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What's better: modern built-in motherboard sound chip or old sound card?
superticker:
This thread has now been moved to the programmer's area since it's about driver design. I have a reply there. https://www.donationcoder.com/forum/index.php?topic=6131.msg43263#msg43263
I know NT isn't real-time, but 50ms for an IRQ to be handled sounds ludicruous. And AFAIK, data processing isn't done directly in the IRQ handler, instead some state information is saved and passed down as an IRP, and the IRQ handler itself finishes quickly. Iirc linux does somewhat the same by having "high" and "low" parts of their IRQ handlers....
-f0dder (November 10, 2006, 04:51 PM)
--- End quote ---
Continued on the new thread.
gjehle:
well,
if it comes to audio you will have, at some point, an analog signal
and unless you're using digital outputs, you'll have these on your soundcard (aka classic speaker/line out)
while digital electronics operate within well defined limites and are fairly well to master, what it comes down to is your digntal/analog converter.
THAT's actually the part a lot of (lower to middle end) cards have a poor (cheap) design.
a lot of cards use a simple R/C combo to smoothen the signal, that's cheap (2 pieces) and easy.
but if you want a good quality, it sucks.
also, (sorry if i'm repeating something that has already been said, i haven't read _all_ of it) on-board cards are more prone to noise, that's one reason why a lot of middle to high-end cards are not even pci(e) but firewire or usb, just to get away from all the nasty noise producing electronics inside the computer case.
if you want sound = get onboard
if you want decent sound = get a card
if you want good sound = get a external soundcard
and if you're DAC sucks (simple R/C smoother) the digital part can be as good as it gets, it'll be ruined just before it reaches your speakers.
and why do a lot of manufactures use simple R/C ?
it's easy to build cheap digital electronics (you really dont have to care about noise that much if it's digital anyways)
but it's hard and expensive to build low-noise analog devices.
they also tend to use up quite a bit of PCB real estate
ok, that's all for my 2ct
superticker:
... why do a lot of manufactures use simple R/C ?
It's easy to build cheap digital electronics (you really don't have to care about noise that much if it's digital anyways).
But it's hard and expensive to build low-noise analog devices.
They also tend to use up quite a bit of PCB real estate
-gjehle (November 11, 2006, 09:04 PM)
--- End quote ---
All the above is true, but there's a more important--theoretical--reason why a cheap first-order RC filter is used, and that has to do with distortion. Any analog filer is a "causal filter" such that it can't know the future. In contrast, digital FIR filters can be non-causal such that the value of the points are known before time zero and after time zero. If we balance the filter coefficients across time zero, then we have a "zero-phase" filter across all frequencies.
Having zero-phase delay across all frequencies creates a distortionless filter for our application, so we really favor digital filtering over analog filtering when it's feasible.
For audio play back, the strategy is to oversample the signal by x4 (or better) using a first-ordered anti-aliasing analog filter (to minimize phase distortion) at ultrasonic frequencies, either 45K or 90KHz. Then we run it through a zero-phase digital filter to achieve a "perfect" cut-off at whatever frequency we want, say 45/2 KHz.
Remember, for a non-causal filter, the cutoff can be perfectly sharp; whereas, for a causal filter (which includes any analog filter), there will always be a response roll off. The higher the order, the sharper the roll off and phase delay (leading to more distortion). So our cheap first-order RC filter produces less ultrasonic distortion than a higher ordered analog one. But the assumption is standard audio equipment won't reproduce those ultrasonic frequencies (say at 90KHz) anyway.
I should add my area is in scientific instrumentation design, not sound card design. Frankly, I couldn't tell you whether the PC sound chips are being design right or not. But it wouldn't cost anymore to design them with x4 oversampling in mind and zero-phase digital filtering at their outputs. I do know that some audio component CD players employ x4 and x8 oversampling and digital comb filtering, but I really can't speak for the sound-chip PC world. Perhaps someone else knows for sure or has a link to a sound-chip spec sheet.
superboyac:
All the above is true, but there's a more important--theoretical--reason why a cheap first-order RC filter is used, and that has to do with distortion. Any analog filer is a "causal filter" such that it can't know the future. In contrast, digital FIR filters can be non-causal such that the value of the points are known before time zero and after time zero. If we balance the filter coefficients across time zero, then we have a "zero-phase" filter across all frequencies.
Having zero-phase delay across all frequencies creates a distortionless filter for our application, so we really favor digital filtering over analog filtering when it's feasible.
For audio play back, the strategy is to oversample the signal by x4 (or better) using a first-ordered anti-aliasing analog filter (to minimize phase distortion) at ultrasonic frequencies, either 45K or 90KHz. Then we run it through a zero-phase digital filter to achieve a "perfect" cut-off at whatever frequency we want, say 45/2 KHz.
Remember, for a non-causal filter, the cutoff can be perfectly sharp; whereas, for a causal filter (which includes any analog filter), there will always be a response roll off. The higher the order, the sharper the roll off and phase delay (leading to more distortion). So our cheap first-order RC filter produces less ultrasonic distortion than a higher ordered analog one. But the assumption is standard audio equipment won't reproduce those ultrasonic frequencies (say at 90KHz) anyway.
I should add my area is in scientific instrumentation design, not sound card design. Frankly, I couldn't tell you whether the PC sound chips are being design right or not. But it wouldn't cost anymore to design them with x4 oversampling in mind and zero-phase digital filtering at their outputs. I do know that some audio component CD players employ x4 and x8 oversampling and digital comb filtering, but I really can't speak for the sound-chip PC world. Perhaps someone else knows for sure or has a link to a sound-chip spec sheet.
-superticker (November 12, 2006, 02:15 AM)
--- End quote ---
yeah.......what he said. ;)
mouser:
my motherboard decided to help me solve this dilemna by not working..
so now i'm off to buy a sound card.. :tellme:
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