Non-ovesampling DACs vs. Oversampling DACs for audio - your feedback requested

Started by Larry McFarren in comp.dsp15 years ago 3 replies

I came across a link today to an article about the merits of a non-OS DAC over an OS DAC. Of course, I'm referring to the filter part of the...

I came across a link today to an article about the merits of a non-OS DAC over an OS DAC. Of course, I'm referring to the filter part of the DAC. Anyway, this article doesn't make much sense to me: http://www.sakurasystems.com/articles/Kusunoki.html What is your take on this? Is the author correct when he says that 16-bit accuracy is not possible if you oversample? I don't believe th...


Oversampling and receiver

Started by Awal in comp.dsp10 years ago 9 replies

Hi all, I would like to know if the folling would be possible. Lets say that my tx use a raised cosine filter to limit the bandwidth of my...

Hi all, I would like to know if the folling would be possible. Lets say that my tx use a raised cosine filter to limit the bandwidth of my transmitted signal. At the receiver I would oversample the received signal by a factor of x and by using a Viterbi algorithm I would be able to approximate every symbol affected by ISI caused by the memory of my raised cosine filter. By doing this would i...


Tx and Rx filters gains and energies

Started by commsignal in comp.dsp6 years ago 5 replies

Hi All, Although it is a basic question but I couldn't find its answer in any available reference. When we filter the signal at the Tx and Rx,...

Hi All, Although it is a basic question but I couldn't find its answer in any available reference. When we filter the signal at the Tx and Rx, how should we adjust the filter gains or filter energies to keep the overall energy figures correct *at every stage*. For example, should Tx filter gain be necessarily set to the oversampling factor, or what should be the Rx filter gain. Thanks. _...


Adding noise to ADC input to increase accuracy

Started by dwjbosman in comp.dsp8 years ago 2 replies

Dear all, I have just been reading about the use of oversampling in a decimator to enhance the resolution of a sampled signal. The idea is...

Dear all, I have just been reading about the use of oversampling in a decimator to enhance the resolution of a sampled signal. The idea is to sample a signal containing 0.5LSB white noise at samplerate of N times the Nyquist rate. It is then possible to extract log4(N) extra bits of resolution by summing N samples. However there is another scheme which can provide much more of an increase...


dsPic adc for audio signals

Started by Anonymous in comp.dsp13 years ago 5 replies

Hi. With a dsPic (ie dsPic30F4013) I need to process two voice band channels (for telephony, fs=8kHz). Can there be problems to use the...

Hi. With a dsPic (ie dsPic30F4013) I need to process two voice band channels (for telephony, fs=8kHz). Can there be problems to use the embedded adc, oversampling 8 times and then filtering? Do you think this may waste more than one half of dsp bandwidth? Why is not usual to use the embedded adc for audio signals? Thank you. pp


Carrier Recovery for QPSK

Started by Ali A Nasir in comp.dsp10 years ago 1 reply

Hi everyone, As a part of problem, I have to implement carrier recovery in some distributed MISO communication system. I was following one...

Hi everyone, As a part of problem, I have to implement carrier recovery in some distributed MISO communication system. I was following one paper to implement that problem which targeted to estimate the f offset in the range of (1 / (2Ts) ) ( for Ts is the oversampling time ). The f offset is estimated with minute error about which author suggested the use of PLL to handle it, but didn't r...


physical layer A/D performance simulation

Started by Laron in comp.dsp9 years ago 1 reply

Hi, I want to simulate the physical layer for CDMA IS-95, i've wrote the bs_signal and the channel model, the rrc filter in the BS site(tx)...

Hi, I want to simulate the physical layer for CDMA IS-95, i've wrote the bs_signal and the channel model, the rrc filter in the BS site(tx) use 8x oversampling with baseband model. The question is how to simulate the receiver-side's frequency error induced during downconverter and the A/D timing issue during A/D? also the digital AGC performance? B.R. Thanks.


Increase resolution through decimation

Started by mariking in comp.dsp12 years ago 3 replies

Hi I need to take a standard 8bit ADC which samples at 40Msps increase the resolution by about 4 or 5 bits. I know that oversampling spreads...

Hi I need to take a standard 8bit ADC which samples at 40Msps increase the resolution by about 4 or 5 bits. I know that oversampling spreads the quantization noise over a wider bandwidth thereby decreasing the SQNR. So i can increase the resolution through decimation? How do I increase the resolution through decimation? It decimates the signal frequency but i don't physically know how to increa...


High pass polyphase filter

Started by Piergiorgio Sartor in comp.dsp10 years ago 4 replies

Hi all, I'm trying to design an high pass polyphase filter. The approach is to take the 5 tap high pass FIR, oversample it by a factor N...

Hi all, I'm trying to design an high pass polyphase filter. The approach is to take the 5 tap high pass FIR, oversample it by a factor N (usually 128 or 256), then subsample it again into the different phases. The oversampling was done by polynomial interpolation and by zero-padding and low-pass. I mean, the two methods were tried, not altogether. In both cases, the results are dis...


symbol synchronization method for SDR in Matlab?

Started by algora in comp.dsp8 years ago 1 reply

hello, any of you has experience with different methods for symbol synchronization so you can help me to implement one of them in a receiver for...

hello, any of you has experience with different methods for symbol synchronization so you can help me to implement one of them in a receiver for bpsk modulation? i've implemented a costas loop for carrier recovery and the code i've found for symbol synchronization is too heavy and takes too much time. this is the matlab code: tnow=del*oversampling+1; tau=0; xs=zeros(1,length(y)); % i...


Timing recovery by oversampling in QAM systems

Started by Calabi_yau in comp.dsp12 years ago 1 reply

Hi, A basic question about QAM. I've noticed that there are timing recovery systems that calculate the 'correct' sample by means of an...

Hi, A basic question about QAM. I've noticed that there are timing recovery systems that calculate the 'correct' sample by means of an interpolation guessing wich one of this intermediate samples is the most suitable instead of closing a loop and recovering and using the recovered clock as a receiver clock.. It seems to me quite interesting but I have a doubt: If we don't synchronize the...


FFT Size calculation.

Started by Anonymous in comp.dsp12 years ago 8 replies

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...

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 3/2. Then per FFT we have 96 samples. And further assume that I know I have 96 samples per FFT (but I...


Query on AIC-23 Codec

Started by Anonymous in comp.dsp13 years ago 1 reply

Hi All, i am going through the datasheet of AIC-23 Codec on TI board, in the datasheet i have come acrros this statement BOSR...

Hi All, 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 = 384 fs where 0 and 1 are the bit to be set in the BOSR Register. My Query is, what actually mean by t...


Design of anti-alias filter for ADC, with oversampling and averaging

Started by Roy in comp.dsp10 years ago 1 reply

Hi, I am trying to design proper anti-aliasing filters for a new hobby project (digital control of a quad-rotor flying robot - see...

Hi, I am trying to design proper anti-aliasing filters for a new hobby project (digital control of a quad-rotor flying robot - see previous post for more questions). Here are the specifics: * My sensors are analog MEMS accelerometers and gyroscopes. I'll have at least 6 sensors to process, at least initially. * I will be using a microcontroller with 10-12 bit built-in ADCs to process t...


Finding maximum of sinc in 0..1

Started by jungledmnc in comp.dsp2 years ago 11 replies

Hi folks, the audio world is sort of obsessed with sort of "true peak level". Unfortunately it is rather tricky to calculate, so it is...

Hi folks, the audio world is sort of obsessed with sort of "true peak level". Unfortunately it is rather tricky to calculate, so it is calculated by oversampling to about 192kHz and taking the normal peak level. Sadly that's really not true peak level. So how about the (nearly) correct value - if we take say 2 * N + 1 samples around each sample S[0] (hence S[-N] .. S[N]), we can calculate ...


Strange fm component in the ECG

Started by paramonte in comp.dsp12 years ago 5 replies

Dear all, We are acquiring the ECG with a high resolution system, sampling frequency 2400Hz. The system is an oversampling sigma delta, 24...

Dear all, We are acquiring the ECG with a high resolution system, sampling frequency 2400Hz. The system is an oversampling sigma delta, 24 bits. In some ECG´s (from normal subjects and patients) we get, added to the ECG, a frequency modulated version of the ECG itself, e.g. a sinus wave with constant amplitude and whose instant frequency is proportional to the ECG signal value in that partic...


Oversampling using Cheby2 IIR FIlter.

Started by Emiliano in comp.dsp13 years ago 13 replies

I have to use a cheby2 IIR filter with 60db bandstop ripple to implement oversamplig procedure. I'm sure that I can calculate the coefficient...

I have to use a cheby2 IIR filter with 60db bandstop ripple to implement oversamplig procedure. I'm sure that I can calculate the coefficient with matlab in few minutes, but in my application I have to calcute it in real time depending on the cut-off frequency. My question is: thers is an analitic rappresentation of the transfer function of cheby2 filter dapending on the frequency in lapl...


How to detect signal from microwave sensor?

Started by gpezzella in comp.dsp9 years ago 4 replies

Dear Friends My application should detect signal that come from MicroWave cavity (the one used in house alarm). When people walk in front a...

Dear Friends My application should detect signal that come from MicroWave cavity (the one used in house alarm). When people walk in front a MW, a little signal of few mV and frequency between 10-50Hz is generated (beat signal). I should detect this! I'm developing on ATTiny micro family. Now: Since AtTiny have 20x amplifier buit in, which is better: a) add 3 new bit by Oversampling a...


OFDM and oversampling

Started by 6.20 in comp.dsp13 years ago 8 replies

Hi ! I would like to know if in practice, it is useful to oversample the signal at IDFT output. The sampling rate at the IDFT output is the...

Hi ! I would like to know if in practice, it is useful to oversample the signal at IDFT output. The sampling rate at the IDFT output is the Nyquist rate (2 times the frequency of the baseband signal) but I believe that it is necessary to oversample in order to perform the Root Raised Cosine Filtering. If I am right, what is the method to oversample : - Interpolation (using an IDFT size 2 times ...


oversampling

Started by CW in comp.dsp10 years ago 1 reply

Assume i receive a signal and perform direct conversion from passband to baseband with a front end analog filter with bandwidth +/- B for...

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 decide to sample the signal at the output of the analog filter at 10B i will have correlated noise on ...