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Dolby's TrueHD 96K Upsampling To Improve Sound On Blu-Rays
Stowie101 writes in with a story about your Blu-ray audio getting better. "The audio on most Blu-ray discs is sampled at 48kHz. Even the original movie tracks are usually only recorded at 48kHz, so once a movie migrates to disc, there isn't much that can be done. Dolby's new system upsamples that audio signal to 96kHz at the master stage prior to the Dolby TrueHD encoding, so you get lossless audio with fewer digital artifacts. The 'fewer digital artifacts' part comes from a feature of Dolby's upsampling process called de-apodizing, which corrects a prevalent digital artifact known as pre-ringing. Pre-ringing is often introduced in the capture and creation process and adds a digital harshness to the audio. The apodizing filter masks the effect of pre-ringing by placing it behind the source tone — the listener can't hear the pre-ringing because it's behind the more prevalent original signal."
May I be the first to say this- fuck Bluray, and fuck Cinavia.
I used to buy Bluray disks. Hell, I own a whole shelf full of them (about 80 titles in total). Every single one eventually got ripped to my NAS in two formats- a relatively lossless MKV file containing the original video and audio streams (up to DTS-HD MA), and a lossy x264 version for playing on crappy devices like the PS3 or 360.
Then Cinavia rolled around, which did two things:
1) It purposefully corrupts the audio stream in an attempt to encode digital information into it (go read their patents- the harder you try to pry Cinavia into an audio stream, the more damage is done to the original quality)
2) It prevented me from playing my legally purchased and legally ripped (it's legal in my country to rip disks and things you BUY) disks off my NAS on my PS3
What pisses me off the most though is that Sony is pushing Cinavia on everyone as hard as they can. AFAIK all new BR players need to be equipped with it, and most of the new BR disks are supposed to have it as well. And they're still advertising the disks as "Lossless", when in fact the audio is NOT lossless- it's lossy, the degradation of which is brought about solely by Cinavia's presence.
Before anyone yells [citation needed] at me, here's your proof straight from the Wikipedia page (http://en.wikipedia.org/wiki/Cinavia):
"Cinavia's in-band signaling introduces intentional spread spectrum phase distortion in the frequency domain of each individual audio channel separately, giving a per-channel digital signal that can yield up to 20 kilobits per second—depending on the quantization level available, and the desired trade-off between the required robustness and acceptable levels of psychoacoustic visibility. It is intended to survive analogue distortions such as the wow and flutter and amplitude modulation from magnetic tape sound recording. On playback no additional audio filters are used to cover up the distortions and discontinuities introduced."
So there you have it. Lossless is no longer lossless, because Sony insists on using this stupid fucking DRM on their stupid fucking format (as usual). Dolby's new gimmicky technology might claim to give you better lossless audio, but none of that matters the moment they drive Cinavia into the stream.
-AC
Question: Was 44.1 kHz chosen in part because the integer 44100 is highly composite? It's divisible by the following factors up to its square root: 1, 2, 3, 4, 5, 6, 7, 9, 10, 12, 14, 15, 18, 20, 21, 25, 28, 30, 35, 36, 42, 45, 49, 50, 60, 63, 70, 75, 84, 90, 98, 100, 105, 126, 140, 147, 150, 175, 180, 196, 210.
Especially interesting is that it's divisible by 7.
Prime factorization of 44100 is 2^2 x 3^2 x 5^2 x 7^2, or (2x3x5x7)^2, or just 210^2. Pretty cool, huh? Coincidence or by design?
You can't improve audio quality of *audible frequencies* by increasing resolution of the horizontal axis (sampling frequency) beyond a rate which surpasses the Nyquist frequency for human hearing.
Nyquist-Shannon notwithstanding, the range of human hearing is wider than 20kHz.
http://www.cco.caltech.edu/~boyk/spectra/spectra.htm (a properly conducted experiment)
That said, doubling the sampling rate isn't going to do anything for a digital signal. At best, the new signal will simply play each of the old signal's samples twice.
After all, I am strangely colored.
If you treat the incoming 44.1 or 48 KHz stream of incoming signals as points on a curve, and apply Curve Fitting calculations to interpolate the intervening data points, you can mathematically recreate some of the detail.
However, this isn't necessarily accurate data -- it's just recreated, the same as when you expand a picture. But like a picture, there are different algorithms and techniques for doing the upsampling, and they "colour" the sound much as an upscaled photo may have jaggies or appear a little blurry.
What I find more interesting is the idea of combining curve approximation with a point mass. You treat the current sample as a point in time, and use acceleration curves to make the "mass" travel a path that intersects all the sample points. If your calculated mass correlates to the actual mass of the drivers in your speakers and the air they move, it should result in a more accurate recreation of the original sound curve.
In fact, I believe Mobile Fidelity got in some hot water with the USG for using just such an approach to encoding 44.1 audio disks, and had to sign a non-disclosure promising they wouldn't use the algorithms for anything other than audio processing. Apparently the USG developed similar algorithms for cruise missile guidance (missiles have mass), so even though it's an obvious and purely physical phenomena being modelled, it's a "military secret." :D
I do not fail; I succeed at finding out what does not work.
"X. Significance of the results
Given the existence of musical-instrument energy above 20 kilohertz, it is natural to ask whether the energy matters to human perception or music recording. The common view is that energy above 20 kHz does not matter, but AES preprint 3207 by Oohashi et al. claims that reproduced sound above 26 kHz "induces activation of alpha-EEG (electroencephalogram) rhythms that persist in the absence of high frequency stimulation, and can affect perception of sound quality." [4]
Oohashi and his colleagues recorded gamelan to a bandwidth of 60 kHz, and played back the recording to listeners through a speaker system with an extra tweeter for the range above 26 kHz. This tweeter was driven by its own amplifier, and the 26 kHz electronic crossover before the amplifier used steep filters. The experimenters found that the listeners' EEGs and their subjective ratings of the sound quality were affected by whether this "ultra-tweeter" was on or off, even though the listeners explicitly denied that the reproduced sound was affected by the ultra-tweeter, and also denied, when presented with the ultrasonics alone, that any sound at all was being played.
From the fact that changes in subjects' EEGs "persist in the absence of high frequency stimulation," Oohashi and his colleagues infer that in audio comparisons, a substantial silent period is required between successive samples to avoid the second evaluation's being corrupted by "hangover" of reaction to the first.
The preprint gives photos of EEG results for only three of sixteen subjects. I hope that more will be published.
In a paper published in Science, Lenhardt et al. report that "bone-conducted ultrasonic hearing has been found capable of supporting frequency discrimination and speech detection in normal, older hearing-impaired, and profoundly deaf human subjects." [5] They speculate that the saccule may be involved, this being "an otolithic organ that responds to acceleration and gravity and may be responsible for transduction of sound after destruction of the cochlea," and they further point out that the saccule has neural cross-connections with the cochlea. [6]
Even if we assume that air-conducted ultrasound does not affect direct perception of live sound, it might still affect us indirectly through interfering with the recording process. Every recording engineer knows that speech sibilants (Figure 10), jangling key rings (Figure 15), and muted trumpets (Figures 1 to 3) can expose problems in recording equipment. If the problems come from energy below 20 kHz, then the recording engineer simply needs better equipment. But if the problems prove to come from the energy beyond 20 kHz, then what's needed is either filtering, which is difficult to carry out without sonically harmful side effects; or wider bandwidth in the entire recording chain, including the storage medium; or a combination of the two.
On the other hand, if the assumption of the previous paragraph be wrong â" if it is determined that sound components beyond 20 kHz do matter to human musical perception and pleasure â" then for highest fidelity, the option of filtering would have to be rejected, and recording chains and storage media of wider bandwidth would be needed."
After all, I am strangely colored.