Like many other things ….
Steffan
> On 19.03.2020|KW12, at 17:01, Ethan Fenn wrote:
>
> So interestingly those two #define's together would have no effect!
>
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As long as we're going off the rails...
This provoked me into learning something new:
https://stackoverflow.com/questions/24177503/how-does-the-c-preprocessor-handle-circular-dependencies
So interestingly those two #define's together would have no effect!
-Ethan
On Thu, Mar 19, 2020 at 7:34
#define analog digital
#define digital analog
and now read again ….
Best,
Steffan
> On 19.03.2020|KW12, at 12:31, Theo Verelst wrote:
>
> Maybe a side remark, interesting nevertheless: the filtering in digital
> domain, as
> compared with the analog good ol' electronics filters isn't the
Hi Eric
I'm sure your filterbank EQ sounds fine. Aliasing should be contained to a
very low level if appropriate windows/overlap are used and the filter
response isn't pushed to any extremes.
But, zero-phase (offline) processing is straightforward to achieve with
FIR. You just do a linear-phase
Hi Ethan,
It's been a few years since I've ran or heard this FFT filterbank EQ. I do
remember it being quite clean, indeed, I chose to work on it precisely
because I realized that it could be designed to be zero-phase (meaning no
phase distortion like you get from traditional FIR/IIR eqs).
The
On Thu, Mar 12, 2020 at 9:35 PM robert bristow-johnson <
r...@audioimagination.com> wrote:
> i am not always persuaded that the analysis window is preserved in the
> frequency-domain modification operation.
It definitely is *not* preserved under modification, generally.
The Perfect
> On March 12, 2020 5:35 PM Ethan Duni wrote:
>
>
> Hi Robert
>
>
> On Wed, Mar 11, 2020 at 4:19 PM robert bristow-johnson
> wrote:
> >
> > i don't think it's too generic for "STFT processing". step #4 is pretty
> > generic.
>
> I think the part that chafes my intuition is more that
Hi Robert
On Wed, Mar 11, 2020 at 4:19 PM robert bristow-johnson <
r...@audioimagination.com> wrote:
>
> i don't think it's too generic for "STFT processing". step #4 is pretty
> generic.
>
I think the part that chafes my intuition is more that the windows in steps
#2 and #6 should "match" in
> On March 11, 2020 6:53 PM Ethan Duni wrote:
>
>
> On Tue, Mar 10, 2020 at 8:36 AM Spencer Russell wrote:
> >
> > The point I'm making here is that overlap-add fast FIR is a special case
> > of STFT-domain multiplication and resynthesis. I'm defining the standard
> > STFT pipeline here
On Tue, Mar 10, 2020 at 8:36 AM Spencer Russell wrote:
>
> The point I'm making here is that overlap-add fast FIR is a special case
> of STFT-domain multiplication and resynthesis. I'm defining the standard
> STFT pipeline here as:
>
> 1. slice your signal into frames
> 2. pointwise-multiply an
On 10/03/2020 19:45, Ethan Duni wrote:
On Mar 10, 2020, at 3:38 AM, Richard Dobson wrote:
You can have windows when hop size is 1 sample (as used in the sliding phase
vocoder (SPV) proposed by Andy Moorer exactly 20 years ago, and the focus of a
research project I was part of around
> On Mar 10, 2020, at 3:38 AM, Richard Dobson wrote:
>
> You can have windows when hop size is 1 sample (as used in the sliding phase
> vocoder (SPV) proposed by Andy Moorer exactly 20 years ago, and the focus of
> a research project I was part of around 2007). So long as the window is based
> On March 10, 2020 11:34 AM Spencer Russell wrote:
>
>
> Thanks for your expanded notes, RBJ. I haven't found anything that I disagree
> with or that contradicts what I was saying earlier - I'm not sure if they
> were intended as expanded context or if there was something you were
>
Thanks for your expanded notes, RBJ. I haven't found anything that I disagree
with or that contradicts what I was saying earlier - I'm not sure if they were
intended as expanded context or if there was something you were disagreeing
with.
On March 8, 2020 7:55 PM Ethan Duni wrote:
>
> Fast
On 10/03/2020 00:41, Ethan Duni wrote:
...
Right, but if you are using length K FFT and zero-padding by K-1, then the hop
size is 1 sample and there are no windows.
You can have windows when hop size is 1 sample (as used in the sliding
phase vocoder (SPV) proposed by Andy Moorer exactly
It is certainly possible to combine STFT with fast convolution in various ways.
But doing so imposes significant overhead costs and constrains the overall
design in strong ways.
For example, this approach:
> On Mar 9, 2020, at 7:16 AM, Spencer Russell wrote:
>
>
> if you have an KxN STFT
> On March 8, 2020 7:55 PM Ethan Duni wrote:
>
> Fast FIR is a different thing than an FFT filter bank.
>
> You can combine the two approaches but I don’t think that’s what is being
> done here?
> On March 9, 2020 10:15 AM Spencer Russell wrote:
>
>
> I think we're mostly on the same
I think we're mostly on the same page, Ethan. Though even with STFT-domain
time-variant filtering (such as with noise reduction, or mask-based source
separation) it would seem you could still zero-pad each input frame to
eliminate any issues due to time-aliasing. As you mention (paraphrasing),
On Sun, Mar 8, 2020 at 8:02 PM Spencer Russell wrote:
> In fact, the the standard STFT analysis/synthesis pipeline is the same
> thing as overlap-add "fast convolution" if you:
>
> 1. Use a rectangular window with a length equal to your hop size
> 2. zero-pad each input frame by the length of
Nowhere was it mentioned that there was an across the frame multiplication
with a scalar as far as manipulating the transform coefficients. That
might make it time variant. My concept was in the domain of audio
engineering which reads a side-chain signal to obtain attenuation factors
in the
On Sun, Mar 8, 2020, at 7:41 PM, Ethan Duni wrote:
> FFT filterbanks are time variant due to framing effects and the circular
> convolution property. They exhibit “perfect reconstruction” if you design the
> windows correctly, but this only applies if the FFT coefficients are not
> altered
so Ethan, what is your definition of time invariance? because you say it's
not time invariant because of time domain aliasing but then you say there
is delay due to compute time. delay due to window and compute time is
unavoidable and not to be factored into time invariance / variance. coding
>
> If the system is suitably designed (e.g. correct window and overlap),
> you can filter using an FFT and get identical results to a time domain
> FIR filter (up-to rounding/precision limits, of course). The
> appropriate window and overlap process will cause all circular
> convolution
On Sun, Mar 8, 2020 at 11:41 PM Ethan Duni wrote:
> FFT filterbanks are time variant due to framing effects and the circular
> convolution property. They exhibit “perfect reconstruction” if you design the
> windows correctly, but this only applies if the FFT coefficients are not
> altered
No, MDCT TDAC is the same. Perfect reconstruction only obtains if the
coefficients are not changed at all. Coding noise causes (uncancelled) time
domain aliasing that is shaped according to the window design. Limiting this
effect is a primary aspect of MDCT codec design.
Ethan
> On Mar 8,
Audio compression by definition 'alters' the transform coefficients and
they get perfect reconstruction with no aliasing due to the transform
alone. In fact 'TDAC' or time domain aliasing cancellation is a hallmark
of the MDCT or DCT type IV which is ubiquitous in audio codecs.
On Sun, Mar 8,
FFT filterbanks are time variant due to framing effects and the circular
convolution property. They exhibit “perfect reconstruction” if you design the
windows correctly, but this only applies if the FFT coefficients are not
altered between analysis and synthesis. If you alter the FFT
The system is memoryless just because it is based on the DFT and nothing
else, which is also why it's time-invariant. unless you alter certain
parameters in real-time like the window size or hop size or windowing
function, etc, any input gives you the same output at any given time, which
is the
Well I believe the system is LTI just because the DFT is LTI by
definition. The impulse response of a rectangular window I believe is that
of a sinc function, which has ripple artifacts. Actually, the overlap-add
method (sorry I don't have time to dig into the differences between
overlap-add and
> On March 8, 2020 2:00 PM Zhiguang Eric Zhang wrote:
>
> it is not causal because the zero-phase system does not depend on past samples
>
>
> On Sun, Mar 8, 2020 at 1:58 PM Zhiguang Eric Zhang wrote:
> > the frequency response is a function of the windowing function
> >
> >
what
it is not causal because the zero-phase system does not depend on past
samples
On Sun, Mar 8, 2020 at 1:58 PM Zhiguang Eric Zhang wrote:
> the frequency response is a function of the windowing function
>
> On Sun, Mar 8, 2020 at 10:34 AM robert bristow-johnson <
> r...@audioimagination.com>
the frequency response is a function of the windowing function
On Sun, Mar 8, 2020 at 10:34 AM robert bristow-johnson <
r...@audioimagination.com> wrote:
>
>
> > On March 8, 2020 10:05 AM Ethan Duni wrote:
> >
> >
> > It is physically impossible to build a causal, zero-phase system with
>
> On March 8, 2020 10:05 AM Ethan Duni wrote:
>
>
> It is physically impossible to build a causal, zero-phase system with
> non-trivial frequency response.
a system that operates in real time. when processing sound files you can
pretend that you're looking at some "future" samples. i
It is physically impossible to build a causal, zero-phase system with
non-trivial frequency response.
Ethan
> On Mar 7, 2020, at 7:42 PM, Zhiguang Eric Zhang wrote:
>
>
> Not to threadjack from Alan Wolfe, but the FFT EQ was responsive written in C
> and running on a previous gen MacBook
if you are implementing an FIR filter using "fast convolution" which uses an
FFT and overlap-add or overlap-scrap (often the latter is called
"overlap-save"), then the window is *rectangular*, but if you do this right,
there is no ripple artifact from the windowing of the time-domain data.
Not to threadjack from Alan Wolfe, but the FFT EQ was responsive written in
C and running on a previous gen MacBook Pro from 2011. It wouldn't have
been usable in a DAW even without any UI. It was running FFTW.
As far as linear / zero-phase, I didn't think about the impulse response
but what
> On March 7, 2020 6:43 PM zhiguang zhang wrote:
>
>
> Yes, essentially you do have the inherent delay involving a window of samples
> in addition to the compute time.
yes. but the compute time is really something to consider as a binary decision
of whether or not the process can be real
Yes, essentially you do have the inherent delay involving a window of
samples in addition to the compute time.
On Sat, Mar 7, 2020, 5:40 PM Spencer Russell wrote:
> On Sat, Mar 7, 2020, at 6:00 AM, Zhiguang Eric Zhang wrote:
>
> Traditional FIR/IIR filtering is ubiquitous but actually does
On Sat, Mar 7, 2020, at 6:00 AM, Zhiguang Eric Zhang wrote:
> Traditional FIR/IIR filtering is ubiquitous but actually does suffer from
> drawbacks such as phase distortion and the inherent delay involved. FFT
> filtering is essentially zero-phase, but instead of delays due to samples,
> you
Sorry I meant Alan :)
On Wed, Jan 15, 2020, 11:20 PM Alan Wolfe wrote:
> probably pretty basic stuff for most people here but wanted to share a
> writeup and demo i made about FIRs.
>
> Post: https://blog.demofox.org/2020/01/14/fir-audio-data-filters/
>
This is a very cool blog, I need to spend some time with it. It's also
interesting to draw parallels between the graphics stuff that Alex writes
about to the audio realm.
Traditional FIR/IIR filtering is ubiquitous but actually does suffer from
drawbacks such as phase distortion and the inherent
On Tue, Mar 3, 2020, at 4:21 PM, robert bristow-johnson wrote:
>
> Like a lotta things, sometimes people use the same term to mean something
> different. A "phase vocoder" (an STFT thing a la Portnoff) is not the same as
> a "channel vocoder" (which is a filter bank thing).
It’s maybe worth
r...@audioimagination.com
>>>
>>> "Imagination is more important than knowledge."
>>>
>>>
>>>
>>>
>>>
>>> Original message
>>> From: Alan Wolfe
>>> Date: 3/3/2020 16:10
gt; --
>> r b-j r...@audioimagination.com
>>
>> "Imagination is more important than knowledge."
>>
>>
>>
>>
>>
>> ---- Original message
>> From: Alan Wolfe
>> Date: 3/3/2020 16:10 (GMT-05:00)
more important than knowledge."
>
>
>
>
>
> Original message
> From: Alan Wolfe
> Date: 3/3/2020 16:10 (GMT-05:00)
> To: A discussion list for music-related DSP
>
> Subject: Re: [music-dsp] FIR blog post & interactive demo
>
>
ination is more important than knowledge."
Original message
From: Alan Wolfe
Date: 3/3/2020 16:10 (GMT-05:00)
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] FIR blog post & interactive demo
Man that's neat. I've been wondering how a v
Man that's neat. I've been wondering how a vocoder worked. I'm looking
forward to reading through your work.
BTW, there is also an IIR demo and blog post now.
http://demofox.org/DSPIIR/IIR.html
On Tue, Mar 3, 2020 at 1:04 PM Zhiguang Eric Zhang wrote:
> this is cool, i can't believe I
this is cool, i can't believe I actually worked on FFT filtering (via phase
vocoder) before learning FIR/IIR filters ... ?
if anyone's interested in that source code it's here:
https://www.github.com/kardashevian
On Wed, Jan 15, 2020 at 11:20 PM Alan Wolfe wrote:
> probably pretty basic stuff
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