sam9,
I might disagree with what you said regarding transient attacks and 20kHz signals, if you mean that signals beyond 20 kHz are non-audible. Many instruments have harmonic overtones that extend beyond 20 kHz. While any number of instruments may have the same fundamental frequency, it is the harmonic overtone package, if you will, that allows us to distinguish a Stradivari from a Guarneri violin. Many of these subtleties are lost with most speakers, let alone the source material (how good were the microphones, the mixing consoles, etc?), but this doesn't reduce the desire or need to have these frequencies properly reproduced. For example, the presence of greater than 20kHz frequencies may influence audible frequencies via modulation. So even if you don't hear the 20kHz signals directly, you may notice indirectly their absence.
I'd refer you to James Boyk, from CalTech ( http://www.cco.caltech.edu/~boyk/ ). He's written a few articles about higher frequencies and their affect on recorded music, such as http://www.cco.caltech.edu/~boyk/inflection.htm and http://www.tnt-audio.com/intervis/boyke.html
However, if you meant that as long as there was no slew-rate limiting at 20kHz of transients, and that such a standard is sufficient for audio, I'd probably agree with you.
Regards, Robert
I might disagree with what you said regarding transient attacks and 20kHz signals, if you mean that signals beyond 20 kHz are non-audible. Many instruments have harmonic overtones that extend beyond 20 kHz. While any number of instruments may have the same fundamental frequency, it is the harmonic overtone package, if you will, that allows us to distinguish a Stradivari from a Guarneri violin. Many of these subtleties are lost with most speakers, let alone the source material (how good were the microphones, the mixing consoles, etc?), but this doesn't reduce the desire or need to have these frequencies properly reproduced. For example, the presence of greater than 20kHz frequencies may influence audible frequencies via modulation. So even if you don't hear the 20kHz signals directly, you may notice indirectly their absence.
I'd refer you to James Boyk, from CalTech ( http://www.cco.caltech.edu/~boyk/ ). He's written a few articles about higher frequencies and their affect on recorded music, such as http://www.cco.caltech.edu/~boyk/inflection.htm and http://www.tnt-audio.com/intervis/boyke.html
However, if you meant that as long as there was no slew-rate limiting at 20kHz of transients, and that such a standard is sufficient for audio, I'd probably agree with you.
Regards, Robert
I can't find a logical or scientific explanation why the higher sampling rates or DSD sound better, but I definitely HEAR it! I would really like to know what is responsible for the analog-like smoothness I hear.
Yeah- 20Khz bandwidth seems like it should be adequate, with no slew limitations. I know that I barely hear up to about 17khz! But, I can surely tell a difference with SACD- there is a dramatic lack of irritation to my ears, and during loud and busy passages in the music, it dosen't sound congested. Cymbals wash and shimmer instead of having that "white noise" effect (I can't remember who described it that way, but that is a wonderful description of cymbals on 16 bit machines). All of this comes from comparisons to my experience playing and listening to live music. Unfortunately, whether anyone who isn't at this site or others like it has a system that will show enough of an difference to give SACD a fighting chance of surviving or not is yet to be seen. I believe most people could tell, even on a decent low-priced system.
I really wish that everyone here would get to a dealer to have a listen, or pick up one of the cheap sony players (I saw the NS500V going for $169 US at best buy!)and start supporting this format before it goes the way of beta, digital compact cassette (a loser to start with), and vinyl. To my ears, it is really that good.
Steve
Yeah- 20Khz bandwidth seems like it should be adequate, with no slew limitations. I know that I barely hear up to about 17khz! But, I can surely tell a difference with SACD- there is a dramatic lack of irritation to my ears, and during loud and busy passages in the music, it dosen't sound congested. Cymbals wash and shimmer instead of having that "white noise" effect (I can't remember who described it that way, but that is a wonderful description of cymbals on 16 bit machines). All of this comes from comparisons to my experience playing and listening to live music. Unfortunately, whether anyone who isn't at this site or others like it has a system that will show enough of an difference to give SACD a fighting chance of surviving or not is yet to be seen. I believe most people could tell, even on a decent low-priced system.
I really wish that everyone here would get to a dealer to have a listen, or pick up one of the cheap sony players (I saw the NS500V going for $169 US at best buy!)and start supporting this format before it goes the way of beta, digital compact cassette (a loser to start with), and vinyl. To my ears, it is really that good.
Steve
sam9
I'm not an expert on human hearing but I do remember reading briefly of some of the discussion around two groups of experiments. In the first one, subjects were asked to listen to two tones the first was a pure sine tone well within the normal audible range. (Maybe it was 8k or 10k.) The second was the same tone with a second tone beyond 20kHz added. The subjects could clearly hear the difference. It was concluded that audio reproduction above 20kHz was important and audible.
The second experiment modified this. The first tone was the same. The second was the same except that after the two tones were combined they were passed through a steep filter that effectively cut off everthing above 20kHz. The subjects could still hear the difference. I believe they were also asked to distinguish between the the second signal from both experients (two tones added but one with and one without the 20kHz filter) and could not distinguish between them. The conclusion was that the subjects could not hear sounds above 20kHz but could hear harmonics and/or modulation artifacts caused by the addition of the >20kHz tone and since they could still hear the artifacts after the combinred signal passed a 20Khz filter the frequencies artifacts must be below 20kHz.
Assuming these two experiments to be sound. I would surmise that a recording scheme limited to 20kHz max is sufficient to reproduce any frequency we can hear even if the sourse includes frequencies above 20kHz.
There is one proviso or caveat: the recording scheme should not produce distortions of any kind in the range below 20kHz because of its inability to deal with tones above that. This is a valid reason why some speakers may respond beyond 20kHz - it may easier to be flat up to a given frequency if it responds somewhat beyond that point. Following this logic, an ADC with a "perfect" conversion algorth should have no problem if it's upper limit is 20kHz. Since algorithms are not always perfect there could be an advantage to a sampling rate that lets the ADC opperate beyond 20kHz. -However, the potential to improve the algorithm could make that redundant.
An aside: audio amplifiers often are designed to have a response far above 20kHz. I'm not that knowledgeable but I gather the reason is that if the circuit is allowed to roll off too soon, distortions will be introduced above 20kHz that will generat harmonics and/or modulations at lower frequencies where the will be audible.
I'm not an expert on human hearing but I do remember reading briefly of some of the discussion around two groups of experiments. In the first one, subjects were asked to listen to two tones the first was a pure sine tone well within the normal audible range. (Maybe it was 8k or 10k.) The second was the same tone with a second tone beyond 20kHz added. The subjects could clearly hear the difference. It was concluded that audio reproduction above 20kHz was important and audible.
The second experiment modified this. The first tone was the same. The second was the same except that after the two tones were combined they were passed through a steep filter that effectively cut off everthing above 20kHz. The subjects could still hear the difference. I believe they were also asked to distinguish between the the second signal from both experients (two tones added but one with and one without the 20kHz filter) and could not distinguish between them. The conclusion was that the subjects could not hear sounds above 20kHz but could hear harmonics and/or modulation artifacts caused by the addition of the >20kHz tone and since they could still hear the artifacts after the combinred signal passed a 20Khz filter the frequencies artifacts must be below 20kHz.
Assuming these two experiments to be sound. I would surmise that a recording scheme limited to 20kHz max is sufficient to reproduce any frequency we can hear even if the sourse includes frequencies above 20kHz.
There is one proviso or caveat: the recording scheme should not produce distortions of any kind in the range below 20kHz because of its inability to deal with tones above that. This is a valid reason why some speakers may respond beyond 20kHz - it may easier to be flat up to a given frequency if it responds somewhat beyond that point. Following this logic, an ADC with a "perfect" conversion algorth should have no problem if it's upper limit is 20kHz. Since algorithms are not always perfect there could be an advantage to a sampling rate that lets the ADC opperate beyond 20kHz. -However, the potential to improve the algorithm could make that redundant.
An aside: audio amplifiers often are designed to have a response far above 20kHz. I'm not that knowledgeable but I gather the reason is that if the circuit is allowed to roll off too soon, distortions will be introduced above 20kHz that will generat harmonics and/or modulations at lower frequencies where the will be audible.
in an analog amplifier, limited HF response usually indicates slew rate limiting (unless the roll-off is deliberate due to a filter). slew rate limiting translates to distortion and poor transient response. most properly-designed amplifier circuits have intrinisic frequency response well past 20kHz though - 100kHz is not uncommon and i've even seen a few flat to 200kHz and beyond.
The higher sample rate means that higher frequencies can be reproducted. Even if you cant hear 30KHz or 50 KHz tones it doesnt mean that you cant hear these frequencies as harmonics. Harmonics can be heard up to 50 or even 100KHz. It makes the difference in sound between different musical intruments.
That why SACD sounds better.
That why SACD sounds better.
i'm not so sure about 50kHz or 100kHz being audible, but what do i know.
are you sure it's the response itself that's important, or the things that go along with extended HF response? one thing SACD and hi-rez PCM give u is greatly improved phase response, even at lower frequencies. this is definitely audible. i think people spend too much time looking at frequency and not enough at phase. timing is everything.
are you sure it's the response itself that's important, or the things that go along with extended HF response? one thing SACD and hi-rez PCM give u is greatly improved phase response, even at lower frequencies. this is definitely audible. i think people spend too much time looking at frequency and not enough at phase. timing is everything.
sam9
My personal impression is that really good CD's such as Chesky, RealTime, MoFi and so on don't have any problem with high frequencies. Or with much else.
My take on SACD, DVD-A and 24/96 discs is that the people making them are simply more attentive to the end product than is the case with run of the mill CD's.
I think this is analogous to the case with LPs. They CAN sound very good. Virtually any produced these days will sound suprisingly better than the mass market LPs of the 60's, 70's and 80's simply because the people making them care more.
My conclusion: there is more at work in these comparrisons than just the technology. The people you make the product and their skill and attitude are even more important.
My personal impression is that really good CD's such as Chesky, RealTime, MoFi and so on don't have any problem with high frequencies. Or with much else.
My take on SACD, DVD-A and 24/96 discs is that the people making them are simply more attentive to the end product than is the case with run of the mill CD's.
I think this is analogous to the case with LPs. They CAN sound very good. Virtually any produced these days will sound suprisingly better than the mass market LPs of the 60's, 70's and 80's simply because the people making them care more.
My conclusion: there is more at work in these comparrisons than just the technology. The people you make the product and their skill and attitude are even more important.
YES!
i've said it again and again, but PHASE PHASE PHASE! everyone talks about bandwidth, dynamic range, freq response etc. etc. but neglect phase. WHEN we hear something is an extremely important component of WHAT we hear. also, human phase accuity is better than originally thought several decades when Bell Labs did their seminal research on human hearing, i have heard based on some military research that humans can distinguish phase well into several kilohertz.
i've said it again and again, but PHASE PHASE PHASE! everyone talks about bandwidth, dynamic range, freq response etc. etc. but neglect phase. WHEN we hear something is an extremely important component of WHAT we hear. also, human phase accuity is better than originally thought several decades when Bell Labs did their seminal research on human hearing, i have heard based on some military research that humans can distinguish phase well into several kilohertz.
People that make them don't care. All they care about is getting your hard earn cash. They invest in more money so the bands pick recording studio that has the best deals.
Phase only comes down on room acoustics and loudspeakers. Using either PCM or SACD only records the frequencies that is produced by vocals and instruments.
Sam9,
have you heard SACD? I find quite a bit wrong with the sound of even the best 16/44.1 There is an obvious difference when listening to an SACD. I would even dare say that there is an obvious difference when listening to a remaster to SACD, even if it's not the best recording. I guess it comes down to one less weak link in the audio chain, as far as those recordings go.
As for phase- it must be one of the most overlooked distortions. It could possibly explain a lot about why it's generally perceived that simpler signal paths sound better. The only problem is that a speaker produces quite a bit of phase distortion. Why would we hear the small distortions caused by a CD player, and yet be able to ignore the huge errors in our loudspeakers?
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