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A good idea for a thread, if a little dangerous!
Yep, probably best to quarantine it now.
Here is the next set of topics regarding myths. These deal mostly with digital vs. analog: https://benchmarkmedia.com/blogs/application_notes/digital-audio-demonstration-video
His discussion of the stepless output is based on pure sampling theory, where each sample is a zero-width Dirac delta function (his 'lollipop diagram').That's correct in theory, but in the real world, a DAC does implement the zero-order hold he mentions, and the output of the DAC chip DOES have steps; it does not produce zero-width delta function outputs. A DAC chip is therefore followed by a Nyquist reconstruction filter, to remove the steps.A real-world ADC will also generate a sampling (or quantisation) error, as, unless you have infinite bits, there will be an error between the real sample, and the nearest ADC quantising value. His lollipops are assumed to be perfect samples, with zero quantisation error.Sampling theory assumes perfect, 'brick-wall' Nyquist filters. In the real world, these do not exist. Real filters have problems like roll-off rates, and ripple in passband amplitude & phase....It would have been illuminating for him to have changed the precision of his samples, from the 16 bits he used, to 8 bits, or even 4 bits. Using 16 bits, the quantisation error will be below the noise floor of the analyser he was using. It's not a good idea to try to claim an effect doesn't exist because you can't measure it....I started my career working on the development of the GSM standard, and the first network and handsets. In particular, the frequency synthesis and modulation. We used a technique called Digiphase, a type of fractional-N synthesiser. It did direct digital modulation by constantly changing the synthesiser frequency. It used a third-order interpolator, combined with digital predistortion to meet the modulation and spectral mask requirements. Essentially, a noise-shaping DAC.
The video has some problems. First, a scope won't reveal if the sine wave is "exactly" and/or a "perfect" sine wave. For instance, if the fundamental frequency is altered slightly, the scope will never reveal it. Even an amp with 5% or more harmonic distortion will be quite difficult to see on a scope. Now consider an entire orchestra, singing. Evidently, we must believe what he says.If other frequencies are introduced, the scope will never reveal it. What do I mean by that? The distortion analyzer shows nothing but the fundamental frequency and harmonics. If the fundamental frequency is slightly altered, it will not be seen.When dealing with bit depth, "stair steps" (16 bit, 65,536 values, 24 bit, we have 16,777,216 values), very rarely is the analog signal going to be exactly on a "value"/"step" during the sample period. The signal will be in between, so which value is chosen, an upper or lower value? Whichever value is chosen, the slope/rise time is altered between samples by definition, thus the fundamental frequency is also slightly altered between samples, said instrument won't reveal to us.We won't have to worry about the slope exceeding 20khz unless the harmonic is quite high in frequency. Eight bit alters the slope/rise time even more, plus 8 bit lacks dynamics, inner detail even more than 16 bit. The inference that 8 bit, even 16 bit is enough for high quality music is the opposite of what Philips engineers believed, but RCA marketed 16 bit players anyway, and the rest is history.Notice the Gibb's effect. It is within 20khz which is to be expected, Notice its amplitude value is high compared to the rectangular wave. Any Intermodulation distortion in the system, whether it be from speakers, electrical components, will cause mixing with this ringing, and cause non musical tones in the audio band.It takes two samples to recreate a sine wave. At 10khz, there are only 4 samples per cycle, 5khz only 8 samples. However, music is not a simple sine wave nor a rectangular wave with equal repetitive waveforms. Music is complex with all sorts of phase relationships between instruments and their waveforms. Think it can reproduce the music perfectly when parts tolerances enter the picture and values are altered?
The elephant in the room is our own minds...
Thought it would be fun to start a thread about audio myths. There are many out there, but let's start with this one:https://benchmarkmedia.com/blogs/application_notes/152143111-audio-myth-switching-power-supplies-are-noisyI've noticed some of the Class D amps have large transformer power supplies, as well as the vast majority of Class A and Class A/B amps. According to this white paper, that is a myth for the reasons outlined. Thoughts?
I have always thought all transformers color the sound. But some of the newer SET tube amps do not seem to be colored. Is that because of the size of the transformer?David Berning and other companies quit making transformer driven tube amps many years ago because of this.
I think this link, presentation, has been addressed in depth by some on another forum, and has some fundamental problems we tend to overlook. I know I did. I would like to post portions of their comments from this link if I may.https://audiokarma.org/forums/index.php?threads/digital-technology-collection.64669/
