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

This is...

[Anonymous Coward]

...worthless shit.

Re:This is...

[Anonymous Coward]

No kidding. A/B/X or GTFO.

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]

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.

[Anonymous Coward]

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.

[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.

[Anonymous Coward]

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.

[Em Adespoton]

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.

[Brett Buck]

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.

[TheGratefulNet]

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.

[jasomill]

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.

[mr9times]

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.

[crashumbc]

LOL, the "reviews" of those are hysterical

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.

[jasomill]

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.

[Anonymous Coward]

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.

[cvtan]

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.

[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.

[__aaltlg1547]

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.

[__aaltlg1547]

I hope you're joking.

Re:You cant hear it anyway.

[Anonymous Coward]

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

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.

[gman003]

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

He's joking.

Re:You cant hear it anyway.

[the eric conspiracy]

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

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

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

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

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

Re:You cant hear it anyway.

[sahonen]

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

Re:You cant hear it anyway.

[sahonen]

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

Re:You cant hear it anyway.

[Anonymous Coward]

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.

[__aaltlg1547]

Does it say how he disproved the sampling theorem?

Re:You cant hear it anyway.

[Anonymous Coward]

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

Re:You cant hear it anyway.

[TheRealMindChild]

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.

[Anonymous Coward]

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.

[the eric conspiracy]

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

"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.

[Jiro]

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.

[afidel]

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.

[squiggleslash]

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.

[jo_ham]

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.

[Anonymous Coward]

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.

[Anonymous Coward]

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.

[timeOday]

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.

[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.

[reve_etrange]

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

Re:You cant hear it anyway.

[old and new again]

or coat hangers

Re:You cant hear it anyway.

[old and new again]

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.

[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/

Re:You cant hear it anyway.

[sr180]

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.

[Anonymous Coward]

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.

[drkstr1]

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.

[bill_mcgonigle]

They're upsampling to 96K to do their filtering work. It'll get downsampled again after the filter chain is done with it.

You definitely can hear clipping damage if filters are done with too low of a sampling rate.

Have you ever taken an 8-bit-per-channel image and converted it to 16-bit to do image editing, then down to 8-bit again for output? If not, try that now, even just fiddling with the levels widget - you'll see what's happening in the histogram.

Re:You cant hear it anyway.

[sr180]

Not true.. There is a good reason, and its reducing your quantisation noise. You will increase your Signal to Noise ration by sampling higher, doing your processing and then filtering back down. In fact, doubling your sampling frequency gives you the equivalent snr increase of more than an extra bit. DSP cycles are dirt cheap in the recording stage, so why not?

Re:You cant hear it anyway.

[Anonymous Coward]

How does a sound engineer get to call themselves an engineer

Maintaining analog radio equipment during the pre-transistor years should be classified as something an engineer might do. Just see what goes into a modern large production to see the requirements of the job. There are so much that the name "engineer" is still fitting. Sound and acoustics related engineering decrees are available in universities. Signal processing is a sufficiently complicated subject as well for an engineer to do.

Re:You cant hear it anyway.

[Anonymous Coward]

Another sound engineer here.

Well, your first argument is a little moot because firstly the quantization noise of 16/48 audio is already well, well below the hearing threshold. Sure, it's higher than the absolute hearing threshold if you set the loud end to 120SPL (ie. to stun), but it doesn't matter. In order for that to matter you would have to live inside a completely sealed chamber with zero ambient noise and not listen to anything for days in order to get your ears sensitive enough to hear the noise, which sort of defeats the whole point.

Come to real life and....
1. The background noise in whatever room you're in is waaaaaaaay higher than the quantization noise, no matter what. Or if you use sealed headphones then the noise of the bloodflow.
2. If you get the noise down to zero by moving to a sealed, floating chamber, it still doesn't matter because even if you like your music very quiet, the music itself will be loud enough to desensitivize your ears from the noise.

And secondly, reducing quantization noise by upping the sampling rate is just plain wrong, and highly wasteful.

The reason is that binary-encoded decimal is log2, which is to say every single bit gets you 6dB less of noise. You add one bit (ie. make it 17bit audio) and the noise goes down 6dB. Note that "6dB down" means "half the linear amplitude" (ie. (1.0 - 6dB) = 0.5) because decibels (and the human ear) are themselves logarithmic.

Time, on the other hand, isn't logarithmic but strictly linear. This means that in order to get 6dB less of noise - twice the linear amplitude - you actually have to make it twice the sampling rate. If you only add one extra sample (ie. from 48000 to 48001Halways been about bullshit... Everyone that is smart is from the Michael Gerzon camp, which sadly failed back in the 70s when it could have actually gained market relevance.

Ironically Michael Gerzon worked on MLP, the compression technique under the skin of Dolby TrueHD.

Re:You cant hear it anyway.

[Anonymous Coward]

Oops, this part got borked up:

(ie. from 48000 to 48001Hz) the difference is going to be almost nonexistent.

Then again, Dolby has always been about bull...

Captcha: sample

Re:You cant hear it anyway.

[fluffy99]

Not true.. There is a good reason, and its reducing your quantisation noise. You will increase your Signal to Noise ration by sampling higher, doing your processing and then filtering back down. In fact, doubling your sampling frequency gives you the equivalent snr increase of more than an extra bit. DSP cycles are dirt cheap in the recording stage, so why not?

Huh? Your sampling SNR is driven by the number of bits not the sampling rate. Higher sampling rates allow you to place your analog filters higher up, and then later digitally filter or decimate down to whatever sampling rate/cutoff you want..

Re:You cant hear it anyway.

[Prune]

Eh? The grandparent posted a link to peer-reviewed research published by the AES, a serious engineering journal. To imply that the AES is going to be shilling for an audio company is like implying that the Lancet is going to be shilling for a medical device manufacturer.
It's really sad how people like you who jump to conclusions and have an ideological axe to grind have brought down the level of discussion on Slashdot over the last few years. This place actually used to be good.

Re:You cant hear it anyway.

[Anonymous Coward]

I disagree, capacitance in the feedback path of an amplifier can provide frequency compensation which leads to stability. Also, negative feedback at AC plays a critical role in active filter design.

Re:You cant hear it anyway.

[Prune]

Just read the paper. Otherwise, this sort of discussion is worthless. This shit isn't even new. Apodizing filters were integrated into the DACs made by several major semiconductor manufacturers years ago after blind tests showed preringing was audible. People also conveniently forget that the fourier transform-analogue that the ear carries out uses a very narrow time window.

Re:You cant hear it anyway.

[Prune]

This has little to do with the 96 kHz part and everything to do with the preringing-reducing filters. The article summary is misleading.

Re:You cant hear it anyway.

[Prune]

The article summary is misleading. This isn't about removing sampling artifacts, but about removing reconstruction artifacts (by the DAC's digital filter)--specifically, preringing. The 96 kHz thing here is a red herring. And as usual, Dolby is way late to the game. A few major semiconductor manufacturers added apodizing filters to their DAC chips years ago after studies showed preringing was audible.

Re:You cant hear it anyway.

[Anonymous Coward]

"Just pay money so we can continue this discussion."

BULLSHIT.

Re:You cant hear it anyway.

[Ed Avis]

I assume the experiment used a high quality 44.1kHz digitizer and DAC. But what about lower quality hardware? If they're done less than perfectly, the digital to analogue or analogue to digital conversions may perceptibly change the sound even at 44.1kHz. One reason to use higher frequencies might be that a 100Hz ADC-DAC pair is always imperceptible, even if built using cheap hardware.

Re:You cant hear it anyway.

[madprof]

I wish I had mod points.

Re:You cant hear it anyway.

[Hognoxious]

50%? It'd make Schubert sound like Slipknot.

Re:You cant hear it anyway.

[Anonymous Coward]

Nah, we all know that the *real* reason for going to 96k is to drive the viewer's dogs crazy without the viewers knowing what's going on ;)

Re:You cant hear it anyway.

[Anonymous Coward]

I have recently upgraded my ATI GPU driver from 12.3 to 12.4 and thought "Wow, much faster and AA quality is better!" -- and then I realized a few days later that the update had failed. :-P

Re:You cant hear it anyway.

[Anonymous Coward]

That paper describes a down-sampling filter: for when you have a 96kHz track and want a 48kHz track without aliasing. It's got pretty much nothing to do with placebo up-sampling like this Dolby system.

Re:You cant hear it anyway.

[Anonymous Coward]

Using it in the wrong room (Or not putting any money into modifying the room).

http://www.cepro.com/article/beatles_pink_floyd_engineer_alan_parsons_rips_audiophiles/

Re:You cant hear it anyway.

[sociocapitalist]

Or Denon ones for that matter...

Re:You cant hear it anyway.

[nothings]

I posted about this on twitter a month ago.

The frequency chosen had to be a multiple of 900 and had to be somewhere in a limited range of frequencies (above 40Khz, below some number I forget). The 900 comes from a factor of 300 (to guarantee it was divisible by 50 and 60 for PAL/NTSC), and a factor of 3 (the preferred number of samples per scanline; 2 was too few, 4 would have been wasteful).

There is no evidence that the specific multiple of 900 from the required range (40Khz to 47Khz) was chosen because of what the factors of the multiplier would be, but rather because the frequency wanted to be as high as possible (giving a wider region between limits of human hearing and the nyquist freq, thus making filtering it cheaper), but higher frequencies would have required encoding samples in the vertical blank part of the signal.

Certainly the fact that 900 itself is already the product of the squares of the three smallest primes is coincidence, since the factors of 300 and 3 were essentially independently motivated--the 3 wasn't chosen because it "completed the set" with the 300. Likewise, I don't believe the additional factor of 49 was chosen because of that factor. (Having more divisibility is useful in some circumstances, but 7 is such an uncommon divisor; 900*48 would be far more useful on the general divisibility front, introducing more factors of 2 and 3. But, in fact, nobody NEEDS this number to be more divisible, as it's not needed to be divisible beyond the factor of 900 that was required.)

There's a wikipedia page about it (do an "I feel lucky" search on google for "44.1").

Re:You cant hear it anyway.

[Anonymous Coward]

...inserted into a high resolution analog audio source

I thought analog stuff had INFINITE resolution anyway... did you mean "low noise" or "high quality" ?

Re:You cant hear it anyway.

[thegarbz]

I shudder to think of the number and the size of polystyrene caps (typically pF range) you would need to replace tantalum / electrolytic caps (typically uF range). :-)

Polypropylene on the other hand I agree with you wholeheartedly.

Re:You cant hear it anyway.

[thegarbz]

The story I heard is that this is the reason why CDs typically were 74 minutes and that specific size, rather than some other more rounded number like the 80 minute discs that came later.

I heard Sony was actually arguing about REDUCING the sampling frequency down to 32kHz as most adults can't hear above about 15kHz anyway. Management wanted smaller discs but were met with resistance from designers who wanted better audio.

Re:You cant hear it anyway.

[pipatron]

I wish more audiophiles would realize this very basic property of the human psychology that you just now described.

Re:You cant hear it anyway.

[TheLink]

You might like reading some of the ones for a related product:
http://www.amazon.com/Denon-AKDL1-Dedicated-Link-Cable/dp/B000I1X6PM

Re:You cant hear it anyway.

[Anonymous Coward]

> NTSC is 59.97Hz, not 60Hz

*Color* NTSC is 59.97, but black and white NTSC is an even 60Hz.

Re:You cant hear it anyway.

[Anonymous Coward]

Or even better, Pear Audio Cables (several thousand $$$ per cable).

Re:You cant hear it anyway.

[Anonymous Coward]

Customers who view this also viewed:

- Tuscan Whole Milk, 1 Gallon, 128 fl oz
- Uranium One
- Canned Unicorn Meat

What?

Re:You cant hear it anyway.

[Anonymous Coward]

NTSC is 59.97Hz, not 60Hz

*Color* NTSC is 59.97, but black and white NTSC is an even 60Hz.

Actually, color NTSC is 59.94 Hz frame rate - double the 29.97 Hz field rate.

Re:You cant hear it anyway.

[Anonymous Coward]

No, he's right, it's because in the early 80s the only practical and cheap way to record digital audio was to use analogue videotape.

The CD sample rate was chosen based on the amount of information you could get in a single frame of video, with either an NTSC or a PAL video recorder.

NTSC: 245 active lines/field × 60 fields/second × 3 samples/line = 44,100 samples/second

PAL: 294 active lines/field × 50 fields/second × 3 samples/line = 44,100 samples/second

Re:You cant hear it anyway.

[squiggleslash]

Schubert didn't compose all the classical music in the universe! Even so, 50% isn't that big a range, when you consider the difference between the extremes and the midpoint.

Beethoven is very liable to be conducted at wildly different rates. Even the waltzes of Johann Strauss, which you'd expect to be conducted at similar rates given the fact they're... well, meant to be danced to, differ wildly.

One of the best stories I heard about this issue was that Holst's Planets tended, until recently, to be conducted "in a hurry" with conductors generally having each part be around 3-4 minutes long. Why? Because Holst is young enough to have conducted his own music and had his performances recorded. On gramophone records. That had a maximum length of... three and a half minutes.

Play the same pieces at a more leisurely rate and they sound rather better, like, perhaps they were intended to be. It's unlikely Holst composed the music for the recordings, but he wouldn't be the first artist to accept compromises on a final performance in the name of technological expediency.

Re:You cant hear it anyway.

[Anonymous Coward]

which supposedly would not of been possible

Congratu-fucking-lations. I didn't think there were any more words that fuckwits would put 'of' to instead of 'have.' You've found another one!

Re:You cant hear it anyway.

[squizzar]

Yup. For once AC is absolutely right and I have no mod points to waste on him...

The two base values for frame rates in video correspond to a 1/1 and a 1/1.001 version (e.g. 60 and 59.94, 30 and 29.97, 24 and 23.98 etc. etc.). If you look at broadcast equipment you will find two (or maybe three if they add a 27MHz one) crystals - a 74.25 and a 74.17 (74.25/1.001) or some multiple of those numbers.

Wikipedia will explain the reason for the 1/1.001 divide - see NTSC Colour encoding. That 1/1.001 ratio is small enough that most PLLs could be pulled to lock to it so existing 60Hz black and white sets could still receive a colour signal at 59.97Hz. Clever solution leading to ridiculous legacy situation now where we have more broadcast video standards than we know what to do with.

Re:You cant hear it anyway.

[Anonymous Coward]

D'oh! I got it reversed, though.

Frame rate is 29.97 Hz.
Field rate is 59.94 Hz, two fields per frame.

Re:You cant hear it anyway.

[mcgrew]

Those studies had many flaws. Had they actually recorded three 15kHz tones with different waveforms (one sine, one sawtooth and one square wave) in both digital 44k (not kHz; it's samples per second, not cycles per second) in both analog and digital the difference would have been apparent to almost anyone. They measured the reaction to music rather than discreet tones.

However, the new Dolby strategy isn't going to sound any different to most people, as most folks simply don't have speakers that good. The speaker is the most important part of your sound system, and today's are usually really BAD, little four inch two ways with a "sub" woofer when older high end speakers had fifteen inch or larger woofers (larger than your subwoofer) and "supertweeters" capable of reaching supersonic frequencies (usually up to about 30 kHz). The supertweeters would be useless to me, as most tweeters can reach close to 20 kHz, farther than I can hear, but some young folks can her tones up to 22 kHz.

Unless you have $1000 apiece speakers, you probably won't hear the difference.

Re:You cant hear it anyway.

[Anonymous Coward]

Only McGrew could nitpick Hertz as a unit for sampling rate (it's common and perfectly appropriate) and go on to say "supersonic" instead of "ultrasonic."

Re:You cant hear it anyway.

[b4dc0d3r]

Pretty sure that's irrelevant to the point GP was making. For the same master, vinyl has lower fidelity than a CD. You're talking about using a different master for different media.

The vinyl preference, for the same master, is usually attributed to being "warmer", or more like sitting there while someone is playing live. The digital media are described as being "cold" or "impersonal". I think this is mostly due to the presence of upper harmonics, giving the sound a more brash sensation.

I think the higher fidelity gives people more audio "data" than they really need. It's almost the same concept as a strip tease (subjectively) being better than actual nudity - your brain fills in the parts that aren't there, and your imagination is usually better than reality. In psychoacoustics, you can play the harmonics of a note and your brain actually can fill in the fundamental, hearing a note lower than any that was played. So it is not unheard of to have your auditory system do this.

Re:You cant hear it anyway.

[Anonymous Coward]

Yeah, they have the stupid music, special effects sounds, & background crap so jacked up you can't hear the dialog anyway. So who the frack cares??????

Re:You cant hear it anyway.

[Anonymous Coward]

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.

I see what you did there...

Re:You cant hear it anyway.

[fluffy99]

I wish I had mod points.

I wish I had a big stick to whack all these so-called experts that are spouting total BS. What makes it worse is that some of these guys claim to be in the sound recording business and really don't understand the electronics or mathematics underlying the equipment they use.

The system I work on cost somewhere around 4 million, has 1500 channels, recording at various rates from 32k up to 192k. We calibrate it end-to-end and certify our measured data to +/- 0.2 db. So some twat on here claiming his system is good to 0.01 dB with no artifacts is really smoking something.

This Dolby system is really just trying to estimate the sampling artifacts and back them out of the digital data. The whole higher sampling rate thing is a smokescreen. As someone else pointed out, Dolby makes their money by licensing and they really want to convince consumers that they need this in their equipment.

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

[loufoque]

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

[jonsmirl]

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

[Anonymous Coward]

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

[Tablizer]

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

[sahonen]

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

[smellotron]

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.

Re:Worthless gimmick with no audible benefits

[DigiShaman]

Question. In theory, you could turn those stair stepped signals into vectored splines. Does anyone already do this?

Re:Worthless gimmick with no audible benefits

[Anonymous Coward]

Lots of software and hardware does this. I personally hate the outcome. There is no guarantee, when sampling frequencies at the high end, that there were supposed to be nice smooth splines. You don't have the resolution, and so you muddy the high end. I always turn resampling off and use the native rate where possible, it sounds more true.

Re:Worthless gimmick with no audible benefits

[Anonymous Coward]

Why deal with polynomials when sines are so easy to produce electronically?

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.

Re:Worthless gimmick with no audible benefits

[Anonymous Coward]

Even a cheap DSP processor or FPGA can apply those sort of filters easily. A signal sampled at 48 kSa/s is a walk in the park for most DSP platforms actually.
I'm going to guess they just upsampled it to 96kHz and then applied an extra low pass filtered to it. Maybe an adaptive filter to smooth out the envelope a bit. And now they're going to claim it's more fluid or something like that.

Re:Worthless gimmick with no audible benefits

[Prune]

The article summary is misleading. You should have done more digging before posting here, such as reading the peer-reviewed research that deals with the topic in question: http://www.aes.org/e-lib/browse.cfm?elib=12992
The issue at hand is removing artifacts on the reconstruction side, specifically, preringing, which audible outside the audio band (the ear uses a very narrow fourier transform window and is highly sensitive to preresponses; follow the references in the paper). Of course, Dolby is way late to this as I remember a major semiconductor manufacturer adding apodizing filters exactly for this reason to their DAC chips years ago. Dolby's approach is also stupid as this technique should be performed purely by the digital OS filter on the DAC side. Instead, Dolby does it during mastering, so that now you need to store 96 kHz data... and this way the apodizing is not nearly as effective than on the DAC side where audio usually is oversampled 8x rather than 2x (normally done so slow-rolloff analog filter can be used to remove the HF images and prevent them from causing nonlinear effects and intermodulation in the subsequent analog electronics).

Re:Worthless gimmick with no audible benefits

[Prune]

The artifacts are not during recording, but during reconstruction on the DAC side. Of course, Dolby is way late to the game as usual. A major semiconductor manufacturer added apodizing to their DAC chips' oversampling digital filter years ago to remove preringing. It makes a lot more sense than Dolby's approach as well, because 1) you don't need to _store_ the audio at 96 kHz the way Dolby is doing it during mastering, and 2) the DAC digital filter usually oversamples to 8x (to make the subsequent analog filter simpler with a slower roloff) so that the apodizing can be done with more precision. The article and article summary are both really bad in this case.

Re:Worthless gimmick with no audible benefits

[Prune]

You don't need a DSP. DAC chips whose integrated oversampling filters had the option to be set to do apodizing to tweak the preresponse have been out for years. A couple of semiconductor manufacturers have been producing such. I think one of those was Wolfson or Burr Brown. This also shows how little sense Dolby's approach makes. They have to oversample during mastering in order to apply the filter, and then you have to store twice the data on the Blu-Ray or whatever. Whereas if it's on the DAC side, you don't have this inefficiency and can feed it the usual 44.1 or 48 k, plus the filter is much better because it's 8x rather than 2x oversampling. On top of this, if your equipment already uses a DAC chip with such apodizing filter, then you feed it this Dolby format, you'll end up with even stronger filtering. This is an epic fail by Dolby, though I suppose not as bad as squandering the only true HDR display technology when they swallowed up Brightside.

Re:Worthless gimmick with no audible benefits

[L4t3r4lu5]

We're all happy to laugh at CSI when they "enhance" a digital photograph... Why is it so difficult to explain why you can't up-sample digital sound?

Analogy: Put a 1280x1024 (1.3MP) resolution photo up on your monitor, and take a photograph of it with your 20MP DSLR. Do you think you now have a 20MP photo of that scene?

Re:Worthless gimmick with no audible benefits

[Anonymous Coward]

Yes, it's a called low pass filter. These can be implemeted in software with an FIR filter, or in analog hardware.

Re:Worthless gimmick with no audible benefits

[jonsmirl]

Your optical cable can do two channels of uncompressed or 5.1 of highly compressed - usually AAC or MP3.

I agree that Dolby was a real benefit back in the stereo days. But the need for Dolby processing disappeared with HDTV and Blueray. Dolby doesn't want to give up the $100M/yr in licensing fees so they 'invented' pointless things to mess with HD audio. They are taking advantage of the brand awareness built back when Dolby was a valuable feature. Every device you get with a Dolby label carries $2-3 in licensing fees. Keep Dolby for stereo processing, but get rid of it on HD audio.

Re:Worthless gimmick with no audible benefits

[Anonymous Coward]

There is no guarantee, when sampling frequencies at the high end, that there were supposed to be nice smooth splines.

Incorrect. Anything that you cannot reconstruct using these "smooth splines" (actually, sums of sinusoids), is, BY DEFINITION, outside the sample rate you chose to begin with. If you wanted to reproduce that sort of detail, you would have chosen a greater sampling rate. It's not that the sampling represents these higher frequencies poorly -- it does not represent them at all, period. They were never captured in the first place (or rather, they were aliased into other frequencies).

In a strict sense it is a matter of not having enough data, but this is not due to the sampling but to the sampling RATE, which is something YOU picked.

Re:Worthless gimmick with no audible benefits

[Anonymous Coward]

A software filter cannot possibly compensate for DAC interpolation noise, as by definition, this intepolation noise is due to the time-quantization of the DAC itself. No digital magic can eliminate it. It is eliminated by an analog low-pass filter at the output.

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

Re:Lossless + Cinavia == Lossy

[Anonymous Coward]

Thanks for bringing up Sony's bizarro DRM tricks. Their new DVDs from 2011 onward probably play nice only on Sony's own DVD and BR players. I'm just trying to play those fuckers, for DRM's sake!

Re:Lossless + Cinavia == Lossy

[trifish]

Before anyone yells [citation needed] at me, here's your proof straight from the Wikipedia page (http://en.wikipedia.org/wiki/Cinavia):

Do you realize that Wikipedia is not a reliable source that can be cited? You could have inserted the quoted text there. Next time try a primary or secondary source.

Re:Lossless + Cinavia == Lossy

[serviscope_minor]

This is a discussion site, not a peer reviewed journal or research paper.

Re:Lossless + Cinavia == Lossy

[mug funky]

well, though it sucks, it's no more lossy than some lossless audio that has been mastered poorly.

cinavia is applied at the sound mix stage as an effect. this is well before it even hits the screen, certainly before it hits blu-ray. so the watermark is part of the mix, and signed off on by the film-makers. in that sense, what's on the blu-ray is a lossless representation of that.

that way even the cam rips will stop playing if the hardware is looking for it.

of course, all the knockoff chinese players will have no trouble. so it's just western media killing western hardware manufacture. (yes, i'm aware that sony is japanese, but lemme tell you, they have the heart and soul of a greedy fat wall street motherfucker)

Re:Lossless + Cinavia == Lossy

[mug funky]

only problem is a rip is indistinguishable from an unencrypted disc burnt for checking. it makes my job much harder if i can't check the disc that's about to get replicated (and have the AACS crap burnt into it) 10000 times.

Damn Dolby...

[wbr1]

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...

[Anonymous Coward]

You blinded me with science!

Different Dolby.

Re:Damn Dolby...

[cvtan]

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

Re:Damn Dolby...

[djdanlib]

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...

[JonySuede]

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...

[djdanlib]

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.

Re:Damn Dolby...

[JonySuede]

Thanks for the information, I have not ask a professional about that and at my next biannual checkup I will ask him an appointment with an audiologist !

And for the guitar you are right about the frequencies and the cheap distortion pedal that introduce a lot of almost random high frequencies. That is why, when I play the bass I use a BOSS-GT6B and feed either LPCM over SPDIF or balanced XLR to the sound board. I keep the jack connection for my clean amp an Ibanez SWR-100 from 2000 with the tweeter turned off ;) Sound console with SPDIF input are not common but when they have it, it is awesome. Never had I a sound man complain about the quality of the signal coming from my pedal however I had complain about my lousy style ;)

If you want to impress me

[Osgeld]

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

[Ogi_UnixNut]

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

[Anonymous Coward]

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

Re:If you want to impress me

[asn]

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

[ourlovecanlastforeve]

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

[PhrostyMcByte]

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

[TigerPlish]

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

[Anonymous Coward]

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

Re:If you want to impress me

[Osgeld]

"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

[Lunix Nutcase]

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

[smellotron]

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.

Re:If you want to impress me

[Anonymous Coward]

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

Re:If you want to impress me

[smellotron]

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

[Osgeld]

I dont have a receiver, some of us dont piss our money away on pointless toys, for a normal person with a TV its obnoxious to have to crank it full blast to hear dialog, and instantly crank it back down to 5% cause they cut to a scene with music, explosions, or whatever

Re:If you want to impress me

[Anonymous Coward]

Shut up you stupid piece of shit. The only thing you like doing is working your troll shift on Slashdot in your cubicle in New Dehli. Eat shit and die, you little faggot.

Pointless

[Anonymous Coward]

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.

Re:Pointless

[Anonymous Coward]

I found one audiophile stunt where they explained that higher frequency minimizes the hearing of the "jaggies" of the digital sound wave. They explained that LP/analog is the purest form and with only millions of Hertz you can reach similar quality.

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.

Re:Apodizing Filter

[slew]

A fancy name, but it seems to me that this mostly just a DSP textbook minimum phase filter. 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"...

Re:Apodizing Filter

[Prune]

Apodizing filters in audio were introduced a few years ago in DAC chip oversampling filters by Wolfson Micro. They are still FIR filters, with a compensation that allows one to smoothly set the amount of preringing reduction (really, shift to postringing so the filter can be continuously varied between linear phase and something similar to minimum phase). What's more interesting is that preringing outside the usual audio band still seems to be perceptible; follow the references in the paper http://www.aes.org/e-lib/browse.cfm?elib=12992

Re:Apodizing Filter

[Prune]

This is stupid since it's much more efficient to do it on the DAC side, instead of having to store twice as much data. Indeed, some DAC chips do exactly this by allowing you to vary whether you want linear phase, no preringing, or somewhere in between (plus, since DACs usually oversample by 8x, the filter there is much more precise).

Digital harshness? Pre-ringing?

[Anonymous Coward]

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

who records 'expensive movies' at 48k?

[TheGratefulNet]

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?

[Anonymous Coward]

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?

[sahonen]

> 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?

[dgatwood]

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?

[wiredlogic]

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.

Re:who records 'expensive movies' at 48k?

[Anonymous Coward]

It's trivial to get audo A/D converters with more then 16 bits of real data, getting more then 20 bits is a bit tougher. TI has a 96K/24bit A/D with 112db s/n for $1.70, hell it's not even expensive to get decent A/D converters anymore.

Re:who records 'expensive movies' at 48k?

[Prune]

In the last few years there have been quite a few ADCs that exceed 20 bit, but achieve that through aggressive noise-shaping and not throughout the full audio band.

Re:who records 'expensive movies' at 48k?

[xiphmont]

Parent post is talking about broadband SNR, not narrowband depth or SFDR. Very few ADCs/DACs reach 20 bits SNR. Almost all exceed 20 bits SFDR. Noise shaping has nothing to do with it; it actually penalizes SNR even though it is generally a [great] benefit to perceived fidelity.

Re:who records 'expensive movies' at 48k?

[Prune]

>Very few ADCs/DACs reach 20 bits SNR.
Huh? For a few years already DACs like those from ESS have exceeded that. ES9018 is beyond 22 bits.

Re:who records 'expensive movies' at 48k?

[xiphmont]

ESS's own specs list it at THD+N of 20 bits, and SNR can't be higher than that.

This is also an 'ultimate' DAC, supposedly the Terminator of the current generation (in ESS's own marketing anyway)

Re:who records 'expensive movies' at 48k?

[Anonymous Coward]

1. Many ADCs perform better at 48k than at 96k, with less noise/THD.
2. Many portable devices (used in the field for movies) can't record higher than 48k.
3. There's no audible benefit for recording at higher than 48k. (In theory yes, in practice it has not yet been demonstrated.)
4. Just because recording at 96k or 192k is so cheap nowadays that everyone can do it doesn't mean it should be done. (higher != better)
5. People who "laugh at" recording at 48k (or 44.1k) should really get educated about this stuff.

Re:who records 'expensive movies' at 48k?

[Prune]

Dustin Forman (who designed the IC) told me in an email it was 22.5. My guess as to the discrepancy is that they're referring to the overall measurement of the eval board, so the limiting factor there is probably the Analog Devices opamp.

Re:who records 'expensive movies' at 48k?

[xiphmont]

Dynamic range (not SNR) is 22.5 bits ("How quiet the chip can get when it's doing nothing/off"). SNR is 20 bits at best given the 120dB THD+N spec.

Re:who records 'expensive movies' at 48k?

[Prune]

I just checked the datasheet and you're right.
Well that's disappointing. It's no better than what I had four years ago when I took four AD1955 and did some analog of dynamic element matching using a Sharc DSP.

Re:who records 'expensive movies' at 48k?

[xiphmont]

I'm actually giggling that 20 bits of real dept is 'disappointing' in any sense of the word. It's bloody fantastic! :-)
(I hadn't meant to trash the ESS chip in any sense. I've not used it, but it looks impressive)

Re:who records 'expensive movies' at 48k?

[Prune]

Well, human hearing, not including intensity levels beyond threshold of pain, covers 120 dB. Moreover, narrowband signals are perceptible when several dB _below_ broadband noise. This makes me say that audio is not quite a solved problem even on the electronics side (let alone on the speakers side). Various things seem to pop up, and not just from crazed audiophiles. Recently I was reading how it turned out that some amplifiers who were specced at very low THD only made that measurement at moderate signal levels, and had very large distortion for low signals (one ironic example was the hugely overpriced Halcro amplifiers). The thermal memory issue may be another thing.
I did an informal ABC/HR test a year and a half ago with four people, which was blind but not doubleblind as the switching was done by telling another person to switch. It was with the DAC I mentioned. I had two IV's which I could switch in after the DAC using a small relay. One was using the AD797. The other one was derived from Hawksford's discrete current feedback IV (figure 4-4 in http://www.essex.ac.uk/csee/research/audio_lab/malcolmspubdocs/C111%20Current%20steering%20transimpedance%20amplifier.pdf ). To get the distortion of the Hawksford IV near -120 dB (checked with rented distortion analyzer), we added extra gain so we could use more feedback. For testing, we tested each person one at a time using Stax electrostatic headphones fed by this hybrid transistor/tube amp, which is about 5 ppm THD: http://gilmore2.chem.northwestern.edu/images5/gilmore4_1.png
We did eight trials for each person. One couldn't tell the reference (5/8). The other three could tell it (averaging 7/8).
Did another test. There were two of the Hawksford IV per channel, set up as differential since the DAC and the amp were both balanced. Connecting the emitters of the current mirror transistors between the two separate IVs dropped the differential output 3rd and 4th harmonics further (slightly raised 2nd for single ended operation) (I'm pretty sure this infringes on a Nelson Pass patent...). I used a relay for that as well, and did the test switching that connection on and off. Two people could tell the difference (averaging 6.5/8).
Small sample size, not double blind, and ultimately anecdotal. Nonetheless it was sufficient to convince me that the audio hobby has more mysteries to reveal than seems at first.

Re:who records 'expensive movies' at 48k?

[xiphmont]

Regarding hte first point, that 120dB broadband noise figure is giving you at least 140dB of SFDR, probably much better, and the depth of any critical band is going to be even better yet. Even 16 bit data with a decent noise shaper is going to be 120dB deep in the 2-4kHz critical bands. (all of which doesn't disagree with anything you said of course)

Re:who records 'expensive movies' at 48k?

[Anonymous Coward]

I know for a fact that most dialogue in holliwood movies are recorderded at 48000 Hz (sometimes 48048 Hz) with 24 bit depth.

I have heard that they mix at 96 kHz with floating point depth. Maybe they record the music at 96 kHz. Sometimes sound effects are also recorded at a high frequency since the amount of effect filters used would benefit from a better quality signal.

Re:who records 'expensive movies' at 48k?

[Anonymous Coward]

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

Umm, try all of Hollywood, and anyone who makes video for TV. 48kHz is baked into the ATSC standard. It's not a secret.

Watermarking

[Anonymous Coward]

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.

[Anonymous Coward]

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?

[TheCouchPotatoFamine]

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?

[Anonymous Coward]

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.

Re:nyquist?

[Alex Belits]

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.

And this is why you filter before you digitize. What all recording equipment already does.

Re:nyquist?

[TheCouchPotatoFamine]

so no further answer? i /am/ curious, as i still don't see the advantage (N-bit mixing headroom is one thing, but i don't see it here. Ah well)

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?

[hairyfeet]

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!

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?

[Taco Cowboy]

... It's crucial that you place the softening strips in your mouth ...

LOL !!

It's fortunate that visitors of /. have the ability to recognize a joke, no matter how heavy it is laced with jargon

If someone were to copy-pasta what you just written onto a dimwit site, someone may just probably put that stuff in their mouth ....

But anyway, thanks for the laugh !

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

[L4t3r4lu5]

Without this step you may as well be using a walmart SPDIF cable, it will be that bad.

HAHAHA they may as well use wire coat hangers XD

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

[msobkow]

There is an easy way to implement such algorithms if you're manufacturing your own equipment: include a tube circuit in the pre-amp. Tubes naturally have electric "mass" that has to be "moved" by the changing signal strength, smoothing out the raw digitial samples into a proper analogue curve. There used to be a few high-end CD players that incorporated mini tubes, and they sounded far more natural and less harsh than pure digital signals.

When dealing with digital inputs, too much accuracy in the audio stream produces harshness and digital fatigue.

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

[johnwbyrd]

This is incorrect. Interpolating an audio signal using using lerp or Bezier or whatever will introduce auditory artifacts in the upper frequencies of the sound. The only mathematically correct way to upsample a signal is to perform the transformation into frequency space and then resynthesize the signal at the desired frequency with a lowpass filter.

See https://ccrma.stanford.edu/~jos/resample/ for more information on why curve fitting is incorrect.

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

[julesh]

Tubes naturally have electric "mass" that has to be "moved" by the changing signal strength, smoothing out the raw digitial samples into a proper analogue curve.

You can say exactly the same thing for a capacitor, and they're much cheaper, less fragile, and get to their operating point much more quickly when the circuit is switched on.

You can prefectly represent anything up to Fs/2

[robbak]

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

[Anonymous Coward]

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

[Man On Pink Corner]

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

[Anonymous Coward]

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 can prefectly represent anything up to Fs/2

[diamondmagic]

A "square wave" at the given frequency is a periodic function(t){return (t<0.5)?1:-1;} that repeats at said frequency. So yes, there is, by definition.

You're correct that because there's harmonic frequencies above the fundamental ad infinitum (for square, triangle, sawtooth, and really any wave with non-differentiable points), it's impossible to perfectly capture all three types of waves/functions with the same method of digitizing.

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

[jedwidz]

I have a sine wave, triangle wave and square wave at 20khz.

What makes you so sure?

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

[jedwidz]

There's no such thing as a square wave at a given frequency.

Not so. 'Frequency' just relates to how often a repeating phenomenon repeats. This isn't limited to sine waves.

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

[Anonymous Coward]

> Can be represented by or is?

There are three factors at work here that make your distinction irrelevant:

1) How air molecules bounce off each other, propagating waves

2) how hair cells in the cochlea respond to vibrations

3) noise in the signal

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.

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

[Man On Pink Corner]

The point is that you cannot distinguish a square wave from a sine wave at the same fundamental frequency, if you can't hear the odd harmonics. You cannot have a square wave at a given frequency without the odd-order harmonics. If you don't have the odd harmonics, you don't have a square wave -- you have a sine wave.

Nitpicking arguments about the frequency of a signal in the time domain are not relevant. Human hearing operates in the Fourier domain -- almost literally, if you understand how the cochlea works -- not the time domain.

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

[Anonymous Coward]

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

Fourier transformation doesn't "break down" a signal into a representation made up of sinusoidal waves, it is basically a different way of looking at the same phenomenon. There's nothing in a time-domain signal that isn't there in the frequency domain; they are mathematically equivalent.

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

[Prune]

The ear operates both in the time and frequency domains, in a manner analogous to using a very short fourier transform window when calculating a waterfall plot. As for sound above 20 kHz not being audible, studies show 120 kHz is perceptible through bone conduction: http://en.wikipedia.org/wiki/Ultrasonic_hearing and also see http://ieeexplore.ieee.org/iel5/5286202/5291232/05291285.pdf?arnumber=5291285 and other related studies showing ultrasound that is not necessarily consciously perceptible does affect perception of music.

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

[maroberts]

Well said. I would mod this up if I had points and is a result of a failure to understand what is being said.
The GP question about representing square, triangle and sine waves at a certain frequency is a common misunderstanding.

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

[Prune]

Of course you're right about the square wave issue. My comment is regarding your last sentence. The DAC reconstruction filter is analog, not digital, and has a roloff with a limited slope. I think you're confusing it with the digital oversampling filter which brings Fs to around 8x in most configurations so that the images are at a high enough frequency that the analog filter response is sufficiently low dB at those frequencies to essentially remove them.

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

[Prune]

The primary mechanism of ultrasound perception seems to be bone conduction: http://en.wikipedia.org/wiki/Ultrasonic_hearing and also see http://ieeexplore.ieee.org/iel5/5286202/5291232/05291285.pdf?arnumber=5291285 and there were some other related studies showing ultrasound that is not necessarily consciously perceptible does affect perception of music. In any case, for this article the 96 kHz thing is a red herring. The audible difference is due to the use of filters other than the usual symmetric FIR filters which cause preringing in reconstruction. Of course, Dolby is way late to the game here, as a few major semiconductor manufacturers added apodizing filters to their DAC chips years ago after people realized preringing was audible even outside of the usual audio band. See the paper on apodizing filters and preringing: http://www.aes.org/e-lib/browse.cfm?elib=12992

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

[stewbee]

You can perfectly represent any signal with a frequency less than half of your sampling frequency.

Sorry to nit pick, but this not entirely true for ANY signal. It only applies to real signals (ie no imaginary part). If you have a signal which has a real and imaginary part, you can use the entire bandwidth up to Fs. A neat trick that I would recommend verifying yourself. I at one time thought the same thing until someone told me this gem.

As a simple experiment, suppose you have a Fs = 2khz. plot the fft of cos(2*pi*1100*t) and the fft of cos(2*pi*1100*t) + j*sin(2*pi*1100*t). What you will find is the the real signal will have two peaks in the spectrum (one at 990Hz, and the other at -990Hz), where as with the second signal you should see a single tone at 1100 Hz.

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

[Anonymous Coward]

Correction: you can imperfectly represent any signal with a frequency less than half of your sampling frequency, and get back a signal with the same frequency. It won't be a perfect representation. It will just be the correct frequency.

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

[Anonymous Coward]

He was sure, until you told him to downsample them to 20khz. Now he's asking you to tell him which was which again. Since they're "prefectly" represented, I'm sure you'll have no trouble.

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

[Man On Pink Corner]

Thanks for the pointers; some interesting stuff there. I am not up to speed on the physiological and psychoacoustical angles.

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

[Anonymous Coward]

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

Actually, that's an interesting comparison. There is a direct analogy from epicycles to sound vibrations in the air.

A 50-Hz sound wave is represented in air by molecules bumping into each other, and the air pressure measurement during compression-rarefaction over time will be sinusoidal.

Add to that a harmonic at 150 Hz. The air compresses and rarefies faster, but the motion from the 50 Hz wave is still there. The new frequency is superimposes on the "jiggle" of the old wave like an epicycle. It too is sinusoidal. This is true for any frequency you transmit through the air.

So in essence, the "waveform" is the composition of all the frequencies spreading through the air at a given point (where you place your ear/mic). Thanks to interference patterns, phase differences, and reflection/refraction, the same "waveform" will look very different at other points in space.

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

[jedwidz]

No problem with your overall post - note I also replied to AC asking if he was really sure if had three different waveforms to begin with, since they should all sound the same to a human.

My point was that AC was fine specifying a square wave and a triangle wave just by their frequency. Or he/she could've specified a period of 50us, which means exactly the same thing.

Reading your comment again, I almost agree with you that 'at a given frequency' suggests a frequency-domain view, as opposed to 'with a given frequency' which would suggest a time-domain view. But that's only by being really pedantic and reading between the lines. Anyway, it's clear to me that AC meant 'triangle wave and square wave with a frequency of 20khz '.

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

[Anonymous Coward]

20kHz square waves and triangle waves are not signals with frequencies less than 1/2 the sampling rate.

You understand that you have to apply a low-pass filter before sampling, right? And that low-pass filter will turn your square and triangle waves into sine waves because the harmonics will get filtered out?

I swear, it's like you morons don't bother to read the first damned thing about sampling theory before jumping into a pissing match.

No upsampling

[Anonymous Coward]

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

Wait a second

[Noitatsidem]

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

Pointless to store upsampling on disc, do it in HW

[guidryp]

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.

lol wut

[Alex Belits]

pre-ringing

Really? In an uncompressed audio? And the solution not only involves upsampling as a part of the process but requires the signal to stay upsampled?

My eyes are rolling at 15krpm.

Re:lol wut

[Prune]

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

Re:lol wut

[Prune]

PS this has absolutely nothing to do with digital compression. The problem is the fourier transform the ear does is with a very narrow window, and a combination of nonlinear and hysteresis effects make preresponse perceptible even when its fundamental frequency is outside the usual audio band. Preringing can be fully avoided if instead of symmetric FIR filters (linear phase) one uses IIR filters (minimum phase). However, this has other problems associated with it, such as audible perception is the phase variation of the latter is not smooth enough. The solutions usually implemented are a FIR filter with some compensation (thus the word "apodizing") which allows one to specify where in between the two extremes it is.

Re:lol wut

[Alex Belits]

The problem is the fourier transform the ear does is with a very narrow window, and a combination of nonlinear and hysteresis effects make preresponse perceptible even when its fundamental frequency is outside the usual audio band.

This sounds like some serious stretching of the possibilities, considering that cochlea is an array of independent mechanical band-pass filters (thus not only narrow window for its fourier transform but also possibly inconsistent window per frequency) placed after a mechanical transmission line and a just as mechanical low-pass filter. No aliasing but plenty of things that have to be compensated or ignored in post-processing that happens in our brain -- both not exactly in favor of this hypothesis.

How about something really mundane -- a combination of loudspeakers' crossover filters and physically different high/low loudspeakers drivers producng inconsistent delay on lower and higher frequencies? This is a common problem and a pain to compensate, so loudspeakers' phase responses are anything but linear. Low frequencies simply reach the air (and the listener) later, creating the impression of distorted high-frequency "pre-ringing" on wide-band signal because higher frequencies arrived earlier, even if there was no distortion in the signal. Delaying high frequencies compensates this effect if it is present, and creates regular echo if it is not. Since delays are not compensated perfectly, you still have distortion (attributed to "phase variation" that shouldn't be perceivable at frequencies this high).

In general, of course, DAC may have to oversample to apply its own filters, but this is confined to DAC itself, signal comes in as 48kHz digital and comes out as analog.

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.)

What for if DTS-HD is lossless already

[Anonymous Coward]

From a human point of view, me owning all kinds of high end analog sound cards and having both a blueray + dts-hd receiver, I can tell you this... When done right there is definately a difference on good quality speakers + receiver when it's in DTS... partly because it only goes through one DAC on the way out (your receiver) so it's much louder and clearer than analog. But this upconverting stuff... this is only good for crap quality mp3's where you can pretty much predict patters of garbage like that crystal garbled digital sounding effect, I could see something filtering that, or maybe enhancing some higher end frequencies by boosting them a bit. This doesn't belong on bluerays...

Now another thing i noticed... you ever have a pair of headphones on, or watching a movie and you hear a sound that makes you turn around because it sounded very real? i've noticed on dts-hd at home that this happens more frequently... especially with things like knocking on doors. Where as before it only happened very rarely. The last movie that did this to me was valkyre, I had my sound up a good deal and me and the other person almost jumped out of our chairs lol... Then again I also have 2 sub-woofers, one sealed 12' for real lows and one ported 10' both properly calibrated to match my main speakers, 2 full range sony towers, and 4 large sattelites (also with pretty wide frequency range) for the rest of the surround (7.1), and one pretty big center speaker with 3 cones.

Will REAL people (spending most of the money) care

[Anonymous Coward]

With the new "wait 10 seconds while we educate you" after waiting for 60 seconds for the BluRay disk / player to "boot", will anyone really care? This is just a waste of effort, there is nothing that is THAT good that can't wait until its on HBO/TNT or Free OnDemand. Let the Movie Companies and Empty Suits/Suitests freeze in the dark. My wife, my family nor I give a crap about some 5% issue that is lost with munching on popcorn, telephones, A/C, heaters, dogs and bad speaker placement.

Anyone have a laser disk they want to use as a fresbie?

WHO CARES

[wintermute000]

Only a tiny proportion of jerkoff audiophiles and home cinema nuts will be able to hear the difference.
I wish they spent all that time, effort and money on improving something that actually needed improving

Re:WHO CARES

[dintech]

What's funny about audiophiles is that the one thing that makes a MASSIVE difference it the one thing they never seem to do. Room Treatment. It's amazing the improvement in sound quality you can achieve with some cheap fiberglass insulation and rockwool mounted in a wooden frame and wrapped in some breathable cloth. There seems little point to me in spending thousands on stereo equipment when you have room modes the size of mountains. Acoustic panels can help control those and makes huge perceived and measurable differences.

money money money mooonney

[Anonymous Coward]

Now we know what slashdot's parent company meant by looking for ways to further monetize their online properties

Re:And Harry Nyquist is rolling around in his grav

[poopdeville]

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.

Re:And Harry Nyquist is rolling around in his grav

[fluffy99]

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 article says nothing about the human hearing range other than making a reference to some other unproven hypothesis. The article does show that instruments produce frequencies will above 20kHz, which was never really in question.

An exercise for the reader

[xiphmont]

Assuming from the press release that this is an apodizing filter that 'removes' Gibbs effect preringning... how many peer reviewed studies can we compile here below my post that indicate anyone can hear these 'artifacts'?

Ready, on your marks, go!

Be careful of publications from the businesses that are pushing these filters (eg, Meridian audio / J. Robert Stuart). AES papers count only if the results have been independently reproduced (the AES is in the business of publishing 'interesting ideas', and the papers run the gamut of careful science to raving lunacy).

"I listened and _totally_ heard a rounder, fuller sound with better staging" is not data. It's what audiophiles have said of every nonsense breakthrough of the past 40 years. The one true breakthrough, digital audio, they generally still roundly pan and feel the need to 'fix' by sprinkling fertilizer all over it. It's like holy penguin pee, only it smells bad.

'Proofs' and explanations are nice, I indulge in them myself, but the blind listening data is the final authority.

Re:An exercise for the reader

[Prune]

> the AES is in the business of publishing 'interesting ideas', and the papers run the gamut of careful science to raving lunacy
Whoa, how did I not notice all those years! Thanks for the warning--I rush now to cancel my subscription!

Re:An exercise for the reader

[xiphmont]

Surely you mean membership...?

Re:An exercise for the reader

[Prune]

I mean subscription. http://www.aes.org/e-lib/subscribe/

Re:An exercise for the reader

[xiphmont]

Oh, what do you know. In that case, go forth and unsubscribe :-)

As for me, it was 'membership'. Peer review means little when a substantial portion of the peers don't believe in the scientific method.

Re:An exercise for the reader

[Prune]

Why do you get that impression about them? I've never seen such a scathing attack on the AES before.
I'm skeptical as the only other time I've seen such an attitude against a professional society is when a colleague was dissing the ACM after his paper got rejected by SIGGRAPH.

Re:An exercise for the reader

[xiphmont]

Well, for the record, I've not been rejected, but I've only published within AES once.

It's not an attack, it's more a statement of truth. The AES publishes all sorts of things. Papers with interesting ideas and no data (eg, the J. Dunn 'equiripple filters cause preecho' paper, which presents a fascinating insight, even if it doesn't work out in practice), papers with data that are effectively WTFLOL (the famous Oohashi MRI paper) and papers that are more careful controlled studies. It runs the whole gamut on both sides, just as I said.

Do you deny that a substantial portion of the membership, including many elders of the group, are not 'bigger numbers are always audibly better' audiophiles? It was Andy Moorer himself who, with no hard data, kicked off the insane sampling rate race that now has some hardcore audiophiles wondering if 192kHz is enough.. they're holding out for 384kHz!

Is the AES a worthless cesspool? Oh heck no. Never said that. But treating its publications as more than a good industry rag (where it's sometimes hard to tell the research from the advertisements).. or perhaps an advanced debating club... is probably not a very good idea. Treating any one AES paper as gospel is just insane.

Re:An exercise for the reader

[Prune]

So where along the spectrum does something like this fall? http://www.aes.org/e-lib/browse.cfm?elib=7497
At first hand it seems implausible that something like this will matter. It doesn't show up in standard THD measurements (though it does show up in Hawksford-style pseudorandom filtered noise measurements). Yet later a few things came out: 1) it's a rediscovery of an effect that was initially confirmed decades ago in tube circuits (though the time constant in tubes is much bigger, far below concern audio frequency), 2) other people measured it, and 3) people built amplifiers that minimize the effect by minimizing variation in power dissipated across the primary gain devices (for example http://peufeu.free.fr/audio/memory/img/complete-schem-1.gif ). In discrete circuits it's not likely to be audible, but one wonders if it might be audible in ICs, especially given the tight thermal coupling between sensitive input stage and high power output stage in the same package. Given this, though we don't know, a perceptible effect to some ears is at least plausible. Just because no one has performed an ABC/HR test to confirm it doesn't mean we should dismiss it.
I guess my point is that it's too easy to make an error when seeing an "interesting idea and no data" and dismissing it. A cursory examination and making a quick call on whether a perceptible effect is audible is bound to lead to sometimes throwing out a baby or two with the bathwater.

Re:An exercise for the reader

[Prune]

Oops, meant to write "whether a perceptible effect is plausible" not audible...

Re:An exercise for the reader

[xiphmont]

Oh! I remember this one :-) I'll be honest here-- this particular debate is outside my core expertise. I have enough background to say "this is all plausible" (I'm an electrical engineer after all), and I've discussed it in person with an author of a few papers on the same subject, but I'm only a dilletante when it comes to building amps.

>I guess my point is that it's too easy to make an error when seeing an "interesting idea and no data" and dismissing it.

I agree with you completely. Interesting ideas should be published; no paper is born in the state of being independently verified. I object to those who take these papers as evidence to support a position when no such validation has taken place. Thinking aloud is useful, but thinking aloud != hard data.

Re:An exercise for the reader

[Prune]

I have one more thought on this: think of how much low level distortion is masked by how crappy speakers are in general. I think many blind tests will have to be revisited when we have truly low distortion speakers. Even electrostatic headphones are not that great. Then there's the French ionic headphones http://membres.multimania.fr/plasmapropulsion/Industrial_issues/Plasmasonic.htm but the bastards didn't measure distortion so one can only guess.
The best performing I've seen is glow discharge plasma. On Google patents you can check US 4,219,705. He used helium to create stable glow discharge (in air, glow discharge is very unstable and inevitably transitions to an arc) and then shaped it in such a way as to get a flat frequency response. Photo: http://3.bp.blogspot.com/_tb6Dp4NFH5w/SxWU_QWONKI/AAAAAAAAAnk/IgXp5VMkZlk/s1600/Plasmacell1.jpg Unfortunately, around 330 Watts in the discharge alone and only above 500 Hz (regular cone speaker for below 500) and you have to refill helium tank at welding shop periodically... But look at waterfall and impulse response: http://tinyurl.com/7d6pdnv and http://tinyurl.com/7xfppsz and THD
I decided to build this without helium. I realized I could do it once I came across microhollow cathode discharges: sandwitch a CRCLC->regulator). 135 uF total at around 3000 V is about 600 Joules or about twice a defibrillator.
One of the huge film-in-oil caps leaked (so much for "designed for pulse discharge") and the oil caught on fire and I was so startled by the mini-explosion that I broke the complex electrode structure. Haven't returned to this project yet, but I think there's something to this approach. There was also virtually no ozone (unlike the usual corona discharge speakers people drive with RF), but UV is definitely an issue to overcome.

Re:An exercise for the reader

[Prune]

Weird, slashdot cut off parts in the middle of my post....
"and THD" should be
..and THD is <0.1% and mostly 2nd order, caused by the single-ended nature of operation.

"microhollow cathode discharges: sandwich a" should be
...microhollow cathode discharges: sandwich an insulator between two conductors, all <1 mm thick, drill a <1mm hole, apply few hundred volts, and you get a stable glow discharge that itself can serve as a compound cathode for a bigger discharge to a third electrode, since it removes the high voltage gradient near the cathode which causes instabilities promoting glow-to-arc transition. I platinum-plated tungsten pieces (Pt wire from eBay jeweller, distilled nitric acid from sulfuric+nitrate, dash of lead acetate) then baked in an evacuated quartz glass tube in DIY kiln so the Pt diffuses into the surface of the tungsten (else it will just flake off). For the insulator layer, I got 1 mm sapphire wafer as a free sample from an optics company. Drilling was a pain--diamond bit on the Dremel and still took a while. Put third electrode--a nail, a couple cm from the sandwich. 2700-2800 V from the bottom plate electrode to the nail, a resistor chain from the nail to the middle electrode (top plate of sandwitch). The first discharge formed in the hole right away and then jumped to the nail as a big discharge. If you blow out the big discharge, it reignites in the same sequence, it was great. With around 200-250 mA going through, I could stretch it to as much as 5 cm; could be more if one blows air through the microhollow. To modulate with sound, I used a transmitter power tetrode tube, one of those metal/ceramic with forced air cooling (used a server blower fan for that). Put the tube in series with the discharge, ran as voltage-controlled current sink. Worked. Then I dropped a screwdriver into the exposed _linear_ power supply I had built, and it shorted filter capacitors (transformer->

Here's the thing

[Sycraft-fu]

It doesn't matter if there's a mathematical difference, it matters if there's a perceptible one. There's a lot out there that you can prove mathematically is more like the actual original sound wave. None of that shit matters to reproduction for human enjoyment. What matters is if the difference is perceptible to humans. The sound wave could be totally different and if humans can't hear the difference it doesn't matter.

That is the whole thing behind lossy compression. You can do an imperfect deconstruction/reconstruction of a sound wave and humans will have trouble telling the difference, or find it impossible at higher bitrates. Telling the difference as an objective matter isn't hard, you can do it on a scope, FFT, with a diff, whatever. Telling the difference listening to it is impossible (with sufficiently high bitrate, like 256k MP3).

Also don't think that just because the AES is the society for audio engineers they are immune to audio voodoo. The world of pro audio is one I play in as a hobby, and there's voodoo that goes around in it to. A friend of mine is a professional audio engineer, and a good one, but he gets in to the voodoo. He has special cables, paints the edge of his CDs green with a marker, and things like that. He's convinced he can hear a difference.

Re:Here's the thing

[Prune]

Read the paper. There are citations for the audibility of preringing outside the audio band. Moreover, see my other post here http://slashdot.org/comments.pl?sid=2857759&cid=40038179

Re:Here's the thing

[Sycraft-fu]

"Read the paper" is not feasible if they want money for it. Sorry, I'm not paying just to argue with you in the Internet.

Re:Here's the thing

[Prune]

You mean like any other professional journal?
By the way, your comment is flippant, and also sad: whether it's worth paying should be judged not on the merits of arguing with me, but on the merits of further educating yourself.

Re:Here's the thing

[Anonymous Coward]

You mean like any other professional journal?

Uh... lol? Please avoid saying obviously wrong things. Maybe typing "Science Commons", "PubMed Central" or "Open Access" in your favourite search engine might help to enlighten you a bit.

Re:Here's the thing

[Anonymous Coward]

I think paying is often judged on the merits of whether there's any disposable money left over after the ramen you already cut back on twice.

Perhaps someone can contact Dr. Craven and inquire if he could provide an official comment for the /. article?

It's called "night mode" turn it on

[Sycraft-fu]

Some devices also call it dynamic range compression. You, well, compress the dynamic range of the soundtrack. Movies are supposed to have large dynamic range, by design, and they (usually) ship with the full range soundtrack on disc. Dialogue can be 30-40dB below peak (how much is actually encoded in the Dolby stream). Theater reference levels are 105dB on the mains, 115dB on the sub. That is how loud it is allowed to peak for big hits and so on.

You can have that at home too. Get a good receiver with its own mic for measuring and configuring our setup and you can have the volume dial configured to cinema reference, hence the negative numbers. 0dB will mean full theater reference, same levels you'd get at the movies.

If that is too much dynamic range, kick on the compressors your gear has. Most receivers call it "night mode" and it'll squash the shit out of the dynamic range. You can also axe the subwoofer, big hits come from it and when a system doesn't have one, it'll further squash them down (as to not blow out your main speakers).

Re:It's called "night mode" turn it on

[Osgeld]

super, show me where it is on my tv, I have better things to do than waste money on toys

Um

[Anonymous Coward]

You can't add information where none exists. If you have only 48,000 samples for a given second of audio, interpolating 96,000 samples from that doesn't make the audio better any more than using an HD camcorder to record an SD television displaying a VHS movie will improve the movie's video quality.

Re:And Harry Nyquist is rolling around in his grav

[dmbasso]

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.

Re:And Harry Nyquist is rolling around in his grav

[Anonymous Coward]

There's no such things as an ideal analog filter. No such fundamental problem exists for digital filters.

Tantalum Microphony

[residents_parking]

I've seen it when the cap was biassed at a few volts, and this was in a cheap measurement instrument using PIC with a 10 bit ADC. I sorted it by designing out the bias, as the product was low cost.

Re:You cant (sic) write properly...

[Anonymous Coward]

"they're probably marginally worse that"

Let me guess: you're an American, right?

It's "worse THAN"...

As usual....

[Tomsk70]

lots of comments from people *insisting* that there's no difference.

TV's upscale low-res (standard) broadcasts to 1080 - does the resulting picture look hi-def? No. Does it look better than directly displaying the lower-res? Yes.

At this point, whether the upscaled picture looks as good as 1080 is irrelevant - it just looks better than if you'd displayed the low-res picture without any processing.

Do you then complain endlessly about how it's not actually 1080? Apparently, lots of the posters here would.

Re:And Harry Nyquist is rolling around in his grav

[Anonymous Coward]

A response ignorant of the problems of digital filtering. Go have a look at the frequency response of a digital filter to see why one would want a damn high resolution before it's applied. It's not a lovely smooth 20db/octave rolloff with a flattening of response due to leakage that you get on an analogue system.

If you are approaching digital and analogue filters the same way then please do everyone a favour and leave the industry.

Re:And Harry Nyquist is rolling around in his grav

[pipatron]

Bullshit. Assuming enough resources (computational time and storage), any 4-bit computer can implement an ideal digital filter with as high resolution and steepness as you wish. Something you can not simply get with an analogue filter due to laws of physics.

In practice, you have limited resources, usually time because you're realtime-constrained, and then you must cheat. If you must cheat, a higher sample rate may or may not benefit you.

Not yet another BD audio format! Enough already!

[GrumpyOldMan]

There are far too many BD audio formats already, AC3, DTS, DCA, DTS-master, etc, etc, etc. With a decent ($3000) surround-sound HT setup and 40 year old ears, I cannot tell much difference between any of them. I wish the BD producers focused more on doing better video transfers. I'd much rather they use the space wasted by these new audio formats on higher bitrate video (and the same goes for the useless, space-wasting extra features).

As far as I'm concerned, the only thing these extra audio formats do is make ripping the files & playing them back via an embedded streaming device more complex. My oldest device cannot handle any of these new fancy formats beyond AC3, so I need to remux newer BDs to add an AC3 sound track to the MKV.

Sigh

My first cochlear implant.

[invertedflyboy]

You forget about the upcoming cochlear implant for ultrasonic localization. I fully expect to realize an return on all my audio "investments". On a side note, all my alien friends already say our movies sound like s&it smells.

Re:And Harry Nyquist is rolling around in his grav

[Lluc]

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.

Just because musical instruments produce frequencies above 20kHz (as shown in your link), it doesn't mean that the average human can hear them. Younger people can hear frequencies up to ~20kHz, and maybe a bit above, but most middle age adults probably cut off around 15kHz or lower. Here's one study showing 18-24 yr olds who can mostly hear 24kHz, but they're generating the sound at 117 dB -- a very dangerous level for more than just a few seconds. (http://informahealthcare.com/doi/abs/10.3109/00206098409070087?journalCode=ija)

Listening to loud sounds (>85dB) for extended periods of time will decrease the high frequency response of the human ear, so I wonder if high frequency hearing in children and teens of the last decade or two will have even worse hearing that their parents due to the ubiquitous white ear buds.

Re:And Harry Nyquist is rolling around in his grav

[poopdeville]

"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."

Re:You cant (sic) write properly...

[BronsCon]

Its probly also a typo. And im probly makeing a few ov them on purpis.

What you are all forgetting

[LeadSongDog]

... is that yours aren't the only ears in the room. Even if you don't need 24/96....

Think of the puppies ! ! !

Supersampling in consumer units

[johnwbyrd]

The thing I don't get about all these consumer-grade audio products hawking 96k sound are why engineers, who should know better, are attracted to them. It's as though we came out with a new TV that displays past the wavelengths of visible colors into the ultraviolet spectrum. Sold, presumably, with the tag line, "Our new plasma television reproduces all the colors that you CAN'T see... but BEES CAN!" Likewise, dogs in your household may appreciate an upgrade to 96k sound, but if humans cannot physiologically perceive sounds above about 22-23 kHz... then why... why...

Remove 98KHz passfilter induced harmonic overtones

[TiggertheMad]

When dealing with digital inputs, too much accuracy in the audio stream produces harshness and digital fatigue.

Citations, Please. This sounds very much like analog fan boy bullshit. "Too much accuracy", really? I have seen quite a bit idiotic pseudoscience used to explain why analog is better than digital. If you just like vinyl over CDs that is fine, just say so, but don't try to snow job me with some jargon filled statement in an effort to back up your personal tastes.

Re:Remove 98KHz passfilter induced harmonic overto

[Anonymous Coward]

The sad fact of it is that people don't like high quality audio.

When people cannot tell the sound is coming out of a box, they can't distinguish artificial sounds from other environmental acoustic sounds, and the effect can be quite unnerving. You don't know if a scream is on the recording or happening in the room next to you. People like to be able to tell the sound is coming from a speaker.