Hi all,
I would like to know about the oversampling used in communication
system. There are 2 technique which commonly used.
1. Using (N-1) zero padding, to get N factor oversampling and finally
use a digital filter to filled up the zeros.
2. Repeat the sample (N-1) times to get N facto...
What is the highest oversampling rate in commercial use on high fidelity (15
kHz) audio nowadays? I'm not looking for anything leading-edge or very
pricey. Just commercially available ADCs that are premium priced but aren't
extremely expensive.
A few years ago I recall 8X oversampling wa...
"I don't see where it says that higher sample rates reduce the
bandwidth."
Well the title for one,
"Option: Reduce noise by oversampling (at expense of bandwidth)"
And the first sentence too, also just type in a oversample rate and you
will see it recompute the lower bandwidth and displ...
As oversampling produces several baud rate signal sequences (for example,
oversampling by a factor of 2 produces two baud rate sequences with an
offset of half baud duration), could these sequences be viewed as a kind
of diversity? and what's its gain?
_____________________________________
...
Hi,
I've spent some time reading about oversampling and I think I understand
the fundamentals. Oversample by some factor > desired bandwidth,
decimate and increase ENOB (with certain assumptions about dithering,
etc).
My signal of interest is slowly varying (1-2 Hz), and I'm using a 10...
> In article ,
> "HSDPA-boy" wrote:
>
> | I see. Since you mention increased accuracy, are you assuming RRC
> | filtering in the digital domain?
>
> Yes.
>
>
> | Could you point me to some literature that discusses these matters?
It
> | would be interesting to see how the o...
I'm working with the new 56F630x series of Motorola DSP's (with an
onboard 12-bit ADC). My application could make good use of 14-bit
data, so I planned on oversampling by a factor of 16 and filtering
with decimation to get an extra 2 bits out of the ADC.
I recently spoke with someone from Mot...
I have tested use only cos(2*pi*fmix*n/fs) to demodulate a BFSK input
signal. And it worked. Question is how much worse will it be? Will it be
3dB worser?
Can I compensate this by oversampling instead? For example will a double
oversampling give 3dB more SNR.
Any answer is welcome!
/Mary
...
Hi,
I've been reading quite a bit about oversampling and it's apparent use
in noise reduction, but haven't been able to tie everything together.
As I probably can't explain my exact question clearly, I'll give an
example, I have a Analog Devices evaluation board with a 1MSPS A/D
with CPU hooked...
I'm new to DSP and also this group.
My question is "what can be the negative effect of Oversamling in terms of
Destortion in original signal?". As in undersampling two spectrum are
overlapped and we may not be able to get original signal.. is that any
kind of effect in oversampling?
...
Eric Jacobsen wrote:
...
> You can always trade off sample rate for sample width, i.e., if you
> want 12-bits of dynamic range, but your storage medium is eight bits,
> crank up the sample rate. When you decimate (or reconstruct) for
> playback you'll get your equivalent twelve bit...
The proper way to perform such oversampling is to put a decimating
lowpass filter just after your ADC. Because of the oversampling, noise
will spread over _wider_ spectrum area and hence when you apply a lowpass
filter you will get higher signal/noise ratio cutting out this (_wider_)
noise.
N...
Hi,
I've got a fundamental question. I have a signal (impulse response of a
room, computed from the original signal and the recorded signal) and
now I want to find out the maximum of that impulse response. This IR
has to be interpolated somehow, right now it is done by oversampling,
but I was ...
Hello,
I have a question regarding the OFDM transmitter. In order to reduce the
burden on the reconstruction filters, oversampling is used before feeding
the data to the DAC.
Oversampling, however, creates images of the baseband signal. To remove
these images, low pass filters are placed after...
hello !
I am working on a design of a Root Raised Cosine Pulse Shaping Filter and
my design of filter taps are on the basis of sampling frequency which is
of 4 times the BB frequency. Now to use this filter in the Tx section, I
need to oversample my input data by 4 times. I would like to have a
s...
Hi,
I have the following problem that I am trying to address. Lets say we
take 64 numbers and apply FFT of size 64. Then we concatenate 100 of
these 64 numbers, and perform D/A conversion. This signal is digitized
at a rate other than the original sampling rate, say we have
oversampling of 3/...
Prasanna Kumar Ganta wrote:
> How about...
>
> (1) Amplitude limiting the digitized FM signal.
> (2) Passing them through a frequency discriminator ( a filter whose
> transition band is linear and wide enough to span the delta F of the FM
> signal)
>
> Now you have an AM signal....
SL wrote:
> Hello all
>
>
>
> I'm wondering about how many bits I can gain if I first sample a signal with
> a sampling rate of fs and have N bits. And then I filter the signal to a
> bandwidth of fc Hz. And then at last I down sample the signal with D.
>
>
>
> ex.)
> ...
When the samples go through the digital to analog converter (DAC), the
spectrum is replicated periodically, as you should now. In order to
remove (suppress) the spectral replicas, a reconstruction filter is
required following the DAC.
If there are no high frequency bins left empty to create a...
Ville Voipio wrote:
>
> Actually, it would be more precise to talk about "oversampling"
> converters, because that's what makes the difference. Not the
> actual converter topology or modulator order.
>
> - Ville
>
Not exactly. The pole in the feedback loop of a delta-sigma conve...
"Tim Wescott" wrote in message
news:11f1okk9m8sna1@corp.supernews.com...
> > Clay
>
> I saw that after I posted the comment - the next observation is that it is
> exact when you're taking the differential, but how exact is it when you're
> approximating the differential with a dif...
Hi Fred,
My application is an OFDM transmitter. The output of the IFFT needs to be
oversampled by a factor of 4 (after adding the CP). I thought of doing the
oversampling by interpolating (inserting "zeroes") and low pass filtering
the output of the interpolator to remove the unwanted images. I d...
"kiki" writes:
> Hi all,
>
> Can anybody suggest some methods of gaining 8-bit ADC/DAC performance with 6
> bit only ADC/DAC?
One can use oversampling, or oversampling along with
noise-shaping. The latter technique is commonly called a delta-sigma
(some say sigma-delta, but they'r...
Hi,
I'm sampling a signal with a 35 MHz BW at a rate of 100Msa/s and 12 bit.
The nature of the signal is something like 100ns Peaks at about 2/3 of
peak value. I need the integration over each signal(100ns) but due to
the sampling rate I only get at about 4-10 samples of every signal, a
number w...
Hi All,
I am going to be selecting a specific sigma-delta converter for an
application and am getting a bit confused on the terminology. I have
read up on the theory and understand it.
Looking at this datasheet: http://focus.ti.com/lit/ds/symlink/ads1251.pdf
On the first page, it says 20...
> On 10 Mrz., 11:05, Rune Allnor wrote:
> > On Mar 10, 9:40=A0am, Andor wrote:
> >
> > > Rune Allnor wrote:
> > > > A far more intersting question is whether a negative group
> > > > delay is non-causal.
> >
> > > You seem to have comletely missed the whole point of the exe...
"maxascent" wrote in message
news:ZpSdnSyNJfpa11jeRVn-pg@giganews.com...
> I am not using a Bessel filter, but an elliptic filter. I dont want to use
> an dds ic as I am going to use an FPGA to do the DDS so I can generate
> other waveforms and arbitrary waveforms. I will have to use a B...
Ron N. wrote:
> Jerry Avins wrote:
>
> > vbbrett wrote:
> >
> > > Why does the 8 samples per cycle make the algoritm easier to implement?
> > > How does the number of samples affect the output? Why can't I just
> > > check 88 samples for a particular frequency? I'm afraid I don't
> > ...
Hi All,
Can anyone provide matlab code fragments to explain to me how to use the
kronecker delta to oversample a signal, s, by a factor of k ?
I'd really appreciate any help anyone can give on this,
Many thanks,
R.
...
> 1. Go to the library
> 2. Get a classic book on multirate processing by Rabiner
> 3. Don't ask any more stupid questions
4. Close all newsgroups.
...
tjuii wrote:
> > Why did you ask your original question?
> >
> > Jerry
> > --
> > Engineering is the art of making what you want from things you can get.
> > ŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻŻ
> >
>
>
> Because I want to know more than I do righ...
bmearns wrote:
> I have a DAC with 12 bit resolution that I want to use to playback an
> audio clip with 16 bits/sample. Obviously, I'm going to loose some
> quality, but I was wondering if it's possible to improve the quality a
> little by increasing the playback rate. For instance, my audi...
Oliver Glassl writes:
> > If so, then you're comparing apples to oranges. A linear PCM signal at
> > Fs samples per second always has a bandwidth of Fs/2 Hz; a 1-bit delta
> > sigma signal at Fs bits per second always has a bandwidth < < Fs/2 Hz.
>
>
> So can you tell me the bandwi...
Don't know anything about Kalman filters, but I have used oversampling
+ averaging to decrease the noise of accelerometers while maintaining
the required bandwidth. You have to have a fast A/D and CPU though, I
use 64 samples.
...
Prasad
> i am going through the datasheet of AIC-23 Codec on TI board, in the
> datasheet i have come acrros this statement
> BOSR Base oversampling rate
> USB mode: 0 = 250 fs 1 = 272 fs
> Normal mode: 0 = 256 fs 1 = 3...
"Symon" :
> Sanjay,
> Stick this into Google.
> sigma delta design maxim
Everybody is getting into that business it seems.
A decade ago it was unclear what new could be said by tutorials on the
subject. (New arrivals haven't always answered that point, and occasionally
even added...
...
>
> I guess that if my image shifts, that means I am effectively convolving
> it with a shifted sinc function, which means I'm multiplying the 2D FFT
> data with a phase-modulated rect function (i.e. somehow disturbing the
> phase of the original data). I don't see how just zero-pa...
I have read a bit about oversampling. Understand most of the theory but
get stuck at the easy bit. If we sample with one a one bit ADC - how do
we get 16 bits back from this single bit to do the processing?
Naebad
...