Slashdot Mirror

← Back to Stories (view on slashdot.org)

The Successor to AC'97: Intel High Definition Audio

Posted by michael on from the hear-hear dept.

An anonymous reader writes "A few days back Intel announced the name to its previously dubbed 'Azalia' next-generation audio specification due out by midyear, under royalty-free license terms. The Intel High Definition Audio solution will have increased bandwidth that allows for 192 kHz, 32-bit, multi-channel audio and uses Dolby Pro Logic IIx technology 'which delivers the most natural, seamless and immersing 7.1 surround listening experience from any native 2-channel source'. The architecture is designed on the same cost-sensitive principles as AC'97 and will allow for improved audio usage and stability."

84 of 428 comments (clear)

  1. It is still onboard sound by Anonymous Coward · · Score: 5, Interesting

    Will it still also suffer from the same effects of background noise from the rest of the voltage going through the motherboard, or have they found a way to block that out also? 32/192 is fine as a standard... but it is still onboard sound. It needs some seperation from the motherboard to maintain a high S/N ratio

    1. Re:It is still onboard sound by UrGeek · · Score: 4, Interesting

      Mmmm, what would really be nice if the DAC's were not on the sound chip but in a sheilded housing if it's own and then some nice connectors. And the sound chip would have that digital audio interface - i forgot what it is called - if it even supports something as insane as 32-bits/192kbps

    2. Re:It is still onboard sound by xlyz · · Score: 2, Informative


      use a DAC out of your case

      just use digital out to a good A/V receiver

    3. Re:It is still onboard sound by xlyz · · Score: 5, Informative

      If only there were some way to have a digital output from the computer, and do the D/A conversion in a dedicated box.
      br> there is

      digital out is common on today pc (either optical or coax) and any good A/V receiver with integrated decoder is able to convert the signal from digital to analog

    4. Re:It is still onboard sound by ethanms · · Score: 2, Insightful

      You're right... but keep in mind that most of the motherboards out there that give out lousy sound from onboard are due to poor layout from the manufacturers... who giving poor layouts because want to save money and physical space on the motherboard, at the expense of analog components like sound...

      more bits and more kHz are useless for onboard until you clean up the analog paths to the jack, and properly isolate the codecs on the motherboards using ground moats. Nothing worse then a company that routes a processor +12V feed trace right under the analog side of the codec... or worse, a noisy signal like PS/2 or NIC.

      Dear Boss: please don't fire me for this post

    5. Re:It is still onboard sound by alienw · · Score: 2, Informative

      There is no physical possibility of having *good* onboard audio. Even with all the above construction techniques, it's damn near impossible to completely isolate the prodigious amounts of digital noise that a typical computer produces.

      A much better idea is to run a digital link to an outboard DAC that has its own power supply and is outside the computer. That would actually give you extremely high quality audio, assuming the DAC box is properly designed.

    6. Re:It is still onboard sound by Detritus · · Score: 2, Interesting

      If they were really serious about noise, they could use RF construction techniques and put the analog components in a shielded can on the motherboard, with bypass capacitors on the power/ground connections. You can shield anything if you are willing to spend some money.

      --
      Mea navis aericumbens anguillis abundat
    7. Re:It is still onboard sound by j3110 · · Score: 4, Informative

      If they are going through that much work, I wouldn't be suprised if there wasn't a seperate card with the DAC that you put in a slot and run cables to. It's been done before, just not for this purpose.

      That said, I actually think 32bit audio may be at least 8 bits overkill. I'm all for 192Khz, because we can actually hear a difference in the resolution of the wave. 16bit audio allowed for 64K levels that were smoothed between. Most audio is pretty smooth sounding, and I doubt you can hear any difference between 16 and 32 bit unless you crank the volumn up to a level that could damage your hearing.

      Also, 32bit DACs are practically impossible to buy last time I checked. A full 16bit DAC is pretty expensive relatively and it's exponentially more complicated with each bit to build a proper DAC. I'm expecting a lot of shortcuts. A 32bit ADC for recording is prohibitively expensive, so I gaurantee you won't be doing any 32 bit recording any time soon on a PC.

      Basically, the 32bit idea is dead in the water. The machine will be long gone before any audio is distributed that takes advantage of it. You probably can't use it for mixing because you probably won't be able to record at 32bit. It's also going to be more expensive in components. Speakers aren't going to be accurate enough to 32bits of resolution. They may shoot for 24 bit, because you can get an OK DAC and ADC for working with 24 bits, but it'll still cost.

      The 192Khz thing is awesome. Right now, you can get 48Khz out of some consumer cards, but 192 would be excellent. Maybe we'll get digital audio up to proffesional quality some day. Right now if you go get a recording from a studio, you get tape (unless you can't afford it). All professional audio equipment is not only analog end-to-end, it's also usually tube based. The average transistor is pure sewage, and even MOSFETs are lacking. There's gotta be a lot more R&D into just transistors before we have professional grade audio going anywhere near digital. This is still going to be helpful to the end user that likes music, but we are still a long way off from having no audible differences. Amazingly enough, I think speaker technology has advanced more over the last decade than digital audio.

      --
      Karma Clown
    8. Re:It is still onboard sound by Hoser+McMoose · · Score: 2, Insightful

      Yeah, it's terrible how nVidia MADE 3Dfx screw up their entire distribution channels, how they made them buy out STB and try to become a card manufacturer. Absolute horrible how they made 3Dfx deliver all their products a year late (or more) and missing much-needed features. And it was especially bad how nVidia made 3Dfx release crappy drivers (or no drivers) for so long.

      Face it, 3Dfx killed themselves, nVidia just moved in to pick up the slack. Even if the whole lawsuit between the companies had gone in 3Dfx's favor it's unlikely that they would have managed to survive long enough to see the results of it. A combination of bad decisions and products that were a day late and a dollar short (but still expensive) killed them, not nVidia.

    9. Re:It is still onboard sound by sfe_software · · Score: 2, Insightful

      It is still onboard sound

      Not necessarily. The specification can be used for PCI cards as well, and in fact AC97 is used on some lower-end audio cards. It's more of a specification for minimum supported features and other specs.

      The fact that it is on-board in itself doesn't mean it is bad. It's all in the implementation. With proper design techniques (ground-loop isolation, etc) you can get quite a good S/N ratio. It doesn't need "separation from the motherboard", rather, it needs a buffered power bus, separate audio and digital grounds, etc.

      The bottom line is, you get what you pay for. If you spend $100 for a motherboard with onboard sound, video, nic, modem, etc... you're likely to get cheap versions of each. If you spend $130 on a PCI sound card, you'll probably get really good specs, whether it is based on AC-97, this new spec, or its own details.

      --
      NGWave - Fast Sound Editor for Windows
    10. Re:It is still onboard sound by ethanms · · Score: 4, Insightful

      I'm guessing you don't work in the industry... it's not only possible, but it's been done on many designs...

      Codec construction is important, for example two major suppliers from Taiwan: C-Media and Realtek, are both pretty much crap even on their high end parts... they've traded features and low BOM cost for audio quality...

      Other codec supplies, like Analog Devices & Sigmatel (or even Wolfson, Phillips, etc) have put audio quality as a priority to feature sets.

      Unfortunately if Realtek rolls out some new feature then the others need to follow or be left behind.

      Using ground layers properly, moats and keeping traces near the edge of the board... or even better, making sure you keep the codec as physically close to the jacks as possible, will yield very good results easily rivaling your average sound card.

      Let's also keep in mind that an AC'97 or "HD-Audio/Azalia" codec goes for between $0.50 and $1.25...

      Where-as a typical SoundBlaster will go from $50-200... they're able to use a lot higher grade support components, and since they are on a PCI card they're able better isolate from the rest of the motherboard (which speaks to your point...)

      As for digital out...

      Many motherboard manufacturers are finding that the masses are demanding SPDIF (digital) output from onboard sound, it's been available for the past several years from AC'97 vendors, even on most of the low end codecs, but adding the TOS (or even RCA) jacks cost too much in BOM and board real estate (surprise, surprise)...

      I think the next big requirement from users will be that SPDIF provide an AC3/DTS signal for all 4/6/8 channel audio. I'm surprised that this wasn't a requirement for Azalia, but we'll see what happens in the near future... After all, AC'97 is currently at version 2.3, there's room for change...

      Currently nearly all (even the $200 SB Audigy2) provide only PCM (2-ch) when playing non-DVD audio (when playing DVD they will all pass the AC3/DTS signal out, but they do not generate their own based on a multi-channel game or sound file).

      This is mainly due to the licensing fees from Dolby to encode AC3/DTS signals, and partly due to the processing overhead that would be required for implementation in soft-audio.

      The exception to this are boards equipped with the nVidia nForce2 audio, they build a DSP into the southbridge(ICH) that encodes AC3 out of any 4/6-ch source being played.

    11. Re:It is still onboard sound by mlyle · · Score: 3, Informative

      Impossible? Impossible why? I don't see why this would be the case. In fact, I imagine that with minimum wiring you could run two 16bit DACs in parallel, one handling the top 16 bits at twice the voltage, the other handling the low 16.

      You mean the top 16 bits at 65536x the voltage, and the other handling the low 16. Else you've just produced a 17 bit DAC.

    12. Re:It is still onboard sound by Tough+Love · · Score: 2, Informative

      If only there were some way to have a digital output from the computer, and do the D/A conversion in a dedicated box.

      Here's one

      This box allows you to use spdif with your existing analog stereo.

      Specs here

      --
      When all you have is a hammer, every problem starts to look like a thumb.
    13. Re:It is still onboard sound by j3110 · · Score: 5, Informative

      Tubes generally have a flatter curve when comparing amps out to signal voltage, but MOSFETs have a flatter RMS Watts out compared to RMS signal level. Basically, MOSFETs screw up the wave form more than tubes, but manage to preserve the loudness at various frequencies better.

      I would rather take the one that can be fixed with an equalizer. :)

      Where transistors really rule though is low power usage. A class A tube amp will keep you warm at night without even actually making noise. We need better transistors, but I'm not saying we need tubes everywhere.

      What you are saying about needing twice the sampling rate is complete BS. Between that remark and the tube vs MOSFET remark, I can tell you care very little about the wave form.

      If I tried to make a 20khz sound with 40khz sampling, I wouldn't hear a thing. Take two samples of one wave equidistant, and you get the middle and the beginning of the wave, which are always the same value. At 4X the sampling, you get a saw wave (or worse, a muffled trapazoid if you phase shift 45 deg). So, if you do 20Khz at 80Khz, you're still screwed. How many points do you need on a wave to make it smooth? I would say at least 8 at high frequencies (and that has a chance of only getting you about 66% of the power). That's about 160Khz for 20Khz sound.

      I hate that someone actually thought the whole 2X the audible range was good enough to begin with. You may not hear a difference, but I can. If you don't, check the frequency response of your speakers.

      If you want to calculate how many samples it takes to get >90% power, you should calculate the distance on the wave that is 90% power, then divide the length of the full wave by that distance, then multiply by 20Khz. So, here's a trusty sine table for you already measured in percentage of wave: .15 : .81 .18 : .90 .20 : .95 .25 : 1 .30 : .95 .32 : .90 .35 : .81 .32-.18=.14
      1/.14 = 7.14
      20KHz*7.14=143KHz

      This isn't RMS calibrated, but so what.

      192KHz: /20KHz=9.6
      2pi/9.6=.65 radians
      sin(pi/2-pi/9.6)= 94.69% power output.
      Basically, it's the sine of half the distance away from the peak to the furthest point out. Why is is it not the average? If you take the average, then you are forgetting that you will miss the right side of the wave completely. The only case better is sometimes when you hit the exact peek.

      Also, you have to consider that this is going to create distortion too. Consider that the resonance of 20KHz with the actual output level, since it varies around the 94% mark.

      --
      Karma Clown
    14. Re:It is still onboard sound by ]ix[ · · Score: 3, Insightful
      Here we go again, I promised myself not to get involved in "audiphile" discussions again but...


      If I tried to make a 20khz sound with 40khz sampling, I wouldn't hear a thing. Take two samples of one wave equidistant, and you get the middle and the beginning of the wave, which are always the same value. At 4X the sampling, you get a saw wave (or worse, a muffled trapazoid if you phase shift 45 deg). So, if you do 20Khz at 80Khz, you're still screwed. How many points do you need on a wave to make it smooth? I would say at least 8 at high frequencies (and that has a chance of only getting you about 66% of the power). That's about 160Khz for 20Khz sound.


      But if you use a 40.00001 kHz and a brick wall low pass filter at 20 kHz you will get the original wave.

      The only real reason to higher than 2X the audible range is that it is difficult to make brick wall filters. In theory a "muffled trapetzoid will still represent the original wave correctly. Nyquist was pretty clear about this. =).

      Linear PCM is just such a waste of space above 44kHz sampling frequency. Everytime you double the sampling frequency you double the amount of data but you only get one octave more information. So if the signal is music, frequency shouldnt be linear. Wavelets for instance are much more efficient at storing data but much more complex to implement. 192x32 is just a brute force way of saying: I cant afford to make decent filters.

      Your "power-calculation" just shows that you have never taken a signal processing class in your entire life. But thats ok, those classes are hard.

      --
      This is my sig, show me yours
  2. OSS drivers? by cyb97 · · Score: 4, Interesting

    Does the royalty free license also imply that we'll see good opensource drivers for a plethora of platforms?

    1. Re:OSS drivers? by dreamchaser · · Score: 4, Informative

      Not necessarily. It's still up to the hardware manufacturers to implement it on their hardware, and then either provide drivers for said hardware or publish their specs as well.

    2. Re:OSS drivers? by Clockwurk · · Score: 4, Insightful

      It depends on how nice intel is feeling. Royalty free doesn't mean that intel doesn't control it. Royalty free only implies free as in beer, not free as in speech.

    3. Re:OSS drivers? by DrEldarion · · Score: 2, Interesting

      I just hope for good drivers period. I can't tell you how many times I've had problems with onboard audio even in Windows. I've seen computers where the audio will work flawlessly in Win2k but not in XP, where it'll work fine in XP, but not in 2k, where it'll work fine until you reformat and reinstall the exact same OS, then be broken, etc. etc.

      I finally got fed up with it and just got a cheapo PCI card and haven't had any problems since. Incidentally, you get better gaming performance when you don't use onboard audio, too.

    4. Re:OSS drivers? by DrSkwid · · Score: 3, Insightful


      Even better would be if turning it off in the BIOS meant that the OS actually ignored it.

      --
      There are places where the networks are not touching,and there are places where they are-Boeing's Lori Gunter
    5. Re:OSS drivers? by ctr2sprt · · Score: 4, Insightful
      Hardware is a fundamentally different beast than software. Software can be copied and modified easily once the initial version has been created. Hardware, on the other hand, continues to bear an associated cost per-copy even after the initial development is finished. Because of the nature of the medium, after-the-fact modifications are extraordinarily difficult. So it's not really valid to compare hardware licensing to software licensing, at least not using the oversimplified "free as in beer/speech" simile.

      In any event, if Intel are letting groups take their spec and implement it in hardware that's meant to be sold for profit... It doesn't get much freer than that. "Free as in speech" doesn't mean you have to give away the farm. You're allowed to keep certain rights for yourself, and make certain restrictions on use, just like open source software does. (And just like there are for free speech, in fact.)

  3. Initial reaction by Firehawke · · Score: 5, Insightful

    The very first thing I thought when I saw the article itself was, "Please don't let this be as bad as AC'97."

    Don't get me wrong, AC97 is cheap, but it really dragged on the CPUs of the timeframe it came out. This one looks like it might be a shot at the Creative Labs end of the market, but with cheaper components (meaning most likely CPU-based)

    I'm sure it'll be on pretty much every board before too long-- well, the non-nForce ones, anyway.

    1. Re:Initial reaction by BoomerSooner · · Score: 2, Interesting

      Agreed, AC97 is a POS. Every computer I've ever seen someone is using it on the driver implementation and quality is pure shit. Just spend the $50 and get an Audigy card.

    2. Re:Initial reaction by dnoyeb · · Score: 3, Interesting

      Yea, integratedness has fallen out of favor with me. At least those things that are human detectable such as audio and video.

      Integrated sound thus far has been a bad failure. It works well if nothing else is taxing the CPU, but otherwise, it can stutter. My nforce stutters when the network is active so no playing mp3s located on my Linux share...

    3. Re:Initial reaction by Anonymous Coward · · Score: 2, Insightful

      Well, that's the fault of your cheap'n'nasty Nforce chipset, not integrated sound per se.
      I've built any number of PCs (all Intel-based) over the last 3 years or so with AC'97 onboard audio, and have never noticed the audio "stutter" under any kind of load.
      Sorry, but that's the truth. Don't blame AC'97 just because your particular implementation of it is sucky..

    4. Re:Initial reaction by treat · · Score: 2

      Could someone please explain exactly what is wrong with AC97? How could the quality be affected if I'm using the SPDIF out? (And why would you complain about quality if you're not?)

    5. Re:Initial reaction by Anonymous Coward · · Score: 2, Informative

      Because AC97 resamples everything internally to 48kHz, including 48kHz streams, so it auto-mangles everything you put through it. If that wasn't enough the Windows sound system (many are afflicted by such voodoo) resamples *everything* through its mixer further mangling the sound before AC97.

      Unfortunately SPDIF is not bit-perfect by no means, you need ASIO for that. An easy way to tell is to play a Dolby Digital or DTS .wav through a board, if it arrives at the AV reciever unaffected then the computer isn't screwing with it.

  4. Progress In Consumer Audio? Yes! by ten000hzlegend · · Score: 5, Interesting

    True progress from Intel, strange but true

    This new system for audio managment is great news for portable devices such as DVD+screen, next-gen PDA devices and even handheld game systems *Gameboy Advance II or PSP?*

    I've long been following PC related audio solutions, all the way from Sonarc to the latest 5 and 6 channel set-ups, my normal set-up is bass speaker, left / right and one for routing system alerts etc... this kind of announcement coupled along with the latest cards supporting the new Dolby processing solutions could well make me upgrade

    More to post...

  5. All the usual concerns. by IGnatius+T+Foobar · · Score: 4, Insightful

    On its face this is a great announcement, but we must have all the usual concerns. Will it work in Linux? Are the hardware API's going to be published, so someone can write Linux drivers? Or is this going to be the next Centrino, needlessly obfuscated to give Intel's friends in Redmond yet another unfair advantage?

    I'm also concerned that a new audio hardware API may introduce way too many opportunities for things like Digital Restrictions Management. Long term, doing that is of course futile because someone will find a way around it, but that doesn't stop some hardware makers from setting out the legal minefield anyway.

    It's a sad state of affairs when politics and litigation are at the forefront of geeks' minds when technology ought to be.

    --
    Tired of FB/Google censorship? Visit UNCENSORED!
  6. Isn't this just a bit much? by UrGeek · · Score: 4, Insightful

    32-bit audio at 192kbps? Why not just stick with 24bit at 96kbps - it is good enough for most studios. And actually 16-bit at 44.1kbps is the most that these old ears are gonna hear anyway - if even that well after sitting front for Jimi Hendrix.

    1. Re:Isn't this just a bit much? by ten000hzlegend · · Score: 4, Interesting

      With modern audio requirements, getting as close to the fidelity of the original is the "flavour of the month"

      Last year, Pink Floyd released Dark Side on SACD, 24-bit audio at 48khz / 96khz, the amount of clarity over a CD, once the benchmark, was remarkable, I attended a launch party at was blown away even in a relatively acoustic poor setting

      I for one welcome consumer 32-bit audio

    2. Re:Isn't this just a bit much? by boa13 · · Score: 2, Interesting

      Last year, Pink Floyd released Dark Side on SACD, 24-bit audio at 48khz / 96khz, the amount of clarity over a CD, once the benchmark, was remarkable, I attended a launch party at was blown away even in a relatively acoustic poor setting

      How much of that clarity was due to the excellent sound engineers they probably hired? How much was due to the stage setup, and the excellent speakers and amplifiers they probably had? How did you compare the clarity over a CD? If they offered a comparison, how do you know the CD was a good one, and not a voluntarily dirtied version?

      I for one am very wary of launch parties.

    3. Re:Isn't this just a bit much? by Jeff+DeMaagd · · Score: 2, Interesting

      The problem is that at 24 bits per channel, it is impossible to fully realize that sort of dynamic range with physical objects.

      The extra eight bits to get to 32 bits is simply a waste. The best I can think of is steganography where you can hide data in the least significant byte and few would catch on unless the data was carefully analyzed.

    4. Re:Isn't this just a bit much? by ten000hzlegend · · Score: 3, Informative

      True, we handed Gary Wright who was announcing the various specifications of SACD at the time of play, a 1984 Dark Side CD, a 1993 20th anniversary CD and finally a copy of Echoes which had the latest digital master before the 30th anniversary re-master

      Clean, no scratches and if I recall, the Japan import 1984 cd was worth a mint

      Anyhow... we played each one and came to the result that the 2 channel 30th anniversary remaster was far superior, even on a great system, and the surround mix was simply amazing to hear

    5. Re:Isn't this just a bit much? by JebusIsLord · · Score: 3, Insightful

      In double-blind tests, people have been unable to tell the difference between the SACD layer of the new release and the 1992 CD remaster. The cd-layer on the 30th anniversary version is needlessly overcompressed, probably just to make it sound different than the SACD layer. Try it double-blind, you'd be surprised at how much placebo comes into effect.

      --
      Jeremy
    6. Re:Isn't this just a bit much? by gidds · · Score: 2, Informative
      The theory for high sample rates (AIUI) is that they allow much gentler filtering, giving less distortion in the audible range.

      Standard CDs are sampled at 44.1Khz, so the highest frequency they could possibly store is a sound at 22.05kHz. However, this doesn't meant that they will reproduce anything less than that with perfect accuracy. Firstly, the sound needs to be filtered to prevent anything over 22.05kHz hitting the convertors (as they'd cause very nasty artefacts); this filtering has a lower cut-off (usually around 20kHz) for safety, and although the filter has a steep response, it's not infinite; it'll reduce some lower frequencies too, and it'll also cause phase changes at lower frequencies. (I gather current filters are much better than those used for early CDs, which were responsible for much of the early complaints.) Filters are also needed in the player, which also affect the sound.

      Greater sample rates would allow much gentler filters to be used, which would have less (or no) effect on audible frequencies, even those above 20kHz.

      Secondly, it's claimed that although we can't hear sound at those higher frequencies, we can detect phase changes and timing changes occurring faster than CD can store; the additional timing resolution would help with that.

      And thirdly, in the studio (and wherever sound is processed) the tiny changes caused by filtering and slight timing shifts can add up, to the point (it's claimed) where they can have a very audible result. The extra frequency and time resolution, just like the extra sample resolution of 24 or 32 bits, allows mixing and other processing to be done with less loss.

      So there are reasons why 96kHz or 192kHz and 24- or 32-bit sound might provide real benefits. I'm unlikely to hear them myself -- I'm a musician, not an audiophile or sound engineer -- but as technology gets more powerful, faster, and cheaper, I'm sure sound quality will only improve.

      (If the RIAA doesn't stop it... Oops, a little bit of politics there, yes indeed.)

      --

      Ceterum censeo subscriptionem esse delendam.

  7. I have particulary fine ears... by Anonymous Coward · · Score: 2, Funny

    So, I think I'll wait for 42.1 with 0Hz to 1GHz (+/- 0.0000001%) bandwidth and 256 bits samples audio hardware, which shouldn't be to far away :o)

    1. Re:I have particulary fine ears... by tloh · · Score: 2, Funny

      Jesus! The constant ding of the everyday sound spectrum must drive you nuts then. I'll bet people look at you funny when they hear nothing but see you shouting "make it stop! make it stop!"

      --
      Stay sentient. Don't drink bad milk.
    2. Re:I have particulary fine ears... by imsabbel · · Score: 2, Funny

      And those audiophiles will still claim that their vinyl sounds better...

      --
      HI O WISE PRINCE. WHT TOOK U SO DAM LONG?
  8. A next gen audio specification? by xankar · · Score: 4, Funny

    Hear hear!

    Pun completely and totally intended.

    --
    ~To choose doubt as a philosophy of life is akin to choosing immobility as a means of transportation. -Yann Martel
  9. the way I look at it by Bubba · · Score: 2, Insightful

    At least they are changing an old standard that has had mixed issues for several years. New input on old (possibly failed in some aspects) standards is always good for sales.

  10. Linux Logo opportunity? by bstadil · · Score: 4, Interesting
    Any idea what it would take to use this as an opportunuty to establish a sort of Azalia Certified for Linux Logo and a set of requirements that goes with it?

    Logo that you could stick on the box and "Journalists" et al could include in the normal fluffy Buzz Word compliance reviews.

    --
    Help fight continental drift.
    1. Re:Linux Logo opportunity? by Sumocide · · Score: 2, Funny


      1. Get a license from Dolby. Good luck with that.

      2. Implement the specs

      3. ???

      4. Profit!!!

  11. That's great! .. by ShadeARG · · Score: 3, Interesting

    .. but when will we see high definition video support with component and dvi i/o?

  12. That's audio ? by Rosco+P.+Coltrane · · Score: 2, Insightful

    The Intel High Definition Audio solution will have increased bandwidth that allows for 192 kHz,

    192 kilo-Hertz? that's more longwave radio than audio. Hell, it's like 5 times the frequency of ultrasounds. Who are they kidding? This smells of marketting bull, or deceptive commercial practices targetted at trendy audio posers ...

    --
    "A door is what a dog is perpetually on the wrong side of" - Ogden Nash
    1. Re:That's audio ? by PatrickThomson · · Score: 3, Insightful

      I gather that with 48khz there are ikky problematic sounds if you forget to filter out high frequecies that reach all the way down into the audible domain - 196khz ensures that these artifacts will be well out of the range of hearing and the abilities of most equipment to reproduce.

      --
      I am one of many. My idea is not unique, nor do I expect my voice alone to sway you. I speak in a chorus of opinion.
    2. Re:That's audio ? by Professeur+Shadoko · · Score: 2, Informative

      Well, actually, 192KHz is the sampling rate.
      Even if frequencies that high cannot be heard, using such a sampling rate will decrease the noise added by analog->digital conversion.

    3. Re:That's audio ? by admbws · · Score: 3, Informative

      192khz refers the the sample rate, how many times per second the sound is sampled, not how many cycles per second. While theoretically, 192khz sample rate does allow frequencies higher than the ear can hear to be recorded, its real purpose is to make the lower frequencies more accurate - for example, a 22050hz sine tone (if you can hear that high!) sampled at 44100hz is only sampled twice per cycle, and would effectively be recorded as a square wave (although, admittedly at that frequency you'd need to be a dog to tell the difference!)

    4. Re:That's audio ? by bbbl67 · · Score: 2, Informative

      I don't really think they mean 192 kiloHertz but 192 kilobits per second. There is a difference in the case of lossy-compressed audio. The higher the bps, the less lossy the quality of the audio is. And this bitrate also includes all of the channels together, not just one channel.

    5. Re:That's audio ? by Anonymous Coward · · Score: 5, Informative


      for example, a 22050hz sine tone (if you can hear that high!) sampled at 44100hz is only sampled twice per cycle, and would effectively be recorded as a square wave (although, admittedly at that frequency you'd need to be a dog to tell the difference!)

      This is completely and utterly wrong. I hear this very often though.

      At 44100Hz sampling, a 22050Hz signal will be reconstructed as a 22050Hz SINE WAVE. The reconstruction of sampled signals is not as simple as you think it is. This is covered in any elementary DSP book.

      With IDEAL equipment sampling at frequency N allows perfect reconstruction of all frequencies N/2 in all cases. The rather = comes about because of the potential of sampling the frequency N/2 at the zero crossings. However, if only two nonzero points are sampled of the N/2 component, it can be reconstructed perfectly.

      Using a higher sampling rate has to do more with counteracting clock jitter and the error introduced by non ideal equipment.

    6. Re:That's audio ? by DarrylM · · Score: 2, Interesting

      192 kilo-Hertz? that's more longwave radio than audio. Hell, it's like 5 times the frequency of ultrasounds.

      Yeah, that is pretty high, but it will allow for a flatter frequency response in the human hearing range than what is possible with 44.1kHz or 96kHz. The reason is that the sampling process has a frequency response of a sync function: sin(x) / x. At a sampling rate of 44.1kHz, the amplitude response of the sample at the high end of the human hearing range will be a fair bit lower than at the low end of the human hearing range. This results in less amplitude (volume) range for the higher frequencies - meaning that the sound won't be quite as close to the original.

      When you sample at a higher frequency, the sync function is, in effect, stretched out so that the frequencies at the high end of the human hearing range have a much better amplitude response. Translation: the sound output should, theoretically, be closer to the original at higher frequencies.

      Other people have also mentioned the benefits of reduced harmonics and such. As for how much of an actual difference to the perceived sound quality this will make, I have no idea. My speakers aren't all that great, anyway. :-)

  13. I can't tell if you're joking or not by roystgnr · · Score: 4, Informative

    But assuming you aren't, just find a sound card with a digital output (I think all the higher end cards have SPDIF now) and plug it in to your home theater.

    1. Re:I can't tell if you're joking or not by tho+1234 · · Score: 2, Informative

      Unfortunately, i doubt there will ever be a digital output for high-res audio.

      Look at any of the commercial DVD-audio or SACD players available- none of them support digital output at 192khz/24bit. If one was available, anyone could bypass the huge amounts of DRM/watermarking on those new discs, and make bit-perfect copies by simply plugging it into your soundcard/dat recorder.

      Anyways, the S/PDIF standard doesn't support bit rates high enough for 192/24 audio, so an entirely new format would have to be made, and somehow i doubt the RIAA will allow one to be made.

      Of course, digital output at 44.1/16, (well AC'97 resamples it to 48khz) will still be available, and that's more than good enough for 99.999% of the market.

      So basically, this high-res stuff is nothing more than a marketing ploy, there is no way you can achieve 24-bit performance on a noisy switching powesupply while blasted by EMI, reproduced by a 99 cent DAC/opamp chip. (well there's no way to achive 24 bit performance period at room temperature, since the johnston resistor noise of any system is greater than the resolution of a 24bit system, but that's another matter altogether)

  14. Not true discrete channels? by SpookyFish · · Score: 5, Informative


    This sounds like it could be more smoke and mirrors, though there really isn't enough information to be sure.

    ProLogic IIx will "synthesize" multiple channels from a stereo or 5.1 source. I sincerely hope Intel isn't thinking "we can do the same old thing (stereo) and marketing folks can call it 7.1 multichannel because we put this Dolby fake surround processing in the chip!"

    Despite how much ProLogic has advanced, it still doesn't hold a candle to true, *discrete* 6+ channel sound (like DD/AC3 or DTS).

  15. DSD Support? by babymac · · Score: 2, Interesting

    When will we see support for the DSD audio format in computer hardware? I have yet to hear this technology for myself, but friends who have heard it say it's incredible. Like analog, only better. The one bit tech behind it is very compelling...

    --
    "War makes me sad." - Me
  16. double-blind, controlled test, please? by bcrowell · · Score: 4, Insightful
    Last year, Pink Floyd released Dark Side on SACD, 24-bit audio at 48khz / 96khz, the amount of clarity over a CD, once the benchmark, was remarkable, I attended a launch party at was blown away even in a relatively acoustic poor setting
    I think you're deluding yourself. Audiophiles make a lot of claims that they can hear certain things, but they never test their own claims using double-blind studies in which the other variables are all controlled for.

    I teach a physics lab class, and in one of the labs, I have students test their own hearing, to see the highest and lowest frequencie they can hear. There's some individual variation, but basically the top end of everyone's range comes out to be no less than 10 kHz, and no more than 20 kHz. I have never had a single student who could hear frequencies above 20 kHz.

    The 44 kHz (IIRC) sampling frequency of a CD means that you can actually record signals with frequencies as high has 22 kHz (half the sampling frequency -- that's a methematical theorem about the discrete Fourier transform). The reason they designed CD audio around that figure was exactly because of the limits of human hearing.

    Even if there was a hypothetical human who could hear 30 kHz, there would be many other things preventing it from being useful musically. For instance, your tweeters most likely can't respond well to those frequencies. Furthermore, the music might sound worse to such a person if the 30 kHz stuff was left in. The musician couldn't hear it, and therefore couldn't adjust his tone to make it sound good. The audio engineer also couldn't hear it, and therefore couldn't judge whether it sounded good or not.

    Another practical issue is that distortion will always introduce high-frequency harmonics, so that even if you could hear those frequencies, a lot of what you were hearing would probably be spurious stuff coming from distortion.

    People who really want to hear good stereo sound should spend their effort on the two things that will make a lot of difference: (1) getting good speakers, and (2) working on the acoustics of the room, the placement of the speakers in the room, and the placement of their own head in the room. Note that all the stuff under #2 is free or cheap. The audio industry would rather have you waste your money on stuff that's expensive, which is why they promote expensive, superstitious ways of improving sound, such as gold monster cable.

    --
    Find free books.
    1. Re:double-blind, controlled test, please? by JebusIsLord · · Score: 3, Insightful

      you were told wrong. use Cooledit or something, remove everything below 11khz on a track and then give it a listen.

      16khz is usually a pretty good cutoff for music though - most MP3 encoders cut out everything over 16khz. I can hear up to 22khz test tones, but have a really hard time telling if an actual song was lowpass filtered at 16khz or not.

      --
      Jeremy
    2. Re:double-blind, controlled test, please? by Anonymous Coward · · Score: 2, Insightful

      Poster surely meant removing everything ABOVE said limits.

    3. Re:double-blind, controlled test, please? by kamelkev · · Score: 2, Interesting

      "The 44 kHz (IIRC) sampling frequency of a CD means that you can actually record signals with frequencies as high has 22 kHz (half the sampling frequency -- that's a methematical theorem about the discrete Fourier transform). The reason they designed CD audio around that figure was exactly because of the limits of human hearing."

      You are referring to the Nyquist criterion, which states that in order to guarantee you are not losing analog signal information you must sample your source at twice the frequency of the source.

      A detailed explanation of the criterion and theory is here

      I don't believe it has anything to do with Fourier, or more likely, it can be understood very simply without any knowledge of advanced mathematics (see the link)

      I both agree and disagree with you on your above points... it seems unlikely that the average person can hear about 20khz, but that doesn't necessarily mean that sampling at a higher frequency is pointless. It seems somewhat intuitive that the lower ranges would be that much more "correct". I.E. it can't hurt to sample faster, but it probably doesn't help so much.

    4. Re:double-blind, controlled test, please? by theLOUDroom · · Score: 4, Informative

      The 44 kHz (IIRC) sampling frequency of a CD means that you can actually record signals with frequencies as high has 22 kHz (half the sampling frequency -- that's a methematical theorem about the discrete Fourier transform).

      Yep, you're denfinately a physics teacher, not an EE.

      44 KHz sampling rate only lets you record frequencies up to 22KHz if you had a PERFECT d/a convertor and a PERFECT filter. It is provably impossible to implement a perfect filter. (One with a perfect cutoff and a perfectly flat passband.) Sampling at 44 KHz allows someone to design a decent recording setup with compenents that actually exist. Sampling at 96KHz gives the engineer even more breathing room when designing the filter in front of the A/D convertor. Instead of going from H(jw)=1 to H(jw)=0 in the space of 2KHz, he now can do it in 20. This means he can use a filter design with a flatter pass band. This means there is less distortion of all those frequencies that you can actually hear.

      Even if there was a hypothetical human who could hear 30 kHz, there would be many other things preventing it from being useful musically. For instance, your tweeters most likely can't respond well to those frequencies. Furthermore, the music might sound worse to such a person if the 30 kHz stuff was left in.

      Actually, it's much easier to build a tweeter than can handle 30KHz, than it is to build a subwoofer that can handle 20Hz. There are plenty of tweeters on the market right now which claim to work at 30KHz.
      Second, your statement about the 30KHz stuff making the music sound worse doesn't make any sense. The goal of an audiophile-quality setup is to reproduce the original audio exactly. We're not talking about adding in some strange 30KHz waveform, we're talking about preserving the signals that were there in the first place.

      People who really want to hear good stereo sound should spend their effort on the two things that will make a lot of difference: (1) getting good speakers, and (2) working on the acoustics of the room, the placement of the speakers in the room, and the placement of their own head in the room. Note that all the stuff under #2 is free or cheap.

      Actually, they should buy a good pair of headphones. For $300 they can buy a pair of headphones that would be tough to beat with speakers at 10X the price.

      --
      Life is too short to proofread.
    5. Re:double-blind, controlled test, please? by hankwang · · Score: 3, Informative
      To prevent high frequencies from messing up your recording, you must place a filter before the A/D convertor. This will block those high frequencies from being digitized, but it introduces a new problem: no filter is perfect.

      Yes, 96/192 kHz sampling is a good thing for recording studios for the reason that you explain. Moreover, >=24 bit recording means that you don't get aliasing problems if the signals are amplified or attenuated during the mixing process.

      However, this is all on the recording side. After sampling at >=96 kHz, you can apply a digital filter with a perfectly flat passband up to 20 kHz and stopband above 22.05 kHz, and then downsample to 44.1 kHz. In any CD player, the opposite process is performed (the famous "oversampling"): it is hard to filter the noise above 20 kHz in the raw 44.1-kHz signal. Therefore, the DAC converts the signal digitally to a 4 to 16 times higher sampling rate and with a slightly higher bitresolution (e.g. 18 or 20 bits). Then, the DAC digitally filters out everything above 22 kHz while leaving everything below 20 kHz.

      The (still digital) signal is now a "smooth line" through the supplied data points at 44 kHz. This signal is converted to a voltage by the true (non-signal-processing part of the) DAC. The part of the spectrum below 20 kHz will be exactly the same independent on whether the original input to the DAC was 44, 96, or 192 kHz. (Note: 1-bit DA convertors use a slightly different approach, but with the same result).

      As far as the bit resolution is concerned: in the final signal, 16 bits is enough for a dynamic range of 92 dB. If the hearing treshold is at 0 dB, that means that for peak levels of less than 92 dB, the resolution is fully sufficient to encode even the softest audible sounds. Note that 92 dB is quite loud: about 4 W power to a typical 87 dB/W loudspeaker at 1 m distance. It is defendable to use a bitresolution higher than 16, if you want to hear a ticking watch in the background while the music is playing at the pain treshold of 120 dB. For that, you need 5 more bits: 21 bits. On the consumer end, 24 or 32 bits is a waste of storage space.

      --
      Avantslash: low-bandwidth mobile slashdot.
    6. Re:double-blind, controlled test, please? by Jeff+DeMaagd · · Score: 2, Interesting

      I am not an audiophile but I will note these things:

      The Nyquist theory is an absolute best-case, and assumed that you sampled at the peaks.

      Even with four samples per wavelength you can get pretty weird looking sample data. IIRC, EEs try to get at least eight samples per shortest wavelength to get a decent waveform representations, less than that and you can get some noticable potential frequency and phase shifting errors. On CD audio, that makes it a little over 5kHz.

    7. Re:double-blind, controlled test, please? by Monkelectric · · Score: 3, Informative
      Wrong wrong wrong... You're assuming the POINT of sampling at higher frequencies is to get a larger frequency response -- its not. It's to REDUCE QUANTIZATION ERRORS and NOISE, and increase DYNAMIC RANGE (the real measure of a sound card).

      Quantization errors occur in the less signifigant bits, a high quality ADC will have an uncerainty of about + or - 4 bits. Think of a 10khz signal on the edge of human hearing like a nice china boy cymbal -- a cycle of a 10khz audio signal will be represented by about 4.41 samples :) I know the nyqist limit/shannons theorom says thats enough, but out here in the real world where there's noise and quantization errors its not enough, which leads me to my next point **the nyquist limit is valid only for situations where there is no noise** in other words: THERE IS NO SITUATION FOR WHICH THE NYQUIST LIMIT IS VALID. The Nyquist limit is at best, a guideline.

      So now the reason you need higher resolution/bigger samples is because that alters the noise floor. + or - 4 bits in a 24 bit recording is alot less signifigant then + or - 4 bits in a 16 bit recording. Also, imagine at 192khz your 10khz signal is now represented by 19.2 samples -- error and noise is MUCH less destructive with more samples.

      I deal with these issues every day in my studio, and the rule with audio is pretty much always, more is better. However, There is a point of diminishing returns -- and IMHO I think that point is 24bit/96khz. It is very difficult to distinguish a 96khz signal from a 192khz signal.

      --

      Religion is a gateway psychosis. -- Dave Foley

    8. Re:double-blind, controlled test, please? by Steve+Franklin · · Score: 2, Insightful

      And why again, beyond playing computer video games, do I need this on my computer? After my first experience with XP SP1 killing my onboard DVD player, I have decided to put my extra AV cash into my non-computerized, non-windowized, non-BillyGatesized, non-rebootized, instant-on audio/video system. Other than running my computer sound through my stereo, the farther away my LCD TV and audio systems are from my computer, the happier I will be. And the fewer profits Mr. Gates receives from my near-term upgrades, the more ecstactic I will be.

      Some of you guys need to wake up from your computer-induced hypnotic states and forget this convergence nonsense. It's all a big mind-freak to let Billyboy take control of an area he hasn't even the intelligence to understand.

      Sorry, I usually try not to be this blunt, but there's a point where geekdom sometimes loses track of the bigger techno-picture.

      --
      Hic iacet Arthurus, rex quondam rexque futurus.
    9. Re:double-blind, controlled test, please? by nathanh · · Score: 2
      Having 96 kHz sampling isn't about recording pitches above 22 kHz; it's about getting a better, smoother approximation of the sound waves. With 44 kHz sampling, a 22kHz sound wave is very discrete and choppy. Good ears would almost certainly detect this.

      No, this is simply wrong. It's hard to explain without the mathematics but basically the "square edges" you see will be completely removed by the obligatory low-pass filter after the DA convertor.

      Nyquist's theorem proves (with mathematics and I've done the derivation myself) that sampling at (slightly more than) 2x the highest frequency can EXACTLY reproduce the original wave. Not approximate. Exact.

      That's all assuming that you have infinite bit resolution. Of course, in practise you don't have infinite bit resolution, so that's (one reason) why SACD and DVDA sample much higher than the Nyquist rate. You can trade bit resolution for higher sampling rates because they are equivalent. But once again, you need to grok the mathematics to understand why.

    10. Re:double-blind, controlled test, please? by be-fan · · Score: 2, Insightful

      What is retarded is your belief that you, as an EE, are of a higher order of magnitude than that of a Ph.D. professor.
      --------
      I'm not an EE, nor am I the original poster.

      I will state this now and at the end of my post...You took my comment out of context for your own benefit.
      -------
      No I didn't.

      Without Physics and Mathematics, what is EE? Nothing. The same goes for Chemistry. They are both applied Physics fields.
      ---------
      While true, that doesn't mean that a physicist necessarily knows jack-shit about chemistry or electrical engineering. We were talking about audio signals, and I am more inclined to take the word of an EE than a physicist, regardless of the taxonomy of the fields. The EE works directly with this sort of thing, while the physicist only understands it indirectly, unless he is a specialist.

      --
      A deep unwavering belief is a sure sign you're missing something...
    11. Re:double-blind, controlled test, please? by nathanh · · Score: 2, Informative
      This assumes that the samples are at the peaks, if the sample 180 degrees out of phase of the signal, sampling the valleys, then you have no signal.

      Wrong. 180 degrees will indeed sample at the valleys instead of the peaks but the magnitude is the same, only the sign is different. Perhaps you meant 90 degrees.

      If an input sine wave is near 1/3rd the sampling rate, you can easily get a bunch of nasty phase and magnitude modulations as part of your output signal.

      Nope. Still wrong.

      For your own education, here is Nyquist's Theorem.

      Nyquist's theorem: A theorem, developed by H. Nyquist, which states that an analog signal waveform may be uniquely reconstructed, without error, from samples taken at equal time intervals. The sampling rate must be equal to, or greater than, twice the highest frequency component in the analog signal.

      Notice the language "without error". There is no error. It's hard to grok, and impossible to believe without doing the maths, but it is 100% true.

      Though as I said before, the real world is more fun because sampling is never exact. Errors in the times when samples are taken and errors in the magnitudes of the samples will screw you.

    12. Re:double-blind, controlled test, please? by nathanh · · Score: 2, Interesting
      Higher sampling rates do capture more accurate detail particularly on the phase accuracy.

      No, you get EXACT reproduction without having to use higher sampling rates.

      In fact, if sampling a 1kHz sine wave at 2kHz, one has to sample at the peaks or get a magnitude error. If the samples happen at the zero crossing (out of phase sample), one gets no signal samples at all:

      That's because you mistakenly think Nyquist's theorem is Fn = 2Fmax. Nyquist's theorem is Fn > 2Fmax. So what you're seeing is aliasing when Fn = 2Fmax. This causes an attenuation in the amplitude proportional to cosine of the phase difference between the sampling frequency and the signal. If you have Fn < 2Fmax then you get a "beating volume" effect as the phase difference shifts over time.

      Don't get all excited. You haven't proven Nyquist wrong. You just didn't understand what Nyquist said.

  17. memory requirements by Saville · · Score: 4, Interesting

    Since you can fit ~80minutes of music on a ~700meg CD you have ~146K/sec for your music. That is at 16bit 44.1KHz stereo songs. Now audio data will take 8.7 times as much memory if recorded in stereo, but if recorded with eight (7.1) channels each song will take almost 35x as much memory thanks to the higher sampling rate and the use of 32bit values instead of 16bit. That is 5.08 megs/sec for your audio.

    I like that this standard is very future proof, but when can we use it? Already CD sound is good enough for all but maybe 10,000 people on the planet. Most people's audio experience is probaby limited by their audio hardware, not the source sound. Hey, most people are quite happy encoding their mp3s at 128k!

    Where will the high quality sound data come from? Audio CDs are still going to be 16bit, stereo, 44KHz. DVDs have compressed audio. Almost all video games use compressed audio of some sort too because we don't have enough memory yet for even CD quality sound.

    I love that it is 7.1 and that it is very future proof, but other than making 7.1 standard it seems to be a standard for marketing to use as an advantage, not something consumers will ever use (by the time they can use it they'll have upgraded anyway). It seems that this beyond CD quality audio is just included because they can and we'll never see it in use this decade :)

    Better to overbuild than underbuild I guess. But I'm not excited about this promise of higher quality audio.

  18. 7.1? by Cyno01 · · Score: 2, Interesting

    I had this discussion the other day with some friends, none of us are audiophiles, but we all have decent setups. I have 4ch surround for my entertainment center and 4.1 for my sterio in my bedroom, but we all understand that the 5th is a front center, and we all assume, but none of us know that 6.1 has a rear center chanel. But none of us could figure out the arrangment of 7.1 surround. Is there an overhead speaker or no front center speaker and 4 evenly spaced in front. Can anyone shed some light on this?

    --
    "Sic Semper Tyrannosaurus Rex."
    1. Re:7.1? by Rufus211 · · Score: 4, Informative

      Quick google found this review that includes nice pictures.

      4.1: Front Left, Right; Mid Left, Right
      5.1: Front Left, Right, Center; Mid Left, Right
      6.1: Front Left, Right, Center; Mid Left, Right; Back Center
      7.1: Front Left, Right, Center; Mid Left, Right; Back Left, Right

      I always thought the mids ended up being farther back than shown in the picture though.

    2. Re:7.1? by EulerX07 · · Score: 2, Informative

      Check it out at dolby.

      It's basically : Left, Center, Right; SurroundX(left,rear left, rear right, right). Total overkill IMHO, 5.1 is good enough for me.

    3. Re:7.1? by geirt · · Score: 2, Informative

      In the movie world, a 7.1 audio mix usually means a 5.1 surround mix plus a conventional 2 channel stereo mix. You can synthesis a conventional stereo mix from a 5.1 surround mix, but the result may vary. That is why some movies are mixed in 7.1, which really is both a 5.1 and a stereo mix.

      When the movie is distributed on DVD or used in cinemas they use the 5.1. When the movie is sent on TV (eg. PAL with NICAM), you get the stereo mix.

      --

      RFC1925
  19. I prefer OSS by MarcQuadra · · Score: 2, Interesting

    I still prefer OSS, even on my 2.6 testbox, ALSA is about two-and-a-half more bitches to set up from scratch. I really hate having to do all the module configs when OSS just seems to work.

    All I really need is playback from my systems, ALSA is overkill for my needs, and I hate recompiling the alsa-drivers package every time I update my kernel (on 2.4 systems).

    Hopefully someone will automate or simplify ALSA for low-end use.

    --
    "Sometimes, I think Trent just needs a cup of hot chocolate and a blankie." -Tori Amos on Nine Inch Nails
    1. Re:I prefer OSS by 0x0d0a · · Score: 4, Informative

      Hopefully someone will automate or simplify ALSA for low-end use.

      The distros that have shipped ALSA as default, like SuSE, have had pretty good dummy-proof setup of ALSA for a while. Probably every major distro will be using ALSA in 2.6, which means that the remaining OSS/Free holdouts, like Red Hat, will be doing up easy-to-use UIs for ALSA.

      I also stopped using ALSA a while ago -- it was just a pain in the ass to recompile the alsa-driver package each time I upgraded the kernel, and all the software I use also supports an OSS interface (and *most* was using ALSA through the OSS compatibility interface). I expect I'll be using it again in 2.6.

      --
      May we never see th
  20. Definitely some fishy Marketing going on here by codifus · · Score: 5, Informative

    First off, 32 bit, 192 Khz, wants to appeal to those very serious about audio. 32 bit cards can have a dynamic range ratio of 144 db. That's beyond what normal humans can dfifferentiate, which is 120 db if we're lucky. Not only that, but professional 24 bit cards far exceed the needs and capabilities of most , if not every, user, with aaround 110 db of dynamic range. And they're going to put this mega high tech onboard? Hmm. 2ndly, the inclusion of Dolby. This is to appeal to the movie guys, but the real serious audio guys know that Dolby encoded audio is like an MP3, lossy compression. Serious audio guys will frown on that aspect. Incorporating these 2 aspects seems somewhat contradictory, which marketers always tend to do when trying to appeal to everyone. I, for one, remain highly skeptical. CD

  21. EAX vs Dolby Pro Logic IIx by BrookHarty · · Score: 2, Interesting

    But isn't Dolby Pro Logic IIx for creating natural surround from stereo for music/movies while EAX allows game developers to create surround sound reflections for 3d enviroments?

    And Creative has breakout boxes, multiple inputs, surround emulation software, accelerated audio, EAX# and A3D compatible, support for most games, etc. (And DRM)

    I don't see this killing off creative, but will hurt its marketshare from non-gamers.

    On the flip side, Creative labs have been quite stale, only minor updates to its Audio card line. They have been adding many other products, they even have mini-pc's, gfx, burners, mice, keyboards, etc..
    -
    Secondlife

  22. Isn't this all for naught? by midifarm · · Score: 3, Interesting
    I mean seriously... Professional recording studios at most record in 24-bit 192kHz. So where would this 32-bit recording come from? Hasn't most of the world been dumbed down to where MP3's sound good or at least good enough? I don't know too many people with a sound system worthy of playing anything 32-bit. Besides what is the point of it all?

    The hottest selling gadget of the "music" world is the MP3 player and the seemingly hottest article of contention is the online music store. None of these are even close to being prepared for 32-bit let alone the sizes of the files necessary to create such a file.

    There are a lot of comments about 6.1 and 7.1 CD's or recordings and it's all rather silly. There's no real precident of a true recording done in surround. Would you really want the lead guitar only coming from the left rear channel? The only time that I would think that it would be cool would be at a live performance, but as far as I know no one has really done anything like this.

    So were looking at several GB of needless information to recreate a CD with most likely marginal musical worth, and Intel is leading the charge? I think they're looking at their dwindling x86 market share (AMD is on the upswing, not pushing my Mac-centric views out there) and trying to find a niche by using it's brand recognition. I think Dolby and DTS will have more to say as to whether this proposed solution will have any legs.

    Remember most of the manufacturers and broadcasters still haven't totally agreed upon an officially acceptable HD format! DVD took too long. CD was all Sony, but took long enough for acceptance. Where is this leaving the consumer? A 32-bit 192kHz audio card in their computer, decoding 7.1 channels of information so they can play video games using samples that have been resampled from their original 16 or 8-bit formats.

    I think the word is overkill and it's needless. Most people can't tell the difference and for those that can, I scoff at you. I've worked with some of the best audio engineers in the world and they wouldn't be able to hear the nuances you claim. There is "air" there in higher fidelity recordings, but most speakers can't play it back any way. Ah well, thoughts?

    Peace

  23. Centrino shares some similarities with WinModems by 0x0d0a · · Score: 4, Insightful

    Centrino's wireless Ethernet controller is roughly the WiFi equivalent of a WinModem. Some of the components that are traditionally done in hardware (I'd guess the same stuff as in WinModems, like the DSP work, though I don't know the exact extent of the "softwarization") are done in software. Intel is not holding back on Linux support to secretly help out Microsoft -- I agree with you there. They're just in the same position as the WinModem vendors. If they supply their product's crown jewels -- open source the software that does a lot of the heavy lifting in their hardware product -- they've funded the R&D for what will be promptly snapped up by competitors and produced more cheaply.

    So, you are right that there is no plot to help out Microsoft, but the grandparent is right that Intel may be cagey about supporting a platform where users are rabid about having source (and much of the architecture works less reliably without source).

    Frankly, I'm frusterated with the whole laptop situation, and I wish, wish, wish that laptop vendors would make some critical mistake in the price wars and accidently commoditize their product, with standard components and form factors, so that things can be built and swapped out a la desktops.

    --
    May we never see th
  24. Protocol vs. controller by Weaselmancer · · Score: 5, Informative

    Don't get me wrong, AC97 is cheap, but it really dragged on the CPUs of the timeframe it came out.

    Well, that's not really AC97's fault.

    AC97 is really nothing more than a 5 wire signal specification. It has more to do with voltages and waveforms on wires. And a register set in the codec that the wires are talking to.

    But that's the idea of AC97 - you don't need to know who made the codec, only that it's AC97. Then it's a drop in replacement, pretty much.

    But controllers - everybody and their brother has a different idea how to talk to an AC97 codec. And it's the controller that determines the performance. Are you bit banging your codec? Then performance will suck. Are you using interrupts? Performance will improve. Using DMA? Performance will improve again. Does your DMA engine suck? Performance will drop.

    If you're having a drag on your cpu due to audio, it isn't AC97 that's at fault. It's someone's lousy idea for a controller. AC97 is a spec, not a gadget.

    Weaselmancer

    --
    Weaselmancer
    rediculous.
  25. The labels will botch it by Animats · · Score: 2, Informative
    Then we'll have the labels compress everything so that it's up near the top of the scale anyway. "Nobody wants to be the softest CD in the changer". Most popular music is compressed so hard it's badly damaged.

    The main reason you need more than 16 bits is because, during soft passages, most of the high bits are zero and you may effectively have only six or four bit audio. Classical recordings that aren't compressed really do suffer from this problem.

    But really, the number of people who buy classical piano recordings is small.

    If the industry can agree that the reference level for popular audio is somewhere well below 100%, this could work out. But that won't happen.

  26. Wow by ajagci · · Score: 2, Interesting

    The Intel High Definition Audio solution will have increased bandwidth that allows for 192 kHz, 32-bit, multi-channel audio

    This is so that my eight-eared mutant pet bat from outer space can finally have a full high-fidelity experience.

    For regular humans, of course, CD-quality audio is already overkill.

  27. Does it have built in DRM ? by havaloc · · Score: 2, Interesting

    I'm surprised no one has brought this up, but does it have any sort of DRM (Digital Rights) built in to it? If so, no thanks!

  28. Re:Can you recommend some computer speakers? by swordgeek · · Score: 2, Informative

    You seem to misunderstand the meaning of speaker wattage. This is the MAXIMUM power the speakers can withstand for a short time (I believe 1/4 second) without blowing up. It has no bearing whatsoever on speaker quality, efficiency (how loud they play at a given volume setting), amplifier requirements, or how loud they're "designed" to be played. Ignore that number entirely; it has no relevance for you.

    That said, I settled on the Logitech speakers for my computer after a lot of listening--they're the only ones I found that sounded like music. I will admit that I didn't listen to the Klipsches, because they were out of my price range. I expect that they'd be quite good, as they make non-computer speakers which are very nice indeed. (Mind you, Altec-Lansing makes stereo speakers too, and their computer offerings are without exception, unmitigated shite!)

    If you have a passable amp, then unpowered stereo speakers are likely to be the best choice. A few years ago, it would have been the only choice, but a few computer speakers are at least considering.

    But ignore the 200watt rating. Even if it were valid (it's not), it's completely meaningless and irrelevant to your shopping.

    --

    "People who do stupid things with hazardous materials often die." -- Jim Davidson on alt.folklore.urban