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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."
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.
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.
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).
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
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.
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
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.
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.
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
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.
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
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.
:)
.15 : .81 .18 : .90 .20 : .95 .25 : 1 .30 : .95 .32 : .90 .35 : .81 .32-.18=.14
/20KHz=9.6
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:
1/.14 = 7.14
20KHz*7.14=143KHz
This isn't RMS calibrated, but so what.
192KHz:
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