Here's a Fourier joke about Joseph Fourier that you might not like, i heard
he rode a bike!
On Thu, Jun 25, 2020, 6:59 PM Emanuel Landeholm
wrote:
> Sorry for being a slowpoke! Is this an efficient implementation of STFT
> (short time fourier transform)?
>
> On Thu, Jun 25, 2020 at 8:49 AM
onvolution. but if you're doing
> non-LTI stuff in the phase vocoder, then that friendly frequency-domain
> behavior is more salient.
>
> --
>
> r b-j r...@audioimagination.com
>
> "Imagination is more important than knowledge."
>
>
> &g
Hi Russ,
Yes. In the previous reference, there is no example for overlap-add. A
sine/cosine framework is a relatively simple one for OLA and fulfills the
necessary requirements. In the case of audio coding, various filterbanks
with different types of windows have been designed for 'perfect
Why add all the unnecessary gripe details? Seems unprofessional.
Maybe someone over at SoundCloud can help - they're in Berlin
Or contact someone I know who used to DJ in California @grenier on twitter
On Thu, May 21, 2020, 2:37 PM gm wrote:
> Hi
>
> I am looking for a combined patent and
hi music-dsp,
just a disclosure that I worked on this whilst studying for my master's
degree at NYU, and was also a summer intern at Eventide. incidentally, one
of the founders at Eventide, John Agnello, has a patent that is similar to
what is being discussed here.
So RBJ thinks by himself and drinks by himself from '98 until. Let them
know it's real son if it's really real, understandable, self-explainable
On Fri, Mar 20, 2020, 3:47 PM robert bristow-johnson <
r...@audioimagination.com> wrote:
>
>
> > On March 20, 2020 2:45 PM STEFFAN DIEDRICHSEN
>
t; Then the regular overlap-add STFT resynthesis is the same as "fast
> convolution", and will give you the same thing (to numerical precision) you
> would get with a time-domain FIR implementation.
>
> On Mar 8, 2020, at 2:04 PM, zhiguang zhang
> wrote:
>
> but bri
according to the window design. Limiting
> this effect is a primary aspect of MDCT codec design.
>
> Ethan
>
> On Mar 8, 2020, at 4:45 PM, zhiguang zhang
> wrote:
>
>
> Audio compression by definition 'alters' the transform coefficients and
> they get perf
framing delay given by the frame size, and algorithmic latency
> given by the overlap. These are the delays that you’d compensate when
> running offline.
>
> Ethan
>
> On Mar 8, 2020, at 2:04 PM, zhiguang zhang
> wrote:
>
>
> The system is memoryless just because it is
Dr Bosi at AES.
-ez
On Sun, Mar 8, 2020 at 4:42 PM robert bristow-johnson <
r...@audioimagination.com> wrote:
>
>
> > On March 8, 2020 3:07 PM zhiguang zhang
> wrote:
> >
> >
> > Well I believe the system is LTI just because the DFT is LTI by
> defin
> > > > 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 Pro from
> 2011. It wouldn't have been usable in a DAW
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
A low-pass filter codec compression scheme; now that sounds like genuine snake
oil to me!
On Fri, Feb 13, 2015 at 7:02 PM, Theo Verelst theo...@theover.org wrote:
Ian Esten wrote:
It's lossy. Definitely not linear.
Right, it's not linear, because probably there are quantization
On a serious note, this may already be employed in hardware samplers. I
couldn’t find much info on it. SNES and Playstation store samples in a
compressed ADPCM format.
http://en.wikipedia.org/wiki/Bit_Rate_Reduction
On Thu, Feb 12, 2015 at 3:02 PM, Theo Verelst theo...@theover.org wrote:
BTW, what does linearity have to do with implementing compressed audio into
samplers? Why not just include a decoder upon playback?
On Fri, Feb 13, 2015 at 7:56 PM, Ian Esten i...@ianesten.com wrote:
Lossy encoding wouldn't necessarily be non-linear in all cases.
Of course it is non-linear.
Re:Pono, what about the DAC in the device? That could make an audible and real
difference. Also, there is undeniably more information in high res downloads,
if the original master was recorded to tape or to hi-res in Pro Tools. So, has
anyone ever considered the sample-level ‘phase’ effect
Lossy codecs are deemed transparent due to perceptibility and annoyance of
artifacts. How do you resolve lossy codecs with HD download shops like
HDtracks?
—
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On Wed, Oct 8, 2014 at 4:05 PM, Ethan Duni ethan.d...@gmail.com wrote:
Comparing neural firing rate to a PCM data
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