15:38PM +0400, Andrew Gaydenko wrote: > And the question is: what is common way to smooth the result? Some offtopic > apps has > something like "1/24, ..., 1/3 octave smoothing". What does it mean? Lowpass (linear phase) filtering of the response. Or windowing the IR w

Re: [linux-audio-dev] LADSPA Extension for Extra GUI Data

On Mon, Jun 19, 2006 at 11:25:52PM +0100, Steve Harris wrote: > On Mon, Jun 19, 2006 at 11:58:43PM +0200, Fons Adriaensen wrote: > > > What worries me is that LV2 is *not* going to solve the problem that > > DR raised w.r.t. my "Moog" filter plugins. > > This pa

Re: [linux-audio-dev] LADSPA Extension for Extra GUI Data

On Mon, Jun 19, 2006 at 10:34:05PM +0100, Steve Harris wrote: > FWIW, I think the "not changing any code" thing is a blind, someone, > somewhere has to change some code if you want new behaviour*. To me the > critical thing is not that, but that a display function or whatever only > solves half th

Re: [linux-audio-dev] Re: LADSPA Extension for Extra GUI Data

On Mon, Jun 19, 2006 at 01:49:14PM -0400, Dave Robillard wrote: > On Mon, 2006-06-19 at 14:15 +0200, Alfons Adriaensen wrote: > > All of them fake. > > Fake like the countless bug reports I get about your filter plugins not > working because they take some silly arbitrary unit instead of Hz for

Re: [linux-audio-dev] realtimeness: pthread_cond_signal vs. pipe write

On Wed, Jun 07, 2006 at 05:42:26PM -0400, Lee Revell wrote: > But, from the original post it seems that pthread_cond_signal is not > realtime safe as it locks a mutex: > ... > How can glibc guarantee that we are not put to sleep if there is > contention? The mutex associated with a CV is held on

Re: [linux-audio-dev] realtimeness: pthread_cond_signal vs. pipe write

On Wed, Jun 07, 2006 at 08:49:38AM -0400, Paul Davis wrote: > nice to hear that they are faster. on the other hand, once again POSIX > screws us all over by not integrating everything into a single blocking > wait call. i've said it before, i'll say it again - this is one of the > few things that

Re: [linux-audio-dev] LV2 library API

On Tue, May 30, 2006 at 05:48:35PM -0400, Dave Robillard wrote: > The function we're talking about pulls this info directly from the data > file (not eg from a loaded Port object which would have a const type > string). The library doesn't load all the stuff from the file into > memory (in which

Re: [linux-audio-dev] LV2 library API

On Tue, May 30, 2006 at 03:10:38PM -0400, Dave Robillard wrote: > > I'd consider any interface that just returns a constant and requires > > a malloc() and a free() to do it plain broken. This data doesn't live > > in kernel space, or does it ? You could just return a const char *. > > It's not a

Re: [linux-audio-dev] LV2 library API

On Tue, May 30, 2006 at 06:07:15PM +0100, Steve Harris wrote: > On Tue, May 30, 2006 at 11:43:57AM -0400, Dave Robillard wrote: > > char* type = lv2_port_get_type(someplug, 0); > > if (!strcmp(type, LV2_DATATYPE_FLOAT)) > > /* ... */ > > free(type); > > Makes sense to me. You could make the A

[linux-audio-dev] Re: [linux-audio-user] Re: LAC-Konzertreport auf SWR2 15.5. 23h (German only)

On Mon, May 15, 2006 at 11:31:00PM +0200, Christoph Eckert wrote: > recording... :) . + oggenc + ftp + mail LAD ??? :-) :-) :-) -- FA Follie! Follie! Delirio vano e' questo!

[linux-audio-dev] [ANN] New release of Aeolus, updates

Hello all, The long announced new release of Aeolus is finally available. Version 0.6.6 is almost a complete rewrite of the previous official release, 0.3.1 (a lot happened in bewteen). This should still be considered a beta release - no doubt some nasty bugs will be uncovered when this version

Re: [linux-audio-dev] LADSPA2: logarithmic hint

On Tue, May 02, 2006 at 02:47:37PM -0400, Paul Davis wrote: > note: the hint didn't say "exponential". it said "logarithmic". If V represents the value sent to the port, and P the corresponding widget position, it means that the function V -> P should be logarithmic wich is the same as saying tha

Re: [linux-audio-dev] LADSPA2: logarithmic hint

On Tue, May 02, 2006 at 05:21:44PM +0100, Steve Harris wrote: > "this goes from 0Hz to fs/2Hz, and I want it to be logarithmic", That's a contradiction. -- FA Follie! Follie! Delirio vano e' questo!

Re: [linux-audio-dev] LADSPA2: logarithmic hint

On Tue, May 02, 2006 at 12:15:20PM -0400, Paul Davis wrote: > saying that the port range is exponential doesn't pin it down very much. > it still requires the host to make decisions about precisely what kind > of exponential curve to use for the range, and it may get it wrong. It does pin it dow

Re: [linux-audio-dev] LADSPA 2

On Sat, Apr 22, 2006 at 03:01:26PM +0200, Lars Luthman wrote: > ... > support for polyphony (you can run several plugin instances as a > polyphony group with a single call to run_multiple() which lets you do > common calculations once). That's not the point. Even in that case each plugin instanc

Re: [linux-audio-dev] LADSPA 2

On Sat, Apr 22, 2006 at 02:26:57PM +0200, Thorsten Wilms wrote: > Distribution / finding plugins: > Stability: > Control/audio rate: > Port grouping: > Port Roles: > Referencing: > Hints: > Presets: > Help / Discription: > MIDI/OSC > GUI lib: To which I'd add: Polyphony/Multiple channels: Plu

[linux-audio-dev] Ladspa rdf

A maybe silly question: where on a typical system are the rdf descriptions of ladspa plugins supposed to live ? I can't find them ! -- FA Follie! Follie! Delirio vano e' questo!

Re: [linux-audio-dev] [OT] First trip to Europe

On Wed, Apr 19, 2006 at 05:39:01PM +0400, Dmitry Baikov wrote: > Moscow > St.Petersurg > -- > Helsinki > Stockholm > Berlin > Koeln > Brussels > London > Paris > Bourdeaux > Madrid > Barcelona,Figeras > Genova > Milan? > Rome > Venice > Vienna > Prague > -- > Moscow If you'd come to Antwerp (45 k

[linux-audio-dev] New release of AMB plugins

A second release of the Ambisonics plugins is now available. * Added cube (8-speaker) decoder. * Removed conflicting port hints. -- FA Follie! Follie! Delirio vano e' questo!

Re: [linux-audio-dev] multiface latency question

On Wed, Apr 12, 2006 at 09:45:42PM +0200, Esben Stien wrote: > Actually, it varies between 206.765 and 206.766 That's about 20 nanoseconds difference, or less than the delay of 10 meters of cable... -- FA Follie! Follie! Delirio vano e' questo!

Re: [linux-audio-dev] Measuring playback and capture quality.

On Sat, Apr 08, 2006 at 08:09:49PM +0100, James Courtier-Dutton wrote: > Neither of those applications work. > I have ardour working fine with stereo in/stereo out. > > jaaa -J seems to talk to jack, but does not capture or play any sound. > jaaa -A fails to even open the alsa device. > > Have t

Re: [linux-audio-dev] Measuring playback and capture quality.

On Sat, Apr 08, 2006 at 05:17:14PM +0100, James Courtier-Dutton wrote: > Hi, > > Are there any linux tools out there that will sample the line in, and > display a detailed spectrum scope of the detected sound? > > E.g. > http://www.pcavtech.com/soundcards/ct462048/SNR_LB_FS.gif > > Eventually I

Re: [linux-audio-dev] Re: GPL Audio Hardware

On Thu, Apr 06, 2006 at 03:58:51PM -0400, Joshua Boyd wrote: > On Wed, Apr 05, 2006 at 07:52:26PM -0500, Richard Smith wrote: > > > Getting that to fly in an FPGA however may be the problem as you start > > running into clock speed limits. And an ASIC suffers from the lack of > > startup cash. >

Re: [linux-audio-dev] multiface latency question

On Thu, Apr 06, 2006 at 02:57:28PM -0400, Lee Revell wrote: > On Thu, 2006-04-06 at 20:52 +0200, fons adriaensen wrote: > > Have you been able to use -p 64 with 2.6.16 on the Thinkpad with > > ACPI ? I had no problems with it on my R51 when using 2.6.8 (SuSE > > 9.2), but si

Re: [linux-audio-dev] multiface latency question

On Thu, Apr 06, 2006 at 08:28:32PM +0200, Jan Weil wrote: > 206.805 > > This is a Thinkpad T43 with an unpatched 2.6.16 with CONFIG_PREEMPT=y, > jackd 0.100.7, [EMAIL PROTECTED] > > Now what does that value tell me? These are samples, I suppose, i. e. > 4.308 msec? Yep, it's samples. > Fons,

Re: Fwd: [linux-audio-dev] LADSPA processing: ams, om, ... Anything else?

On Sat, Mar 18, 2006 at 12:02:40PM -0300, Denis Alessandro Altoe Falqueto wrote: > My /etc/hosts was like this: > > 127.0.0.1 localhost.localdomain localhost > > I changed it to: > > 127.0.0.1 bach bach > > And it all worked. I found the solution in the hexter homepage, at

Re: [linux-audio-user] Re: [linux-audio-dev] [ANN] netjack-0.9rc1

On Mon, Mar 13, 2006 at 11:59:15PM +0100, stefan kersten wrote: > as paul stated, network byte order is defined to be > big-endian, so yes, you have to convert 32 bit floats (and > doubles, for that matter) on intel, because they are stored > lsb first. of course it would be perfectly valid for ne

Re: [linux-audio-user] Re: [linux-audio-dev] [ANN] netjack-0.9rc1

On Mon, Mar 13, 2006 at 05:25:04PM -0500, Paul Davis wrote: > On Mon, 2006-03-13 at 23:10 +0100, fons adriaensen wrote: > > Is it true on the common platforms that using ntohl and htonl on > > floats will always result in compatible data on the wire or in a > > file ? In ot

Re: [linux-audio-user] Re: [linux-audio-dev] [ANN] netjack-0.9rc1

On Mon, Mar 13, 2006 at 10:21:39PM +0100, [EMAIL PROTECTED] wrote: > On Sun, Mar 12, 2006 at 03:50:26PM -0500, Lee Revell wrote: > > > Why do you use big-endian on the wire, requiring a double swap for x86 > > <-> x86? Wouldn't LE make more sense, especially as PPC Macs become > > unavailable? >

Re: [linux-audio-dev] Re: Karlsruhe

On Sat, Mar 04, 2006 at 05:06:49PM -0500, Lee Revell wrote: > Where can I find a decent map of Germany? The whole country is blank on > Google Maps... Try http://www.de.map24.com/ -- FA

Re: [linux-audio-dev] Re: Which widgets?

On Mon, Feb 27, 2006 at 01:16:17AM +0200, Jussi Laako wrote: > On Sun, 2006-02-26 at 20:38 +0100, Albert Graef wrote: > > Canvases give you much more than just rendering. They also manage the > > graphical objects that you created and, if anything changes, rerendering > > the changed parts happ

Re: [linux-audio-dev] Re: [linux-audio-user] Re: Free Software vs. Open Source: Where do *you* stand?

On Mon, Feb 20, 2006 at 11:41:43PM +0100, David Kastrup wrote: > > They have paid for a license to use it, and for nothing else. > > Well, then they might have some expectation to be able to use it, no? > Without the ability to adapt the software to different devices or > applications, or fix err

Re: [linux-audio-dev] Re: [linux-audio-user] Re: Free Software vs. Open Source: Where do *you* stand?

On Mon, Feb 20, 2006 at 11:01:10PM +0100, David Kastrup wrote: > Lee Revell <[EMAIL PROTECTED]> writes: > > > By this logic, locking my doors is immoral because it diminishes > > people's freedom to roam around my house. > > Those people have not paid for access to your house. Purchasers of > pr

Re: [linux-audio-dev] dmix and jack

On Mon, Jan 30, 2006 at 08:05:42PM +, James Courtier-Dutton wrote: > It is due to the fact that the alsa programming in jackd has been > implemented wrongly. The poll revents are not handled correctly. > This results is OK operation when using hw:0,0, but likely to fail for > more exotic als

Re: [linux-audio-dev] Additional chunks in WAV files with libsndfile ?

On Tue, Jan 24, 2006 at 09:02:23AM +1100, Erik de Castro Lopo wrote: > fons adriaensen wrote: > > > What I need in particular is some way to calibrate the time > > axis - i.e. to say frame #N corresponds to t = 0, and some > > other similar info, mostly sample indices. &

[linux-audio-dev] Additional chunks in WAV files with libsndfile ?

Hi all, is there a recommended way to write / read additional chunks in WAV files, using libsndfile (assuming it's possible at all - I didn't find any hints to this in the docs) ? What I need in particular is some way to calibrate the time axis - i.e. to say frame #N corresponds to t = 0, and so

Re: [linux-audio-dev] detune

On Sun, Jan 15, 2006 at 03:20:56AM +0100, Tim Goetze wrote: > You can periodically window and granulate the signal, resample the > grains and resynthesize -- that's the faster way, doing it all in the > time domain. You'll get some comb filtering from the time-stretched > phase cancelling out d

[linux-audio-dev] [ANN] First (alpha) release of JACE

JACE is a Convolution Engine for JACK and ALSA, using FFT-based partitioned convolution with uniform partition sizes. I wrote it mainly as a 'proof of concept' for something more complicated, to be announced at the next LAC. But it could be useful as it is, hence this release. Main features: -

Re: [linux-audio-dev] VST compiled for linux / gui message loop

On Sat, Jan 07, 2006 at 04:45:21PM +0100, [EMAIL PROTECTED] wrote: > why dont you open a separate display connection for the plugin ? > then you can even move the gui updates to a different thread and there > you go... > > look into gtkplug.c and gtksocket.c on how this works. Does X allow multi

Re: [linux-audio-dev] Interaction bug between zynaddsubfx and muse.

On Sun, Jan 01, 2006 at 06:42:02PM +0100, Robert Jonsson wrote: > Indeed, there are several issues at work here... > In anycase, MusE has from 0.7.2pre2 a fix that enables synths with identical > names to be used. > Also, in the case with ZynAdd, another option is to use only one instance. > Zyn

Re: [linux-audio-dev] Audio/Midi system - RT prios..

On Sat, Dec 31, 2005 at 08:03:06AM -0500, Paul Davis wrote: > On Sat, 2005-12-31 at 00:04 +0100, fons adriaensen wrote: > > 1. If things have to be timed accurately, it seem logical to concentrate > > this activity at one point. At least then the timing will be consistent, >

Re: [linux-audio-dev] Audio/Midi system - RT prios..

On Fri, Dec 30, 2005 at 05:10:44PM -0500, Paul Davis wrote: > On Fri, 2005-12-30 at 22:27 +0100, Pedro Lopez-Cabanillas wrote: > > On Friday 30 December 2005 17:37, Werner Schweer wrote: > > > > > The ALSA seq api is from ancient time were no realtime threads were > > > available in linux. Only a

Re: [linux-audio-dev] Audio/Midi system - RT prios..

On Fri, Dec 30, 2005 at 11:54:56AM -0500, Paul Davis wrote: > you don't know the correct priority to use. i imagine an api along the > lines of: > > jack_create_thread (pthread_t*, void* (thread_function)(void*), > void* arg, int relative_to_jack); > > the last

Re: [linux-audio-dev] db gain controls.

On Fri, Dec 23, 2005 at 01:37:31AM +, James Courtier-Dutton wrote: > I have a question for some audio professionals out there. > What is the smallest sensible gain control step in dB. > Is it 0.5dB ? > I am asking, because if one is using a digital gain control in a 24bit > fixed point DSP, o

Re: [linux-audio-dev] High-order Ambisonic coder/decoder in JACK/LDASPA?

On Thu, Dec 15, 2005 at 09:04:28PM +0100, Georg Holzmann wrote: > >Regardless of the JACK / LADSPA question, seventh order Ambisonics using > >16 speakers is just ridiculous. Either it's horizontal only, and in that > >case using 7th order is just a waste of resources and effort (3th order > >will

Re: [linux-audio-dev] High-order Ambisonic coder/decoder in JACK/LDASPA?

On Thu, Dec 15, 2005 at 04:56:04PM +0100, Asbjørn Sæbø wrote: > Some colleauges of mine do need a tool for coding and decoding of > high-order Ambisonic for their research. They are aiming for seventh > order, played back over sixteen loudspeakers. They are now planning to > implement this, usin

Re: [linux-audio-dev] xruns

On Sun, Nov 20, 2005 at 08:29:37AM -0500, Paul Davis wrote: > however, this is not necessarily the right approach to handling xruns. > its worth trying. a better start is to check if the xruns go away or > occur less frequently with a larger buffer size. USB cards often work a lot better (allow m

Re: [linux-audio-dev] Mixer controls

On Sun, Nov 06, 2005 at 12:21:30PM +, James Courtier-Dutton wrote: > The problem I have is what should I > display for the 0.0 gain_multiplier setting. I.e. When it effectively > mutes the sound output at it's minimal slider setting. "Off" ??? -- FA

Re: [linux-audio-dev] Mixer controls

On Sun, Nov 06, 2005 at 10:59:57AM +, James Courtier-Dutton wrote: > My question is really what should I do when the gain_multiplier is 0.0 > > Do I: > a) Limit the range of the gain control to 0dB to -40 dB and have a > separate Mute control. > b) When the gain control has a gain_multiplier

Re: [linux-audio-dev] Re: Radio receiver.

On Sat, Nov 05, 2005 at 09:23:05PM +0200, Juhana Sadeharju wrote: > > Thanks for the tip on "diversity reception". Yes, I got the > idea from astronomer's systems. Normally 'diversity reception' means to combine the signals from 2 or more receivers to obtain a result that has a better S/N ratio (

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