9

If you play 16-bit audio at 48kHz, you need the DAC analog reconstruction filter to pass 20kHz and attenuate 96dB at 24kHz, which is quite steep and requires complex multistage analog filter. The advantage of using oversampling is moving the sampling rate much higher, for example oversampling by 4x means the DAC runs at 192kHz, and the analog filter only ...


3

But for D/A conversion, normally there is no quantization Well, yes there is. In general, because DACs have a certain resolution (8-bit, 10-bit, etc.). But specifically to your question, in a sigma-delta modulator, there's a lot of quantization -- a sigma-delta modulator is based on the notion of a "one-bit" DAC whose output is either $v_{max}$ ...


2

Well, you cannot completely avoid images, you can only suppress them sufficiently. How much "sufficiently" is is completely up to your application! So, we can't tell you this. However, thanks to the fact that you're comfortably oversampling your 400 kHz, the first aliases would appear at 2.93 MHz (=lowest frequency in signal, i.e. -400 kHz, plus ...


2

We're still a couple steps behind on emulating good tube distortion and even some other effects you find in some stomp boxes (although a lotta stomps have digital innards). Don't you have multitrack recording? Why not record both the clean guitar as well as the great-sounding guitar coming outa the Mesa Boogie?


2

It is not a mistake. In section 4.8.4 the authors introduce the reconstruction of $x_r(t)$ from $x[n]$ in equation $(4.140)$; and in the paragraph below, the authors consider the input sequence and output signal to the ideal D/C converter as $x[n]$ and $x_r(t)$ respectively, as shown below $$x[n]{\longrightarrow}\boxed{\textit{Ideal D/C converter}}{\...


1

I don't think it's a typo, but it has to do with the sampling process. As shown in Eq. $(4.30)$ in the 3rd edition of Discrete-time Signal Processing by Oppenheim and Schafer, (ideal) sampling involves scaling by $1/T$, where $T$ is the sampling period. If the continuous-time signal $x_c(t)$ is ideally band-limited, the spectrum of the discrete-time signal $...


1

Sampling Time offset is the disparity is the starting point of sampling, ie , sampling did not start at point "0". Sampling frequency offset (SFO) is the error in the practical sampling frequency from what is required in paper. It is mainly due to practical limitations in generating perfect sampling pulses. Carrier frequency offset (CFO) is the ...


1

It depends and typically there are more important aspects to consider: Make sure your effects are in the right order. I'm guessing you do NOT want a reverb before your distortion pedal regardless of whether any of these is analog or digital. Some effects must be part of the recording and cannot easily be applied through post-processing. For example if you ...


1

Sinc interpolation can exactly reconstruct an above-Nyquist-sampled strictly bandlimited signal from noiseless samples. See the Whittaker-Kotelnikov-Shannon reconstruction or resampling theorem: https://en.wikipedia.org/wiki/Whittaker–Shannon_interpolation_formula and https://ccrma.stanford.edu/~jos/resample/Theory_Ideal_Bandlimited_Interpolation.html For ...


1

To implement a digital-to-analog converter, all you need is an ideal low-pass filter to filter out the periodic frequency response that $\varOmega \geq \varOmega_s/2$. $$ H(j\varOmega) =\left\{ \begin{aligned} T,\ \ \ \ |\varOmega|<\varOmega_s/2 \\ 0,\ \ \ \ |\varOmega|\geq\varOmega_s/2 \end{aligned} \right. $$ The impulse response of an ideal low-pass ...


1

The OP is trying to implement a notch filter with a 3 tap FIR filter (as restricted by the problem he is trying to solve). It must be noted that this would result in a very poor filter implementation: It will reject 60 Hz perfectly but will have unavoidable attenuation over a very broad section of the desired passband range. An actual practical ...


1

First, the wiktionary definition of "interpolate", definitions 1 & 2, is spot-on here. Interpolation just means to fill in the blanks, correctly or not. What is meant by "order" here in interpolation? Is it referring to derivative?? Dan's covered this -- it's the order of the polynomial you need to affect the reconstruction. I.e. ...


1

I'd just like to understand if this is a rabbit hole or the elusive unobtainium. A little bit of both. Ethernet is really NOT a great choice for real time audio. You would need a fairly beefy processor that can handle a network stack (TCP or UDP). Getting the latency down to something that's playable is extremely challenging. Even if you get it on Ethernet, ...


1

i would suggest that there is a definite advantage to 1-bit DACs that have a good balanced output (that is if one terminal is high the other must be low) going into an analog difference amp, that the 1-bit DAC suffers no non-linearity distortion. with a 1-bit, two-level output DAC, there are no intermediate steps in the DAC input/output relationship. there ...


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