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Talk:Videos/Digital Show and Tell

5,541 bytes added, 09:59, 25 August 2013
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You talk about discrete values (whether the analog sample points, or the infinitesimal image pixels). BUT, these are in some way, averages. In a digital camera, the pixel value is the integral across about 90% of the pixel-pitch. In analog audio, is it an instantaneous sample, or an average over the preceding sample-interval, or is it sometimes even more "blurred" than that? Also, when performing DAC, how do we get rid of the stairstep so perfectly without distortion? --[[User:RichardNeill|RichardNeill]] 07:51, 4 March 2013 (PST)
: The pixel values in a camera are area averages exactly as you say, a necessary compromise in order to have enough light to work with. The sensor sits behind an optical lowpass filter that is intentionally blurring the image to prevent much of the aliasing distortion (Moiré) that would otherwise occur. Despite that, cameras still alias a bit, and if you _remove_ that anti-aliasing filter, you get much more (I have such a camera, danged filter was bonded to the hot filter, so both had to go to photograph hydrogen alpha lines).
: Audio does in fact use as close to an instantaneous sample as possible. The 'stairsteps' of a zero-order hold are quite regular in the frequency domain; they're folded mirror images of the original spectrum extending to infinity. All the anti-imaging filter has to do is cut off everything above the original channel bandwidth, and it doesn't even have to do a great job to beat a human ear :-) --[[User:Xiphmont|Xiphmont]] 02:56, 12 March 2013 (PDT)
What's the correct way to plot a reconstructed waveform? If I have an array of samples and play them back through a DAC, the oscilloscope shows a smooth curve. But plotting them with eg matplotlib shows a stairstep. Thanks --[[User:RichardNeill|RichardNeill]] 07:58, 4 March 2013 (PST)
: well, a fully reconstructed waveform is equal to the original input; it's a smooth continuous waveform. OTOH, if you want to plot an actual zero-order hold, a zero order hold really is a staircase waveform.
: If you want to plot the digital waveform pre-reconstruction, a mathemetician would always use lollipops, an engineer will use whatever's the most convenient. --[[User:Xiphmont|Xiphmont]] 02:56, 12 March 2013 (PDT)
I have a strange issue with the gtk-bounce program - on (k)ubuntu 12.10, spectrum and waveform work just fine, but if I scroll into the gtk-bounce panel, the cursor disappears. Anyone seen that behaviour? - Julf
: Edit out the calls to "hide_mouse()" in gtk-bounce-widget.c. It's hiding the mouse on purpose because it's supposedly a touch application :-) --[[User:Xiphmont|Xiphmont]] 02:56, 12 March 2013 (PDT)
One issue I'm not sure you have covered, relates content with short, loud sections (e.g. more like a movie or maybe classical music, less like pop music).
Is this a real problem?? Is this ever an audible?
[[User:Klodj|Klodj]] 21:59, 11 March 2013 (PDT)
: the bit depth doesn't affect the correctness or 'fineness' of the reconstruction, it only changes the noise floor. It is the same and sounds the same as what happens when recording quieter-than-full-range on analogue tape. Compare a 16-bit digital signal recorded at -20dBFS to the same signal recorded on tape at -20dBV. Both will be smooth and analogue, but the tape will be noisier ;-) --[[User:Xiphmont|Xiphmont]] 02:56, 12 March 2013 (PDT)
:: Thank you for your reply! Ok, I understand that, when comparing digital to "analog" (tape), this is a non-issue. The tape noise floor is higher. But could you clarify a related point that is entirely in the digital domain? Lets say we have a 16bit recording of the 1812 Overture where the canons don't clip. The average level is going to depend on how dynamic range is compressed to handle the canons, but lets say it -36dBFS. If I adjust volume to suit the average level, then won't I effectively be hearing a noise floor equivalent to 10 bit quantization (16 bits - 36dB/6db_per_bit) for the majority of the recording (dither aside). --[[User:Klodj|Klodj]] 19:13, 13 March 2013 (PDT)
::: Yes. --[[User:Xiphmont|Xiphmont]] 00:54, 14 March 2013 (PDT)
I've really enjoyed your videos, many thanks for your work producing them. In keeping with your message in the video about breaking things I was curious to see what would happen as you pushed the signal up to and beyond the Nyquist frequency. Would the filters mean that it would simply fade out smoothly (what order filter is employed?)? I suppose I could check this out myself, but would be interested to hear your answer. Also is the Gibbs phenomenon audible to any degree? --[[User:Stuarticus|Stuarticus]] 14:00, 11 June 2013 (PDT)
: If the filter is smooth, it will fade smoothly. If the filter is sharp, it will drop off suddenly. Some DACs put the transition band of the anti-imaging filters straddling the Nyquist frequency or slightly past, so you might even see the signal fold back about Nyquist as it drops off (this used to be more common about 10 years ago). So long as no significant aliasing reaches back under 20kHz, this is just one of many arbitrary design decisions that don't really affect the audio quality. The order of the digital filter used in the upsampling stage can be nearly anything, but they're not likely to be as huge as many software resampling filters, where a linear-phase FIR of 512 or 1024 taps is common. The analog filter stages, if they're there at all, are unlikely to be more than a handful of taps. Detailed spec sheets should say outright, and if they don't they should at least usually mention the approximate slope. --[[User:Xiphmont|Xiphmont]] 03:45, 15 June 2013 (UTC)
I just watch the video and has one question. During demonstration which Monty has feed generator at various frequency and convert to digital at 44.1KHz and convert back to analog and show the analog result at second oscilloscope which is the good way to show the whole digital way of encode and decode. At 1KHz sine wave input, when sampling at 44.1KHz, there will be 44.1 sampling per sine wave or 22 sampling per half sine wave which is not too bad to represent the sine wave and the output from digital-to-analog should still be quite close to original sine wave. However, at input of 20KHz, there will be only 2.2 sampling per sine wave or just 1 sampling per half wave and it merely enough to represent sine wave. My question is if there is just one sampling per half wave, how can the output from digital-to-analog is still be very good sine wave. If 20KHz is sampling at 192KHz, there will be 9.6 sampling per sine wave or 4.8 sampling per half wave which is far from perfect but still better than one. Does this mean that we should have better output if we increasing the sampling from 44.1KHz to 192KHz? --[[User:Somchaisis|Somchaisis]] 05:37, 25 August 2013 (UTC)

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