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."

Comments

You cant hear it anyway.

[Hatta]

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.

Re:You cant hear it anyway.

[Anonymous Coward]

I guess that experiment failed to use monster cables then

Re:You cant hear it anyway.

[Anonymous Coward]

The hearing limit is actually about 20kHz. You need more than 40kHz sampling if you want to capture a sine wave at 20kHz.

The purpose of capturing at higher than 48kHz is to prevent sounds at frequencies above 20kHz being captured at a too-low sampling frequency, and appearing as audible frequencies. These can be removed by analog filtering, but only about one octave above the cutoff frequency. Analog filters are not ideal brick-wall filters, so 96kHz sampling is useful.

However, once the audio is acquired and digitized, software can provide a true brick-wall digital filter. This is impossible to do in analog hardware. After applying the brick wall filter, it can be sampled down to 48kHz or 44kHz with no loss. So, there is absolutely no reason to put 96kHz on disc.

The article isn't clear whether it's 96kHz on just the master, or the disc also.

Re:You cant hear it anyway.

[aardvarkjoe]

That's just because your hardware sucks. If you use the correct equipment, anyone with a discerning ear will be able to hear the difference.

Re:You cant hear it anyway.

[MightyYar]

This might be my favorite review ever on Amazon:

These cables deliver crisp clear sound and are worth every penny. The sound, in all ranges, is amazing. My panoramic eq has never sounded better. I just have one gripe. My Television sometimes won't turn off ever since I've started using these cables with my stereo surround system. In fact it's on right now despite the fact that it's not even plugged in to the electrical outlet. I'm not sure how but these cables are supplying independent power to my television and stereo receiver. It's really cut down on my electricity bill even though, at times, I've lost the ability to control my TV.

Another downside is that, occasionally, there will be high pitched shrill sounds through the speakers. Almost as if a young woman is screaming. It doesn't happen all the time though. Usually it's around 3am when the TV turns itself on. I'm not sure why. It always turns on this show called "Hell Beast". Tivo is not set to record it but, without fail, it turns on every night at 3:33 am. I'm not sure what it's about. There's some sort of gargoyle or mutant goat or something. I think it's a monster movie show. Although they never show a movie and the goat monster guy just says "I want you" over and over. I think it's British or something. I don't really understand the humor. I'm usually tending to my newborn daughter who's routinely wakes up crying because of the screaming coming out of the television. It's funny too because that goat character on the show sometimes yells the name Shannon and that's the name of my daughter. LOL...

Other than those few issues I'm really enjoying the free electricity. It's helped with $$. Especially after all the money I had to drop re-soding my lawn after some teenagers burnt a star into my front lawn. Some stupid neighborhood gang. They're calling themselves 9-9-9.

Re:You cant hear it anyway.

[SimonTheSoundMan]

I'm a sound engineer and you are totally right.

Going back in history. 44.1kHz was chosen because it syncs with PAL video frames, 48kHz syncs with NTSC. If you were doing linear editing, you can dub and cut the audio perfectly to the half frame.

44.1kHz stuck because Umatic, an analogue videotape that you could buy a PCM head as an optional extra, was chosen to create the master copies for CDs to be sent to duplication in to pressed CDs.

Re:You cant hear it anyway.

[Anonymous Coward]

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?

Re:You cant hear it anyway.

[dgatwood]

Actually, polystyrene caps can make a huge difference over electrolytic or tantalum caps in certain parts of some circuits. For example, in condenser microphones, the coupling capacitor between a microphone element and the first FET stage is a critical part of the circuit in which the signal level is very, very weak. Thus even tiny amounts of noise from cheap capacitors can have a significant effect on the final result. A fair number of cheap Chinese microphones sound dramatically better if you replace the cheap dipped tantalum caps they use with a film cap or poly cap.

We're not talking about a small difference here, either. We're talking night and day. A deaf person could just about hear the difference. :-) Replacing just the handful of tantalum capacitors in those microphones can make the difference between a muddy sound with a harsh, brittle top end and a fairly clean, accurate representation of what is being recorded... all for about five bucks and a few minutes of soldering. (Even better, the most important one—the FET coupling cap—is usually direct-wired between the capsule mount and the FET's lead, so you don't have to worry about lifting traces....)

Capacitors within the feedback path of an amplifier circuit can also degrade the sound noticeably. Admittedly, this isn't as much of an issue these days with the rise of modern, chip-based amplifier circuits, but it is still worth keeping in mind, particularly given that most condenser microphones still use transistor-based amplifier circuits.

Just to be clear, though, it doesn't have to be polystyrene film. The difference between a polystyrene cap and a traditional metal (polyester) film cap is negligible compared with the difference between film caps and electrolytic or (*shiver*) tantalum caps. Tantalum caps simply should not be within a city block of any trace that carries an audio signal.... Okay, slight exaggeration, but you get my point.

And, of course, it doesn't make sense to replace every capacitor. If it isn't in the signal path, it usually won't make much difference (though the absence of capacitors in the right places on power supply rails can cause some fun problems), and even if it is, it may or may not make much of a difference, depending on where the capacitor is in the signal path.

Re:You cant hear it anyway.

[guidryp]

Here is a link to the original paper by Dr. Peter Craven where he mathematically proves that an apodizing filter can make audible improvements in sound reproduction.

You can't mathematically prove something sounds better. Most adults can't even hear 16KHz, let alone 20 KHz and beyond, or detect subtle variations in those ranges.

You have to do double blind testing. Double blind testing has shown even real 24/96KHz can't be discerned from 16/44.1KHz by audiophiles and recording pros.

Anything they are trying to sell beyond this is placebo snake oil.
http://mixonline.com/recording/mixing/audio_emperors_new_sampling/

Worthless gimmick with no audible benefits

[sahonen]

Dumb, dumb, dumb. An ideal sample rate upconversion results in something that *is* identical to the source. Mathematically. It's like re-encoding a 64kbps MP3 to 192kbps. If anything you are going to *lose* quality due to inherent errors in the process.

Re:Worthless gimmick with no audible benefits

[PhrostyMcByte]

Mod parent up!

A lot of people will see a graph of PCM and think up-sampling will help make the stair-stepping be finer, less noticeable, and thus improve quality. Unscrupulous audio companies love to take advantage of this belief with up-sampling tech.

That belief is, of course, complete bullshit—the stair-stepping of PCM is merely a digital encoding which DACs use this to reproduce a full, fluid signal. There's literally nothing for up-sampling to do that could add any quality! The only thing it will do is introduce even more errors.

In some cases DACs have even behaved worse at higher sample rates—meaning in that case you'd not only have more errors from upsampling, but also more errors from the DAC.

Re:Worthless gimmick with no audible benefits

[Anonymous Coward]

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.

Lossless + Cinavia == Lossy

[Anonymous Coward]

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

Apodizing Filter

[Josuah]

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.

Unsampling ... then re-sampling in 96KHz?

[Taco Cowboy]

Oh, c'mon !!

This is one thing that simple does NOT make any sense

If the thing was recorded in 48KHz, it's at 48KHz, and no matter how one can "un-sampling" that shit and then re-recording it in 96KHz (even at 96MHz or 96GHz), it does not boost _anything_ !!

Re:Unsampling ... then re-sampling in 96KHz?

[Austerity Empowers]

Obviously you can't unsample and re-record an audio stream to reduce pre-ringing or de-apodizing all but the smallest apods. All you are doing is essentially taking harsh audio and putting it on qualudes. This simply produces depressed, harsh audio, like Norm Macdonald.

Time and money would be better spent using gold plated, neon injected, forward biased monster cables, with gallium arsenide softening strips. The justification for using gold plated, neon injected, forward biased monster cables is well known by audiophiles. The gallium arsenide softening strips work by absorbing the harsh pre-ringing frequencies by actually siphoning out the high frequencies. Silicon engineers have long known that gallium arsenide is ideal for conducting the highest frequency signalling. Used in this application it acts as sort of a apod rejection filter, allowing the ringing to be thrown free of the cable, before it can manifest as ringing, or even chirping.

However it is important to stress that the gallium arsenide softening strips must be calibrated to your eardrums carefully before use. You will need a vector analyzer and a sound meter. It's crucial that you place the softening strips in your mouth, while providing the four port network the VNA requires using both your feet and hands. You must suck on the strips until the sound meter absorbs all the s-parameters in your body, which are being slowly drained by the gallium arsenide strips. You must maintain this until the sound meter gets down to at least 3dB (or 1dB if you have especially sensitive ears). Without this step you may as well be using a walmart SPDIF cable, it will be that bad.

Re:Unsampling ... then re-sampling in 96KHz?

[msobkow]

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

Re:You can prefectly represent anything up to Fs/2

[Man On Pink Corner]

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.

And Harry Nyquist is rolling around in his grave

[Gordo_1]

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.)