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Comment Re:The bit depth does matter (Score 0) 841

That is ridiculous to the nth degree. There is no mathematical basis for what you say. When you sample, all below nyquist is reproduced 100%. Peaks and valleys do not need to line up. To the extent that an analog source has frequencies above nyquist they will simply alias, and one can easily predict whether they will be audible. Your "phase" concept has no signal processing basis.

Regarding bit depth:
Bit depth provides dynamic range. 16 bit means you can hear your home theatre hiss during quiet passages when it is really cranked. With 20 bit you can't. With 18 bit you probably won't. But 16 bit is NOT ENOUGH.

Regarding sample rate:
Reproduction at a higher sample rate means simpler filters in the DAC. That has some importance in spite of what the article incorrectly states about ease of digital filter design.

Comment Re:Why the hell is audio linearly quantized? (Score 1) 841

Any non linear code will create distortion products, both harmonic and intermodulation. Don't know what sigma delta has to do with it. Conversion to non linear codes would be a totally separate process from filtering/decimation of sigma-delta sequences.

Image a loud and soft tone of widely different frequencies. With linear coding you can separate the two tones by a filter. With non linear coding the loud tone will interfere with the coding of the soft tone such that after filtering intermodulation products will exist. (imaging the loud tone going on/off and its effect on soft tone quantization)

Only linear codes can re-create complex waveforms without intermodulation and harmonic distortion.

The closest you can come is "long term" volume compression/expansion such as that used by dolby. In that case you can limit the dynamic range before coding and try to expand it back afterwards, but the data samples must be linear coding. Even this has side effects though.

Comment Article is So Wrong InSo So Many Ways (Score 0) 841

Article is full of crap that is just plain wrong and misguided and the analogies suck. I'm not sayin 192KHz 24 bit is needed but the article is weird and says things that just are not true.

For example, saying that a stairstepped sine wave is mathematically the same is wrong --- stairsteps are impulses convolved with a square wave "impulse". This creates a roll-off at high frequencies. Basic signal processing. If you don't understand it, don't worry. Many don't. But the resulting sine wave will be the wrong amplitude. Sampling theory is based on infinitely fast impulses at each sample point, not stairsteps. A subtle point, but he misses many subtle points.

As for 192K 24Bit, there are reasons it is useful as opposed to 48Khz or especially 44.1KHz 16Bit:
1. Dynamic range. 20 bits gives 120dB +_ a few. But 16 bits (96dB) is not enough. 24 bits is way overkill, but doesn't hurt anything except storage space. Home theatre systems with 16 bits make audible noise when you turn them up. Put you ear next to the speaker when it is quiet and you will hear hiss. It may hurt your ears if they are there when when some sound comes through, but that depth is audible. His assertion that 16 bits is enough is not science, it's his opinion. (maybe even 18 bits is enough, but 18 is on the edge)
2. Simplicity of DAC - 192KHz means that dac filters can easily remove images. 96Khz is high enough to make the filter job simple, but 192Khz is simpler yet. His assertion that doing steep filters in digital is no issue means he doesn't really understand digital filters. Steeper slopes means higher lobes and more passband ripple.
3. All his talk about ultra sonics is laughable. Design a bad amp and it will sound bad. So what? Oh --- put in a bad signal so it will sound better?
4. My only point of full agreement is that you need good equipment first, 192/24 second. And I partially agree in that 192/24 is overkill.

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