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kiki - 2004-11-15 14:47:00
Hi all,
I've heard a lot of aliasing. What do they look like in image and
audio/speech/music with aliasing?
I did not see/hear any real aliasing stuff, so the concept of aliasing
looks abstract to me, although I know how it occurs in terms of
mathematics...
I am also wondering about the ...
On Tue, 1 Apr 2008 23:35:21 -0700 (PDT), dbd wrote:
> "Get over it" is not an attitude. It is a constructive suggestion for
> someone who has difficulty perceiving the use of "anti-aliasing
> filter" for filters that perform anti-aliasing.
>
> I'm not annoyed, I'm disappointed.
>
> I'v...
Kevin T - 2006-09-12 14:51:00
Looks like Treo and maybe other Windows Mobile based devices have huge
(~15% THD) wideband aliasing products when you play any non 44.1 file (
MP3 or .Wav) back. Windows resamples all to 44.1 k and seems to have
poor anti-aliasing filters. Whats up with that? My Laptop (XP) AC97 is
OK on ...
Green Xenon [Radium] - 2008-05-25 00:01:00
Hi:
Is spatial aliasing caused by an insufficient pixel-density?
Also, some video websites say that spatial aliasing causes jaggies,
other say it causes the Moire effect, still others say it causes
pixelation. Who is right?
Thanks,
Radium
...
Please refer to
www.intersil.com/data/fn/fn3651.pdf
page 10 top and page 9 bottom.
Please correct me if I assume something wrong.
On figure 12 (pg 10) you see the aliasing profile of the CIC and how
it does it translate to baseband. The filters performance can
therefore be estimated. Quit...
Green Xenon [Radium] - 2008-05-12 20:20:00
Hi:
Aliasing is a digital entity. What is the analog equivalent of aliasing?
Thanks,
Radium
...
lovejet - 2009-06-28 11:56:00
Hi, I'm junior in the signal processing and I have a question. Can anybody
help to answer me? Suppose there's a analog signal bandwidth is B Hz and
with noise. There is no higher frequency componet outside the B Hz but the
noise. If I sample and quantize the analog signal by the sampling rate 2*B
Hz...
sorenbirk - 2006-10-09 07:27:00
I am making an application where several inputs has to be filtered. The
filtering is a simple moving average filtering with a variable filter
length. The problem is what happens when I connect these moving average
filters in cascade? I know that I will get some aliasing, but how big is
the aliasing ...
Hi,
For a single carrier system, rooted raised cosine pulse shaping filter
is generally used in both transmitting and receiving. I have read a
paper which said the FFT bin outside of signal plus excessive bandwidth
were set to zero to suppress aliasing signal component. I find that in
frequency...
M.L. wrote:
> Hello Jerry,
>
> I am not sure what you mean.. Why will the effect that you describe be
> introduced?
It's called "aliasing". see
http://www.dsptutor.freeuk.com/aliasing/AliasingDemo.html
Jerry
--
Engineering is the art of making what you want from things you can ge...
2007-03-31 02:24:00
If you are estimating time-delay bewteen two microphones would it be
better to use a Bessel filter for anti-aliasing rather than a
Butterworth due to the approximate linear phase.
Wang King
...
Vista - 2007-06-02 16:31:00
Hi all,
I need your help on the following difficulties:
I am designing something like "automatic" detection of aliasing,
programmatically.
For a given signal discrete time signal x(n), possibly infinite support, we
know its values are all positive, they are absolutely summable, so DTFT ...
hrh1818 - 2009-07-27 16:44:00
The following document says on page 5 an anti aliasing filter is used
on the A/D input channels of an energy meter. Warning the document is
31 pages long, only click on the link if you have a broadband internet
connection. http://focus.ti.com/lit/an/slaa409a/slaa409a.pdf. My
understanding is ...
2009-01-15 09:05:00
Why exactly does MDCT introduce time-domain aliasing? From the 2M
samples, the MDCT transforms to M samples, before the IMDCT transform
back to 2M samples. From my understanding going from the frequency
domain to the time domain using IMDCT, the original 2M samples are now
only represented by M ...
Mark - 2006-04-27 16:42:00
How to prevent aliasing caused by non-linear function implemented in
the digital domain
Consider an audio input signal band limited to 20 kHz and sampled at
44.1 kHz for example. No problem. But suppose I want to implement a
limiter or other non-linear function in the digital domain. The
...
twain - 2006-03-18 01:16:00
I have a large bunch of WAV files sampled at 44.1kHz and at 48kHz, which
I would like to convert to 8kHz sampling rate, efficiently/quickly but
professionally & accurately (i.e. no aliasing etc.)
Could anyone recommend me a tool for doing that (free software or Matlab)?
(For example I know...
I'm working on a paper about interpolation that I threatened to do long ago.
I'm trying to say that a spectrum that has nonzero (or not small) samples at
or near fs/2 is problematic.
But I'm having a bit of trouble saying why it's bad necessarily.
One thing one can say is that it's likely t...
nikkogta4 - 2009-04-13 09:48:00
Hi,
i hav just studied the impulse invariance method & bilinear transform.
In Impulse Invariance there is "many to one mapping" and hence aliasing
takes place.
while in Bilinear Transform there is "one to one mapping" and hence no
aliasing.
Actually what is the "one to one mapping" & "many to ...
2007-04-10 21:24:00
I am talking about geophysical data here, where you have a record
consisting of a few hundred traces, so the data has two dimensions
time and distance (or t-x in short).
If an event has a dip or slowness less than one sample per trace,
it is not aliased. If it has greater dip, it would alias, u...
John182 - 2006-12-28 08:20:00
hey guys - i can't seem to find find this statement any where - but it
makes sense to me - could someone please verify.
When you find the DFT of a signal say: x[0]=2, x[1]=4, x[2]=6, x[3]=8 -
you will get four discrete points in frequency (per period) now if you
zero pad such that tou get x[0]=2, x...
Got this from a friend, who was mostly interested in it because it
looks cool, and we both play an on-line game that includes this type
of helicopter.
http://www.break.com/index/helicopter-blades-stop-moving.html
It's a pretty interesting example of visual aliasing. I wonder what
the rota...
Tom - 2004-06-07 03:24:00
I have been reasing a paper that says that for acoustic beamformers (I
suppose the same applies to EM beamformers), the distance between
microphones must be
d ...
How does a sound card set its anti-aliasing filters? After all, you can
program a sound card to read at say 44.1kHz or 22,050Hz or half of that
again so how do the ani-aliasing filters change? Switched cap filters are
sampled filters so they would not be good and digital filters are no good
eith...
2004-08-23 09:53:00
Here's something I don't remember about aliasing - someone please
verify whether or not this is correct. Assume the sample rate is
2000 Hz for the sake of illustration.
Conventional wisdom tells us that a 1200 Hz signal will look
like an 800 Hz signal due to aliasing. I say it may
look the s...
Jerry Avins wrote:
(snip, I wrote)
> > It is the FFT that creates negative frequencies. You could,
> > for example, use the Hartley transform instead.
> Can I cite you as authority for the notion that negative frequencies are
> created by (artifacts of) Fourier transforms? Please...
pwaiaung - 2006-09-07 09:54:00
Hello everyone,
I want to design a real time DSP system.Band of interest extends from o to
4kHz.I will use 12 bit ADC and 3rd order butterworth lowpass filter.
What I want to know is how I can estimate the minimum stopband attenuation
for the anti-aliasing filter, minimum sampling frequency an...
> That is correct, you can't. I would question whether a length = 3
> filter is going to do you much good for decimating data by a factor of
8
> -- you would at least need a length = 8 filter and probably more.
I forgot to mention one more thing abt the filter. It is not normal
anti-aliasi...
2006-05-19 01:25:00
banton wrote:
> Mmh.. allright. So the answer to the question, if I can use the
> shift theorem to achieve arbitrary phase shifts for real valued
> signals is "NO!"?
You can use a form of the shift theorem for arbitrary non-integer
shifts. However, there is a complication that arises for...
2006-04-05 17:39:00
Two additional thoughts.
If there is noise in the adc, then sampling at a higher rate will cause
that noise to be averaged over three samples rather than just the one.
This might improve the snr a little.
A long time ago Crystal (now Cirrus Logic) shipped two different
versions of their aud...
pompano - 2008-01-30 18:23:00
> My apologies for the bad link....it is actually:
>
> http://s188.photobucket.com/albums/z65/liznbren/stuff/?action=view¤t=envelope.jpg
>
> Thanks....
> Brendan
>
> If I sample at just fractionally above the Nyquist rate,
That is the problem right there, and what you are seein...
mudskipper - 2007-02-12 09:57:00
hello,
i'm trying to program a wavetable-based synth in c++. I have some
waveforms and different envelopes for them and want to switch between
settings while playing. I use a 4096 samples buffer. My question is, how
do i now implement anti-aliasing? I thought of multiplying envelope &
waveform, ...
The amount of the distortion due to aliasing equals to the sum of power
for all components higher then Fs/2.
Indeed the speech spectrum envelope falls at about 12dB/oct at high
frequencues. The amount of the high frq content in the clear voice is
very low. So just the direct sampling at...
Given a stationary discrete-time stochastic process x(n) with
discrete-time Fourier transform X(e^jw) and spectrum Sx(w). If x(n) is
downsampled by two to give y(n) it is well known that
Y(e^jw) = 0.5 [X(e^jw/2) + X(e^j(pi+w/2))]. (A)
My question is, what is the resulting spectrum, not just...
Given a stationary discrete-time stochastic process x(n) with
discrete-time Fourier transform X(e^jw) and spectrum Sx(w). If x(n) is
downsampled by two to give y(n) it is well known that
Y(e^jw) = 0.5 [X(e^jw/2) + X(e^j(pi+w/2))]. (A)
My question is, what is the resulting spectrum, not just...
On Tue, 01 Apr 2008 14:41:06 -0600, jim
wrote:
> Eric Jacobsen wrote:
> > The more fundamental issue as I understand it, is whether or not a
> > filter would change the original, uninterpolated input samples in the
> > output, i.e., are the interpolated samples distinguishable from ...
somenoob - 2006-06-21 08:54:00
I'm pretty new to the world of frequency domain as proven by this
question:
Is there any correlation between the quality of resampler required to make
aliasing "imperceptible" and the magnitude of sampling rate change?
I'm currently playing with various resampling algorithms, running some
44...
2007-01-05 22:24:00
Ok first post here and I thought I'd bring up something that might be
thought-provoking... have you ever looked at the workings of the human
visual system? It does some fairly amazing things that, on the
surface, seem to violate sampling theory.
#1 - Hyperacuity
In the eye, rods and cones ar...
nikolatesla20 wrote:
> This morning I came up with a solution that seems to work for
> what I want. What I'm doing now is just always upsampling by 5,
> then I filter, and then I use a varying downsample rate to get
> the ratio I want.
> My filter is set to cut -80db at 10000hz runnin...
Tim Wescott wrote:
> I'm trying to think of examples where using an anti-alias filter is a
> bad idea, or must at least be approached with extreme caution. I
> already know about control systems and video applications, and I believe
> that this is a big issue with EKG machines.
>
> Do...
"HelpmaBoab" wrote in message
news:ph6Pf.4437$JZ1.96604@news.xtra.co.nz...
> Are you saying that a SB has a fixed anti-aliasing filter at 20kHz even
> you
> you sample at say 22050Hz? Surely not.
> I understood they used over-sampling and down-sampling.
If I record from, say, an SB ...
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