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The Successor to AC'97: Intel High Definition Audio

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."

41 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: 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

    3. 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.

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    4. 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.

    5. 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.

    6. 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.

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    7. 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.

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  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 DrSkwid · · Score: 3, Insightful


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

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    4. 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 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...

  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.

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  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 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

    3. 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
  7. A next gen audio specification? by xankar · · Score: 4, Funny

    Hear hear!

    Pun completely and totally intended.

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  8. 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.

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

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

  10. 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.

  11. 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).

  12. 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.

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  13. 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!)

  14. 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.

    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 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.

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    3. 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.

    4. 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.

      --

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  15. 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.

  16. 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.

  17. 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

  18. 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.

  19. 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

  20. 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.

  21. 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.

  22. 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.