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Waveform horror! Is a 44.1 kHz sampling rate sufficient? 

I used Audacity to generate a 19 kHz tone, planning on burning it to a

> CD and playing it in my car stereo for... um... well, let's not get > into that. I then zoomed in on the waveform to take a peek, prior to > burning the waveform. What horror! >=20 >
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>=20 > The waveform is severely distorted. I could see where the points > mathematically > would be correct, but given the sparse sampling points, the speaker > would not be instructed to swing rail-to-rail at, say, between 15.0000 > and 15.0005 seconds. >=20 > A FFT gives a spread of frequencies centered around the 19 kHz, but, > yuck! >=20 > I tried various other frequencies: 9 kHz gives a pretty distorted > waveform as well. >=20 > What are the implications as far as accurate sound reproduction at a > 44.1 kHz > sampling frequency, as used by CDs? >=20 > Michael

No sir, not if you want to sample the behavior of the republican = c*ck-suckers in here, they will never admit being a million innocent = killers, they kept calling their opponent communist when in fact they = are the real communist by not allowing others to speak or let them do = their job. Every day all you hear is bashing from them. So you need = higher sampling rate in order to keep up with these shitty jerks.

Reply to
Speeders & Drunk Drivers are M
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20X sampling would be far better. Not however, the ear tends to "smooth out" HF garbage.._listen_ to the result and if not acceptable then change the sampling rate until it sounds good and do not look at it..
Reply to
Robert Baer

19 kHz ?? Can you hear the harmonics ? If so, then you may have some nonlinearities to find too. boB
Reply to
boB

If it is really is a 19kHz tone few can hear it. The second harmonic would be 38kHz and third, 57 kHz. Nobody hears these components so the waveform is immaterial. In otherwords the audio channel plus the ear filters out the higher harmonics. Some people may claim they can hear the difference but likely they would be keying off level differences or noise and not on the actual waveform purity.

You mentioned an FFT centered around 19 kHz? A periodic waveform no matter what it's shape should have components at the fundamental frequency, 19 kHz and above, 38 kHz, etc. There should be no components centered around the fundamental. If there are, it means the waveform is being modulated by something else, it has intermodulation distortion. If this spray of modulation products gets down into the audio band, say 10kHz or lower, the signal will definitely be audible. Furthermore the spray of non-harmonic garbage will make a tone sound bad or weird.

The 44.1 kHz sampling rate is, in effect a modulating, signal on the audio input. Modulation creates sum and difference frequencies and their multiples. 44.1 was picked so that no sum or difference products or their multiples fall into the audio band. With over sampling and proper filtering these products called aliasing should not appear in the audio stream.

Reply to
Bob Eld

The strange sampling rate has to do with the early use of VCRs as a digital recorder. The PCM adapters packed three samples from the left and three samples of the right channel into a single video line and added perfect vertical and horizontal waveforms, so that it could be recorded on any standard video recorder.

Since usable data could not be recorded on the invisible field synch lines, some buffering was needed to get the audio data into visible lines only. Thus the sample rate is not three times the horizontal line frequency, but slightly below that.

Paul

Reply to
Paul Keinanen

The problem is that Audacity seems to use first order interpolation ("connect the dots with lines") to display some waveform, while in the real world, the values are only defined at the sample points (dots). Feeding a pulse train consisting of the actual sample values at the sample points into an analog or digital 20 kHz low pass filter will produce quite different results than the first order interpolation used by the Audacity display.

Only with the audio signal which has simple harmonic relation to the sampling frequency, you will get FFT display with all energy concentrated into a single FFT bin.

Paul

Reply to
Paul Keinanen

As an experiment, I just genned a 19 KHz tone with CoolEdit 96, and it draws the waveform "splined" correctly.

-- Les Cargill

Reply to
Les Cargill

But that's the most interesting part of this thread!

--
Nemo
Reply to
Nemo

This shows what a difference a display algorithm makes. If you zoom in so only a few dozen sample points are displayed, Cool Edit correctly brick-wall filters the waveform at 1/2 the sampling rate, and uses the resulting values (in this case a 19kHz sine wave) to lterally "connect the dots" between the sample points. It looks like a drawing right out of a DSP text.

The 19k tone (at 44.1k sampling rate) I just generated in Cool Edit

2000 looks as I describe above. I saved it as .wav and loaded it in Audacity, where it indeed looks very ugly - the waveform is drawn as a straight line between sample points.

Pick some "good" windowing for the FFT and it should be pretty close to a single bin.

And just for fun, here's someone who doesn't have a clue what he's talking about:

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Reply to
Ben Bradley

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