Over on the proaudio list (where the degreed EEs and similar wonks hang out) there was a raging debate over Sampling Rate Conversion, and whether upsampling has made the numbers moot.
One highly respected lister (who also designs and sells DACs) said, "As a rule, less calculations is better for quality. In this case, the 96 to 44.1 takes more calculations." The inference (based on other posts in the thread) is that it is better to record at 44.1 or 88.2 (integer multiple) if audio CD is the delivery mediium. It is for me, with the exception of orchestral projects for a label that also releases them in SACD and DVD-A. I should also add that they have taken material that was originally 16/44.1 and released it in SACD and DVD-A formats.
The purpose of this thread is to ask what SR you record, how you get to 16/44.1, and if you did listening tests to determine your methodology?
Rich
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r8 brain Pro here... http://src.infinitewave.ca/
r8 brain Pro here...
Rich, I use 96K for all critical projects now, because some of t
Rich, I use 96K for all critical projects now, because some of them have unknown release media, ie some will be used for DVD-V release, which is 48 and of course everyone wants a CD.
In my reading of SRC algorithms, 88 is never drop sampled to 44, it goes through the standard upsampling to 88 X 44 or 96 X 44 then re-interpolated. Correct me if you know of a specific example where 88 is drop sampled.
Gunnar-- if you read my post carefully you would see that the pe
Gunnar-- if you read my post carefully you would see that the person suggesting the integer multiple "myth" IS someone who designs, builds, and sells ADCs. I will not mention his name as he was not asked about about quoting him, but it is a name any pro engineer would recognize and respect.
I will email him and ask him to take part in this thread.
David-- I am not sure about all SRC algorithms, but as I almost never record anything that is destined for a video format, I work at 44.1 and 88.2 integer multiples. It certainly will sound as good as other SRs and possibly be more efficient.
Looks like the "compare and decide" poll is 1 did the test and the others have not.
Hmmmm......
Rich
Not even close.. R8brain is a much cleaner sounding SRC- basical
Not even close.. R8brain is a much cleaner sounding SRC- basically sounding much closer to the original rather than the coloration introduced by Sequoia's SRC...
Now, SRC is one of the improvements in Version 9 of Sequoia. To be honest, though, I haven't really tested the new version.
--Ben
I read Glenn Meadows mastering board but I can't post to it (luc
I read Glenn Meadows mastering board but I can't post to it (lucky for all), as they still can't do this for GMail accounts. But I went through most of the long threads about testing the SRC in all the big DAW's as well as some of the offline SRC's like the R8BrainPro one.
Wavelab 6's new Crystal resampler tested one of the best, I think they awarded it "most improved" in the end, over the very ordinary performance of the internal WL5 version. Before WL6 I didn't do much SRC'ing but used the free version of R8Brain and found it excellent.
ghellquist wrote: Personally I run at 44.1 through Lavry Blue. T
ghellquist wrote: Personally I run at 44.1 through Lavry Blue. The results are plenty good enough for me.
As for SRC - there seems to be a widespread myth that going from 88.2 to 44.1 is easier than going from 96 to 44.1. This myth is generally spread by people that has not designed and tested SRC algorithms. It certainly must be much easier is the lay persons guess.
Wikipedia has a small article telling you a bit about how an SRC works. One important thing to notice is that if you write a good SRC it can translate between any rate of sample rates. There are even algorithms available that allows the ratios to be variable. This of course is a bit beyond the normal high-school level math.
http://en.wikipedia.org/wiki/Sample_rate_conversionI should say that I have done some sample based algorithms, but no SRC-s. Frankly the part about sinc functions is a bit above my level of math.
What I am trying to boil it all down to is that the only way to select sample rate is by measuring and listening to the actual equipment and algorithms you are planning to use. If you start apply theories about things that you do not know intimitely it is common for the conclusions to be wrong.
Gunnar
Awesome, thanks for the link, been wondering what to set my logic projects at as I'm a recording noob, but a Math Major all the same. Kind of saddened to see logic pro not looking so hot. Once I get something else to compare side by side, I'll see if I notice. Good to see science dispelling myth. Any of these processes that depend on precision are going to benefit from more accurate processing algorithms as well as better floating point precision. Damn these sloppy binary floating point limitations, we need quantum computing. :)
Also, here is a decent pdf link from Bias that has some graphed tests with information that seems relevant at a glance.
http://
Yeah, David's right - there is no "easy way out" with SRC. Conv
Yeah, David's right - there is no "easy way out" with SRC. Converting from 88.2 to 44.1 is just as intensive as going from 96 kHz to 44.1. You can't simply drop the doubled samples and get to 44.1.
Granted, I don't write these algorithms, but I do have a team of scientists that do it for me and they all agree, there's no easy way out. (We have to alter SR for use of forensic purposes - capturing voice samples and transmitting them from hostile, network-constrained environments back to CONUS for comparison. The original file MUST be high quality, but for quick comparison, limited SR is perfectly acceptable. Bear in mind, compressed files are not admissable for what we do - it's considered an alteration of the file and therefore cannot be used. Therefore, we use a limited bandwidth version - seeing as how you only need 2x the bandwidth, not the maximum frequency - this little fact saves us a lot of bandwidth.)
For my SRC, I find reconversion to be the best. I almost always feed out of the DA at the recorded rate into the analog realm and then back into the PC in 44.1. This is usually accompanied by some kind of processing (basic EQ or something - Crane Song - sounds OH so smooth even with no cuts or boosts.) Besides, the 44.1 on the Aurora actually sounds pretty darned good!
J.
I think that Pulse Code Modulation sucks as a recording method!
I think that Pulse Code Modulation sucks as a recording method! It is however hugely convenient and inexpensively executed. The truly good digital recording system is the Sony/Philips "DSD" (direct stream digital) system and that IS what the "SACD" CD is! Everything else is just PCM, which sounds like PCM, which sounds like PCM and all of the conversions require bad math. So either record at 44.1kHz PCM consistently and convert for video or start with something good like DSD. Mathematics only impress mathematicians. The rest of us want to hear something good.
3 + 3=33??
Ms. Remy Ann David
ptr wrote: Is there even an editor that works with DSD/DXD nativ
ptr wrote: Is there even an editor that works with DSD/DXD native...? All the ones I've heard of pops it in to PCM for editing... Much use for the beeing better then...?
/ptr
Pyramix. (DXD - which, BTW, is PCM - 32 bit/384 kHz)
Sonarerec wrote:
Users of DSD have reported fried power amps from the ultrasonic noise that is "out-of-band" for our ears, but not so for the amps that are valiantly atttempting to reproduce it.
I don't buy that. First, the noise is no greater in amplitude than the program material, just higher in frequency.
Secondly, I've been listening to SACD (DSD) for years - almost immediately after it was introduced and my amps (ranging from Onkyo Receivers, HK power amps, Technics receivers, Rotel, B&K, Hafler, etc.) haven't ever issued one breath of a complaint.
Besides, don't you think there would be some kind of class-action lawsuit against sony and philips if they put out a product which was knowingly causing damage to amplfiers and loudspeakers.
I have to ask you now - have you ever actually listened to an SACD on something other than the crappy display at Circuit City? Most people who bash DSD usually haven't actually heard it in a real system.
I'm with Remy 100% on this one.
The HF noise is so high in nature that most components don't even see it and if it does happen to make it down the wire to the speaker, the actual frequency of the noise is naturally filtered out by the capacitance of the air between the tweeter's magnet and voice coil (a natural crossover/rolloff if you will).
J.
I am neither bashing nor defending DSD, as it is a moot point fo
I am neither bashing nor defending DSD, as it is a moot point for anyone whose delivery format is 44.1/16. Clients are sometimes curious about quad until they hear the cost, but DSD/DXD/SACD etc is meaningless to most, and except for the majors is dead in the water.
Much of what is on the shelves started life as PCM-- that is certainly the case with my Naxos stuff that becomes SACD, but at least it began as 88.2 rather than 44.1, which is the case with more material than the consumer realizes.
As to HF noise, you can tell Tony Faulkner it isn't there, as he fried a Levinson with it. He further stated:
"Sonoma does not convert DSD to PCM for any main audio path activities. Nor does SADiE. Both Sonoma and SADiE create PCM files for the purpose of creating waveform profiles for the PC's display GUI, both Sonoma and SADiE create PCM files for the creation of the CD Red Book layer. That is where it stops. I am not completely up to date with Pyramix DSD, but certainly to begin with it could do no DSD processing (including edit crossfades) without conversion to/from 352.8k/32bit for the duration of the processing or crossfade. A retrograde step in my judgment, for a format targeting itself at being squeaky clean. Early unreleased prototypes of the first SADiE 2ch
DSD workstation used pcm sidechains in the DSD processing for eq, but this is no longer the case in production models. Sonoma and SADiE use proprietary DSD chips for processing, and they work well. The only problem always in the background with DSD is the amount of high frequency error noise, and whatever you do it can get in the way - DSD workstations include assignable and configurable low pass filters to stop the noise getting out of hand.
He has actually worked with DSD for release. Has anyone here who can share their experiences?
Rich
Sonarerec wrote: As to HF noise, you can tell Tony Faulkner it i
Sonarerec wrote: As to HF noise, you can tell Tony Faulkner it isn't there, as he fried a Levinson with it. He further stated:
It seems that it would be impossible to diagnose this. How could one discern that HF caused an amp to blow? It would be far more likely (and I'm sure the engineers at Levinson - no longer Mark himself - would agree) that it is far more likely that improper load or spike/surge is the more likely cause.
Sonarerec wrote:
"Sonoma does not convert DSD to PCM for any main audio path activities. Nor does SADiE. Both Sonoma and SADiE create PCM files for the purpose of creating waveform profiles for the PC's display GUI, both Sonoma and SADiE create PCM files for the creation of the CD Red Book layer. That is where it stops. I am not completely up to date with Pyramix DSD, but certainly to begin with it could do no DSD processing (including edit crossfades) without conversion to/from 352.8k/32bit for the duration of the processing or crossfade. A retrograde step in my judgment, for a format targeting itself at being squeaky clean. Early unreleased prototypes of the first SADiE 2ch
DSD workstation used pcm sidechains in the DSD processing for eq, but this is no longer the case in production models. Sonoma and SADiE use proprietary DSD chips for processing, and they work well. The only problem always in the background with DSD is the amount of high frequency error noise, and whatever you do it can get in the way - DSD workstations include assignable and configurable low pass filters to stop the noise getting out of hand.
Unless something has changed within the past 3 months on both of these systems, that's not true. Both (based on the requirements of 1 bit science) MUST convert to PCM to perform crossfades. Additionally, very few plug-ins are capable of processing on DSD.
This is why all of the editing that Telarc does is performed within the analog domain (converted from DSD to analog, processed, then reconverted back to DSD.)
Sonarerec wrote:
He has actually worked with DSD for release. Has anyone here who can share their experiences?
Sure. But I record in PCM first and then upconvert later. I've done one project recently with a local university whose CD (and SACD) will be released spring of 07 and I'm working with a choral ensemble right now for an SACD. I record in 176.4 and then send the individual tracks to the duplication house (in both cases, it is/will be Oasis.)
J.
Cucco wrote: [quote=Sonarerec]As to HF noise, you can tell Tony
Cucco wrote: [quote=Sonarerec]As to HF noise, you can tell Tony Faulkner it isn't there, as he fried a Levinson with it. He further stated:
It seems that it would be impossible to diagnose this. How could one discern that HF caused an amp to blow?
Hardly impossible. The output of some early dsd dac's did output easily measurable vhf signals. In some units the output looked practically like an i-m distortion test. If the following amp has extremely extended freq. resp, the electronics very well may try (in futility) to reproduce it, possibly generating a lot of heat. This has probably been solved in contemporary units.
It has been over a year since I have done any tests, but my experience was that the best pcm conversion was preferred over dsd. In our tests the dsd was further sonically from the analog source than the pcm. I always had the feeling that I was being micro-waved when listening to dsd. But the technology is very young. Its possible that many improvements have been implemented since that time.
Zilla wrote: [quote=Cucco][quote=Sonarerec]As to HF noise, you c
Zilla wrote: [quote=Cucco][quote=Sonarerec]As to HF noise, you can tell Tony Faulkner it isn't there, as he fried a Levinson with it. He further stated:
It seems that it would be impossible to diagnose this. How could one discern that HF caused an amp to blow?
Hardly impossible. The output of some early dsd dac's did output easily measurable vhf signals. In some units the output looked practically like an i-m distortion test. If the following amp has extremely extended freq. resp, the electronics very well may try (in futility) to reproduce it, possibly generating a lot of heat. This has probably been solved in contemporary units.
It has been over a year since I have done any tests, but my experience was that the best pcm conversion was preferred over dsd. In our tests the dsd was further sonically from the analog source than the pcm. I always had the feeling that I was being micro-waved when listening to dsd. But the technology is very young. Its possible that many improvements have been implemented since that time.
I didn't say it was impossible for it to occur, rather, it's nearly impossible to diagnose as a surplus of high-frequency content. I stand by this firmly. I also find it very hard to believe that a Levinson amp is not capable of such heat dispersion. The HF noise is not such as to cause EXTREME high heat - it's not like the HF noise is so high in amplitude as to actually compete with signal level.
As for the comparison from DSD vs. conversion from PCM, this is purely subjective. Personally I differ radically in opinion. But, obviously, subjective matters are purely that and certainly cannot be proved.
Cucco wrote: This is why all of the editing that Telarc does i
Cucco wrote:
This is why all of the editing that Telarc does is performed within the analog domain (converted from DSD to analog, processed, then reconverted back to DSD.)
Huh??? That makes absolutely no sense. Care to describe the process in detail? Telarc owns SADiE, Sonoma, and Sequoia. Why would they edit tape (the only way to edit analog) with all that power there?
--Ben
So I thought I would join this fracas now as well. Having been
So I thought I would join this fracas now as well. Having been a very busy technician over the years I have discovered many ridiculous audio faux pas, innocent as they may be, they have caused similar problems and conversations like this here.
For instance, it's not uncommon for an operational amplifier to oscillate in the supersonic regions or even cause a huge DC offset, when they have a tendency to go bad. I certainly can't hear that, you can't certainly hear that but your amplifiers and speakers can. Sometimes your ancillary pieces of equipment may object to that vehemently and thusly will cause undue misery to other pieces of equipment.
Never mind about the morons who believe they can watch the football games on television even if the sound is turned down without any interference to their recordings is dead wrong. Televisions and computer monitors because of their high voltage circuits absolutely broadcast both electromagnetically and acoustically high frequency trash from 15kHz to 38kHz at substantial levels that condenser microphones have no problem picking up.
So generally I find your complaints probably mostly due to operator error as it usually is? No insult to anyone here, just some things that people don't necessarily think about that can cause these disastrous, equipment damaging problems to occur, while remaining innocently chagrined.
And please stop cleaning your jewelry while making a recording
Ms. Remy Ann David
It's sort of amazing how far "quotes" and info can wander from t
It's sort of amazing how far "quotes" and info can wander from the source and lose or change their meaning...
I'll set some things straight from my perspective:
1) I've never said that anyone at Telarc edits DSD sources in the analog domain. If you're seeing that somewhere, I certainly didn't write it. It's not true.
2) All DSD sources are edited here on either Sonoma or SADiE DSD workstations. We prefer these DSD workstations because the DSD data stays as DSD as long as we don't change anything on it, i.e. level. At the edit point ONLY, the workstation sends the DSD signal through the Sony "E-Chip" to perform the crossfade ONLY. The surrounding DSD data is untouched. No DSD signal is changed to DXD or analog at any point during editing. DXD is NOT used in Sonoma or SADiE DSD workstations.
The SADiE DSD workstation is capable of taking the DSD project from editing all the way through to final SACD Cutting Master authoring. I make the CD masters on the same system. Very cool!
3) The vast majority of Telarc-produced/engineered sessions are recorded with EMM Labs converters to a Sonoma DSD workstation. Our classical and many jazz sessions are recorded "live-to-2" and "live-to-surround." Some jazz and other non-classical sessions are recorded to multi-track Sonoma (up to 32 tracks). After editing (All on Sonoma in DSD!), the Sonoma multi-track source is used for a mixdown session through a high-end analog board with the stereo or surround mix recorded directly back into the Sonoma system via the EMM converters.
4) I've recorded to every available pro format over my career except Wire and Wax - otherwise I've used them all. I love 30 ips analog, no NR on a great analog deck such as ATR-24 or Studers. I've had the opportunity to have available for comparison the session source as compared to: DSD via EMM converters, DSD via dCS converters, DSD via Genex converters, DSD via Tascam converters, 96 & 192 kHZ via dCS converters, 96 & 192 kHZ via PT HD, 96 & 192 kHZ via Apogee, 96 kHZ via Prism AD2 converters, 96 & 192 kHZ via Pacific Microsonics converters. My preference is hands down the EMM DSD converters with Sonoma. That combo has been the closest to my analog console (or preamp) session source as compared to any of the above formats. The multi-track EMM/Sonoma system has given me the advantages of tracking on a better-than-analog system. I don't use plug-ins and I don't do pitch fixing, so ProTools-like systems don't fit my production style.
5) pcm sources at any fS by any manufacturer, later transferred to DSD by any means available will NEVER, ever sound as good as a DSD source of the same session recorded with a high-end DSD converter.
6) The workstations we use and love are:
3- Sonoma DSD
1 - SADiE DSD
1 - Sequoia pcm
3 - SADiE pcm
and several old pcm and one DSD Sonic still creaking along
7) I've played lots of DSD sources through lots of amplifiers and systems. No piece of gear has ever been fried by DSD material. Some amplifiers with unstable signal paths have exhibited oscillation due to the UHF noise of unfiltered DSD. A well-designed amplifier has never had a problem with the DSD signals. Incidentally, most every SACD player employs a gentle 50 kHZ filter on its output.
8) I've recorded with DSD since 1996. I have no imicrowave burns, although my feet do hurt occassionally. I wonder if THATs a side-effect of DSD ;-)
Hope this helps!
With Best Regards,
Michael Bishop
Wow Michael! Thanks for joining us here. Personally, I'm truly
Wow Michael! Thanks for joining us here. Personally, I'm truly humbled.
I apologize for mis-quoting. I could have sworn that I read you didn't edit in the box.
As for the workstation use, I made an assumption. I own literally every DSD and most every PCM disc in your catalog and I've never seen any other DAW credited except for Sonoma. Also, I've only seen info about Sonoma and Meitner on your website.
Thanks for sharing your thoughts on DSD!
J. (not worthy...)
Thanks for the enlightening post Mr Bishop! I belive the bigges
Thanks for the enlightening post Mr Bishop!
I belive the biggest problem with recording in DSD is pedagogical. The developers of the system have left most of uss bottomfeeders in a state of vacuum as the availible information on the format and how the editors work the files is rather poor.
I have some experience with Pyramix foe instance, an that littel tuch tells me that a workstation that handles DSD in a non native way is not the way to go.. Pyramix has becom very poular in my roame (Sweden) amon high end sound engineers, but most of them do nopt record in DSD, beacuse of the translating to PCM for editing fact..
Where do I find relevant information about Sonoma? (Looking at Saidies site, they do not seem to support DSD any more??)
/ptr
ptr wrote: Where do I find relevant information about Sonoma? (
ptr wrote:
Where do I find relevant information about Sonoma? (/ptr
Contact Gus Skinas at the Super Audio Center in Boulder, Colorad
Contact Gus Skinas at the Super Audio Center in Boulder, Colorado, USA
MixItUp wrote: Contact Gus Skinas at the Super Audio Center in B
MixItUp wrote: Contact Gus Skinas at the Super Audio Center in Boulder, Colorado, USA
I listed that site in the post right before yours..
:-?
Not too interested in DSD anymore here. Had the DSD equipment an
Not too interested in DSD anymore here.
Had the DSD equipment and sold it because no one wanted to produce
in the format and pay for the expensive equipment.
This, coupled with listening tests at Emil Berliner Studios in Berlin, that showed no detectable difference between 96 K and DSD in blind listening tests. They dumped it. Same at Teldex Studios. They dumped it.
Tony Faulkner dumped it. Even Sony dumped it!
However, I have enjoyed transferring the edited 1/2" tape to DSD on the
Tascam 1000 through the Neve DPD (with its inbuilt Prism converter.)
Sound is excellent and this can then be sent to a SACD mastering house.
I'd much rather they go out of business supporting DSD and SACD than have
my business affected by supporting a narrow niche format.
Telarc, Pentatone, Ondine and some specialists continue to support and release in real DSD/SACD. More power to them.
In my opinion, DSD and SACD have lost ALL of their snob appeal which
was really the only thing going for it.
My own belief is that the CD is not going to go away any time soon.
We record in the sample rate required for release and try to avoid
any sample rate conversion.
Personally I run at 44.1 through Lavry Blue. The results are ple
Personally I run at 44.1 through Lavry Blue. The results are plenty good enough for me.
As for SRC - there seems to be a widespread myth that going from 88.2 to 44.1 is easier than going from 96 to 44.1. This myth is generally spread by people that has not designed and tested SRC algorithms. It certainly must be much easier is the lay persons guess.
Wikipedia has a small article telling you a bit about how an SRC works. One important thing to notice is that if you write a good SRC it can translate between any rate of sample rates. There are even algorithms available that allows the ratios to be variable. This of course is a bit beyond the normal high-school level math.
http://en.wikipedia.org/wiki/Sample_rate_conversion
I should say that I have done some sample based algorithms, but no SRC-s. Frankly the part about sinc functions is a bit above my level of math.
What I am trying to boil it all down to is that the only way to select sample rate is by measuring and listening to the actual equipment and algorithms you are planning to use. If you start apply theories about things that you do not know intimitely it is common for the conclusions to be wrong.
Gunnar