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Why are sound cards marketed wrong? - Page 2

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[Yes, you behave too]

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post #14 of 53
jerry.gif


On topic. Its no surprise that sound cards are marketed in the wrong way. Marketing people after all are rarely audiophiles them selves. The same holds true for most enthusiast grade hardware. Audiophile's don't need the guy on newegg to tell them anything as they them selves know how to interpret the specs and if need be look up competent reviewers if they want/need a more subjective viewpoint.
Edited by Bit_reaper - 4/23/13 at 11:29am
    
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post #15 of 53
Quote:
Originally Posted by airisom2 View Post

Well, afaik, dynamic range and SNR are different, with SNR being like you said, and dynamic range being lowest and loudest parts that the DAC is actually able to process. Yeah, they're similar, but not similar enough to be able to change one out for the other. Most DAC's already have a good enough dynamic range, so its usually blanketed over like most of the other specs.
Yeah they're different. With SNR, tell me the conditions under which the signal was measured and the ones under which the noise was measured. Noise level depends.

Actually, just read this:
http://nwavguy.blogspot.com/2011/09/noise-dynamic-range.html
Quote:
You also have to take into consideration on the sensitivity of the headphones, the quality of the power delivery of the amp, the efficiency of the headphone's drivers, and the "synergy" that the headphones have with the amp. You probably know that lower impedance headphones need more mW than volts, and the opposite is true for higher impedance headphones. I've read that the amps on these sound cards aren't able to really power higher impedance headphones without it sounding weak and whatnot (underpowered). All the "600ohm mode" is, is a gain setting anyway. I had a Schiit Magni, which is capable of outputting 1.2w into 32ohms. In reality, I couldn't get it to go louder than either my receiver or my FiiO E17 because it would start clipping the sound (keep in mind that I eq up lower frequencies). The Magni is roughly 4 times more powerful than an E17. Although some amps have high power ratings, doesn't mean that you're actually able to use it all, which is another thing I'm trying to point out with these sound cards.
Yes, it's about the electrical capability of the amp, the headphone sensitivity, headphone impedance, listening volume.

Anything that claims to be able to handle 600 ohms (anything can handle 600 ohms, it's just a matter of volume, performance, and so on) just has a relatively high (voltage) output. It's kind of vague because it depends on so many other things.

Anyway, an important point like you said is that most people aren't using the power their amps are capable of.

We're interested in the performance of the amp under the conditions we're operating them: the headphones, input signals (audio-range frequencies, certain levels), volume output, etc. You can argue sound quality if you want. That has to do with the frequency response, distortion level, and everything else. Just note that for a wide range of reasons, peoples' impressions may not really be the greatest indicators of actual differences in sound produced.
Quote:
As far as the compression and stuff goes, I was under the impression that if the DAC isn't able to completely play files above it's rated dynamic range, that the chip does some kind of process that cuts off, or compresses, the ranges that it isn't able to convert.
No it doesn't. Regarding dynamic range, see above link. If a DAC can only do 16-bit output (not 24-bit output), then info needs to be truncated—or really, dithered—to 16-bits before being sent to the DAC. That's one matter, and it's not even the DAC's responsibility.

In any situation, the DAC tries to output an electrical signal that corresponds to the information it gets. Give it a 16-bit word, and it'll try to output that, with some level of accuracy. Give it a 24-bit word, and it'll try to output that too, if it can. In either case, the full-scale level is similar; extra bit depth just gives extra information, analogous to having more numbers after the decimal point.

Let's say you have a real-world 24-bit DAC and you're feeding it 24-bit information. Then the noise floor is effectively about as great as signals in the least 5-9 significant bits or so, depending on the actual device's performance. So any info in say bits 21-24 of a 24-bit word are pretty much drowned out by the noise floor anyway. There isn't any kind of compression or anything. It's just that the information is effectively as good as lost because of the hardware limitation: the noise level.
Edited by mikeaj - 4/23/13 at 12:53pm
post #16 of 53
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Quote:
Originally Posted by mikeaj View Post

In any situation, the DAC tries to output an electrical signal that corresponds to the information it gets. Give it a 16-bit word, and it'll try to output that, with some level of accuracy. Give it a 24-bit word, and it'll try to output that too, if it can. In either case, the full-scale level is similar; extra bit depth just gives extra information, analogous to having more numbers after the decimal point.

Let's say you have a real-world 24-bit DAC and you're feeding it 24-bit information. Then the noise floor is effectively about as great as signals in the least 5-9 significant bits or so, depending on the actual device's performance. So any info in say bits 21-24 of a 24-bit word are pretty much drowned out by the noise floor anyway. There isn't any kind of compression or anything. It's just that the information is effectively as good as lost because of the hardware limitation: the noise level.

Alright, I think I'm getting this now. So basically, it outputs all 24-bits, but because of the noise floor, some of the extra bits from the file gets absorbed by it. I was thinking that the maximum SNR was set in stone, and you couldn't go above it. So, let's take the Xonar STX for instance. It has a 124dB SNR on the line-out port, and a 24-bit file needs a ~144dB SNR in order to to be correctly reproduced. From what I've read, for every bit you have, the SNR increases by 6dB. Do the math, and 3.333->bits are being absorbed by the DAC's inadequate noise floor, so you're really only listening to a ~21-bit file. I guess my question now is how the bit depth correlates to what your hearing, and what bits in a file corresponds to what. Is it like network packets to where you have some identifying bits, and the other data bits, or is it something totally different?

And here's another thing I don't understand (I think). By following the Nyquist theory, the peak frequency that is created is roughly half the sample rate. So, if you have a 44.1k file, the frequency range will peak at ~22k. You apply that to a 96k file, and it jumps up to 48k. The average human ear can only hear up to around 20k, so what's the point of sampling higher than the threshold of human hearing if we can't hear the differences? It kinda reminds me of some of the headphones reviews I read. Some are rated from like 15Hz to 41KHz, yet we can't necessarily hear those frequencies. People usually say that because the frequency range is more extended at either end, it gives the perceptible sound range greater clarity. Is that what's basically going on with a 96+ sample rate? That sounds more subjective than anything.

Another question that I have is how the bit depth relates to the frequencies being played. Here's my stand on it: Having a 16-bit file is like looking at a 20-band frequency spectrum (the visualization that looks like bar graphs), while a 24-bit one is like looking at a 80-band frequency spectrum. Both are playing the same thing, but the 24-bit file has greater levels of resolution that allows for a greater sense of separation between the frequency levels. With that said, do 16-bit files, although being able to reproduce the 20Hz to 20KHz frequency spectrum, have some grey areas in it? For instance, a 16-bit file may be able to reproduce a sound at 12,945Hz, but will it be able to reproduce a sound at 12,945.3463456Hz? Is that where the benefits of a 24-bit file come into play? Then again, no DAC is able to reproduce all 24 bits because of circuitry limitations, so is that sense of separation diluted even further (separation/sonic difference between 16bit and 24bit and the separation between precise frequencies)?

Well, I guess this is turning more into an audio information thread, oh well. It helps out the community regardless, so I guess it isn't a bad thing.
post #17 of 53
Quote:
Originally Posted by airisom2 View Post

Alright, I think I'm getting this now. So basically, it outputs all 24-bits, but because of the noise floor, some of the extra bits from the file gets absorbed by it. I was thinking that the maximum SNR was set in stone, and you couldn't go above it. So, let's take the Xonar STX for instance. It has a 124dB SNR on the line-out port, and a 24-bit file needs a ~144dB SNR in order to to be correctly reproduced. From what I've read, for every bit you have, the SNR increases by 6dB. Do the math, and 3.333->bits are being absorbed by the DAC's inadequate noise floor, so you're really only listening to a ~21-bit file. I guess my question now is how the bit depth correlates to what your hearing, and what bits in a file corresponds to what. Is it like network packets to where you have some identifying bits, and the other data bits, or is it something totally different?
Raw audio data is almost always Linear Pulse-Code Modulation. 24 bits of 1s and 0s indicate some level. There's one bit for the sign (positive or negative) and the rest are in decreasing importance. Music files have some identifying headers and so on, that aren't part of the raw data. But if you look in like a .wav file, you'll see that kind of information. In compressed audio, it's stored a more efficient way.
01011101 11011011 is close to
01011101 11011000 but quite different than
00011101 11011011.

Except in extreme scenarios, even 16 bits is pretty overkill. Most recordings have noise from the mics / environment that's over the 16th bit anyway. There's a demo in this video: video (Click to show)
with reduced bit depth, starting from 16 bits and going own. Check the audio files here (the YT is lossy of course):
http://www.ethanwiner.com/aes/

Quote:
And here's another thing I don't understand (I think). By following the Nyquist theory, the peak frequency that is created is roughly half the sample rate. So, if you have a 44.1k file, the frequency range will peak at ~22k. You apply that to a 96k file, and it jumps up to 48k. The average human ear can only hear up to around 20k, so what's the point of sampling higher than the threshold of human hearing if we can't hear the differences? It kinda reminds me of some of the headphones reviews I read. Some are rated from like 15Hz to 41KHz, yet we can't necessarily hear those frequencies. People usually say that because the frequency range is more extended at either end, it gives the perceptible sound range greater clarity. Is that what's basically going on with a 96+ sample rate? That sounds more subjective than anything.
It's not the peak frequency that is created, but the highest frequency that can be mathematically represented. The frequency content is a result of whatever is making the sound. You can sample at 4 MHz a bass playing—you just won't get any supersonic frequency content out of it (supposing we have imaginary mics and an ADC that can record that high).

Average young person's ear might detect 20 kHz, and that's with a relatively loud sound and probably with nothing else playing to mask it. Average person falls well short of that.

Because of the limits of hardware ADC performance (sharper low-pass filters are harder to do well), maybe so you can avoid a sample rate conversion later on, and for some small practical concerns, it could be better to sample at higher than 44.1 kHz, which already allows for a bit over 20 kHz of audio. Not much benefit over say 50-60 kHz or so (and even then, for all practical purposes these days, 44.1 kHz is fine), but the lowest commonly-seen rates are 88.2 kHz and 96 kHz. Harmless overkill, mostly, except that it means larger file sizes.

Or maybe you're recording stuff for analysis or non-human consumption.

Quote:
Another question that I have is how the bit depth relates to the frequencies being played. Here's my stand on it: Having a 16-bit file is like looking at a 20-band frequency spectrum (the visualization that looks like bar graphs), while a 24-bit one is like looking at a 80-band frequency spectrum. Both are playing the same thing, but the 24-bit file has greater levels of resolution that allows for a greater sense of separation between the frequency levels. With that said, do 16-bit files, although being able to reproduce the 20Hz to 20KHz frequency spectrum, have some grey areas in it? For instance, a 16-bit file may be able to reproduce a sound at 12,945Hz, but will it be able to reproduce a sound at 12,945.3463456Hz? Is that where the benefits of a 24-bit file come into play? Then again, no DAC is able to reproduce all 24 bits because of circuitry limitations, so is that sense of separation diluted even further (separation/sonic difference between 16bit and 24bit and the separation between precise frequencies)?
The ability to separate frequencies is a result of the sampling rate, not the bit depth. If it's under 1/2 the sampling rate, it can be represented. (Too close to exactly 1/2, and practical real-world hardware may have issues / cut it off.) Any frequency under it can be represented: 12,945Hz or 12,945.3463456Hz or whatever, no matter the bit depth. Yes, and the phase correctly as well.

Okay, you might ask where all that information is coming from? How can you tell those frequencies apart? It's from the number of samples. Suppose you had an infinite (or a whole lot) of samples of each. Eventually you'd see that one of them is bobbing up and down a little more frequently than the other. (That's the non-calculus, non-systems theory response.)

Here's a video on digital audio fundamentals, may be worth a look for you:
http://xiph.org/video/vid2.shtml
Edited by mikeaj - 4/23/13 at 6:32pm
post #18 of 53
Quote:
Originally Posted by mikeaj View Post

Raw audio data is almost always Linear Pulse-Code Modulation. 24 bits of 1s and 0s indicate some level. There's one bit for the sign (positive or negative) and the rest are in decreasing importance. Music files have some identifying headers and so on, that aren't part of the raw data. But if you look in like a .wav file, you'll see that kind of information. In compressed audio, it's stored a more efficient way.
01011101 11011011 is close to
01011101 11011000 but quite different than
00011101 11011011.

Except in extreme scenarios, even 16 bits is pretty overkill. Most recordings have noise from the mics / environment that's over the 16th bit anyway. There's a demo in this video: video (Click to show)
with reduced bit depth, starting from 16 bits and going own. Check the audio files here (the YT is lossy of course):
http://www.ethanwiner.com/aes/
It's not the peak frequency that is created, but the highest frequency that can be mathematically represented. The frequency content is a result of whatever is making the sound. You can sample at 4 MHz a bass playing—you just won't get any supersonic frequency content out of it (supposing we have imaginary mics and an ADC that can record that high).

Average young person's ear might detect 20 kHz, and that's with a relatively loud sound and probably with nothing else playing to mask it. Average person falls well short of that.

Because of the limits of hardware ADC performance (sharper low-pass filters are harder to do well), maybe so you can avoid a sample rate conversion later on, and for some small practical concerns, it could be better to sample at higher than 44.1 kHz, which already allows for a bit over 20 kHz of audio. Not much benefit over say 50-60 kHz or so (and even then, for all practical purposes these days, 44.1 kHz is fine), but the lowest commonly-seen rates are 88.2 kHz and 96 kHz. Harmless overkill, mostly, except that it means larger file sizes.

Or maybe you're recording stuff for analysis or non-human consumption.
The ability to separate frequencies is a result of the sampling rate, not the bit depth. If it's under 1/2 the sampling rate, it can be represented. (Too close to exactly 1/2, and practical real-world hardware may have issues / cut it off.) Any frequency under it can be represented: 12,945Hz or 12,945.3463456Hz or whatever, no matter the bit depth. Yes, and the phase correctly as well.

Okay, you might ask where all that information is coming from? How can you tell those frequencies apart? It's from the number of samples. Suppose you had an infinite (or a whole lot) of samples of each. Eventually you'd see that one of them is bobbing up and down a little more frequently than the other. (That's the non-calculus, non-systems theory response.)

Here's a video on digital audio fundamentals, may be worth a look for you:
http://xiph.org/video/vid2.shtml
Interesting stuff, I just watched the last video. I understand it for the best part except he slid down to 1/4 bit-depth? How do you have less than one bit in a sample? All I can visualize is you only have a bit every fourth sample, but that's a lower sample rate not bit depth.
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post #19 of 53
Quote:
Originally Posted by Darren9 View Post

Interesting stuff, I just watched the last video. I understand it for the best part except he slid down to 1/4 bit-depth? How do you have less than one bit in a sample? All I can visualize is you only have a bit every fourth sample, but that's a lower sample rate not bit depth.

Erm, it's been a while, since I saw it, but I poked around until I found that part and I think I know what you mean.

The amplitude of the input (I forgot the setup, was it the output of the sound card back into its input?) was scaled to be 1/4 of a bit at 16-bit depth for illustrative purposes. It's ___ volts input into an ADC, whatever the right number is. That's just an input level. That could be from a generator, a mic, or some sound card output—say, either by using a 24-bit output, by scaling down with an attenuator or amp, or something else. It's not as if somehow you're getting an 18-bit output of a 16-bit system.
post #20 of 53
Thread Starter 
After I watched the two videos (1, 2) from xiph.org, I understand the "basics" now. I would summarize it, but you guys would probably rather watch the videos instead of trying to dissect my text.

This page explains a whole lot more info too.

Although I'm not trying to become an audio engineer or anything like that (well,not right now, at least. I'm just learning this stuff for fun), this stuff is very interesting. So, if anyone would like to throw in some more information, then it would be very much appreciated by the OCN community, and more importantly, by me biggrin.gif
Edited by airisom2 - 4/24/13 at 1:24am
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