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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."
44.1khz audio is already transparent to the human ear. Blind studies have been done where a 16 bit 44.1khz ADC-DAC pair was inserted into a high resolution analog audio source. No significant difference was observed.
Don't waste money on the placebo effect.
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The title is misleading if the actual goal of this is to apply an apodizing filter. I suspect the reason it's called "Advanced 96K Upsampling" is because that's much easier to get people to buy into that than a "Apodizing Filter" sticker.
The article explains how the audible benefit comes from the application of the Meridian apodizing filter, which changes the analog signal reproduced from digital data by reducing the pre-ringing. IIRC the trade-off is that post-ringing increases. The claimed benefit is that since the ringing now occurs after the "real" music of larger amplitude and as a result the ringing is masked or could be considered like an acoustic echo that naturally occurs.
The 96K upsampling is just a side-effect of wanting the extra samples when you are applying the filter.
Here's a decent summary of what is supposed to happen to the analog audio signal as a result of the filter application: Technical analysis of the Meridian Apodizing filter.
That being said, from what I've read over the past few years I think people are kind of mixed on whether or not the filter makes things better, worse, or just different but not better.
Mathematically you've got 5 choices when you want to sample and have sound almost all the way up to nyquest:
1 a brick wall, phase linear filter. Mathematically the best, and yes it has pre and post ringing - the less roll off you allow the more ringing
2 you can do less filtering and allow aliasing instead. In that case you'll get a mirror image of the spectrum above the nyquest rate that wont matter much because only dogs and small children can hear that high. And less ringing
3. You can let the treble roll off a bit. In fact 48k sampling rate is more than cd just so that the roll off from 20 to 24 is longer than that from 20 to 22 and you'll get less ringing. A little roll off never killed anyone
4 you can use an old style filter with some phase shift. It just trades off preringing for postringing and delays some frequencies more than
others and is overall less efficient. Frankly the frequencies being discussed are so high no one will notice the delays. In theory you can mess up the imaging and sound a little that way. There's a reason that the industry has preferred linear phase digital filters to older style analog filters, but no doubt in the digital domain you can optimize a filter with phase delays just like you can optimize one without.
5. You can have an adaptive filter that decides between options 1 2 3 and 4 depending on some unimportant critera like masking. It's unimportant because only children can hear high enough to detect even a hint of ringing or aliasing above 20khz and as far as I know they're not the market.
Try a high but more audible frequency.
It may be less confusing if I put it this way: If you can't hear a sine wave beyond, say, 20 kHz, then you are not going to be able to tell the difference between a sine wave at 7 kHz and a square wave whose fundamental frequency is 7 kHz. That's because the lowest harmonic in the square-wave signal will be at 21 kHz. Your ears will filter it out, just as the antialiasing filter in the recording system would need to do.
Now, that being said, the argument has been made that intermodulation effects in the human ear can allow us to perceive sounds beyond the usual 20 kHz limit when they mix with each other. To the extent these effects occur when listening to the source material at a given level, you could argue that the ultrasonic parts of a performance should be captured and reproduced along with everything else, and that would require a higher sampling rate.
The showstopper for this argument is that any desirable sonic content resulting from IMD at ultrasonic frequencies could only be reproduced "properly" at a specific volume level, because distortion products by definition are generated by nonlinear processes.
Not that this whole thing isn't absurd for the reasons already discussed above, but what no one bloody well seems to understand it that an audio stream is not a godamn bitmap picture. 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. Assuming a high quality anti-aliasing filter is used and excellent quality recording and playback equipment, audio sampled at 48kHz can be unambiguously represented up to about 24kHz. 96kHz is a waste of bits.
Vertical resolution (# of bits) is the only theoretical way to improve actual audio quality further... and beyond about 16-18 bits, it's also beyond the ability of even the most diehard audiophiles to discern (in properly conducted experiments.)
The signal is only "stair stepped" because they chose to graph it that way. The audio signal coming out of the DAC does not look like that. Those stair-steps are happening at frequencies more than half the sampling rate -- they are eliminated from the analog output by a low-pass filter. This is essentially performing this "splining" you are talking about.
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.
Actually upsampling can be useful when you apply digital filters. There is no such thing as an ideal filter, so if you modify one frequency band (e.g. in a equalizer) you end up modifying all others. The higher is the sample rate the lower is this sideband interference.
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Preringing is what the linear-phase oversampling filter in the DAC chip in the player creates. Which is also the place to fix it, by putting an apodizing filter there, and some semiconductor manufacturers do exactly that (Wolfson Micro, etc.). Dolby's approach makes no sense--they oversample 2x during mastering (needed or the apodizing filter doesn't work) and then you have to store twice the data. Why? If the DAC is doing it, then you can just feed it the usual 44.1 or 48 k. Moreover, since the DAC's filter usually oversamples by 8x to allow simpler analog filters post-DAC, it can do the apodizing much better anyway. Once again Dolby takes legit technology and implements it poorly into a lousy gimmick to sell. Instead of reading dumb marketing material and even dumber article summary on slashdot, read some peer reviewed papers discussing preringing and apodizing filters, say http://www.aes.org/e-lib/browse.cfm?elib=12992
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