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Dolby's TrueHD 96K Upsampling To Improve Sound On Blu-Rays

samzenpus posted more than 2 years ago | from the let-the-arguing-begin dept.

Media 255

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."

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This is... (0)

Anonymous Coward | more than 2 years ago | (#40035599)

...worthless shit.

Re:This is... (2, Insightful)

Anonymous Coward | more than 2 years ago | (#40035635)

No kidding. A/B/X or GTFO.

Unsampling ... then re-sampling in 96KHz? (4, Insightful)

Taco Cowboy (5327) | more than 2 years ago | (#40036307)

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? (2)

hairyfeet (841228) | more than 2 years ago | (#40037235)

Exactly, this is like colorization. You can't recreate what wasn't recorded in the first place, all you can do is add shit on top. The funniest part? Ask teens and early 20s and they will tell you they LIKE the "sizzle" of MP3 because that is what they have grown up with. so not only are you adding shit that isn't there but the "artifacts' they are complaining about are ENJOYED by the younger generations which is the big target demographic everyone shoots for!

You cant hear it anyway. (4, Informative)

Hatta (162192) | more than 2 years ago | (#40035601)

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. (5, Funny)

Anonymous Coward | more than 2 years ago | (#40035637)

I guess that experiment failed to use monster cables then

Re:You cant hear it anyway. (1)

cvtan (752695) | more than 2 years ago | (#40035987)

And polystyrene capacitors and solid silver speaker wires and wood-encased iron laminations placed on the power supply transformer and aluminum pyramids to hold up the speakers and...

Re:You cant hear it anyway. (1)

Anonymous Coward | more than 2 years ago | (#40036069)

What ? No Monster Cables? You probably don't even have a warranty. Do you?
Do You?

Re:You cant hear it anyway. (1)

afidel (530433) | more than 2 years ago | (#40036549)

The only one of those that makes sense is the pyramids and they're probably marginally worse that normal spikes if perhaps a bit more fashionable.

Re:You cant hear it anyway. (5, Informative)

dgatwood (11270) | more than 2 years ago | (#40036661)

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. (1)

old and new again (985238) | more than 2 years ago | (#40036783)

or coat hangers

Re:You cant hear it anyway. (1)

Anonymous Coward | more than 2 years ago | (#40035693)

It is important to consume bandwidth for movies delivered over the internet. That way, price tiers can be established, an ISPs can be motivated to play ball with Big Media.

Re:You cant hear it anyway. (0)

Anonymous Coward | more than 2 years ago | (#40035711)

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.

There are lot of people in this business wire article that indicate an improvement:

http://finance.yahoo.com/news/dolby-elevates-quality-lossless-audio-190100832.html

Though, this may be due to changes made outside of the realm of 96khz:
Besides enabling optimum 96k upsampling, this technology features a unique apodizing filter that “masks” the unwanted digital artifacts known as “preringing,” which is introduced during the content-capture and content-creation process

Re:You cant hear it anyway. (5, Informative)

Anonymous Coward | more than 2 years ago | (#40035719)

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. (1)

jasomill (186436) | more than 2 years ago | (#40035861)

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

Actually, it is. To wit, FTA,

In order to enjoy the benefits of a 96kHz disc, you need an AV receiver capable of playing it.

Re:You cant hear it anyway. (5, Informative)

SimonTheSoundMan (1012395) | more than 2 years ago | (#40036039)

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. (5, Interesting)

Anonymous Coward | more than 2 years ago | (#40036135)

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. (1)

sahonen (680948) | more than 2 years ago | (#40036263)

PCM audio was originally recorded on video tape, and you needed to be able to record it in both PAL and NTSC standards. So the fact that the number is highly composite is largely a consequence of having to be divisible by both 50 and 60. More here: http://en.wikipedia.org/wiki/44.1_kHz [wikipedia.org]

Re:You cant hear it anyway. (1)

Anonymous Coward | more than 2 years ago | (#40036487)

Your justification is irrelevant, as 48000 is also divisible by both 50 and 60, and also NTSC is 59.97Hz, not 60Hz, though I suspect the sample rate of both 48 and 44.1kHz audio is adjusted slightly to synchronise with NTSC video frames.

Obviously 44100 was selected because it is highly composite, but it has little to do with syncing to video. Also it specifically does not sync with 24Hz frame rates, as used in film, which is probably why 48kHz and not 44.1 is used in film.

I have heard another story, which I'm doubtful to believe: that a lower samplerate was chosen for CDDA, because one of the executives as Sony wanted the entire of Beethoven's 9th Symphony to fit on a single CD, which supposedly would not of been possible at the previous standard rate of 48Khz. I haven't investigated whether this claim has merit.

48kHz is still standard in film, digital radio and television broadcast, and pretty much every other application outside of CDDA and reproductions of such as MP3, and I guess DVD-A.

Re:You cant hear it anyway. (1)

squiggleslash (241428) | more than 2 years ago | (#40036579)

I have heard another story, which I'm doubtful to believe: that a lower samplerate was chosen for CDDA, because one of the executives as Sony wanted the entire of Beethoven's 9th Symphony to fit on a single CD, which supposedly would not of been possible at the previous standard rate of 48Khz. I haven't investigated whether this claim has merit.

It doesn't. The problem with it is that different performances are different length. It's possible though highly unlikely the Sony executive was refering to a specific performance, but performances can vary in length by more than 50%, depending on the style of the conductor.

Re:You cant hear it anyway. (2)

old and new again (985238) | more than 2 years ago | (#40036833)

he refered to karajan performance that is 73 minutes long, the story is real, but its was not BECAUSE of this that they choose 44.1

Re:You cant hear it anyway. (0)

Anonymous Coward | more than 2 years ago | (#40036621)

NTSC is 59.97Hz, not 60Hz

60 Hz is equivalent to 59.97 Hz. 60.00 Hz is not. In other words if you are factoring 48000 that is not necessarily the same as 48 khz it is the same as 48.000 khz.

Re:You cant hear it anyway. (2)

drkstr1 (2072368) | more than 2 years ago | (#40037237)

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?

./ need to let you keep some mod points in a reserve so you can use them when you come across some fine gems like these! :D

Re:You cant hear it anyway. (1)

jo_ham (604554) | more than 2 years ago | (#40036589)

Ah, Umatic tapes, those were the days.

I used to shoot and edit on those bad boys, using car batteries to run both the recorder and the camera since you could run for much longer without having to swap out PAG batteries all the damn time, assuming you didn't need to be *too* portable - recorder on one shoulder, camera on the other then some chump to carry the battery.

By the end of their life, our linear edit decks were really showing their age, and could be +/- 3 or 4 frames around your actual edit point, but we skipped right over Betacam and went right into NLE with Media 100. It was never quite the same after that, with the edit suite being much quieter with just the sound of computer fans rather than the clanking, clunking and whirring of those old dinosaur decks, and the distorted audio during jog/shuttle. Editing is just too sterile now!

Re:You cant hear it anyway. (2)

sr180 (700526) | more than 2 years ago | (#40036935)

How does a sound engineer get to call themselves an engineer? Im not having a go, Im just asking...

However, for those of you quoting Nyquist, you only have half the answer. One of the side benefits of a higher frequency is lower quantisation noise - and hence a better signal to noise ratio. When you take a sample of sound, you then fit it to 16 bits. Obviously an analogue sound pressure level wont fit perfectly into a 16 bit value - so you have to fit it to the nearest one. The difference then becomes noise - which can generally be approximated as white noise (I know mathematically this is possibly incorrect, but practically its true) with its energy spread over the available frequency. Filter this noise out (which your ears will do for anything above 20-25khz) and you reduce the effective quantisation noise being heard (you have filtered out half of the noise's power) - improving the signal to noise ratio.
This obviously will not work in the case of material already sampled - as the quantisation noise is already there in its sampled form, however, it will have a similar effect for the encoding - if the encoding poduces white noise as part of its process - which (not having researched their encoding thoroughly) is likely.

Will it truely make a difference? I doubt it. TrueHD is already damn good - and the limitations are really going to be in the amplifiers and the speakers, particularly the cheap power supplies modern home amps seems to carry. I'm sure this is really just more about planned obsolescence.

Re:You cant hear it anyway. (1)

Shavano (2541114) | more than 2 years ago | (#40036051)

It's impossible to remove sampling artifacts because once in the digital domain there is no way to distinguish between sounds that were originally on an inaudible range and correctly digitized sounds.

Re:You cant hear it anyway. (1)

sahonen (680948) | more than 2 years ago | (#40036173)

Actually, modern delta-sigma A/D converters are capable of near-brickwall lowpass performance at nyquist.

You can prefectly represent anything up to Fs/2 (1)

robbak (775424) | more than 2 years ago | (#40036675)

That is just (not-so-)simple math. You can perfectly represent any signal with a frequency less than half of your sampling frequency. Audiophiles don't like this, but it doesn't change the fact. The greatest reason for confusion is the 'stepped waveform' graphic often used to explain sampling, which is badly misleading.

40Hz is ample. Anything more is overkill. All you get from 96kHz sampling is ultrasonics that need to be filtered out to prevent distortion from your speakers.

Re:You can prefectly represent anything up to Fs/2 (0)

Anonymous Coward | more than 2 years ago | (#40036949)

Ok Jack,

I have a sine wave, triangle wave and square wave at 20khz. You sample them at 40khz and reconstruct them perfectly.

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

Man On Pink Corner (1089867) | more than 2 years ago | (#40037009)

There's no such thing as a square wave at a given frequency. A square wave is the sum of the fundamental and all odd harmonics, and a triangle wave is represented by another, similar series.

You might have sine, triangle, and square waves whose fundamentals are all at 20 kHz, but both the square and triangle waves will sound exactly the same as the sine wave if they are sampled and reproduced properly at 44.1 kHz. The antialiasing filter will remove the harmonics before the signals are digitized, resulting in three recordings of a sine wave.

Higher sampling rates allow you to use cheaper antialiasing filters, but that's hardly a constraint worth worrying about in a modern digital signal chain.

Re:You can prefectly represent anything up to Fs/2 (0)

Anonymous Coward | more than 2 years ago | (#40037153)

Can be represented by or is? The orbit of planets must also be composed of circles?

Secondly can you hear the difference between a sine wave, square wave and triangle wave? Try a high but more audible frequency.

What are you saying when the audio track has only frequencies lower than some number or that you cant hear a frequency?

Re:You cant hear it anyway. (5, Funny)

aardvarkjoe (156801) | more than 2 years ago | (#40035727)

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

Re:You cant hear it anyway. (1)

mr9times (681984) | more than 2 years ago | (#40035825)

Make fun if you like, but my Rebecca Black mp3s never sounded so good as with my AudioQuest K2s.

Re:You cant hear it anyway. (1)

crashumbc (1221174) | more than 2 years ago | (#40035833)

LOL, the "reviews" of those are hysterical

Re:You cant hear it anyway. (5, Funny)

MightyYar (622222) | more than 2 years ago | (#40035859)

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. (1)

Anonymous Coward | more than 2 years ago | (#40036457)

Great. I know I'm not sleeping tonight, anymore.

Re:You cant hear it anyway. (1)

Shavano (2541114) | more than 2 years ago | (#40036067)

I hope you're joking.

Re:You cant hear it anyway. (1)

gman003 (1693318) | more than 2 years ago | (#40036139)

He linked to a *cable*. And all the reviews are supremely sarcastic or mocking.

He's joking.

Re:You cant hear it anyway. (0)

Anonymous Coward | more than 2 years ago | (#40035729)

Has nothing to do with bandwidth. It is transient distortion that is being removed by the new filter, as pre-ringing will make transient sounds less defined. Do a search to learn what pre-ringing is, as it is a parasitic component of the FIR processes that compressed digital audio uses.

Re:You cant hear it anyway. (1)

Em Adespoton (792954) | more than 2 years ago | (#40035737)

while sound sampled and played back at 44.1khz should be good enough for anyone, it's a lossy representation of the original waveform. The problem with this comes with post-sampling manipulation of the original waveforms, at which point the gaps tend to pile up in patterns that can be detectable by the human ear, due to our innate ability to find patterns in just about anything, even when it doesn't actually exist.

The problem here is that they need the extra data in order to properly reconstruct a 44.1khz equivalent after overfiltering and munging the original waveform until it is in a state that no longer looks like the original. It doesn't matter that it's still a decent 44.1khz sample -- it's one that is easier to find engineered patterns in than the original, and so it sounds different. They need more data to make it sound better.

They could fix this by not doing so much post-processing on the sample, but they'd rather look like they're doing something important to "improve" the sound, and then attempt to hide that fact by overlaying it with a strong signal from the original. Think of it as the audio equivalent of layering an original image over a copy that's been gaussian blurred. Easier than re-creating the image from the blurred copy, and if you only have half the samples and you've munged ALL of them, you no longer have something to do that with.

Re:You cant hear it anyway. (1)

Brett Buck (811747) | more than 2 years ago | (#40035755)

Depends on how the filters that knock out the 44.1 kHz are implemented. It is MUCH easier to filter out 96 khz than 44.1. It's not just a math problem, you have to actually build the D/A and analog section

Re:You cant hear it anyway. (0)

TheGratefulNet (143330) | more than 2 years ago | (#40035791)

88k or 96k is actually a good break-point for digital audio for end-users.

for recording, you want to capture as high as possible (192k is reasonable).

but like you don't view raw photo images at higher than 8bit/color, you don't need more than 24/96 in end user audio.

but 44.1 is NOT enough for some tech reasons. filtering is the main one, not audio cutoff at 20k (which is likely what you are thinking of).

Re:You cant hear it anyway. (0)

Anonymous Coward | more than 2 years ago | (#40037141)

There is no good reason to record higher then 44.1, and filterng is not one either. Oversampling has been common place for a few decades now. 44.1 vs 96 has been tested over and over, double blind single variable tests nobody can ever tell the difference. Now days storage and processing is cheap so all you are wasting is DSP cycles, so knock yourself out. Now for live sound there is a reason to run at higher sample rates but is not for audio quality.

Re:You cant hear it anyway. (1)

jasomill (186436) | more than 2 years ago | (#40035817)

It may not actually be a placebo effect. This is only true if you assume the goal is high fidelity, not merely results that "sound good." Recall that some people prefer vinyl to digital audio, in spite of its inarguable lower fidelity, and this has nothing at all to do with the placebo effect. It is, after all, trivially easy to identify vinyl vs. digital in a blind test. But if, as stated in the article, the result must then be distributed as a 96KHz track to yield the "improvements", that's almost surely bullshit.

Re:You cant hear it anyway. (1)

Jiro (131519) | more than 2 years ago | (#40036547)

In modern times the "loudness war" can actually make the vinyl version better, when the studio doesn't compress the range on the vinyl version.

Re:You cant hear it anyway. (0)

Anonymous Coward | more than 2 years ago | (#40035895)

Sorry, but I can tell the difference between a Beethoven piano sonata performed live in front of me and a pre-recorded performance emanating from two speakers, regardless of the sampling quality used.

I hate recorded music.

Re:You cant hear it anyway. (3, Informative)

the eric conspiracy (20178) | more than 2 years ago | (#40036157)

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.

http://www.aes.org/e-lib/browse.cfm?elib=12992 [aes.org]

Unfortunately it's behind a paywall. but take my word for it's a pretty impressive piece of work.

Most people, whose sensibilities are not trained to the point where they are discriminating enough won't likely notice the difference. However the fact of the matter is the differences are measurable, the principle is based on sound math, and the results are in the perceptible audio spectrum.

Peter Craven made several important contributions to digital recording. He and Michael Gerzon did a lot to push forward the early development of surround sound technology, and made other significant contributions to the process of digital recording. In particular their work on dithering has had a big impact in improving the quality of CD recordings.

http://en.wikipedia.org/wiki/Michael_Gerzon [wikipedia.org]

http://www.aes.org/e-lib/browse.cfm?elib=5872 [aes.org]

http://www.aes.org/e-lib/browse.cfm?elib=6777 [aes.org]

http://www.aes.org/e-lib/browse.cfm?elib=6647 [aes.org]

Re:You cant hear it anyway. (0)

Anonymous Coward | more than 2 years ago | (#40036393)

This needs to be higher. Serious work is behind the concept, and the usual over-simplified "44.1kHz ought to be enough for everybody" mantra just doesn't fully encapsulate the concepts.

Re:You cant hear it anyway. (1)

Shavano (2541114) | more than 2 years ago | (#40036429)

Does it say how he disproved the sampling theorem?

Re:You cant hear it anyway. (1, Informative)

the eric conspiracy (20178) | more than 2 years ago | (#40036535)

The Nyquist Shannon theorem makes some assumptions that are not necessarily valid for digital recording of music.

http://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem [wikipedia.org]

"The theorem assumes an idealization of any real-world situation, as it only applies to signals that are sampled for infinite time; any time-limited x(t) cannot be perfectly bandlimited. Perfect reconstruction is mathematically possible for the idealized model but only an approximation for real-world signals and sampling techniques, albeit in practice often a very good one.

The theorem also leads to a formula for reconstruction of the original signal. The constructive proof of the theorem leads to an understanding of the aliasing that can occur when a sampling system does not satisfy the conditions of the theorem."

Re:You cant hear it anyway. (1)

timeOday (582209) | more than 2 years ago | (#40036657)

What? My recollection is a time limited sample is only limited in how low it can go; e.g. a 1 second recording can only represent down to 1 hz (or half or twice that, I don't remember) because, obviously, at some point you only get 1 or 0 samples during the interval, and you need two samples to say anything about frequency (i.e. how often something is happening).

Of course this limitation is totally irrelevant to music, since the source signal (the song itself) has finite duration, and you can't really "hear" anything much below 20 hz (if anything, strong signals just below that range just make your guts feel queasy).

Is there more to it?

Re:You cant hear it anyway. (4, Insightful)

guidryp (702488) | more than 2 years ago | (#40036859)

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/ [mixonline.com]

Re:You cant hear it anyway. (2)

TheRealMindChild (743925) | more than 2 years ago | (#40036461)

My mother always told me that I can't taste the Tuna in her chicken cassorole. I don't care WHO couldn't taste it, I could.

Re:You cant hear it anyway. (1)

Anonymous Coward | more than 2 years ago | (#40036617)

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.

Higher sampling rates aren't only about audio reproduction quality (as you rightly point out). What it allows engineers to do is design less 'severe' filters, which may have other benefits.

To sample at 44.1 kHz (call it 44), you need to cut things off at 22 kHz. You want to get all the data unto the (theoretical) human hearing limit of 20 kHz, that means you only have about 2 kHz to drive down the signal. If, however, the sampling rate was at 48, you now have have to cut things off at 24, and have 4 kHz worth of room to filter your signal.

So you can use a less expensive filter to achieve the same result, or the same filter would reduce higher frequencies even more (since it has more 'room' to do its thing), which could help reduce aliasing.

Re:You cant hear it anyway. (1)

reve_etrange (2377702) | more than 2 years ago | (#40036727)

I think the reason that ideas like this keep showing up is that people do not understand the sampling theorem [wikipedia.org] .

Worthless gimmick with no audible benefits (5, Insightful)

sahonen (680948) | more than 2 years ago | (#40035691)

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 (4, Insightful)

PhrostyMcByte (589271) | more than 2 years ago | (#40035891)

Mod parent up!

A lot of people will see a graph of PCM [wikipedia.org] 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 (1)

loufoque (1400831) | more than 2 years ago | (#40035911)

Try re-reading the summary. It is not a normal upsampling, it applies a special filter that is supposed to compensate for artificats during recording.

Of course, that filter could just be applied during playback of 48kHz audio, but it would probably require significant horse power to do so in real time.

Re:Worthless gimmick with no audible benefits (2, Informative)

Anonymous Coward | more than 2 years ago | (#40036665)

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.

Re:Worthless gimmick with no audible benefits (1)

Tablizer (95088) | more than 2 years ago | (#40036923)

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.

Well, they kind of are. My daughter used to complain about the high-pitch squeal in older B&W TV tubes. A customer may return an item that bothers their child. May be same for complaining dogs. They may not have wallets, but they can gripe to those who do.
       

Re:Worthless gimmick with no audible benefits (1)

sahonen (680948) | more than 2 years ago | (#40037215)

Modern DSP techniques can implement a brick wall filter with a phase response anywhere from linear phase (equal pre and post ringing) and minimum phase (100% post-ringing). You can even do maximum phase (100% pre-ringing) if you're crazy.

Re:Worthless gimmick with no audible benefits (1)

jonsmirl (114798) | more than 2 years ago | (#40036633)

The purpose of these Dolby "enhancements" is to ensure that every receiver and TV manufactured has to pay Dolby a big license fee so that they can recover the source material that has been Dolby encoded. I wish they'd just leave HDMI level audio in uncompressed PCM. But then we wouldn't need to license these Dolby decoders.

Re:Worthless gimmick with no audible benefits (1)

smellotron (1039250) | more than 2 years ago | (#40037227)

I wish they'd just leave HDMI level audio in uncompressed PCM. But then we wouldn't need to license these Dolby decoders.

We'd also need to reserve more space for audio on our discs. Dolby's and other encoding schemes compress audio data. This factors into bandwidth as well: IIRC, my PS3 sends "bitstream" audio data to my receiver because the optical cable can't handle 5.1 LPCM at the source frequency. HDMI cables are higher-bandwidth than optical, but it was not an option for me.

It's also convenient to use the Dolby matrix-encoded audio to get surround sound out of plain old RCA cables. I discovered this with Guitar Hero on my Wii, and the fact that Nintendo and my receiver's manufacturer both payed Dolby meant that I could get surround sound without switching out the entire audio path.

But yeah.. TrueHD is probably overkill. I expect the room to have a bigger impact than a lossless codec for all but the most anal home theaters.

Lossless + Cinavia == Lossy (5, Interesting)

Anonymous Coward | more than 2 years ago | (#40035725)

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

Re:Lossless + Cinavia == Lossy (-1)

Anonymous Coward | more than 2 years ago | (#40036335)

"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 secondâ"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."

Sounds like some kind of watermarking to me, is that what you are bitching about?

Well shit, don't buy movies if it is that important to you. Is it really shocking that a Bluray player wont play ripped Blueray content? Honestly?

Damn Dolby... (1)

wbr1 (2538558) | more than 2 years ago | (#40035767)

You blinded me with science!
All jokes aside, few if any people can hear the difference between 44.1, 48, or 96khz sample rates. Under the Nyquist limit (half the given sample rate) all are equally precise in recording (an hence rendering the sound). What a higher sample rate does do is make for simpler ADC/DAC chips that sound good, at the expense of more bits. And it allows audio manipulating software (plugins and such) more accurate (but good software design can engineer around that). So aside from mixing and mastering (a little), it makes no difference at all to the ear in the final mixed down track.

Re:Damn Dolby... (0)

Anonymous Coward | more than 2 years ago | (#40035821)

You blinded me with science!

Different Dolby.

Re:Damn Dolby... (1)

cvtan (752695) | more than 2 years ago | (#40036023)

Give a listen to "Aliens Ate My Buick". One of my favorites.

Re:Damn Dolby... (1)

djdanlib (732853) | more than 2 years ago | (#40036133)

I can hear the difference, but I have been a studio sound engineer in the past, and built an audio system that can reproduce it. I don't think most people have that combination. In fact, most people don't have speakers that can produce tones that span the common human's range of hearing of 20 Hz - 20 KHz, and don't know what that even sounds like. Some people can't hear the higher frequencies in that range, either, and sensitivity to the highs tend to drop off with age for some percentage of people. For some reason, I haven't had that happen yet, although by all accounts it should have.

It's a lot better to mix at high sample rates, since you can keep a higher amount of detail around for your mixing, effects and mastering software to work with, which means you have better a quality master file before you downsample. You do eventually downsample to CD quality after you're done mastering.

Re:Damn Dolby... (1)

JonySuede (1908576) | more than 2 years ago | (#40036883)

and some people are extremely annoyed by them, I am in my 30's and I still have to lower the sound above 16Khz by about 6db else I feel attack by the sound.

I hate to be near a metal guitar amp and yet I like live metal rehearsal when I am closer to the bass while being further away from the guitar amp...

Re:Damn Dolby... (1)

djdanlib (732853) | more than 2 years ago | (#40037255)

We're in the same ballpark. Yeah... I know what you're talking about with the metal guitar. I've done that and recorded that and am somewhat of an electric guitar enthusiast. Most metal guitar players have really crappy setups or really good setups that they've made crappy (mostly with distortion pedals, especially those who run directly out of that into a PA) and have played themselves half deaf. I haven't seen a guitar amp that could output much at that frequency - they usually have full-range speakers that produce lots of bass and midrange, with enough highs to get in the neighborhood of 10 KHz. Even the super expensive ones. It's always some really horrendously scooped EQ, but darn if they love that chugga-chugga-weeee sound. Sounds like your favorite bands need a new sound crew. Earplugs are worth a $1/pair to save your hearing and too many guitarists ignore that.

You know, if it's really painful though, maybe you should look into a hearing exam. Hypersensitivity to high pitches could be a sign of developing tinnitus or something else. It couldn't hurt to ask a professional if you haven't already.

If you want to impress me (1)

Osgeld (1900440) | more than 2 years ago | (#40035773)

make a system that amplifies dialog to the same level as everyfucking thing else in the movie so I dont have to constantly fiddle with my remote. Why is it in music we have the loudness wars where all sound is mashed into mindless noise at the peak of volume, but in movies there HAS to be a 100db difference between scenes

Re:If you want to impress me (3, Insightful)

Ogi_UnixNut (916982) | more than 2 years ago | (#40035849)

I'd prefer if they kept movies with the "100db difference". It is far easier to apply a dynamic compressor plugin than it is to undo studio-mastered dynamic compression. In fact, I hope they do the same with music as well, so that eventually we can apply as much compression as we want for a given environment/situation.

Re:If you want to impress me (1)

PhrostyMcByte (589271) | more than 2 years ago | (#40035927)

It is far easier to apply a dynamic compressor plugin than it is to undo studio-mastered dynamic compression.

The cool thing is, a lot of TVs, audio equipment, and software have already had something like this built in for years. Usually they call it a user-friendly name like "night mode", so it can be a little difficult to find, but at least it's there. Why audio can't take the same path is beyond me.

Re:If you want to impress me (0)

Anonymous Coward | more than 2 years ago | (#40035857)

Are you saying that explosions are as loud as dialog in real life too?

Re:If you want to impress me (0)

Anonymous Coward | more than 2 years ago | (#40035991)

75 foot tall talking robots arent real, doesnt mean they have to be to be enjoyable

Re:If you want to impress me (0)

Anonymous Coward | more than 2 years ago | (#40037279)

A nuclear explosion on TV won't blind you, so why should it deafen you?

Re:If you want to impress me (1)

asn (4418) | more than 2 years ago | (#40035865)

This is because the sound SHOULD have a large dynamic range in a movie -- conversations are not as loud as things exploding.

If you want to compress the range, most modern receivers have an option for that...

Re:If you want to impress me (1)

ourlovecanlastforeve (795111) | more than 2 years ago | (#40035885)

THX has a technology that is available on some laptops and home tuners called Smart Dialog Plus that does that. Also if you're using surround sound the dialog usually comes from the center channel and the left and right speakers. You can enhance the clarity of the dialog by increasing the volume of the center channel and adjusting the center channel width to a lower setting.

Re:If you want to impress me (2)

TigerPlish (174064) | more than 2 years ago | (#40035937)

make a system that amplifies dialog to the same level as everyfucking thing else in the movie so I dont have to constantly fiddle with my remote. Why is it in music we have the loudness wars where all sound is mashed into mindless noise at the peak of volume, but in movies there HAS to be a 100db difference between scenes

So let me get this straight - you're cutting down music because of the loudness war, but you want THE SAME THING in movies? Shoot, you already have it! Just pick the mix with the most letters and acronyms in the name!

I'll give you one example, and I hope you have this dvd and a shit-hot hi-fi to go with it so you can duplicate it.

2007's Titanic release, the 3-disk set in the blue case. This one has a "5.1 dolby mix" that I wager most people use -- this is what I call the "muggle mix." For people who don't know any better. THe dialog and music are fairly close -- in fact, the dialog is too loud. This mix is compressed, just like pop music. I play this one with the volume at -52db. (95db 1w 1/m speakers.) It sounds "meh". Sure, you hear everything, and everything's fairly close, but it's "meh". Just like compressed pop.

THen there's the 2.0 Dolby Stereo mix. This is the one you want, if you want it to sound like it did in a theater. This one's uncompressed. To get a natural dialog level, I set the volume at -36 or -34, depends on my mood. At this level the sound is completely natural. WHich means when people whisper, they whisper. When people talk, they talk. When they yell, its getting loud. When Rose makes her trek down E-Deck to bust Jack free, the whole house shakes along with the boat -- and is one of the best demo bits I've ever heard for movie sound.

Same with classical music. I play most of it on the same rig as above at -36 or -34. It's soft when the orchestra's soft, and it's fucking LOUD when the conductor sticks the baton up the orchestra's collective ass.

But when I play compressed pop, it's down to -52 for moderately compressed stuff (squirrel nut zippers) and -62 for MECO's Star Wars disco thingy, which is probably the most compressed music I have.

Movies have huge dynamic range. You can either accept this, or play the muggle track.

Or, get into your receiver's or source's setup, pick DRC = ON compress the snot out of it yourself. Every DD / DTS receiver or prepro has it. It may be called different things, but it's Dynamic Range Compression.

And it's the Devils Work. It should be banned from all recordings.

Re:If you want to impress me (0)

Osgeld (1900440) | more than 2 years ago | (#40036019)

"So let me get this straight - you're cutting down music because of the loudness war, but you want THE SAME THING in movies?"

yes I would like to enjoy a movie without having to constantly fuck with the remote every two seconds, or go though a zeroing process tween media and playback devices( that appears to be longer than the install instructions for ubuntu) for every disk that pops up in the mailbox.

Re:If you want to impress me (1)

Lunix Nutcase (1092239) | more than 2 years ago | (#40037169)

Then use the dynamic compression mode offered by your receiver. Some of us actually like movies to have dynamic range.

Re:If you want to impress me (1)

smellotron (1039250) | more than 2 years ago | (#40037317)

THen there's the 2.0 Dolby Stereo mix. This is the one you want, if you want it to sound like it did in a theater. This one's uncompressed.

This is interesting; I have always assumed that the stereo mix was just a mixdown of the surround mix, and I've not done any A/B tests. Do you know if this is a common phenomenon?

Re:If you want to impress me (1)

smellotron (1039250) | more than 2 years ago | (#40037275)

make a system that amplifies dialog to the same level as everyfucking thing else in the movie so I dont have to constantly fiddle with my remote.

Audyssey is one system that does this, but I'm sure there are others. If I crank my receiver's "dynamic volume" to "heavy" it substantially reduces the dynamic range, which is good for Michael Bay movies while the baby sleeps. It also destroys any music that has dynamic range, so I'm quite glad to have source material with a wide dynamic range and an optional compressor in my playback device.

Pointless (1)

Anonymous Coward | more than 2 years ago | (#40035775)

Purely a marketing stunt. Audio has been recorded at 44kHz for ages now because a signal sampled at that rate can be accurately reconstructed up to 22kHz (Nyquist theorem). Human hearing peaks out around 20kHz at best. Even 22 kHz is a pretty lenient upper bound, most of us will not be able to hear frequencies in the upper teens kHz (think of those mosquito ringtones). 96 kHz is severe overkill and nothing more than superfluous data.

It sounds like they're using the extra spectrum to do some processing on the signal, but there's no reason not to do the processing and then just downsample back to 44 kHz for storage/streaming/what have you.

Apodizing Filter (4, Informative)

Josuah (26407) | more than 2 years ago | (#40035811)

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 [wordpress.com] .

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.

Re:Apodizing Filter (3, Informative)

slew (2918) | more than 2 years ago | (#40036029)

A fancy name, but it seems to me that this mostly just a DSP textbook minimum phase filter [wikipedia.org] . A minimum phase filter is just a causal filter which has minimal group delay means it can be made to sound more "analog" (since analog filters are usually mostly causal meaning there is no filter contribution from the "future"). This is of course basically tipping the hat to the sound those vinyl/tube-amp purists claim is the "best" sound (not that vinyl/tube-amp purists would actually like this as it is still an soul-less digital approximation, although perhaps listening through oxygen-free copper speaker wire might help ease them to this "approximation")...

I guess having "minimum" in the name isn't a good marketing technique thus "apodizing"...

Digital harshness? Pre-ringing? (0)

Anonymous Coward | more than 2 years ago | (#40035839)

Next they'll be saying I need Monster cables to give my audio a truely analog-sounding experience.

who records 'expensive movies' at 48k? (2)

TheGratefulNet (143330) | more than 2 years ago | (#40035853)

do you have a cite for that? I don't believe it.

even home recording is laughed at (technically) if you are not using 24/96. recording at 48k is just absurd. playback at 48k is fine, though; but I'm not at all convinced that million dollar (at least) movies capture audio at 48k.

if that really is true, then people have been ripped off on their blue ray purchases. one of the supposed benefits is 'better sound' and if you still get 48k (and likely 16bit audio too; as its not common to use 48/24 mode) at record time, nothing the BD can do will ever make it better than dvd. yes, dvd uses compression on dolby 5.1 or dts but its compression is actually nearly lossless *compared* to most consumer playback (not a huge S/N dac+preamp+amp) systems.

Re:who records 'expensive movies' at 48k? (1)

Anonymous Coward | more than 2 years ago | (#40036053)

Recording at 48kHz is absolutely acceptable. There is absolutely no audial difference in playback. 24/48 is the standard for this reason.

The lower sample rate becomes a problem in production and mixing of the audio. Because some DSP can introduce Aliasing noise, using a higher sample rate will move the noise into supersonic frequencies which are generally known to be inaudible. Then, the signal is Downsampled to 48kHz, and in this downsampling, the aliasing noise is removed by a lowpass filter. Presto, cleaner audio for the final media. It is assumed that any DSP process after the downsampling is alias-safe, or else noise will be generated again.

Any audible noise you hear in compressed digital recordings today is the result of encoding. The filter discussed here is designed to rid some of that. Low quality encodings will always generate noise, and high quality encodings like used in some films have far less of it.

Re:who records 'expensive movies' at 48k? (1)

sahonen (680948) | more than 2 years ago | (#40036591)

> even home recording is laughed at (technically) if you are not using 24/96

This is mostly home recordists one-upping each other. Actual professionals in the audio industry, especially people working on gigantic projects like movies where halving your DSP/CPU/HDD needs is a direct benefit of 48k over 96k, recognize that there are very very few actual audible benefits to 96k over 48k.

Re:who records 'expensive movies' at 48k? (1)

dgatwood (11270) | more than 2 years ago | (#40036817)

even home recording is laughed at (technically) if you are not using 24/96.

Home recording is also usually laughed at if they go higher than 24/96, e.g. 24/192. 96 kHz is a sweet spot for a lot of reasons. In particular, it produces fewer artifacts and better accuracy when performing pitch detection and correction, and it correctly reproduces up to the maximum hearing range of the human ear instead of (in many cases) rolling off well below the Nyquist limit. I would also expect better quality when doing other things that involve Fourier transforms, such as convolution reverbs (if memory serves). Pedantically, you could achieve the same results by doing SRC to 96 kHz, applying the effect, and then reducing the sample rate back to 48 kHz, but in practice, the plug-ins don't do that, because those effects are computationally expensive enough without all that extra work.

Also, by recommending 96 kHz, you encourage people to buy gear that is relatively recent. The quality of audio amplification has improved significantly in the past twenty years. It's amazing how much better most newer preamps are than the off-the-shelf components were just a decade or so back—lower noise floors, lower THD, etc. Even if those folks decide to record at 48 kHz in the end, they're likely to get better sound than they would from most gear that maxes out at 48 kHz.

Re:who records 'expensive movies' at 48k? (2)

wiredlogic (135348) | more than 2 years ago | (#40036891)

Mixing at 24/96 has some merit in the name of reducing cumulative errors. You'll be hard pressed, however, to find an ADC that produces more than 16-bits of useful, noise-free data at 96KHz for recording.

Watermarking (0)

Anonymous Coward | more than 2 years ago | (#40036011)

The whole reason why there is any industry push for audio over 44.1KHz is to implement watermarking in the frequency ranges above 20KHz.

Same reason they want 32 and 48 bit color.

Placebo alert. (0)

Anonymous Coward | more than 2 years ago | (#40036041)

Unless it was recorded at 96k then this is about as good as upscaling from a dvd player is!

I don't care what filters they use, upsampling is just a gimmick, best to get 96k at the source.

nyquist? (1)

TheCouchPotatoFamine (628797) | more than 2 years ago | (#40036081)

Psst hey, nyquist called and wanted to ask you, what's the frequency, kenneth? //got nothing ///nothing like how your ear can perceive frequencies above 22k or so nothing. ////+3,000$, so your dog can enjoy a TrueHD experience, too! A bargain!

(really, if anyone wants to enlighten me as to why their technique of de-apoizing /requires/ that sample rate, please, let us know)

Re:nyquist? (0)

Anonymous Coward | more than 2 years ago | (#40036585)

Digital sampling / playback at any frequency can have the side effect introducing signals at playback that were inaudible in the original. A high frequency (>20KHz) signal that is sampled can look identical to a lower frequency signal. At playback you hear the lower alias of the original. So an orignal at about 45Hz will be aliased back right into the very audible 1KHz band with the standard 44.1 KHz sampling.

Audiophiles (of which I am *not* one) claim that the high fequency harmonics contribute greatly to the *feel* of the reproduced sound, which is why (they claim) that high quality analogue recording/playback is better.

No upsampling (0)

Anonymous Coward | more than 2 years ago | (#40036685)

Upsampling is not going to fix shitty sound engineers and their crappy encodes.

Wait a second (1)

Noitatsidem (1701520) | more than 2 years ago | (#40036711)

April 1st was a while back... Just saying.

Pointless to store upsampling on disc, do it in HW (1)

guidryp (702488) | more than 2 years ago | (#40036993)

This is pure snake oil.

There is absolutely no point to store the up-sampled audio on disk. It is just a waste of space (and more licensing fees for Dolby)

It is extremely common for output DAC HW to do up-sampling and digital filtering these days. This already removes the ringing without the need for storing the up-sampled data, which is completely pointless. I doubt there is any modern DAC HW that is still using native 44.1/48 and analog filters in the output stage.

So this is total redundant nonsense.

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