AAARRRGGHHH! That was supposed to go to jamin-devel, sorry folks.
Apologies to the Jamin gang, a bad case of driving mail client without
looking at screen.
Sorry about that.
Regards, Dan.
Send instant messages to your online friends http://uk.messenger.yahoo.com
Hi all,
While hacking around with aliasing effects in digital compressors (Yes
it is real, yes you can hear it!), I happened to run a 10Khz sine wave
into jamin with an instance of Jaaa hooked up to the output.
The results were 'interesting' as it appears that jamin introduces
easily measurable
--- Fons Adriaensen <[EMAIL PROTECTED]> wrote:
> One of the things missing in all the LADSPA
> compressors I've seen so
> far is a feature called 'release gate'. This freezes
> the release (i.e.
> the gain stops rising) if the input signal drops
> below a set threshold.
> It helps a lot in avoidi
--- Steve Harris <[EMAIL PROTECTED]> wrote:
> > But that puts potentially expensive gain
> > calculations into the fast sr code, also I was
rather planning
> > on using the impulse used for the upsampler to
> >provide the band limiting for free.
>
> the gain calculations are relativly cheap, its
--- Steve Harris <[EMAIL PROTECTED]> wrote:
> These are aliasing artifacts in the sidechain
> though, right? So they will
> show up as modulations in the output, rather than
> directly audible
> aliasing.
The last thing almost all software compressors do is
something of the form output = gain *
--- John Rigg <[EMAIL PROTECTED]> wrote:
> If the control signal is derived from the upsampled
> input
> to the compressor, that is taken care of.
But that puts potentially expensive gain calculations
into the fast sr code, also I was rather planning on
using the impulse used for the upsampler t
--- Erik de Castro Lopo <[EMAIL PROTECTED]> wrote:
> You need a low pass filter on the control signal. It
> should
> be somewhere well below 1kHz.
Agreed that you need the filter, but a 'brick wall' at
1Khz means that anything faster then 50ms or so as an
attack time (and there are legit uses fo
--- John Rigg <[EMAIL PROTECTED]> wrote:
>
> Don't have a pointer, but wouldn't a simple linear
> interpolator followed by a basic LPF do it fairly
cheaply?
>
It might if we only needed to do the control signal,
but at 44.1, there just is not the bandwidth in either
the control or AUDIO channels
--- John Rigg <[EMAIL PROTECTED]> wrote:
> While we're on this subject, has anyone analysed the
> aliasing behaviour
> of the LADSPA compressors? This post reminded me of
> an article I read
> in the AES journal a while back, and a search on
> aes.org found it:
I just hacked up a little experime
On Thu, 2006-10-05 at 20:14 +0200, David Olofson wrote:
> I don't know how fast it *actually* is, but FWIW, my old Behringer
> compressor/limiter has a lowest attack setting of 0.1 ms, and a
> lowest release setting of 50 ms.
If those sort of settings are to be available in a software compresso
--- Dominique Michel <[EMAIL PROTECTED]>
wrote:
> With jack, mixers as jackeq or jackmix are already
> using db scales.
Ys, but they deal with gain which is unitless in
the DB system.
What is being discussed is how to reference things to
external signal levels (measured in volts, SPL or
wha
--- conrad berhörster <[EMAIL PROTECTED]>
wrote:
> Whoow , that was fast
> maybe i was a little bit misunderstandable.
>
> when i wrote about jack, i meant the jackserver. i
> write an app, which has an
> microphon input. now i will take this microphon
> signal and show it as a gain
> met
--- conrad berhörster <[EMAIL PROTECTED]>
wrote:
> Hello all,
>
> Does anybody know, how i can scale the incoming jack
> signals to dbSPL,
> which is in the range of 0 to 120. An is it possible
> to calculate from dbFS
> (which is used in normal soundapp in range -inf to
> 12db) into dbSPL.
On Thursday 15 December 2005 18:06, Phil Frost wrote:
> There is truth in what you say, but given the problem described, this is
> likely more confusing than helpful to someone new to signal processing.
>
> If the problem is that the input is a low frequency square wave, from a
> Low Frequency Osc
On Thursday 15 December 2005 04:26, Paul Coccoli wrote:
> Here's a poorly written description of my problem (the code in
> question is written from scratch in C++, BTW):
>
> I have a simple gain function that takes a number between 0 and 1 and
> multiplies each input sample by that number. If I us
On Monday 07 November 2005 13:37, Fred Gleason wrote:
> This seems to be one of the differences between regular
> 'pro-audio/recording' boards and those targeted for use in radio broadcast
> environments.
Most serious live sound consoles have half way decent mute automation, and
some get very s
On Monday 07 November 2005 04:05, Loki Davison wrote:
> I was just adding my opinion from a very different understanding of
> mixers/mixing. Desk automation vs live dj / turntablist style mixing.
> Check out http://www.ecler.es/download/img_general_hak320.jpg if you
> want a pic of it. Notice the
On Monday 07 November 2005 00:43, Loki Davison wrote:
>
> My mixer (Ecler HAK320) has selectable curve on all faders, but option
> i use is nonlinear, log style which is a very nice feature to
> preserve. It's got mute switches as well, but i never use them.
>
I don't know it, but how do you hand
On Sunday 06 November 2005 18:20, Tim Orford wrote:
> On Sun, Nov 06, 2005 at 03:21:54PM +, James Courtier-Dutton wrote:
> > It was just an example. The actual range depends on the sound card
> > hardware, but the typical limit is something like -60 dB or -80 dB.
>
> Ah, ok, yes i wasnt sure wh
On Tuesday 01 November 2005 19:24, Juhana Sadeharju wrote:
Hate to burst your bubble
> was discussed at 1997/98 here or elsewhere, and I invented the
> innovation (2) at 1993.
Google for diversity reception, I think you will be surprised by just how far
back this trick goes. I know people w
On Sunday 30 October 2005 11:11, Florian Schmidt wrote:
>
> Also the basic problem of signalling (in this case the disk thread that
> there is work to do) still persists even with lockless ringbuffers. The
> other thinkable approach would be to make the diskthread wakeup
> regularly and check wheth
On Saturday 29 October 2005 22:25, fons adriaensen wrote:
> Frequency shifting will give a few dB extra, not more. It's based on the
> fact that a typical room response will have many very narrow peaks only
> a few Hz or less apart, and shifting the frequencies will smooth out the
> response as th
On Friday 28 October 2005 12:26, Benno Senoner wrote:
> Hi all,
> I would like to route a microphone through a sound card and back to
> powerful amplified speakers.
>
> As we know in analog PA gear you have the microphone feedback problem
> (usually it comes in form
> of high pitched whistle sounds
On Sunday 04 September 2005 22:12, Jussi Laako wrote:
> On Sat, 2005-09-03 at 21:14 +0100, Dan Mills wrote:
> > I don't think this is right, a signal **level** can be measured against a
> > known reference level, and for metering it is important to know what this
> > i
On Saturday 03 September 2005 22:00, James Courtier-Dutton wrote:
> As far as I can tell, all sound cards volume controls are based around
> gain. +dB for gain, and -dB for attenuation.
>
> Examples of sound card hardware datasheets:
> Record Volume control:
> "The range is 12dB to -33dB in steps
On Saturday 03 September 2005 20:30, Jussi Laako wrote:
> On Sat, 2005-09-03 at 19:19 +0100, Dan Mills wrote:
> > > annoying. IMO, all volume adjustments should be in dB (or dBu or some
> >
> > db, as this is a gain change (which is unitless).
>
> It could also be
On Saturday 03 September 2005 18:20, Jussi Laako wrote:
> Percent as volume scale is not very intuitive and is actually pretty
> annoying. IMO, all volume adjustments should be in dB (or dBu or some
> other commonly used unit). Thus you would know what the adjustment
> really is, no matter what the
On Friday 11 March 2005 03:03, ben racher wrote:
> Hello,
>
> I'm starting a student radio station at IUPUI in Indianapolis, Indiana
> and I want our entire audio infrastructure to be based on Linux. I've
> got a rough sense of all the apps we need and what apps to setup on
> which computers, but I
On Sunday 31 October 2004 17:13, Steve Harris wrote:
> > I remember a CVS version of JAMIN used to do something similar?
>
> Yeah, it still does AFAIK. I never tracked it down, but didnt put too much
> effort into it.
>
> It only happens if jack is not running in realtime mode.
Guess again, JAMIN
Hi all,
I am working on a project that makes very heavy use of the gtkmeter code
(borrowed from JAMIN), and seem to have a hard to track down problem
I am using jack so there are a few threads involved and the meters are updated
from a g_timeout_add timer callback, the update_meters function
On Thursday 14 October 2004 15:10, Dave Robillard wrote:
> On Wed, 2004-10-13 at 16:45, Dan Mills wrote:
> > Hi All,
> > I have a small utility (jackswitch - a clickless audio switch
> > for the jack enviroment) and was wondering if there is a site for hosting
> >
On Wednesday 13 October 2004 23:32, Steve Harris wrote:
> > Steve, would plugin.org.uk take it?
>
> Sure. Mail me a tarball or something and I'l look at it in the morning.
It's OK, it looks like it may be going to become an example jack client.
If it does not make it into the jack distribution
On Wednesday 13 October 2004 22:15, Jack O'Quin wrote:
>
> Sure, as long as it's a "good" example. ;-)
>
> It should be something that will not cause us a maintenance burden.
> Large or exotic library dependencies should be avoided. We don't want
> to force people to install GTK or Qt just to com
Hi All,
I have a small utility (jackswitch - a clickless audio switch for
the jack enviroment) and was wondering if there is a site for hosting this
kind of small simple utility program.
Sure I could create a web site for it but it seems rather silly to write a
website that is probably going
On Monday 20 September 2004 22:41, Eric Dantan Rzewnicki wrote:
> On Sun, Sep 19, 2004 at 10:49:01AM -0400, Taybin Rutkin wrote:
> > JACK is a low-latency audio server, written primarily for the GNU/Linux
> > operating system. It can connect a number of different applications to
> > an audio device
>
> http://www.spamblock.demon.co.uk
How embarassing! I got the web page wrong!
http://www.spamblock.demon.co.uk/louderbox.html
is the correct site!
Regards, Dan.
--
** The email address *IS* valid, do NOT remove the spamblock
And on the evening of the first day the lord said...
..
Hi All,
I am pleased to announce that I finally have an alpha release of a louderbox
that is (hopefully) reasonably fit to show its face in daylight
What I have is an 8 band compressor/limiter/clipper/pre em thing written as a
jack client. It borrows very heavily from jamin but as f
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