pinnahertz
Headphoneus Supremus
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I seen no point in developing your own word definitions. "Broken" does not mean "design deficiency".
Broken
Broken
...the original design is flawed ... that includes millions, possibly most of the DACs out in the field. So your definition of "broken" is also the definition of the norm. We need to be aware of the average and norm. That means, "broken" or not, we must work with it if we don't want our audio clipped at the DAC..
Disagree. Any properly designed DAC will properly manage all valid Nyquist information (information that is unclipped in reconstructable Nyquist space). Yes, that's a tautology, but it also happens to be true. Proper DAC design is not flawed. Garbage in, garbage out. In other words, if we have digital program that shows no clipping in a properly designed (reconstructive) digital meter, then a properly designed DAC will deliver a non-clipped analog reconstruction. I believe there may be a caveat here for "synthetic data" created to purposely confuse an anti-imaging filter, but still researching that.
I seen no point in developing your own word definitions. "Broken" does not mean "design deficiency".
[1] A properly designed digital mixing engine, or any digital processing system, will maintain unclipped signal integrity unless the operator pushes the program level beyond full-scale. In a properly designed digital metering system, all peaks are identified. In a poorly designed digital metering system, some peaks are missed, which is missed inter-sample information. Poorly designed tools, nothing more.
[2] We also know that most mixing engines are based on a 32-bit IEEE float topology, which assures that all 24 audio audio bits are maintained perfectly (the actual mixing algorithm is a different story, which is why different DAWs sound different after a mix), hence any "missed clipping" after mixing results from either insufficient tools or operator error.
1. IRRELEVANT! True peak meters and true peak limiters only became available a few years ago. By your definition, all music is broken, all music producers are broken/incompetent, so are all mix engineers and mastering engineers and they all have been for more than 30 years. The only people who aren't producing broken audio content are TV re-recording mixers. So what relevance is there to your statements? None whatsoever, we have to deal with the situation as it exists, not with some engineering utopia of perfect tools and equipment.
2. Nope, most are 64bit these days and have been for a number of years. The summing engine is pretty much identical in all DAWs and they all sound identical. And, not only DAWs but also between DAWs and hardware mixers. Of course, there is a get deal of difference between the functionality, the plugin processors and how different producers/engineers employ those processors but the summing engines are effectively identical.
This whole conversation reminds me of the two kings in Gulliver's Travels arguing over which side of an egg to break.
And if you think all DAW summing engines sound the same, you've probably not done any ABX comparisons, as we have. But I'll let you believe whatever you wish, as "what sounds best" gets into the realm of personal opinion.
[1] The tools to identify and manage all PCM peak data are available to us ... [1a] Those who aren't using such tools should be extra careful, or give their project to an engineer who has the tools to do the job right.
[2] I mentioned that "most mixing engines are based on a 32-bit IEEE float topology" to which he/she replied "Nope, most are 64bit these days and have been for a number of years." He/she didn't even know the difference between DAW hardware (64-bit) and DAW software (32-bit). Sigh.......
Yes, most DAWs today are written for a 64-bit processing environment (Intel, etc.), but virtually all DAWs process audio using a 32-bit IEEE float engine (Avid, Logic, Sonar, etc.), and as such are limited to 24-bit audio.
Well, that was written by someone in marketing.
Well it can go a lot higher than 3 dB if you try. If you use a signal like [... +1 -1 +1 +1 -1 +1, ...] you can repeat the pattern to make the intersample peak go arbitrarily high.
This is highly contrived, though. I think it unlikely that you'd see more than a few dB with real music.
The reconstructed analog signal exists between the samples, so if the samples are already at FS, the reconstructed signal can exceed this in level.
Has nothing to do with Nyquist. Nyquist limit for 48 kHz is 24 kHz.
Both are under Nyquist limit.
- A sine wave at 12 kHz with samples [+1, +1, -1, -1, +1, +1] will have intersample peaks of +3 dBFS top and bottom.
- A DC-shifted sine wave at 16 kHz of [+1, +1, -1, +1, +1, -1] will have intersample peaks of +4.5 dB on top side only.
Or worse. I found this thread because I'm designing a product with a DAC that doesn't clip for intersample peaks, but instead becomes "uncontrolled" and produces random signals in that region instead. So if you're playing at 96 kHz, and play a 32 kHz signal as described above [+1, +1, -1, +1, +1, -1], it should be completely inaudible ultrasound, but the intersample peaks produce very audible white noise/distortion instead. So I'm trying to figure out how much I need to attenuate the digital signal to make sure this never happens in realistic situations. -3.5 dB is probably good enough?
Though this experiment found 3% of songs with >4 dB intersample peaks, though they couldn't say which songs they were, or whether those samples of the songs were already at FS due to clipping.
Well, that was written by someone in marketing.
Well it can go a lot higher than 3 dB if you try. If you use a signal like [... +1 -1 +1 +1 -1 +1, ...] you can repeat the pattern to make the intersample peak go arbitrarily high.
This is highly contrived, though. I think it unlikely that you'd see more than a few dB with real music.
The reconstructed analog signal exists between the samples, so if the samples are already at FS, the reconstructed signal can exceed this in level.
Has nothing to do with Nyquist. Nyquist limit for 48 kHz is 24 kHz.
Both are under Nyquist limit.
- A sine wave at 12 kHz with samples [+1, +1, -1, -1, +1, +1] will have intersample peaks of +3 dBFS top and bottom.
- A DC-shifted sine wave at 16 kHz of [+1, +1, -1, +1, +1, -1] will have intersample peaks of +4.5 dB on top side only.
Or worse. I found this thread because I'm designing a product with a DAC that doesn't clip for intersample peaks, but instead becomes "uncontrolled" and produces random signals in that region instead. So if you're playing at 96 kHz, and play a 32 kHz signal as described above [+1, +1, -1, +1, +1, -1], it should be completely inaudible ultrasound, but the intersample peaks produce very audible white noise/distortion instead. So I'm trying to figure out how much I need to attenuate the digital signal to make sure this never happens in realistic situations. -3.5 dB is probably good enough?
Though this experiment found 3% of songs with >4 dB intersample peaks, though they couldn't say which songs they were, or whether those samples of the songs were already at FS due to clipping.