>In this graph, the signal frequency seems to be 250 Hz, so this graph
>shows the equivalent of about 22000/250 = 88x oversampling.
That graph just shows the frequency responses of various interpolation
polynomials. It's not related to oversampling.
E
On Thu, Aug 20, 2015 at 5:40 PM, Peter S
wr
In the case of variable pitch playback with interpolation, here are
the frequency responses:
http://musicdsp.org/files/other001.gif
(graphs by Olli Niemitalo)
In this case, there's no zero at the original Nyquist freq, rather
there are zeros at the original sampling rate and its multiplies.
So i
In the starting post, it was not specified that resampling was also
used - the question was:
"Is it possible to use a filter to compensate for high frequency
signal loss due to interpolation? For example linear or hermite
interpolation."
Without specifying that variable rate playback is involved,
Let me just add, that in case of having a non-oversampled linearly
interpolated fractional delay line with exactly 0.5 sample delay (most
high-frequency roll-off position), the frequency response formula is
not sinc^2, but rather, sin(2*PI*f)/(2*sin(PI*f)), as I discussed
earlier.
In that case, th
>If all you're trying to do is mitigate the rolloff of linear interp
That's one concern, and by itself it implies that you need to oversample by
at least some margin to avoid having a zero at the top of your audio band
(along with a transition band below that).
But the larger concern is the overa
As far as the oversampling + linear interpolation approach goes, I have to
ask... why oversample so much (512x)?
Purely from a rolloff perspective, it seems you can figure out what your
returns are going to be by calculating sinc^2 at (1/upsample_ratio) for a
variety of oversampling ratios. Here's
Here's a graph of performance in mflops of varying length FFT
transforms from the fftw.org benchmark page, for Intel Pentium 4:
http://morpheus.spectralhead.com/img/fftw_benchmark_pentium4.png
Afaik Pentium 4 has 16 KB of L1 data cache. If you check the graph,
around 8-16k the performance starts
Thanks to all the participants in this thread, I hope it was at least a
little educational, except maybe for some that seem to take everything
as a test to their imaginations of themselves being little computers,
and not human being with normal associations and lasting affections for
serious su
Let's analyze your suggestion of using a FIR filter at f = 0.5/512 =
0.0009765625 for an interpolation filter for 512x oversampling.
Here's the frequency response of a FIR filter of length 1000:
http://morpheus.spectralhead.com/img/fir512_1000.png
Closeup of the frequency range between 0-0.01 (cu
On 20/08/2015, Ethan Duni wrote:
>
> Wasn't the premise that memory
> was cheap, so we can store a big prototype FIR for high quality 512x
> oversampling? So why are we then worried about the table space for the
> fractional interpolator?
And the other reason - the coefficients for a 2000-point w
On 20/08/2015, Ethan Duni wrote:
>
> Wasn't the premise that memory
> was cheap, so we can store a big prototype FIR for high quality 512x
> oversampling? So why are we then worried about the table space for the
> fractional interpolator?
For the record, wasn't it you who said memory is often a c
On 20/08/2015, Ethan Duni wrote:
> But I'm on the fence about
> whether it's the tightest use of resources (for whatever constraints).
Then try and measure it yourself - you don't believe my words anyways.
-P
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Hi,
A suggestion for those working on practical implementations, and lighten
up the tone of the discussion with some people I know from worked on all
kinds of (semi-) pro implementations when I wasn't even into more than
basic DSP yet.
The tradeoffs about engineering and implementing on a pl
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