Also a good starting place for beginners are the xiph show-and-tell videos (probably been posted here before, but whatever):
https://xiph.org/video/vid2.shtml E On Wed, Jun 3, 2015 at 3:05 PM, Ethan Duni <ethan.d...@gmail.com> wrote: > "Perfect" sinusoids/square waves/etc. only exist as mathematical > abstractions. A good starting point would be to get a feel for what, say, > the "square wave" coming out of an analog synthesizer actually looks like - > the noise floor, the distribution of harmonics, frequency jitter, > under/overshoot, etc. I think you'll find that modern digital equipment > generally can produce much, much closer approximations to these ideals than > we ever see from analogue synthesizers. > > But more generally the answers to your questions depend on the specifics > of the A/D/As in question. There are many different designs with different > oversampling and interpolation stages, different noise shaping strategies, > etc. So the question of where the quantization noise ends up doesn't have a > general answer - it depends in on both the design of the A/D/A components, > and on the specifics of the signal in question (these are nonlinear, > time-variant operations we're talking about). > > Likewise, the effect of applying a half-sample delay depends heavily on > how you do that. If you're talking about signals that are simply computed > directly or delay in the analog domain, then you'll simply see a > half-sample delay at the output. But if you're talking about sampling a > signal and applying a half-sample delay, then it depends on what technique > you use to implement that delay. > > Also, note that the function a*exp(b*x+c) is not, by itself, a > well-defined test signal, since it has infinite energy. The sampling > theorem does not apply to such signals in the first place. You would need > to apply some kind of window/truncation to ensure finite energy and > bandlimiting. > > More generally, these issues are all quite well studied and understood, > and not particularly relevant to music-dsp as such. So I'm not sure what is > the point of repeatedly bringing them up in this context. It sounds like > what you want is more like a book or class on mixed-signal circuits. The > main reference I use for that is "Analog Integrated Circuit Design" by > Johns and Martin. It's a bit dated by now, so it's not going to cover the > latest-greatest converter designs (not sure that any book really does), but > the chapters on the noise modeling, theory of oversampled conversion, etc. > are all still valid for learning the theoretical underpinnings of this > stuff. > > E > > On Wed, Jun 3, 2015 at 1:47 PM, Theo Verelst <theo...@theover.org> wrote: > >> Hi, >> >> Playing with analog and digital processing, I came to the conclusion I'd >> like to contemplate about certain digital signal processing considerations, >> I'm sure have been in the minds of pioneering people quite a while ago, >> concerning let's say how accurate theoretically and practically all kinds >> of basic DSP subjects really are. >> >> For instance, I care about what happens with a perfect sine wave getting >> either digitized or mathematically and with an accurate computer program >> put into a sequence of "signal samples". When a close to perfect sample (in >> the sense of a list of signal samples) gets played over a Digital to Analog >> Converter, how perfect is the analog signal getting out of there? And if it >> isn't all perfect, where are the errors? >> >> As a very crude thinking example, suppose a square wave oscillator like >> in a synthesizer or an electronic circuit test generator is creating a near >> perfect square wave, and it is also digitized or an attempt is made in >> software to somehow turn the two voltages of the square wave into samples. >> >> Maybe a more reasonable idea is to take into account what a DAC will do >> with the signal represented in the samples that are taken as music, speech, >> a musical instrument's tones, or sound effects. For instance, what does the >> digital reconstruction window and the build in "oversampling" make of a >> exponential curve (like the part of an envelope could easily be) with it's >> given (usually FIR) filter length. >> >> In that context, you could wonder what happens if we shift a given >> exponential signal (or signal component) by "half" a sample ? Add to the >> consideration that a function a*exp(b*x+c) defines a unique function for >> each a,b and c. >> >> Anyone here think and/or work on these kinds of subjects, I'd like to >> hear. (I think it's an interesting subject, so I'm serious about it) >> >> T. Verelst >> >> -- >> dupswapdrop -- the music-dsp mailing list and website: >> subscription info, FAQ, source code archive, list archive, book reviews, >> dsp links >> http://music.columbia.edu/cmc/music-dsp >> http://music.columbia.edu/mailman/listinfo/music-dsp >> > > -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
