I would be OK with forcing the list to be html free, as long as I got
a notification that my emails were refused because they were in html.
I have sent messages to the list several times and been surprised that
I didn't get a response!
On Thu, Aug 8, 2013 at 10:51 AM, Eric Battenberg wrote:
> I'l
the big problem i am dealing with is people singing or humming and
changing notes. i really want to encode those pitch changes as new
notes rather than as a continuation of the previous note (perhaps
adjusted with MIDI pitch bend messages). there is not sufficient
change in amplitude or even in t
resolution and increase the density to get better time
resolution when you find a note on. That would be pretty low cost.
On Thu, Aug 8, 2013 at 11:23 AM, robert bristow-johnson
wrote:
> On 8/8/13 11:05 AM, Ian Esten wrote:
>>>
>>> On Mon, Aug 5, 2013 at 1:01 PM, robert bristo
Octave or Matlab. Or even Mathematica. It would be very interesting to see
the transfer function of your filter on the same graph as the 'ideal'
analog filter.
Ian
On Thursday, February 5, 2015, Peter S wrote:
> What do you guys use to turn your impulse responses into fancy FFT
> diagrams? If y
It's lossy. Definitely not linear.
On Thu, Feb 12, 2015 at 4:33 PM, robert bristow-johnson
wrote:
> On 2/12/15 3:02 PM, Theo Verelst wrote:
>>
>> Hi all,
>> Just a thought I share, because of associations I won't bother you with,
>> suppose you take some form of audio compression, say Fmp3(wav) w
> Lossy encoding wouldn't necessarily be non-linear in all cases.
Of course it is non-linear. Lossy encoding does not satisfy the
conditions of linearity:
f(a + b) = f(a) + f(b)
f(a.b) = a.f(b)
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On Sun, Apr 5, 2015 at 3:32 PM, robert bristow-johnson
wrote:
> On 4/5/15 5:21 PM, Theo Verelst wrote:
>>
>> In the context of synthesis, or intelligent multi sampling with
>> complicated signal issues, you could try to make the FFT analysis and
>> filtering a targeted part of the synthesis path,
A Leslie emulation (or effect similar to that) might well need one,
depending on how you modeled it. Same statement applies for tape delay
style effects too. As you say, I bet there's plenty of others, too.
Anyone else got any other effects to add to the list?
Ian
On Fri, Nov 19, 2010 at 1:07 PM
Which makes me think of a specialisation of this: waveguides for
physical modeling.
Ian
On Fri, Nov 19, 2010 at 3:41 PM, Ross Bencina
wrote:
> robert bristow-johnson wrote:
>>
>> there is a basic audio process called a "precision delay".
>
> Another requirement of precision delay might be tempo
I was going to ask the same question, until I looked at the webpage.
The features are listed out nicely.
On Fri, Sep 4, 2015 at 9:58 AM, Brad Fuller wrote:
> On 09/04/2015 09:42 AM, Andrew Kelley wrote:
>
> libsoundio is a C library providing cross-platform audio input and output
> for real-time
Thanks for sharing. Looks nice!
A question: I see that the write callback supplies a minimum and maximum
number of frames that the callback is allowed to produce. I would prefer a
callback that instructed me to produce a given number of samples. It is
simpler and more consistent with existing APIs
On Fri, Sep 4, 2015 at 10:58 AM, Andrew Kelley wrote:
> On Fri, Sep 4, 2015 at 10:43 AM Ian Esten wrote:
>>
>> Thanks for sharing. Looks nice!
>>
>> A question: I see that the write callback supplies a minimum and maximum
>> number of frames that the callba
This discussion is a refreshing change from some recent topics.
Constructive, respectful, not insulting. This is how it should be.
On Sun, Sep 6, 2015 at 2:41 AM, Ross Bencina wrote:
> Hello Andrew,
>
> Thanks for your helpful feedback. Just to be clear: I maintain the PortAudio
> core common cod
On Wed, Sep 9, 2015 at 8:42 AM, Theo Verelst wrote:
> Alsa is a relatively low level interface that doesn't do much such as
> dealing with multiple applications and resampling, which pulseaudio can do
> I think, not necessarily what people want.
>
Alsa can deal with multiple applications no prob
Look up spectral flux.
On Wed, Jul 6, 2016 at 7:24 AM, Danijel Domazet <
danijel.doma...@littleendian.com> wrote:
> Hi music-dsp,
> How does one implement an envelope adjustment algorithm that is triggered
> only on transients, rather than on a loudness threashold which is used in
> conventional
That would also be my first choice. It might also be worth looking at
modern algorithms in that family of methods. A lot of effort has gone into
designing exponentially damped sine methods for voice compression and
transmission. They will be more robust to noise than Prony's method. Some
methods ca
ave some references for the time-varying methods? Just
> curious...
>
> On Jul 31, 2017 12:04 AM, "Ian Esten" wrote:
>
>> That would also be my first choice. It might also be worth looking at
>> modern algorithms in that family of methods. A lot of effort has gone into
>
The Kurzweil K150 is the first product I can think of that did it. To
create custom sounds for it required the use of software that modeled the
sound using partial amplitudes over time. It's a very powerful technique
for synthesising certain types of sound, such as a piano, where frequencies
of par
The problem you are experiencing is caused by the fact that after changing
the filter coefficients, the state of the filter produces something
different to the current output. There are several ways to solve the
problem:
- The time varying bilinear transform:
http://www.aes.org/e-lib/browse.cfm?eli
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