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
A few years ago I recall 8X oversampling wa...
"I don't see where it says that higher sample rates reduce the
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?
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,
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?
> | Could you point me to some literature that discusses these matters?
> | 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
Can I compensate this by oversampling instead? For example will a double
oversampling give 3dB more SNR.
Any answer is welcome!
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...
Can anyone tell that what is the effect of doing oversampling before pulse
shaping on the transmission bandwidth?
My concern is that as books describe that doing upsampling and then
filtering results in reduction in the Bandwidth but at the same time if we
view it from different angle, we are thr...
I have a communications system that is streaming analog data. On the
receiver side, this data is converted by an ADC with an oversampling of
4 related to the conversion rate of the DAC on the transmitter. As it is
a system under testing (with variable speed), we don't have a proper
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?
I have a question on OFDM. As we know, most OFDM systems nowadays do
not use all the available tones (subcarriers) for data modulation.
Let us say we have 64 tones in total and we might end up with 48 data
tones, 4 pilot tones, 3 DC tones and 9 null tones, which is kind of an
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_)
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 ...
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...
I have IR preamp with a gain of 64dB, the output noise is about 50mV peak
to peak for a signal of 4V peak peak (max) and the bandwidth is 350kHz.
I have a AD7626 (16 bits 10MSPS ADC) which is largly suffisient for my
I would want to know what would be the best method to f...
> > > 2)the effect of sampling frequency at ADC on SNR is ignored in the
> original equation of 6.02n + 1.76
> you can read this equation as follows:
> For a full-scale sine wave signal, the predicted noise across the whole
> bandwidth is 6.02n+1.76 dB below the power of the fu...
I've been reading a lot about increasing ADC quantization gain through
oversampling and then applying an interpolation filter.
I understand that for every 4x oversampling you can increase the SNR by 6
dB or 1-Bit and that lowers the quantization noise floor.
However, say my ADC is sa...
Assume i receive a signal and perform direct conversion from passband
to baseband with a front end analog filter with bandwidth +/- B for my
signal. In my ADC I sample at 2B and have a signal plus noise limited
to the band +/-B with each noise sample assumed to be uncorrelated.
If i instead dec...
On Mon, 15 Apr 2013 06:22:36 -0500, SRB wrote:
> > Hi All
> > My question relates to the effect of oversampling on the SFDR obtained
> > an ADC. I understand that the SNR of an ADC, considered over a certain
> > bandwidth of interest (BW), can be increased relative to the overal...
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
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/...
Plz help me for ofdm simulation. I have tried to simulate ofdm signal with
64 subcarriers, with an oversampling of factor 4 in frequency domain. Now
I've got 256 QPSK symbols with zero paddings.I'm doing IFFT of 256 points.
But I need to modulate my data on 64 subcarriers. then how would...
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
> Now you have an AM signal....
> 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.
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...
Scott Gravenhorst wrote:
> The technique as I am currently applying it does not completely
> eliminate the effect. It becomes audible around 2000 Hz
> fundamental. Below 2000 Hz, it's not audible (to me). I may be
> able to extend that by adjusting the filter.
At fundamental frequency...
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
> > 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...
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...
> 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...
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
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
> 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
> > ...
In every book and article aobut ADSL processing the A/D and D/A sample
rates are given as 2.208 MHz. For a transmitter implementation at this
sample rate you would get the conjugate symmetric image spewing out of your
D/A since one may use up to bin 255 and there is absolutely no