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This is about the recent recording of the Mahler 3rd with Fischer and the Budapest Festival Orch. I'm looking to buy a copy, possibly by download.
The 2 channel version is offered in:
DSD256, 128 and 64; and FLAC192, 96 and 41KHz. It is also offered as an SACD which I take to be DSD64.
The engineering description given by Channel Classics is:
The recording was originally digitized using the Grimm AD1, which operates at DSD64. The original session tracks were edited and rebalanced (which meant going through the mixer) in the only available format for that purpose; the Pyramix 352.8KHz/24bit PCM (DXD). Prior to the advent of direct digital delivery, the next step in the production process from 352.8KHz/24bit PCM would be the DSD64 edited master for SACD production. What we have done now is also make a direct conversion to DSD128 and DSD256 from that original DXD edited master, without going through any interim processing steps. Those DXD to DSD conversions are not up-samplings, as they would be going from one PCM sampling rate to another, for they are different encoding systems. PCM is a digital value sample based system, and DSD is a digital bit density modulated system. Conversion from any PCM sample rate to any DSD bit rate system is a remodulation, not an up-sampling. We feel there is an audio advantage to this process in using the original files so we give you the choice and you can decide. Jared Sacks
So, if I am reading this right the recording starts out as DSD64 and gets converted to PCM and back to DSD64 for the SACD. [I presume a .dsf rip from the SACD would be the same as the DSD64.]
Then the PCM file is converted to higher order DSD versions and presumably to the FLAC files.
As the original recording was in DSD64 is there any benefit to having the DSD128 or 256 versions?
As the mixdown was in PCM would it be better to stay with the FLAC versions?
Would appreciate if someone could shed some light here. Though Channel Classics distributed a sample in any version, it was an out-take with almost no music on it.
And I think it really illustrates why you just can't release a recording as a one-shot experience with no editing - you HAVE to edit! About 26 seconds in, one of the horn players misses one of his notes - it's not THAT audible, but it's there. Just imagine listening to this mistake repeatedly, anticipating that it's going to be coming up, and losing your focus on the music in the meantime. Not good.
But OTOH, it does raise the same question which Mel raised (way down below in this thread): if you have to edit anyway, why not just record the work in DXD from the get-go, without all this conversion back and forth. Sure, I know the argument that the actual amount of time taken up by the edits is not that large. But Jared has told us that some movements of his previous releases contain a VERY LARGE number of edits (and there's nothing unusual about that - it's just the normal way classical studio recordings are made these days - and even the so-called "live" performances are a patched-together result of a series of performances, edited together).
So, how about it, Jared? I know you and Tom are believers in DSD, but might we hope that, at some point, we might have a DXD-sourced recording or two from Channel Classics? ;-)
The answer to your "why not just record in DXD in the first place" is there is no such thing as a direct DXD (or for that matter any PCM) recording method or hardware. Not at least with any A/D converter made in the last fifteen years used by any recording production outfit. The last really good direct PCM (20 bits on a really good day) A/D converter made was the Pacific Microsonics Model One/Two. These two products have been out of manufacture by at least fifteen years, but can be heard used today (by one of its inventors) by Keith O. Johnson on Reference Recordings.
PCM recording, including DXD (352.8KHz/24 PCM) is today accomplished by A/D conversion performed by Delta-Sigma modulation (a 1-bit or multibit DSD like Pulse Density Modulation process), then converted on-the-fly in the converter box to any sample rate PCM supported. The disadvantage of recording in DXD in the A/D converter are two:
1- The digital processing power available within the A/D converter is not as much as available in a reasonable PC, thereby limiting the robustness of the filter and conversion algorithms used. Plus, offline conversion in post does not have to be in realtime, further adding to the conversion choices.
2- This first PDM to PCM conversion, which is a lossy process, can not be undone in the future. Since native DSD content will increase in value in time as the format becomes the industry standard, the future value would be undermined.
If this sounds self serving, I'm happy to demonstrate to anyone visiting Boston the degradation of even one DSD > DXD conversion. You can experience this yourself with the DSD256 and converted DXD files available on NativeDSD. It helps of course to perform the demonstration with surround files, where the localization cues are much more recognizable, and using a DAW like Pyramix to facilitate A/B switching on the fly.
Until DSD post processing becomes available and commonplace, the best practice for complex large scale recordings is to record in DSD, post process in DXD, then offer deliverables in all playable formats. When DSD processing is made available, the original DSD masters will be available for reprocessing, and the DXD edited masters discarded.
. . . and I think that some of your claims (DXD edited masters will be discarded - really?) will be news to companies like 2L! ;-)
Check out the "Original Source" column on the far right:
Yes, I'm afraid that's to 2L's unfortunate disadvantage if they actually track the DXD output from their converters, and not the DSD doing an offline post processing conversion to DXD.
Morton has always been a DXD proponent, through the days of using their DAD AX24 and AX32 A/D converters, to their current use of Merging's Hours. All use Delta-Sigma modulators as front ends (the actual analog to digital conversion element), and employ onboard realtime 352.8KHz 24 bit converters and decimation filters for the DXD conversion, if that output format is chosen. This results in a lossy second generation output.
I have no knowledge of 2L's workflow, but I do hope they actually track in the fastest bit rate DSD (DSD256 in the case of Horus), then do an offline DXD conversion using Pyramix's excellent Aprodizing conversion algorithms, keeping the original DSD for potential future use. Any other approach makes little sense; throwing out the original higher definition PDM (DSD) format in the A/D converter box to keep the working editing format (DXD) output, where you could have both for just the cost of storage.
As I mentioned in my previous post, if you create two projects in Pyramix (a DXD and DSD) and load a DSD file in each, you can switch between each on the fly. Listening to the original DSD, then the same DSD file converted on the fly to DXD, and played out through the DXD Project Mixer, you immediately sense the added sharpness and reduction of some spaciousness details inherent in PCM. Glad to demonstrate to anyone passing through Boston.
Thanks again Chris
Of course you're going to encounter some deterioration on the converted file!
But actually, at this level of resolution (either DSD256 or DXD) I'd be surprised if the deterioration IS really audible and not just a product of expectations and/or psychology. But whatever. I guess I'll just have to pay you a visit the next time I'm in Boston (a city I've missed so far in my life) - I'd love to hear those Sound Labs! ;-)
Same as occurs within an A/D converter outputting DXD instead of DSD, via the conversion within the converter. You're always welcome to visit Chris and judge for yourself :)
I apologize for cutting and pasting - this is from John Siau of Benchmark Digital:Modern PCM sigma-delta converters produce much lower error signals than 1-bit sigma-delta DSD converters. The errors in the DSD system are due to the 1-bit quantization that occurs in 1-bit sigma-delta DSD converters. Multi-bit PCM sigma-delta converters can be fully dithered and do not suffer from this un-dithered truncation. Every added bit reduces the noise signal by 6 dB. A 4-bit sigma delta converter is 24 dB quieter than a 1-bit sigma-delta DSD converter. Right from the start, 1-bit DSD signals have higher losses than multi-bit PCM signals.John wrote all this in 2015. Have things changed in the meantime?
Conversion from 1-bit DSD to multi-bit PCM is a lossless process inside the audio band. The only thing that is removed is the out-of-band noise above the Nyquist limit of the PCM system. Nothing else is lost. . .
Conversion from multi-bit PCM to 1-bit DSD is always a lossy process. The loss is due to the 1-bit truncation. This truncation introduces a very large ultrasonic error signal that makes the ultrasonic region unusable for audio.
Processing a 1-bit signal to create a 1-bit signal is also always a lossy process. A volume control is one of the simplest processes in a multi-bit PCM system, but it creates a large error signal when applied in a 1-bit DSD system. The same is true for any other 1-bit to 1-bit DSP process. The lossy part of these DSP processes is the quantization back to 1-bit. Cascaded 1-bit truncation processes can rapidly degrade the audio quality. For this reason, DSD is always processed as multi-bit PCM.
Any DSP process applied to a 1-bit signal produces a multi-bit signal. No loss of information occurs until this is quantized back to a 1-bit signal. Why incur the loss by going back to a 1-bit signal after the processing chain?
All practical DSD systems require some sort of DSP processing (gain control, mixing, filtering, etc.) and all of these processes produce multi-bit PCM results. Taking these lossless multi-bit results and adding loss by truncating them back to a 1-bit DSD signal makes absolutely no sense. DSD complicates the processing and adds unnecessary losses to the signal path. DSD does not simplify the signal path. There is absolutely no truth to the marketing hype that claims that 1-bit DSD is a simpler system than multi-bit PCM. The exact opposite is true.
Nothing has changed since 2015, and there's no such thing as a "PCM Sigma-Delta Converter". PCM is an acronym for Pulse Code(ed) Modulation, and PDM for Pulse Density Modulation, also referred to as PWM (Pulse Width Modulation).
You'd have to ask John what he's talking about, but as I read it, there's two possibilities:
1- He's substituting the acronym PCM for the actual PDM (two very different formats), in which case his argument makes some sense.
2- He's conflating two independent processes and calling that a "PCM sigma-delta" converter (which coincidentally are the separate processes inside a pro A/D converter). The first process, (on the A/D side), is the Delta-Sigma modulation of the audio signal; producing either multiple parallel Pulse Density Modulation (PDM) bitstreams (multi-bit PDM, which appears to be John's inaccurately naming multi-bit PCM), or much less frequently a 1-bit PDM bitstream (DSD as with the Grimm A/D Converter). If PCM is desired/required, then these PDM bitstreams (1 to 8 typically) are then followed by low pass filtering (decimation) and conversion to parallel X-bit wide Pulse Code Modulation binary words. That latter process is most certainly a lossy process. Once decimation filtered, you can no longer reconstruct the original PDM (DSD if 1-bit) bit stream.
But so what? My original statement was, and remains that all acoustic audio recordings are made using A/D converters, and the vast majority of those A/D converters are all front ended with Delta-Sigma modulators, producing an actual DSD 1-bit bitstream (as with the Grimm A/D), or from two to eight parallel (multibit) bit streams of Pulse Density Modulation. And, any conversion to any other format for whatever reason (typically post process sweetening and/or conversion to a deliverable format) is a post process from the original A/D PDM bitstream(s) conversion, and is lossy. So, if you can archive the original A/D conversion before performing the PDM to PCM/whatever format, why not do so for future processing possibilities? And especially, if you have the option, why not do those format conversions offline using more powerful conversion algorithms with more powerful processors than are available in the A/D converter box?
Seems self evident to me. But on the other hand, I don't have products to sell, or customers to convince.
But, as I read it, I think that your differences with what John wrote seem to lie in what each of you thinks is lost when going from DSD to PCM.
When John says:Conversion from 1-bit DSD to multi-bit PCM is a lossless process inside the audio band. The only thing that is removed is the out-of-band noise above the Nyquist limit of the PCM system. Nothing else is lost. . .And you say:
Conversion from multi-bit PCM to 1-bit DSD is always a lossy process. The loss is due to the 1-bit truncation. This truncation introduces a very large ultrasonic error signal that makes the ultrasonic region unusable for audio.If PCM is desired/required, then these PDM bitstreams (1 to 8 typically) are then followed by low pass filtering (decimation) and conversion to parallel X-bit wide Pulse Code Modulation binary words. That latter process is most certainly a lossy process. Once decimation filtered, you can no longer reconstruct the original PDM (DSD if 1-bit) bit stream.
So. . . as I read this, I conclude that you're definitely removing SOMETHING in going from DSD to PCM. But what is it? Is it the out-of-band noise and nothing else, as John seems to suggest? I mean, in a way, you guys are kind of saying the same thing. Right?
Chris, you're asking a great question that has been discussed with many thousands of words. I'll give you the short answer, and you can research the underlying discussions. I'd suggest if you're interested, the many postings at Computer Audiophile. Do a search on Miska, the provider of HQ Player, particularly on the DAC Forum.
The short answer to your concluding question is yes, the decimation filter has to eliminate the out of band noise, and in doing so, injects phase artifacts (unfortunitly required by the steepness of the filter) that render the resulting PCM words not convertible back to the original PDM/DSD bitstream(s). Therefore, it's a lossy process.
It's the effects of the necessary decimation filter required in the DSD > PCM conversion to preclude folding back into the audio band any energy above half the sampling rate of the converted PCM result. Since the vast majority of energy in any PDM (DSD) bitstream lies above the audio band, being almost all the energy approaching the bit rate frequency, the decimation filter has to be very steep approaching the upper frequency of the audio band of interest. The decimation filter shaping, with its subsequent phase distortions and nonlinearities (necessary to achieve the required filtering shape), make the reconstruction process impossible back to the original bitstream. Therefore, it's lossy, and in my experience quite perceivable.
A PDM (DSD if 1-bit) bitstream is almost entirely uncorrelated white noise. It's a useable A/D converting format through the process of noise shifting (an activity of the Delta-Sigma Modulator), where the uncorrelated noise is shifted to above the audio band, then filtered out when the audio content is retrieved in playing. Since PDM/DSD is already an analog bitstream, it simply requires low pass filtering to retrieve the audio content (there's many ways to do this, which I won't go into here, but the filter has to be either very steep (not good) if the audio is to be retrieved at the original bit rate, or the bit rate upconverted to a much higher bit rate, and a gentler shaped filter employed).
But for digital post production processing, since PDM/DSD is an analog representation of changing input analog signal levels, it contains no digital values ala PCM. It only contains analog signal level changes represented by the density of bits in the bit stream. Since no values are represented (actually, the density of bits is directly proportional to percent of modulation, which in turn CAN be translated to absolute signal level values in the conversion to PCM), the bit stream must be converted to a value based format (PCM) to be further digitally processed.
So, back to my original point; if as a recording producer you can archive the A/D converters original front end Delta-Sigma modulation bitstream(s), in the form of a 1-bit DSD bit stream(s) at the original recorded bit rate, why not do that? Then you can convert to whatever required/desired 2nd generation format meeting the needs/desires of the producing engineer/label, and have as backup the original PDM/DSD content for future possibilities.
2L's Morten Lindberg, as well as Bert van der Wolf of Northstar/Turtle/Challenge Records (among others) say they prefer the sound quality of DXD, and so record in that format (they say). That's their prerogative and choice, but it is a 2nd generation conversion.
I'm not sure I agree with everything you're setting forth in your post, but I don't want to get into a multi-day discussion of PCM vs DSD - that argument has been going on for a couple of decades now! ;-)
BTW, I've seen some of those discussions with Miska, but I'm sure there are many more which would be useful to follow.
I grew up with a performance of Messiah (Sargent? Sir Thomas?) replete with a horn flub in the Hallelujah Chorus. It became normative. Nothing since sounds right.
I think that happened on the Barbirolli Mahler Fifth - the LP album was release that way (i.e., with the missing horn part - unfortunately, I've forgotten exactly where it was). But when EMI was preparing one of the reissues, they allowed the horn player to come back into the studio and record his missing notes, which the engineers then edited back into the original recording! ;-)
Try here (all DXD albeit not Channel Classics). BTW does this look like it has a commercial future to you? It's in its third year. As I doubt that anyone can hear any difference between DXD and 24/192 I will remain sceptical:
. . . there doesn't seem to be a single title there which is in multi-channel. Not that I would get too excited about Michaela Petri (oops! I guess it's Michala Petri - have I been pronouncing her name wrong all these years?) and her recorder anyway. ;-)
But in any case, I appreciate the thought! And I'm not sure I could tell the difference between 96K and DXD myself! . . . angels dancing on the head of a pin. . .
So, other than that, Mrs. Lincoln, how did you enjoy the play?
Down below I brought up the comparison of multi mike recordings vs the old 3 mike recordings. Also the old recordings were done analog with tube mikes and tube recorders. Remember my listening is 2 channel. I sampled a bunch of Ivan Fischer Channel recordings on Tidal. Mahler, Wagner, Bruckner, and Stravinsky. There definitely is a house sound. I then listened to a Mercury Anatal Dorati recording mainly of Stravinsky. I much prefered the Mercury. More presence,greater dynamics and much better hall ambience. The channel sounds laid back with much less dynamics. Remember I spent my recording career using all analog tube equipment and when I did record classical orchestras I used the 3 mike Decca tree arrangement. Just my opinions. Also I don't necessarily believe it is a digital thing. More of an engineering thing. I have heard some great digital. Professor Johnsons Reference recordings for instance. Also Brothers in arms.
. . . the dynamics and hall ambience. I'm sure that at least part of this impression is accounted for by the two-channel vs. five-channel systems each of us is experiencing the performances through. Was the Mercury Stravinsky recording you listened to the Dorati/LSO Firebird? The general consensus is that that's one of the best recordings Mercury ever made. (BTW, I think I have about 75% of all the Mercury recordings which were issued on CD or SACD, plus a couple of the Plangent-process hi-rez downloads which derived from the later volumes of the big Mercury boxes, such as the Tchaikovsky Suites with Dorati and the New Philharmonia Orchestra, the Paray/Detroit Organ Symphony, etc.)
All other things being equal however, yes, I do prefer more minimally microphoned recordings, and in that sense I agree with you. Minimally microphoned multi-channel recordings, such as the ones on the Nishimura label (NLA unfortunately), have been produced - but even here, not all of these recordings hit the sonic bulll's eye! OTOH, I have to say that multi-microphoned recordings have gotten better and better over the years, and I generally find the ones made in the last decade or so to be quite enjoyable.
The most important thing to consider here is that this is one truly excellent performance. I loved it the first time I heard it, including especially the Alto, Gerhild Romberger. The relative qualiies of the various formats are mere quibbles, I am sure, compared to the excellence of this performance.
Eat your hearts out guys, but I have a non-commercial, native DSD256 Mch version of the same performance made by a friend and Grammy-winning engineer who works with Jared. It is completely unedited, and the performance is remarkably clean without any obvious faults. He used his own special mikes and equipment in parallel with Jared's. The result is spectacular.
However, my Mch system needs DSP for speaker distance correction, bass management and Room EQ. The DSD256 version sounds amazingly realistic but somewhat distorted spatially on my system. Merging Technologies had been using this non-commercial recording as a demo for their NADAC at shows.
So, I play it as I normally do DSD material via conversion to PCM176 with full DSP processing, where it still ranks as the best recording I have ever heard. My good friend and recording critic for TAS, Andy Quint, agrees.
I am going to also acquire one of these commercial versions also for comparison. This thread now gives me something to think about as to which one. Maybe, it will just be the SACD.
Incidentally, my friend recently gave me a file of the last movement of a Budapest Das Lied Von der Erde also with Gerhild Romberger. Awesome and perhaps even a tad better as a recording. My engineer friend agrees. Really looking forward to the whole release.
To provide some background for the recordings I make with Jared of his large orchestra projects; they're experimental and archival, using minimalist microphone alignments. They serve the purpose of providing an alternative minimalist ITU surround microphone alignment recording, as well as until recently, recording at DSD256. In addition, they serve as a session backup, and are the property of Channel Classics.
Channel Classics uses the proven microphone technique for recording orchestras used by all labels; a main array supplemented by accent spot mics placed within the orchestra. This technique, unlike the majority of Channel's much smaller scale projects with fewer microphones, requires mixing and balancing to derive stereo and multichannel recordings. This in turn, recorded in DSD, requires conversion to DXD for mixing. Once edited and mastered, the DXD edited master is converted to the various deliverable formats.
The recordings I make with Jared of the same sessions require no mixing and balancing for surround, since they're only five mics for the five surround channels. DXD conversion is not required for editing, with the exception of the average one tenth second of the edit crossfade interval.
Jared is contemplating editing the DSD256 recorded minimalist mic version, and making it available as a download. The edits I've made of the many takes (typically three playings plus patches) are without consideration of music values or interpretation, just that content that pleased me. Jared of course could not allow the orchestra to be publically presented unedited, so a released minimalist mic version will include the appropriate producer's edits, without DXD conversion, except for the mix to stereo,
A free mixed down stereo and native DSD multichannel sample of my version of the 1st movement's beginning is available on nativedsd.com in all the DSD bit rates and DXD:
Chris - I have to respect the engineer's and other commercial interests. But, there is a sampling of that special version here:
Perhaps but there may be legal/contractual issues in the way.
The good news here is that with this particular release, they will sell you the DXD (352.8KHz/24bit PCM) MASTER.
Why do I call it a MASTER? Because, according to Jared's comments as quoted in your post, that is the format that they used to master the album in and EVERYTHING they sell including the SACD seems to come from that DXD formated file as it came out of the Pyramix mixer.
Or so it would seem.
Thank you. The Channel Classics page I was on did not show that.
And it sounds great! ;-)
The original recording was DSD64. All other rates are either upsamples in the DSD domain or PCM transcodes. None will contain more musical information than the original DSD 64.
Upsampling and transcoding can (and usually do IMO) produce unwanted artefacts which may be very subtle but may neveretheless be audible. So the simplest answer is to keep to the original recording format. However this is not possible here because, as with the the majority of DSD recordings, it has had to be converted to PCM for editing and then back to DSD. The latter process does not necessarily mean that any artefacts from transcodimg are obviated.
So, as all of the formats offered are compromised and as there can be no additional musical information in the higher rate conversions, two other factors become important.
The first is hardware i.e the DAC and how good the digital filter or filters ( if you have a choice) are with the particular format. This will have more effect upon the resultant sound than the format itself. In fact as far as I can work out the only advantage to upsampling is the ability to place the pole of the filter further outside of the audioband ( which may also allow use of a gentler filter characteristic albeit with lower suppresion of aliases). As the filter seems to be the trump card this accordingly provides an unpredicable situation for recommending formats on a general basis.
The second factor is cost. As the DSD64 version is less costly than the others and as it was the original format unless I had experience that one of the other formats always offered better sound with my particular equipment I would save my money and go for the DSD64.
Chris, that does not mean that your DXD version does not sound great. It does, but with your equipment. No guarantee that it will be better or best elsewhere.
NB: Out of interest I spent a merry hour or so last month converting redbook files to both DSD 128 and DXD. The result when going back to the original redbook was similar to removing a stone from a shoe. So nice when it (the upsampling) stops.
Here are Jared's own words:The recording was originally digitized using the Grimm AD1, which operates at DSD64. The original session tracks were edited and rebalanced (which meant going through the mixer) in the only available format for that purpose; the Pyramix 352.8KHz/24bit PCM (DXD). Prior to the advent of direct digital delivery, the next step in the production process from 352.8KHz/24bit PCM would be the DSD64 edited master for SACD production. What we have done now is also make a direct conversion to DSD128 and DSD256 from that original DXD edited master, without going through any interim processing steps.
Those DXD to DSD conversions are not up-samplings, as they would be going from one PCM sampling rate to another, for they are different encoding systems. PCM is a digital value sample based system, and DSD is a digital bit density modulated system. Conversion from any PCM sample rate to any DSD bit rate system is a remodulation, not an up-sampling.
We feel there is an audio advantage to this process in using the original files so we give you the choice and you can decide.
Otherwise, I agree with what you said.
Your comments about upsampling vs. native resolution intrigued me. I have an Esoteric SACD player, and like several players these days the Esoteric offers multiple sampling/filtering options for RBCD playback to keep tweakers occupied. I played around with these settings right after I bought the machine, but ended up selecting DSD conversion as (seemingly) the best-sounding option for CDs. Are you implying that CDs would sound better using the straight 16/44 setting? I'll have to experiment to see how this plays out in my own system.
If I've misread or misinterpreted your post, my apologies.
" Are you implying that CDs would sound better using the straight 16/44 setting? ".
Not quite as I am saying that the digital filter characteristics will determine the "ranking" of any one format over another where sample rate conversion is involved. This may not be the case using native hi-rez formats as they will contain actual additional musical information.
If Esoteric provide various filters that may be used with straight 16/44.1 files it may be that use of one will sound better with this format than upsampling it to DSD with whatever choice of filters you have there. However as each player manufacturer can implement its own filter designs there is no common ground. So even if I believe that upsampling results in certain subtle sonic abberations the net result may still be preferable over redbook in a given system . It depends upon your particular set-up ( and preferences). If you are fortunate to have access to an asymmetric or apodising filter for redbook then I would expect the native resolution to be preferable. However given a typical 22Khz brick wall filter with its attendant pre-ringing, maybe not.
My view would also be that if possible where upsampling is considered necessary is to keep it in the same domain as the original and to use only multiples of the original sampling frequency. So ideally I would avoid transcoding from PCM to DSD and for 16/44.1 use 88.2 or 176.4 sample rates. Hopefully your machine also adds 8 digital zeros to provide 24 bit depth. Again no additional information is created but this can be an advantage with many digital processors where they are designed to process 24 bit words.
I can only suggest that if you have been using DSD upsampling for a long time without question it may be worthwhile switching back to 16/44.1 just to check that your decision to use DSD was correct - it may well be but it will depend on the filters and the upsampling algorithm that Esoteric have provided.
"None will contain more musical information than the original DSD 64."
Sadly not. Sony never completed development and sale of a commercial mixing system for DSD.
DXD is as close as you're gonna get, at least in this case. For all intents an purposes it is the 'master' as it came off of the mixing board.
The thing is, Jared is being very honest about his process so you can choose your best strategy to get as close a possible to that original recording.
I just wish he would offer the DXD files as download options in the future.
I intend to download the Mahler 3 DXD at some point, soon as I figure out how to play it. ;-)
The link is to the Channel Classics catalogue. I know that my vision is impaired but I can't see any reference to a DXD issue of the Mahler 3. Jared refers only to DSD 128 and 256 files transcoded from the DXD file.
I will remain sceptical as to any advantage of these over the DSD 64 version. Although I am always interested in the proof of the pudding I won't be buying the DSD64 and DSD128 versions to compare, mainly because I don't like Mahler much!
and only one has the DXD for sale?
Maybe. I suspect a record company cock up. Interestingly one site is the Channel Classics catalogue, the other is Native DSD which I think is owned by or otherwise related to Channel Classics but offers other labels.
Interestingly if you buy from the catalogue website (without a DXD offer) your order will be processed by Native DSD ( where the DXD file is available) as they are linked. I'll look for a handy nearby wall for some head banging.
I'll jump in soon as I am yet to figure out how to get DXD to play on iTunes (with Bit Perfect plug in). The Bit Perfect plug-in for iTunes does not support FLAC so I have to convert to something Bit Perfect will play while maintaining the sample rate of DXD.
All very complicated for an rapidly aging brain.
. . . although, come to think of it, I haven't actually tried playing multichannel files while Audirvana was still operating in "compatibility" mode. Maybe I should give it a try and report back.
back when I dawdled with Mac.
. . . because iTunes has a WAY BETTER user interface (IMHO) than, for instance, HQ Player does. Too bad about all of iTunes' other limitations. As my library of multi-channel files continues to grow, I find myself (of necessity!) using iTunes less and less.
I did try it a couple of times, almost by default on a couple of machines. However, I found the UI totally counter-intuitive and the format/program constraints unacceptable for classical music.
That was a long time ago, so things might be a bit different now but I use programs that are much more configurable and which have little or no constraints on how I use them or on what files I can play (including the rare AIFF).
I know of nothing about it to tempt me.
iTunes will not allow a multi-channel aiff file to be added to its database, so the only way that Audirvana can play multi-channel files is via its own database. (So compatibility mode needs to be switched off in order to play the multi-channel file.)
prefer the SQ of iTunes/Bit Perfect.
And stereo headphones so I don't care.
I will be downloading the DXD Stereo version anyway.
It never occurred to me that there actually were 2 channel DXDs! Duh.
most of us were born with only two ears. You guys seem to have FIVE or even MORE! ;-)
And yes, navieDSD will sell me the DXD file in multi or stereo but from their web site:
DXD = 32/352.8k PCM - 8X Redbook [file type is .wav and sometimes .flac]
I can import .wav into iTunes but not .flac. :-(
AFAIR, if you import a 24-bit WAV file into iTunes, iTunes will simply truncate the last 8 bits, resulting in a 16-bit file which is not optimized for 16 bits - unless something has changed in the last few years.
via the USB input (Amanero Combo384 Module).
Not sure I get the benefit of the 24 bit depth but who needs to hear 'Brownian' motion? :-)
As the DXD file is FLAC I will need something to convert it with.
Any suggestion for freeware and is there any other CODEC other than FLAC that will store 24/352.8?
I used to use a product called "Max" - although never for DXD. It also ran afoul for me when converting multichannel Chandos flacs to aiffs. (Two-channel flacs were always OK. And even with the multi-channel problems I had - with the wrong signals ending up in the wrong channels in the converted files - ONLY Chandos flac files were affected - the conversions from BIS flacs were always fine - go figure!)
I also have a conversion app called XLD, but I haven't used it yet and I don't know what its capabilities are. I also think that dbPoweramp might work too.
I don't understand paying more money for a DXD file and then converting it to something else in order to play it.
24/96 flacs converted to 24/96 aiffs.
I only have a couple of DXD files (so far anyway), and I play them as flacs (i.e., as downloaded).
Or are you talking about ivan? I'll let him answer for himself.
No, It is my understanding that Channel masters in dxd and then will sell you different versions derived from the DXD master. Seems overly complicated
in Stereo or Multi-Channel.
Sadly, this is NOT the case with most other Channel Classics releases. :-(
Some listeners have personal preferences for one format or another, and are willing to argue about it for years! ;-)
I actually applaud Jared for trying to accommodate these different types of listeners in this release.
To me it looks like these various formats are just a lot of technical hype for marketing purposes. It raises the price of distribution, and ultimately the price to the consumer. I don't believe you can go from one format to another without losing something! Whether it's audible or not, and to whom, might be the issue.
If they wanted to give us the very best, the "mix" would have been in the original miking as was done on the so called "non-commercial" DSD256 version written of above. If DSD64 is what they were working in (and I guess it could have been DSD256) that could have been the principal release. In the glory days of vinyl, were not many recordings made that way? (Too many mikes these days?)
So then they convert to 352.8KHz/24bit PCM (DXD). Question: If they felt they eventually need to go to the Pyramix, why did they not use 352.8KHz/24bit PCM (DXD) for the original live recording? Then they could have done the "required" mixdown with no conversion and that would represent the best that it would be possible to offer. [But they would lose the marketing cachet of having recorded in DSD v. the "old fashioned" PCM.]
But having converted to 352.8KHz/24bit PCM (DXD), how could the fidelity improve by doing yet another conversion? I would have to conclude that 352.8KHz/24bit PCM (DXD) is the best they can now make available.
Given the sophistication of the playback equipment likely used by those considering either the 352.8KHz/24bit PCM (DXD) or one of the DSD versions, is it likely that a DSD version will play back more accurately than the DXD? Is it not accuracy that is the goal? Or is it meeting the personal preference of an end user unfamiliar with the original recording experience?
Finally I might express the view (which a multi-channel listener could feel free to correct) that those of use who listen in just two channels may feel a greater need for accurate aural cues for the space and placement of instruments.
In this case, I would conclude that too, but, as I mentioned, not all listeners feel the same way.
in the recording industry is that Jared actually reveals how he makes the sausage.
You can record in DSD but you can't mix in DSD so you convert the DSD to the very best PCM format available which I trust Jared uses in the Pyramix.
On can argue all day long about formats, PCM vs. DSD and bit depths and sample rates but the truth of the matter is, the problems are largely DECODING issues in the various DACs and NOT in the information contained in the various formats.
In other words, if the recording engineer got the recording part right and it wasn't screwed up in the mixing board, it's on YOU to figure out the best way to play it back.
I've heard Jared's recordings played back on Gigabuck gear and can safely say he is producing some of the best recordings being produced today.
"In other words, if the recording engineer got the recording part right and it wasn't screwed up in the mixing board, it's on YOU to figure out the best way to play it back."
Interesting, and just how are we to figure it out, aside from theoretically, to wit, the DXD files.
I appreciate the samples and I downloaded 2 of them, the DSD256 and the DXD. Problem is that the DSD file has little more than adjusting the bass drum sticks. The DXD has more. Perhaps I should try others. But these do not represent the miking and mix of the final product.
Better still if Jarad would distribute a six minute sample of the final mix that he thought best illustrated his work, and in the various formats.
IMNSHO, it's gotta be user hardware dependent.
There are so many choices in how to decode a digital file that I doubt anyone would ever agree.
I use a multi-bit ladder DAC that likely has a real bit depth of about 18 bits or so. It's based on 8 each PCM1704's and would by many be considered obsolete as TI doesn't sell the chips anymore.
Others have DSD DACs fully capable of decoding DSD-512, some use software to up-sample to that rate, some DSD DACs up-sample internally to even higher sample rates and even deeper bit depths and on and on and on.
Who is to say what's best for you?
I use iTunes with a $10 Bit Perfect plug-in so WTFDIK
You're correct, the free samples on NativeDSD are not the micing and mix of the released album, but are a single take snippet using different mics, recorder, and placement. But the samples do provide accurate comparitive examples of the different formats (PCM/DXD and DSD rates) as how they will play and sound on different playback systems.
Your's is a reasonable request of providing a six or so minute sample of the released Channel Mahler 3 recording in the various delivery formats/rates, so we'll see how best to distribute these.
Thanks for your suggestion!
I spoke with Jared, and he'll add the Mahler 3 5th Movement to the NativeDSD Just Listen label page, in all the deliverable forms next week. This movement is particularly useful as a format comparison, as it is very rich sonically containing not only the orchestra, but the women's and children's chorus as well as the alto Gerhild Romberger.
I'll keep you all informed.
Due to vacation schedules it was not possible to place the Mahler 3 5th movement on NativeDSD this weekend. Until we can get it placed, if anyone wanting the movement earlier would mail me at the Audio Asylum tailspn address with the format and speed desired, I'll E-mail a We Transfer link to it.
and iTunes with the Bit Perfect plug-in won't play FLAC. The nativeDSD site does not specify which file type the download.
DXD is 32/352.8k PCM and it doesn't necessarily have to be a .flac file, could just a easily be a .wav file and it would remain a DXD file and would still be 32/352.8k PCM
See link below:
Or, it could even be a 24/352.8 aiff file. ;-)
if there is such a thing?
My freeware conversion only goes to 192 :-(
Do all these different formats really sound significantly different or are they small. For me it is still the original quality of the recording that matters. For example I really don't care for the sound of the Channel Classics recordings. The don't have the immediecy of for instance the old Decca, RCA and Mercury recordings. I think they are using to many mikes.Read a review of one of there recordings which said they used 17 mikes. Now it is really up to the mixer to create the sound they are looking for.
if they are truly lossless.
The real issue here is with the playback software you use, will it even play it, how much CPU time it takes to convert it to play, how much space the particular lossless CODEC takes on the hard drive, how much it can be compressed for streaming or downloading, etc.
They're very small - as I always say, it's like arguing about how many angels can dance on the head of a pin. (FLAC files are losslessly compressed, whereas AIFF and WAV are usually uncompressed.) And you're right that it is still the original recording quality which matters most, regardless of the particular incarnation.
Yeah, I like the old Decca, RCA and Mercury (and EMI!) recordings too. But OTOH, I believe that great recordings can be made with different microphoning philosophies, and, certainly (or at least IMHO), Channel Classics invariably gives us outstanding examples of their recording philosophy with each new release - really state of the art for their particular approach.
We can't possibly be defining state of the art dependent on what recording method is used? Can we? State of the art should be the absolute best, period!
You say that "State of the art should be the absolute best, period". And yet you also ask "We can't possibly be defining state of the art dependent on what recording method is used? Can we?". If we're talking about absolute best, then I guess we'd have to say that, yes, one and only one recording method must be superior. (Or, one and only one microphone set-up - or whatever.) I don't believe this myself, so I guess my definition of "state of the art" is not as narrow as yours?
I don't really disagree with you. There are a lot of great recordings out there. Having listened to tons of recordings and also recorded both multi mike and the 3 mike Decca tree method I think there are more great sounding recordings using 3 mikes than any other system. The move away from 3 mikes as the technology changed was the famous statement Will fix it in the Mix. Works sometimes and also fails sometimes. As far as multi mike recordings go I think Reference Recordings have done the best work.
or buy what you can play.
which I understand is a fairly common occurrence or there wouldn't be all of that conversion software available.
But the conversion freeware program I use won't go past 24/192 and I'm not certain what file type I should be using.
Bit Perfect WILL handle up to and including 384kHz sample rates, all I have to do is figure out how to get it into iTunes in a format that Bit Perfect can play it.
iTunes is for folks with two ears, not five. :-)
lossy 256k content. :)
will play just about any sample rate you choose, up to and including DXD. But not from a FLAC file.
But I also have an Audirvana plug-in that plays nice with iTunes
of being 'with Kal'. :-)
his previous comments of course. :)
I do run an Acoustat based MC/HT system although the only stats are the mains.
I'm trying to sell my original non-PX cores to a guy with three U-1s and spare backplates who wants to create a five channel SL system. Not quite as much panel area as Ray Kimber's ten Prostat 922 multi-channel system, but still a bunch!
I've got too many coffin sized boxes in storage. :)
Worth the drive to Denver (I lived in Utah just down the road from Kimber Kable during many of those years).
If you're gonna do multi-channel, I guess ya gotta do it right!
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