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What happened to perpetual technologies?
I cannot locate their web site, I am just re-directed to the av123
site which sells the units. Are they out of business? What about the room correction SW which P1A users are waiting for? What about active lodspeaker-toom correction? It is all silent and no appearent interest in the audio communicty, which seems to continue to tweak the system using cables while missing the big point )the room) Thanks marco stanzani |
#2
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What happened to perpetual technologies?
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#3
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What happened to perpetual technologies?
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#4
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What happened to perpetual technologies?
"Dick Pierce" wrote in message
... (andy) wrote in message ... I cannot locate their web site, I am just re-directed to the av123 site which sells the units. No, Perpetual Technologies is alive and, apparently well. AV123 is run by the same guy who runs Perpetual Technologies, which is why you're directed to the AV123 website. That's a strange name for an audio company. Was it supposed to be "perceptual"? Norm Strong |
#5
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What happened to perpetual technologies?
"normanstrong" wrote in message
news:Gbtub.181649$ao4.610833@attbi_s51... That's a strange name for an audio company. Was it supposed to be "perceptual"? Nope, it was Perpetual. Run by a guy Mark Shifter (sp?) who harks back to Audio Alchemy. Snake oil is probably a tad harsh for AA...but there was some debates about their claims and products (DAC-in-a-box for one). Perpetual Technologies/AV123 on the other hand is supposed to have some good stuff. Their Rocket line of speakers enjoys a large internet following and review very well. They also have a high end reference series and some smaller HTiB systems. Designed here in the states and manufactured in China they are supposed to be very high quality for the buck. A "perpetual" comment is the Rockets hold their own both cosmetically and performance wise with speakers several times their price. The crossovers (patent pending) and drivers were designed by none other than Dick Pierce who is without reproach. As his client it appears Mark actually listening to what Dick said and got a good product as a result. |
#6
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What happened to perpetual technologies?
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#7
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What happened to perpetual technologies?
"Mkuller" wrote in message
news:58Pub.246217$HS4.2182771@attbi_s01... Snake oil is more than a "tad" harsh to call AA. Mark owned the company and had a very talented designer in the days when digital was evolving. They produced excellent products at great prices and their DTI jitter buster was a breakthrough product that is still in use today by people using separates. That may be true but I still think they weren't totally honest with some of their products. There should be no need for jitter correction in CD audio as has been pointed out by Dick Pierce and others that CD's can't have jitter. If a jitter buster helps then something is wrong with the CD player. Another example is a quote by Tom Nousaine in this newsgroup a few weeks ago: "I'm guessing that my experience with an Audio Alchemy outboard DAC might be illustrative. Using that device for a level matched test I discovered that the output of the AA was +10 dB compared to the analog output of a Marantz CD-63 player. Inside the case there was a jumper with 0 dB and +10 dB settings. Moving the jumper to the 0 dB position and, guess what, the output was still +4 dB. So to an end-user the device always delivered a higher output level. I'm guessing that this kind of level de-match accounts for practically all, if not exactly all, of the reported cd-player sound differences." |
#8
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What happened to perpetual technologies?
"Rusty Boudreaux" wrote in message
... "Mkuller" wrote in message news:58Pub.246217$HS4.2182771@attbi_s01... Snake oil is more than a "tad" harsh to call AA. Mark owned the company and had a very talented designer in the days when digital was evolving. They produced excellent products at great prices and their DTI jitter buster was a breakthrough product that is still in use today by people using separates. That may be true but I still think they weren't totally honest with some of their products. There should be no need for jitter correction in CD audio as has been pointed out by Dick Pierce and others that CD's can't have jitter. If a jitter buster helps then something is wrong with the CD player. Another example is a quote by Tom Nousaine in this newsgroup a few weeks ago: "I'm guessing that my experience with an Audio Alchemy outboard DAC might be illustrative. Using that device for a level matched test I discovered that the output of the AA was +10 dB compared to the analog output of a Marantz CD-63 player. Inside the case there was a jumper with 0 dB and +10 dB settings. Moving the jumper to the 0 dB position and, guess what, the output was still +4 dB. So to an end-user the device always delivered a higher output level. I'm guessing that this kind of level de-match accounts for practically all, if not exactly all, of the reported cd-player sound differences." Well, if this is your belief as well as Tom's guess, then perhaps this will persuade you otherwise. I use an AA DTI Pro jitter buster. Had it wired into a Proceed PDP using balanced cable and then into Aux two on my preamp next to Aux one, direct feed from the cd player. Identical cables used. The outputs were matched within .5db on all three cd players I used over the decade with this arrangement in my system. Compared to my Phillips 880, the sound was the same but more transparent (it should be since the PDP was 18 bit and the DTI Pro featured noise-shaping specifically designed for 18-bit making it sound like 20-bit, while the 880 was designed at a time when 16 bit multibit DACs were not particularly linear below -80db). Compared to my Marantz 63SE (one-bit pcm), the sound was slightly more neutral (the Marantz a tad "lean") and more natural sounding and about the same in perceived transparency. Compared to my Sony C222ES, the sound was very similar but had a slightly more dynamic, warmer, and more natural sounding bass, and about the same transparency. On a casual level they can all "sound the same". In fine detail, they all have subtle differences. So if the AA DAC had higher output and people did not level adjust, then that may explain some specific results with that DAC. But it is a big leap from there to saying that all DACs were liked better because they had higher output. There were other things at work, especially in the early days when outboard DACs first became popular...different DAC chips, better power supplies, better analog outputs, etc. |
#9
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What happened to perpetual technologies?
OK, some of us knew that. OTOH, answers to the other questions would be more interesting and useful. Kal Thanks Kal did somebody experienced the DeQX devices? Is SigTech still opearting with new prodocts (I still see Copyright 1996-2000 Cambridge Signal Technologies on their URL) Is TacT available with some more affordable products? Is Perp. tech providing the room correction services )this was their claim in Y2K but nothing happened so far (and AFAIK) Overall the BOM cost of digital correction systems is VERY low, so we are going to pay for the IP (which is OK). Still I feel very promising for the medium- to low-end system the technologt fallout in the near future. Still it does not seem to happen so near .. Thanks |
#10
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What happened to perpetual technologies?
"Harry Lavo" wrote in message news:YVqvb.264201$HS4.2355064@attbi_s01...
I use an AA DTI Pro jitter buster. Had it wired into a Proceed PDP using balanced cable and then into Aux two on my preamp next to Aux one, direct feed from the cd player. Identical cables used. The outputs were matched within .5db on all three cd players I used over the decade with this arrangement in my system. Harry, You've got to match to within 0.1 dB, or less than 1 percent difference between voltage levels, in order to rule out effects due to level differences. In your test, level differences can still be a factor in what you heard. I'm not saying they were for sure, but there is uncertainty to whether your experiment tested only jitter differences or jitter differences combined with level differences. --Andre |
#11
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#12
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#13
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What happened to perpetual technologies?
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#14
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What happened to perpetual technologies?
"Nousaine" wrote in message
... "Harry Lavo" wrote: "Rusty Boudreaux" wrote in message ... "Mkuller" wrote in message news:58Pub.246217$HS4.2182771@attbi_s01... Snake oil is more than a "tad" harsh to call AA. Mark owned the company and had a very talented designer in the days when digital was evolving. They produced excellent products at great prices and their DTI jitter buster was a breakthrough product that is still in use today by people using separates. That may be true but I still think they weren't totally honest with some of their products. There should be no need for jitter correction in CD audio as has been pointed out by Dick Pierce and others that CD's can't have jitter. If a jitter buster helps then something is wrong with the CD player. Another example is a quote by Tom Nousaine in this newsgroup a few weeks ago: "I'm guessing that my experience with an Audio Alchemy outboard DAC might be illustrative. Using that device for a level matched test I discovered that the output of the AA was +10 dB compared to the analog output of a Marantz CD-63 player. Inside the case there was a jumper with 0 dB and +10 dB settings. Moving the jumper to the 0 dB position and, guess what, the output was still +4 dB. So to an end-user the device always delivered a higher output level. I'm guessing that this kind of level de-match accounts for practically all, if not exactly all, of the reported cd-player sound differences." Well, if this is your belief as well as Tom's guess, then perhaps this will persuade you otherwise. I use an AA DTI Pro jitter buster. Had it wired into a Proceed PDP using balanced cable and then into Aux two on my preamp next to Aux one, direct feed from the cd player. Identical cables used. The outputs were matched within .5db on all three cd players I used over the decade with this arrangement in my system. Compared to my Phillips 880, the sound was the same but more transparent (it should be since the PDP was 18 bit and the DTI Pro featured noise-shaping specifically designed for 18-bit making it sound like 20-bit, while the 880 was designed at a time when 16 bit multibit DACs were not particularly linear below -80db). Compared to my Marantz 63SE (one-bit pcm), the sound was slightly more neutral (the Marantz a tad "lean") and more natural sounding and about the same in perceived transparency. Compared to my Sony C222ES, the sound was very similar but had a slightly more dynamic, warmer, and more natural sounding bass, and about the same transparency. On a casual level they can all "sound the same". In fine detail, they all have subtle differences. So if the AA DAC had higher output and people did not level adjust, then that may explain some specific results with that DAC. But it is a big leap from there to saying that all DACs were liked better because they had higher output. There were other things at work, especially in the early days when outboard DACs first became popular...different DAC chips, better power supplies, better analog outputs, etc. Ok here's the rest of my story. In a blind experiment the analog output of a Maratntz CD-63 was fed into the analog input of a Marantz CDR610 CD-recorder. The recorder was placed in "record" and the signal at the headphone jack of the CD-R was fed into the system and level matched to the Audio Alchemy with the level control of the headphone jack (whereas the analog output of the player had been subject to a complete extra AD-DA cycle through the CDR610s internal chips.) I found it it illustrative that several experienced enthusiasts were then unable to reliably differentiate between them by sound alone. First, are you sure you weren't just bypassing the ad-da converters out the headphone jack? That's how my Panansonic DAT's work. Otherwise the sound would not be synched. Second, what is this supposed to prove about the sound of DAC's in general and whether or not they would improve upon a given cd player...that was the original point of this post. All this does is suggest that your AA and your Marantz sounded alike (neither the epitome of high-end sound IME). Perhaps your answer to the original poster should be that you doubt the AA DAC would be an improvement...perhaps he should borrow a Wadia, or a Mark Levinson, or an MF, or at least a Benchmark to listen to. No? And based on your dismissal of antidotal discussion of listening tests, your experienced enthusiasts might or might not have heard a difference based on their expectations, and in either case their opinions cannot be trusted. Perhaps, another set of "experienced enthusiasts" might have heard an equally untrustworthy difference. Since you dismiss antidotal evidence as worthless, those "experienced enthusiast's" opinions we should accept as the last word? Given that they were acquaintances of yours, how are we to know that they have been led to have negative expectations of differences by the known opinions, of you their friend. Perhaps another set of "enthusiast" might have reached a different conclusion. But it doesn't count anyway, right? |
#15
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What happened to perpetual technologies?
Hello Andre,
(Andre Yew) wrote in : ... I disgree about the cost of room correction systems. Research seems to indicate that at least 1 second of room correction (or 1 Hz correction resolution) is desirable. At 44.1 kHz, and done with FIR filters, this amounts to about 44.1k*44.1k = 1.9 billion multiply and additions (MACs) per second, and over 8 billion MACs per second for 96 kHz processing, a sample rate at which many receivers and surround prepros are operating at today. That is well beyond affordable, and even achievable. Top-of-the-line Pentiums and Athlons can barely achieve Dhyrstone MIPS at half these numbers, and those numbers are unrealistic and inflated anyway, given that Dhrystone isn't a realistic, or even meaningful benchmark. ... I think you should take a look at this site: http://www.ludd.luth.se/~torger/brutefir.html Realtime FIR filtering with better than 1 Hz resolution at audio sampling rates is available since at least 5-8 years. A top-of-the-line processor should be able to run at least 30 channels at 96 Khz using a program like that. The main problem is that this convolution method is patented by Lake Audio, so it cannot be used outside free programs like the one above, but AFAIK the patent is going to expire within few months. If you use the program above with the following one: http://freshmeat.net/projects/drc/ (incidentally developed by me you can build a good quality room correction system almost for free. You can see some example results achieved with the program above, along with traditional room treatment, at the following URL: http://www.avsforum.com/avs-vb/showthread.php?s= 24df513b860d46cb4eb85577a6528c9a&threadid=283878&p erpage=20&pagenumber=16 And that's just for the implementation of the correction playback side. The measurement side that determines what needs to be corrected has its own set of challenges, which include a simple enough user ... I agree, making a good quality RCS is difficult, making it easy to use is near to impossible (DRC is definitely difficult to use). Bye, -- Denis Sbragion InfoTecna Tel: +39 0362 805396, Fax: +39 0362 805404 URL: http://www.infotecna.it |
#17
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What happened to perpetual technologies?
"Andre Yew" wrote in message
... I disgree about the cost of room correction systems. Research seems to indicate that at least 1 second of room correction (or 1 Hz correction resolution) is desirable. At 44.1 kHz, and done with FIR filters, this amounts to about 44.1k*44.1k = 1.9 billion multiply and additions (MACs) per second, and over 8 billion MACs per second for 96 kHz processing, a sample rate at which many receivers and surround prepros are operating at today. That is well beyond affordable, and even achievable. Top-of-the-line Pentiums and Athlons can barely achieve Dhyrstone MIPS at half these numbers, and those numbers are unrealistic and inflated anyway, given that Dhrystone isn't a realistic, or even meaningful benchmark. I agree that it's not as cheap as the previous poster was claiming. However, your argument using general processors is not valid. In the realm of DSPs and ASICs this level of processing power is certainly achievable at modest cost. |
#18
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What happened to perpetual technologies?
"Rusty Boudreaux" wrote in message ...
However, your argument using general processors is not valid. In the realm of DSPs and ASICs this level of processing power is certainly achievable at modest cost. I disagree. If we're using brute-force FIR techniques, name me an affordable computation system that can achieve 16 billion 32-bit integer MACs per second. That's for a two-channel, 96 kHz system for 24-bit audio, and assumes 32 bits are enough accumulation precision for such a long filter. The newly announced TI C6412 claims to do 2 billion 16-bit MACs per second, so if we generously assume things scale linearly, you'd need at least 16 of them to achieve the required computational throughput. They're about $40 in quantity, so parts cost for the CPU alone is $40*16=$640, which translates into approximately 6*$640=$3840 MSRP difference, not counting support circuitry, the case, power supply, R&D time, etc. to support the extra CPUs. Now if we want to do it for a typical home theatre system with 6 or 8 channels, multiply that by 3 or 4. I don't think that's a modest cost. One solution, as Denis points out, is smarter algorithms that require less computation load. Optimized FIRs, or decimation are two ways to getting there. The Lake DSP stuff was what I was thinking about actually for reducing latency. --Andre |
#19
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What happened to perpetual technologies?
Andre Yew wrote:
|| "Rusty Boudreaux" wrote in message || ... ||| However, your argument using general processors is not ||| valid. In the realm of DSPs and ASICs this level of processing ||| power is certainly achievable at modest cost. || || I disagree. If we're using brute-force FIR techniques, name me an || affordable computation system that can achieve 16 billion 32-bit || integer MACs per second. That's for a two-channel, 96 kHz system for || 24-bit audio, and assumes 32 bits are enough accumulation precision || for such a long filter. The newly announced TI C6412 claims to do 2 || billion 16-bit MACs per second, so if we generously assume things || scale linearly, you'd need at least 16 of them to achieve the || required || computational throughput. They're about $40 in quantity, so parts || cost for the CPU alone is $40*16=$640, which translates into || approximately 6*$640=$3840 MSRP difference, not counting support || circuitry, the case, power supply, R&D time, etc. to support the || extra || CPUs. Now if we want to do it for a typical home theatre system with || 6 or 8 channels, multiply that by 3 or 4. I don't think that's a || modest cost. || || One solution, as Denis points out, is smarter algorithms that require || less computation load. Optimized FIRs, or decimation are two ways to || getting there. The Lake DSP stuff was what I was thinking about || actually for reducing latency. Andre, maybe a FIR-approach is not the right way here, at least not a very long FIR. Basically, as the correction signal is radiated through the speakers as well, it will be reflected by the room the same way as the original, and so will be the signal to correct the reflections of the correction signal and so on. This seems to be more the work of an IIR-filter, which is computionally very very cheap. So the FIR is only needed for a limited period, maybe 100ms or even less which is a lot easier to compute. I have been using Angelo Farina's Aurora plugins for CEP(32bit float) to get the FIR-part and then Matlab to model the IIR-part, but I couldn't get a satisfactory result. Everything sounded a bit dull and distant, but I might have made mistakes. Maybe the whole correction is conceptionally a wrong approach. I would rather try something else, by positioning additional speakers on the first reflection points on the wall/ceiling to cancel the first reflections to get a longer ITD(initial time delay gap). In the moment I'm doing this and the result is much more promising. I have also placed active bass "suckers" in the room corners to cancel the standing waves creation. Unfortunately I do not have enough hardware to create all the signals needed in real time. This is also a bit much of a task for a single person. -- ciao Ban Bordighera, Italy electronic hardware designer http://www.bansuri.my-page.ms/ |
#20
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What happened to perpetual technologies?
"Nousaine" wrote in message
... "Harry Lavo" wrote: snip, too much..read earlier posts in thread for background First, are you sure you weren't just bypassing the ad-da converters out the headphone jack? That's how my Panansonic DAT's work. Otherwise the sound would not be synched. Yes I'm sure because I spoke with the Marantz people prior and without the CDR being placed in record mode no sound gets to the headphone output. That's how my dat works too...but it doesn't go through the convertors. It simple routes the signal to the headphone outlets as well as to the convertors....there is no way to "read" the data off a DAT or a CD-R in "real time". That is just so you can hear what the DAT or CD-R is being fed. Second, what is this supposed to prove about the sound of DAC's in general and whether or not they would improve upon a given cd player...that was the original point of this post. All this does is suggest that your AA and your Marantz sounded alike (neither the epitome of high-end sound IME). I'm glad you regognize that Audio Alchemy was high-end hyperbole. Well, then I guess you think the Marantz SE was also high-end hyperbole. Perhaps your answer to the original poster should be that you doubt the AA DAC would be an improvement...perhaps he should borrow a Wadia, or a Mark Levinson, or an MF, or at least a Benchmark to listen to. No? Why bother; there's nothing they could possibly do but degrade an already transparent signal. How do you square this with your assertion above....since you seem to think the original signal was a transparent as possible. Then either both the SE and the AA were transparant, or they were both high-end hyperbole. which is it? And based on your dismissal of antidotal discussion of listening tests, your experienced enthusiasts might or might not have heard a difference based on their expectations, and in either case their opinions cannot be trusted. But this was a double blind test and the numbers show they were unable to reliably identify them. I didn't rely on their 'opinions.' It would have helped if you had mentioned this in your post...instead of alluding to a nebulous listening session. Perhaps, another set of "experienced enthusiasts" might have heard an equally untrustworthy difference. IMO another set of listeners would have heard (or not) exactly the same thing. Might or might not, in part depending on what their going-in expectations were. Since you dismiss antidotal evidence as worthless, those "experienced enthusiast's" opinions we should accept as the last word? Given that they were acquaintances of yours, how are we to know that they have been led to have negative expectations of differences by the known opinions, of you their friend. Perhaps another set of "enthusiast" might have reached a different conclusion. But it doesn't count anyway, right? Well at least half of them seemed heavily biased in the other direction, at least in comments. But, as you well know, I'm not one to accept opinions when true sonics can be verified. You mean biased to hear differences? Or biased in that they thought they heard differences? Don't forget that when the experiment is double blind there is no way that I can prevent listeners from hearing true differences. Ah, I see. The test set them straight, right? Assuming the test protocol doesn't interfere....which as you well know you have not proven to the satisfaction of those of us who have asked for controls in order to provide such proof. |
#21
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What happened to perpetual technologies?
Andre Yew wrote:
"Rusty Boudreaux" wrote in message ... However, your argument using general processors is not valid. In the realm of DSPs and ASICs this level of processing power is certainly achievable at modest cost. Not true. I disagree. If we're using brute-force FIR techniques, name me an affordable computation system that can achieve 16 billion 32-bit integer MACs per second. This is the problem with these outrageous sample rates. Why do we need to process at a rate in which over 50 percent of the available bandwidth is unused? A single TI TMS320C6416 can achieve 2.88 billion 16-bit MACs/ second. http://focus.ti.com/lit/ds/symlink/tms320c6416.pdf It has a clock speed of 720 MHz and does 4 16x16 multiplies per cycle. If you use one per channel you can achieve a 1 second FIR at 44.1 kHz. However, even this won't do what you actually asked for, which is 32-bit MACs, not 16-bit. In any case, at over $300 a pop, they aren't cheap. I agree, Rusty is wrong - this level of processing is currently not available at modest cost. One solution, as Denis points out, is smarter algorithms that require less computation load. Optimized FIRs, Optimized FIRs? Never heard of that phrase. One way of efficiently computing a convolution is "frequency domain filtering." Essentially you use the convolution == multiplication property of Fourier transforms, converting your time data into the frequency domain, performing the multiplication, and then inverse transforming. That algorithm is o(N*log(N)), instead of o(N^2) as the brute force method is. or decimation are two ways to Yes, decimation is reasonable. 48 kHz was a fine sample rate - we didn't need to throw it away and go to 96 kHz. -- % Randy Yates % "...the answer lies within your soul %% Fuquay-Varina, NC % 'cause no one knows which side %%% 919-577-9882 % the coin will fall." %%%% % 'Big Wheels', *Out of the Blue*, ELO http://home.earthlink.net/~yatescr |
#22
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What happened to perpetual technologies?
Randy Yates wrote in message ...
This is the problem with these outrageous sample rates. Why do we need to process at a rate in which over 50 percent of the available bandwidth is unused? I think there are valid reasons to process at high sample rates: 1. There's some software out there now (DVD-A) that delivers content at high sampling rates, and decimating it to lower rates seems to defeat the purpose of having such high-sample-rate content in the first place (assuming such high sample rate content is audibly different due in part to their high sampling rate). For the really optimistic, some day there may even be a standardized hi-res digital interconnection which may pass along hi-res data unaltered. 2. There're DSP operations which require high sampling rates, such as non-linear algorithms like compression, because non-linear algorithms generate harmonics which may alias. One could argue that such algorithms should upsample, do their non-linear processing, and then downsample, but for the sake of fidelity through simplicity, it may be easier to implement a good-sounding system if the digital processing were kept to a minimum. Optimized FIRs? Never heard of that phrase. One way of efficiently computing a convolution is "frequency domain filtering." Sorry. That's what I meant --- faster ways to do FIRs than just brute force time-domain computations. Yes, decimation is reasonable. 48 kHz was a fine sample rate - we didn't need to throw it away and go to 96 kHz. The stuff that has been implemented by Michael Gerzon and Peter Craven in the early 90s for the B&W correction system is a multi-rate system, where the bass, which needs the most precision and longest filters, be decimated to 1 kHz, the mid-range from 500 Hz to 3 kHz would be decimated to 6 kHz, and everything above that is running at 48 kHz. The difficulty then moves to making the filters for the downsampling and upsampling transparent and efficient. Here's a link to the Gerzon article: http://www.audiosignal.co.uk/Digital...alisation.html --Andre |
#23
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What happened to perpetual technologies?
"Randy Yates" wrote in message
... Andre Yew wrote: "Rusty Boudreaux" wrote in message ... However, your argument using general processors is not valid. In the realm of DSPs and ASICs this level of processing power is certainly achievable at modest cost. Not true. snip In any case, at over $300 a pop, they aren't cheap. I agree, Rusty is wrong - this level of processing is currently not available at modest cost. Yep, mea culpa. I wasn't paying attention I read million, not billion. Our ASICs do upwards of 20 million 32-bit MACs at a fraction of the cost of a comparable DSP solution. However, these ASICs are specialized for the communications equipment market and probably not directly comparable to a higher feature set DSP. |
#24
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What happened to perpetual technologies?
Andre Yew wrote:
Randy Yates wrote in message ... This is the problem with these outrageous sample rates. Why do we need to process at a rate in which over 50 percent of the available bandwidth is unused? I think there are valid reasons to process at high sample rates: 1. There's some software out there now (DVD-A) that delivers content at high sampling rates, and decimating it to lower rates seems to defeat the purpose of having such high-sample-rate content in the first place (assuming such high sample rate content is audibly different due in part to their high sampling rate). Yeah, that's a big assumption. Why not sample at 4 GHz, ASSUMING there is content that would make an audible difference? For the really optimistic, some day there may even be a standardized hi-res digital interconnection which may pass along hi-res data unaltered. So it can be thrown away by the time it reaches the listener's brain? 2. There're DSP operations which require high sampling rates, such as non-linear algorithms like compression, because non-linear algorithms generate harmonics which may alias. You mean like an algorithm I designed into our (Sony Ericsson's) mobiles? One could argue that such algorithms should upsample, do their non-linear processing, and then downsample, Not if you want to waste processing power needlessly performing operations that will have no audible effect. but for the sake of fidelity through simplicity, Ahh yes - fidelity through simplicity... it may be easier to implement a good-sounding system if the digital processing were kept to a minimum. Not if you're hamstrung from doing significant operations down the line because you've got your sample rate up too high! Common Sense 101. Let me tell you why we've got a 96 kHz sample rate system (and - aggh! - 192 kHz too): It makes some people some $$$. At the outside, these high sample rate systems may have been slightly beneficial because the anti-aliasing or reconstruction filtering for some data conversion systems were improved. However, there's no need to require this through the entire chain! Just do it at the conversion (e.g., oversample the A/D, then decimate digitally). -- % Randy Yates % "...the answer lies within your soul %% Fuquay-Varina, NC % 'cause no one knows which side %%% 919-577-9882 % the coin will fall." %%%% % 'Big Wheels', *Out of the Blue*, ELO http://home.earthlink.net/~yatescr |
#25
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What happened to perpetual technologies?
Andre Yew wrote:
2. There're DSP operations which require high sampling rates, such as non-linear algorithms like compression, because non-linear algorithms generate harmonics which may alias. Fine, and the way that's done, and has been done for quite some time in the professional world, is to oversample, process, filter back to the original base bandwidth, then decimate. Poof! aliased harmonics are gone because they were never there to begin with. |
#26
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What happened to perpetual technologies?
Randy Yates wrote in message ...
[re. non-linear DSP algorithms] One could argue that such algorithms should upsample, do their non-linear processing, and then downsample, Not if you want to waste processing power needlessly performing operations that will have no audible effect. I don't understand your statement. Non-linear processing will produce harmonics of the same order as the non-linearity's highest power-series expansion. Therefore some upsampling is required in order to perform such DSP without aliasing. Downsampling is then necessary to return to the playback rate. Level compression is an example of a non-linear process. Whatever we may feel about high sampling rates, they are here, and devices will need to deal with them. If you want to decimate them all to 48 kHz, that's fine. I'd rather avoid one more filter in the processing chain --- it's just one more place to screw up. Another issue that hasn't been addressed yet are the possible ameliorating effects of higher sampling rates on processing errors and bugs. --Andre |
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What happened to perpetual technologies?
Randy Yates wrote in message news:bTXxb.250002$9E1.1349810@attbi_s52...
Are they? If you walk into the living rooms of most people at this point in time, you will find CDs - not DVD-A or SACD. And I think the average consumer, who finds compressed MP3s sound just fine and makes around $30,000/year, isn't going to be willing to shuck out a few thousand dollars for new equipment and replacement media to gain an extra 20 kHz of bandwidth that he never missed in the first place. Yes this true, but many receivers and surround pre-pros are running internally at higher sampling rates already. Some Pioneer receivers run at 88.2 kHz, for example, and current Lexicons run at 96 kHz. They often use A/Ds at high sampling rates, so analog sources are digitized at high rates and are processed that way. CDs are dealt, usually, in their native rates. I would also suggest that most high-end audio design isn't aimed at your average consumer. I'm not saying it's necessarily economically viable, but I think DVD-A and SACD compatibility looms large on their design specs. Besides, didn't you know Meridian upsamples MP3s? :-) Frankly, I hope these (SACD and DVD-A) formats fail. CDs are more than adequate for audio reproduction in any venue barring perhaps a laboratory, and creating a profuse array of formats does nothing but confuse consumers and dissipate resources. I hope not. DVD-A and SACD are very useful in providing high-quality, non-lossy encoded multichannel music. Processing them digitally is another matter altogether unfortunately. DSP engineers know how to properly design and implement a low pass filter. Filtering is one of the most basic tasks for a DSP engineer, and lowpass is the most basic type of filter. Yes, but there are so many tradeoffs to make --- will they make the right choices, perceptually speaking? One of our fellow rahe members, Bruno Putzeys, recently described on the pro-audio list his experiences with a TI SRC4192 chip, where in the inband ripple produced possibly large amounts of pre- and post-echo, but TI thought their customers couldn't deal with the large group delay if they lowered the in-band ripple. I would copy his post, replete in technical detail, here, but the pro-audio list restricts such usage. Perhaps Bruno would repost his report here --- the original article appeared on November 18, 2003 if you have access to the archive. This was in a discussion about using ASRCs as jitter-suppression devices, and their possible consequences if cascaded with other ASRCs or other processing. It is my experience that bugs are best dealt with by repairing rather than covering up. Yes, but only if they're under your control. Shoddy mastering and engineering practices abound. Higher sampling rates, and deeper bit depths can deal with some common audio engineering problems. --Andre |
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What happened to perpetual technologies?
Andre Yew wrote:
Randy Yates wrote in message news:bTXxb.250002$9E1.1349810@attbi_s52... Are they? If you walk into the living rooms of most people at this point in time, you will find CDs - not DVD-A or SACD. And I think the average consumer, who finds compressed MP3s sound just fine and makes around $30,000/year, isn't going to be willing to shuck out a few thousand dollars for new equipment and replacement media to gain an extra 20 kHz of bandwidth that he never missed in the first place. Yes this true, but many receivers and surround pre-pros are running internally at higher sampling rates already. Forgive me for not taking your word for that. If you can provide some verifiable evidence, I would like to see it. Some Pioneer receivers run at 88.2 kHz, for example, and current Lexicons run at 96 kHz. They often use A/Ds at high sampling rates, so analog sources are digitized at high rates and are processed that way. Sampling at a high rate in no way obligates the rest of the data path to operate at that rate. Again, you're making some assertions I don't believe are true. CDs are dealt, usually, in their native rates. This would make no sense. If the remainder of the data path you speak of above is operating at 88.2, then it would make sense to upsample the CDs to 88.2. I suspect your data is wrong - the internal rate is 44.1 while the converter only runs at a high rate to mitigate antialias filter requirements. I would also suggest that most high-end audio design isn't aimed at your average consumer. Your statement was that high sample rates are here. I did, and still do, challenge that remark, if by "here" you mean in widespread market use. Sure, a small percentage has DVD-A or SACD. (I myself have purchased a DVD-A - and was abysmally disappointed.) There will always be a small part of the market buying the most expensive products available. That wasn't my point, and I don't think it was yours either. I'm not saying it's necessarily economically viable, but I think DVD-A and SACD compatibility looms large on their design specs. Whose design specs? If you mean the average digital audio equipment maker, then I'm not sure you're correct. But if they do, the market will decide with their pocketbooks, and I'm betting which choice they're going to make. Frankly, I hope these (SACD and DVD-A) formats fail. CDs are more than adequate for audio reproduction in any venue barring perhaps a laboratory, and creating a profuse array of formats does nothing but confuse consumers and dissipate resources. I hope not. DVD-A and SACD are very useful in providing high-quality, non-lossy encoded multichannel music. They may provide multi-channel music, but we've already got high quality stereo music. It's called "CD." These formats provide *no* (zero) practical advantage in music quality over a CD. DSP engineers know how to properly design and implement a low pass filter. Filtering is one of the most basic tasks for a DSP engineer, and lowpass is the most basic type of filter. Yes, but there are so many tradeoffs to make --- will they make the right choices, perceptually speaking? Yes, they will. It is easy to design and implement a half-band filter using polyphase filtering techniques with fraction-saving or even noise-shaping that will perform extremely well. One of our fellow rahe members, Bruno Putzeys, recently described on the pro-audio list his experiences with a TI SRC4192 chip, Irrelevent. Generic asynchronous sample rate conversion is a far, far more complex task than a simple half-band lowpass filter interpolation (resampling) filter. You're comparing apples to oranges. It is my experience that bugs are best dealt with by repairing rather than covering up. Yes, but only if they're under your control. Shoddy mastering and engineering practices abound. Higher sampling rates, and deeper bit depths can deal with some common audio engineering problems. A mastering engineer can easily screw anything up, even DVD-A and SACD, if they're not careful or don't know what they're doing. You'll never overcome ignorance with more technology - only with education. -- % Randy Yates % "...the answer lies within your soul %% Fuquay-Varina, NC % 'cause no one knows which side %%% 919-577-9882 % the coin will fall." %%%% % 'Big Wheels', *Out of the Blue*, ELO http://home.earthlink.net/~yatescr |
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What happened to perpetual technologies?
Randy Yates wrote in message news:QmAyb.377482$Fm2.393962@attbi_s04...
Some Pioneer receivers run at 88.2 kHz, for example, and current Lexicons run at 96 kHz. They often use A/Ds at high sampling rates, so analog sources are digitized at high rates and are processed that way. Sampling at a high rate in no way obligates the rest of the data path to operate at that rate. Again, you're making some assertions I don't believe are true. Believe what you want. Lexicon prepros operate up to 96 kHz internally, as confirmed in conversations with Lexicon engineers, including David Griesinger, and in this Q and A done for the release of the MC-12: http://www.smr-home-theatre.org/Lexicon/mc12/qa1.html "SMR: Lexicon owners are familiar with the outstanding performance afforded by Logic 7, clearly the most popular of all the available processing modes. I believe it has been updated and refined yet further, is that correct? "Andy Clark: Yes. The MC-12 uses four 32-bit Analog Devices SHARC® DSP engines, which provide enormous processing power. We have taken full advantage of this and have re-written the Logic 7 algorithms for 96kHz internal processing." CDs are dealt, usually, in their native rates. This would make no sense. If the remainder of the data path you speak of above is operating at 88.2, then it would make sense to upsample the CDs to 88.2. The internal architecture of at least the Lexicon, if not other prepros, operate at multiple sampling rates, and have done so since the DC-1 introduced in the mid 90s. This is a well-known fact, since some of its processing modes cannot process above certain sampling rates. For example, in the DC-1/DC-2/MC-1 architecture, Panorama and the ambience synthesis modes (Church, Concert Hall, etc.) could not process 48 kHz sampling rates, and could deal only with 44.1 kHz. In the MC-12/MC-8 architecture, DTS Neo:6 cannot go above 48 kHz. If everything were upsampled to 96 kHz, then clearly Neo:6 could not work at all. Further, since the MC-12/8 can accept 96 kHz digital inputs, it's trivial to check that there is no decimation and upsampling happening around the Neo:6 code. Your statement was that high sample rates are here. I did, and still do, challenge that remark, if by "here" you mean in widespread market use. Sure, a small percentage has DVD-A or SACD. (I myself have purchased a DVD-A - and was abysmally disappointed.) There will always be a small part of the market buying the most expensive products available. That wasn't my point, and I don't think it was yours either. We will just agree to disagree on this. That the internal processing modes of receivers is already running at 88.1 or higher is a good enough condition for "here" for me, and that these receivers also have digital links for DVD-A and SACD with supporting players available makes it more solid for me. Frankly, I hope these (SACD and DVD-A) formats fail. CDs are more than adequate for audio reproduction in any venue barring perhaps a laboratory, and creating a profuse array of formats does nothing but confuse consumers and dissipate resources. They may provide multi-channel music, but we've already got high quality stereo music. It's called "CD." These formats provide *no* (zero) practical advantage in music quality over a CD. Technically speaking, it is incontrovertible that stereo is far inferior to multichannel. I don't even know how that can be an issue for discussion, but we can get into it if you like. You had also originally said: "CDs are more than adequate for audio reproduction in any venue barring perhaps a laboratory". Tell me how a two-channel speaker array creates a lateral moving soundwave. Tell me how a two-channel speaker array gets rid of comb-filtering effects of phantom imaging. Two channel has been barely adequate for audio reproduction, and this has been known since the early 30s. Yes, they will. It is easy to design and implement a half-band filter using polyphase filtering techniques with fraction-saving or even noise-shaping that will perform extremely well. Consider silicon resources vs. latency vs. precision. The choice is not so easy to make, especially since your typical audio company isn't going to spin their own ASIC, much less have the technical know-how to even know such choices exist. Irrelevent. Generic asynchronous sample rate conversion is a far, far more complex task than a simple half-band lowpass filter interpolation (resampling) filter. You're comparing apples to oranges. If you had lots of resources to throw at the problem, I agree, however, most companies don't. This is proven by the vast majority of companies who use upsampling in their products using the Cirrus Crystal ASRC part, which makes this relevant. Looking at the specs published in data sheets for DACs, the fact that they print only THD+N proves to me that the designers perhaps don't know or care that much about the human hearing system. And that in turn leads me to view choices and tradeoffs they make with suspicion. A mastering engineer can easily screw anything up, even DVD-A and SACD, if they're not careful or don't know what they're doing. You'll never overcome ignorance with more technology - only with education. I agree, but that doesn't change the fact that hi-res audio does ameliorate many engineering sins, and if that's a way we can get better sounding music, then so be it. --Andre |
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