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Dolby's TrueHD 96K Upsampling To Improve Sound On Blu-Rays 255

Stowie101 writes in with a story about your Blu-ray audio getting better. "The audio on most Blu-ray discs is sampled at 48kHz. Even the original movie tracks are usually only recorded at 48kHz, so once a movie migrates to disc, there isn't much that can be done. Dolby's new system upsamples that audio signal to 96kHz at the master stage prior to the Dolby TrueHD encoding, so you get lossless audio with fewer digital artifacts. The 'fewer digital artifacts' part comes from a feature of Dolby's upsampling process called de-apodizing, which corrects a prevalent digital artifact known as pre-ringing. Pre-ringing is often introduced in the capture and creation process and adds a digital harshness to the audio. The apodizing filter masks the effect of pre-ringing by placing it behind the source tone — the listener can't hear the pre-ringing because it's behind the more prevalent original signal."
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Dolby's TrueHD 96K Upsampling To Improve Sound On Blu-Rays

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  • by Hatta ( 162192 ) on Thursday May 17, 2012 @07:42PM (#40035601) Journal

    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.

    • by Anonymous Coward on Thursday May 17, 2012 @07:45PM (#40035637)

      I guess that experiment failed to use monster cables then

      • by cvtan ( 752695 )
        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...
        • by afidel ( 530433 )
          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.
        • by dgatwood ( 11270 ) on Thursday May 17, 2012 @09:40PM (#40036661) Homepage Journal

          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.

    • by Anonymous Coward on Thursday May 17, 2012 @07:52PM (#40035719)

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

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

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

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

      • 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.

      • by SimonTheSoundMan ( 1012395 ) on Thursday May 17, 2012 @08:22PM (#40036039) Homepage

        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.

        • by Anonymous Coward on Thursday May 17, 2012 @08:31PM (#40036135)

          Question: Was 44.1 kHz chosen in part because the integer 44100 is highly composite? It's divisible by the following factors up to its square root: 1, 2, 3, 4, 5, 6, 7, 9, 10, 12, 14, 15, 18, 20, 21, 25, 28, 30, 35, 36, 42, 45, 49, 50, 60, 63, 70, 75, 84, 90, 98, 100, 105, 126, 140, 147, 150, 175, 180, 196, 210.

          Especially interesting is that it's divisible by 7.

          Prime factorization of 44100 is 2^2 x 3^2 x 5^2 x 7^2, or (2x3x5x7)^2, or just 210^2. Pretty cool, huh? Coincidence or by design?

          • by sahonen ( 680948 )
            PCM audio was originally recorded on video tape, and you needed to be able to record it in both PAL and NTSC standards. So the fact that the number is highly composite is largely a consequence of having to be divisible by both 50 and 60. More here: http://en.wikipedia.org/wiki/44.1_kHz [wikipedia.org]
          • 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

          • I posted about this on twitter a month ago. [twitter.com]

            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 becau

        • by jo_ham ( 604554 )

          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, bu

        • by sr180 ( 700526 )

          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

      • 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.

        • by Prune ( 557140 )
          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.
      • by sahonen ( 680948 )
        Actually, modern delta-sigma A/D converters are capable of near-brickwall lowpass performance at nyquist.
      • 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.

    • by aardvarkjoe ( 156801 ) on Thursday May 17, 2012 @07:52PM (#40035727)
      That's just because your hardware sucks. If you use the correct equipment [amazon.com], anyone with a discerning ear will be able to hear the difference.
      • LOL, the "reviews" of those are hysterical

      • by MightyYar ( 622222 ) on Thursday May 17, 2012 @08:05PM (#40035859)

        This might be my favorite review ever on Amazon:

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

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

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

      • I hope you're joking.

    • 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

    • 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

    • 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 surel
      • by Jiro ( 131519 )

        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.

    • by the eric conspiracy ( 20178 ) on Thursday May 17, 2012 @08:33PM (#40036157)

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

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

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

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

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

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

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

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

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

      • Does it say how he disproved the sampling theorem?

        • Re: (Score:2, Informative)

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

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

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

          • 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 "h

            • by Prune ( 557140 )
              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.
      • by guidryp ( 702488 ) on Thursday May 17, 2012 @10:09PM (#40036859)

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

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

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

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

        • by Prune ( 557140 )
          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.
          • 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/

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

  • by sahonen ( 680948 ) on Thursday May 17, 2012 @07:49PM (#40035691) Homepage Journal
    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.
    • Mod parent up!

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

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

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

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

        • by Anonymous Coward on Friday May 18, 2012 @02:22AM (#40038297)

          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.

    • 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: (Score:2, Informative)

        by 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

        • by sahonen ( 680948 )
          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.
    • 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.

      • 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

  • by Anonymous Coward on Thursday May 17, 2012 @07:52PM (#40035725)

    May I be the first to say this- fuck Bluray, and fuck Cinavia.

    I used to buy Bluray disks. Hell, I own a whole shelf full of them (about 80 titles in total). Every single one eventually got ripped to my NAS in two formats- a relatively lossless MKV file containing the original video and audio streams (up to DTS-HD MA), and a lossy x264 version for playing on crappy devices like the PS3 or 360.

    Then Cinavia rolled around, which did two things:
    1) It purposefully corrupts the audio stream in an attempt to encode digital information into it (go read their patents- the harder you try to pry Cinavia into an audio stream, the more damage is done to the original quality)
    2) It prevented me from playing my legally purchased and legally ripped (it's legal in my country to rip disks and things you BUY) disks off my NAS on my PS3

    What pisses me off the most though is that Sony is pushing Cinavia on everyone as hard as they can. AFAIK all new BR players need to be equipped with it, and most of the new BR disks are supposed to have it as well. And they're still advertising the disks as "Lossless", when in fact the audio is NOT lossless- it's lossy, the degradation of which is brought about solely by Cinavia's presence.

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

    "Cinavia's in-band signaling introduces intentional spread spectrum phase distortion in the frequency domain of each individual audio channel separately, giving a per-channel digital signal that can yield up to 20 kilobits per second—depending on the quantization level available, and the desired trade-off between the required robustness and acceptable levels of psychoacoustic visibility. It is intended to survive analogue distortions such as the wow and flutter and amplitude modulation from magnetic tape sound recording. On playback no additional audio filters are used to cover up the distortions and discontinuities introduced."

    So there you have it. Lossless is no longer lossless, because Sony insists on using this stupid fucking DRM on their stupid fucking format (as usual). Dolby's new gimmicky technology might claim to give you better lossless audio, but none of that matters the moment they drive Cinavia into the stream.


  • 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 m
    • 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 som

      • 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...

        • 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 th

  • 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

    • by Ogi_UnixNut ( 916982 ) on Thursday May 17, 2012 @08:05PM (#40035849) Homepage

      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.

      • 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.

    • 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

      • 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?

    • 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 c

    • 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 ow

  • Apodizing Filter (Score:5, Informative)

    by Josuah ( 26407 ) on Thursday May 17, 2012 @08:01PM (#40035811) Homepage

    The title is misleading if the actual goal of this is to apply an apodizing filter. I suspect the reason it's called "Advanced 96K Upsampling" is because that's much easier to get people to buy into that than a "Apodizing Filter" sticker.

    The article explains how the audible benefit comes from the application of the Meridian apodizing filter, which changes the analog signal reproduced from digital data by reducing the pre-ringing. IIRC the trade-off is that post-ringing increases. The claimed benefit is that since the ringing now occurs after the "real" music of larger amplitude and as a result the ringing is masked or could be considered like an acoustic echo that naturally occurs.

    The 96K upsampling is just a side-effect of wanting the extra samples when you are applying the filter.

    Here's a decent summary of what is supposed to happen to the analog audio signal as a result of the filter application: Technical analysis of the Meridian Apodizing filter [wordpress.com].

    That being said, from what I've read over the past few years I think people are kind of mixed on whether or not the filter makes things better, worse, or just different but not better.

    • Re:Apodizing Filter (Score:4, Informative)

      by slew ( 2918 ) on Thursday May 17, 2012 @08:22PM (#40036029)

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

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

  • by TheGratefulNet ( 143330 ) on Thursday May 17, 2012 @08:05PM (#40035853)

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

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

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

    • by sahonen ( 680948 )
      > 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.
    • by dgatwood ( 11270 )

      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 o

    • 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.

  • 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)

  • 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 (Score:4, Funny)

    by Alex Belits ( 437 ) * on Thursday May 17, 2012 @11:38PM (#40037405) Homepage


    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 (Score:4, Informative)

      by Prune ( 557140 ) on Friday May 18, 2012 @03:47AM (#40038673)
      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 [aes.org]
  • 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

  • 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 pap

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