On Tue, Mar 19, 2013 at 1:02 PM, Fons Adriaensen f...@linuxaudio.orgwrote:
The code below will do the trick
Brilliant, thanks for sharing. Will be implementing learning from this
later, appreciated!
-Harry
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[Fons Adriaensen]
On Tue, Mar 19, 2013 at 04:26:21AM +0100, Tim Goetze wrote:
A 2nd-order IIR filter is often called a biquad; at musicdsp, look
for that instead.
Not really. A biquad is one way to implement a 2nd order IIR, and
in many cases related to audio DSP, not really the best way.
On Tue, Mar 19, 2013 at 03:02:19PM +0100, Tim Goetze wrote:
for (i = 0; i nframes; i++)
{
g1 += w * (gt - g1 - a * g2);
g2 += w * (b * g1 - g2);
out [i] = g2 * in [i];
}
Surely you realise this version executes exactly as many additions and
multiplications per sample
[Fons Adriaensen]
On Tue, Mar 19, 2013 at 03:02:19PM +0100, Tim Goetze wrote:
Surely you realise this version executes exactly as many additions and
multiplications per sample as a biquad?
Yes. In this case it's possible to remove one multiplication:
a = 0.07f;
b = 1 + a;
// ...
gm = b * gt;
On Tue, Mar 19, 2013 at 04:46:10PM +0100, Tim Goetze wrote:
However, I still take issue with the rather exaggerated claim that
using a biquad to smoothen gain changes would be giant overkill. It
may be slightly less efficient than your now optimised version, but
certainly not enough to
[Fons Adriaensen]
Exactly the same with the form I proposed, w, a, b need to be computed
just once, not for every gain change. In fact only w depends on the
sample rate, a and b are fixed constants.
Ah yes, sorry, I see that now.
If that extra operation comes around to bite hard enough, I'll
On Tue, Mar 19, 2013 at 08:15:41PM +0100, Tim Goetze wrote:
[Fons Adriaensen]
Exactly the same with the form I proposed, w, a, b need to be computed
just once, not for every gain change. In fact only w depends on the
sample rate, a and b are fixed constants.
Ah yes, sorry, I see that now.
On Tue, Mar 19, 2013 at 2:15 PM, Tim Goetze t...@quitte.de wrote:
[Fons Adriaensen]
Exactly the same with the form I proposed, w, a, b need to be computed
just once, not for every gain change. In fact only w depends on the
sample rate, a and b are fixed constants.
Ah yes, sorry, I see that
On Tue, Mar 19, 2013 at 05:26:27PM -0500, Charles Z Henry wrote:
You guys are splitting hairs... kind of misses the forest for the trees.
Here's my nit to pick: two identical 1st-order lowpass filters in series
are only equivalent to a 2nd-order lowpass filter when the quality factor
is 0.5.
On Tue, Mar 19, 2013 at 5:57 PM, Fons Adriaensen f...@linuxaudio.orgwrote:
On Tue, Mar 19, 2013 at 05:26:27PM -0500, Charles Z Henry wrote:
You guys are splitting hairs... kind of misses the forest for the trees.
Here's my nit to pick: two identical 1st-order lowpass filters in series
are
On Tue, Mar 19, 2013 at 06:12:21PM -0500, Charles Z Henry wrote:
If it was always a fixed time over which you need to fade in, I think you
could find a good analytical function to use, or make a table that always
has a predictable effect.
Raised cosine is a good solution, but you don't need
Hi again. Looking for any advice, tips, tricks, anecdotes etc.
I want to eliminate or reduce 'zipper' noise on volume changes.
So I'm looking at two techniques:
Zero-crossing / zero-value signal detection, and slew-rate limiting.
Code is almost done, almost ready to start testing each technique.
On Mon, Mar 18, 2013 at 5:50 PM, Tim E. Real termt...@rogers.com wrote:
Hi again. Looking for any advice, tips, tricks, anecdotes etc.
I want to eliminate or reduce 'zipper' noise on volume changes.
So I'm looking at two techniques:
Zero-crossing / zero-value signal detection, and slew-rate
On Mon, Mar 18, 2013 at 05:50:39PM -0400, Tim E. Real wrote:
If I use a zero-crossing/zero-value detector and apply volume changes
only at these safe points, that's a much more desirable 'perfect' system.
Zero crossings are not 'safe'. You avoid a discontinuity in the signal
but there will
On March 18, 2013 06:47:16 PM you wrote:
On Mon, Mar 18, 2013 at 5:50 PM, Tim E. Real termt...@rogers.com wrote:
Hi again. Looking for any advice, tips, tricks, anecdotes etc.
I want to eliminate or reduce 'zipper' noise on volume changes.
So I'm looking at two techniques:
Zero-crossing /
On 18/03/13 Tim E. Real termt...@rogers.com wrote:
On March 18, 2013 06:47:16 PM you wrote:
On Mon, Mar 18, 2013 at 5:50 PM, Tim E. Real termt...@rogers.com
wrote:
Hi again. Looking for any advice, tips, tricks, anecdotes etc.
I want to eliminate or reduce 'zipper' noise on volume changes.
On March 18, 2013 07:04:52 PM Tim E. Real wrote:
On March 18, 2013 06:47:16 PM you wrote:
On Mon, Mar 18, 2013 at 5:50 PM, Tim E. Real termt...@rogers.com wrote:
Hi again. Looking for any advice, tips, tricks, anecdotes etc.
I want to eliminate or reduce 'zipper' noise on volume changes.
So
On Mon, Mar 18, 2013 at 07:43:32PM -0400, Tim E. Real wrote:
Ah, I may have answered my own question when I said:
(One cannot simply wait for the current data value to be 'zero' because
for example with a perfect square wave signal the 'current' value will never
approach zero, hence the
On March 18, 2013 11:58:48 PM Fons Adriaensen wrote:
On Mon, Mar 18, 2013 at 07:43:32PM -0400, Tim E. Real wrote:
Ah, I may have answered my own question when I said:
(One cannot simply wait for the current data value to be 'zero' because
for example with a perfect square wave signal
On Mon, Mar 18, 2013 at 11:45 PM, Fons Adriaensen f...@linuxaudio.orgwrote:
A critically damped
second order lowpass with a rise time of 30 ms or so will eliminate all
audible artefacts. It's very low on CPU and you only need to run it while
the gain is changing.
Although I understand the
[Tim Goetze]
[Harry van Haaren]
How is the rise time determined here?
As a function of the filter's damping (zeta = 2*Q) and frequency:
http://en.wikipedia.org/wiki/Rise_time
Sorry, zeta = 1 / (2*Q).
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On Tue, Mar 19, 2013 at 3:26 AM, Tim Goetze t...@quitte.de wrote:
tradition (like RBJ's lovely cookbook at musicdsp) will ask for filter
Q, which is 0.5 for critical damping.
Nice mention, checked out the code there, will play around with it a bit :)
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