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 Post subject: DSD vs PCM - and a small truth about SACD
PostPosted: 27 Oct 2015, 21:24 
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Ok...so I know I've been inactive for a while....and I came here to mostly check up to see what's happened during my long absence (plus I have a mildly related LD post to write in a little while). But I was excited to see a forum for SACD; because as an audio guy I've spent a lot of time studying the format and whatnot. I've actually authored an SACD disc a time or two using the so-called "SACD-R" format. Most of you might know this, some of you might not....so I'm going to share what I know.

Ok, so as you probably know, SACD uses the DSD format to store it's audio; and you've probably read where it's a 2.8 mhz sampling rate. "Wow!" you might think; "that's a lot more than 44.1khz i'm using on my lousy CDs"...well calm down there Fido. There's one little thing people tend to really overlook when discussing DSD.

DSD is a 1-bit format.

So what this mean? This means that rather than storing audio using a PCM method; in which discreet samples store an integer value that correspond to a voltage; 1-bit formats store the "change" in your waveform. It's functionally similar to delta-sigma modulation; and 1-bit is also how 99% of DAC's actually convert the digital audio to analog audio. This was done becuase, back in the 80's; building a DAC that worked directly from PCM to analog was difficult; plus the technology of the era was very limited and dynamic range was crap. The term for this is "oversampling DAC". The DAC converts your PCM signal in to a 1-bit delta-sigma stream; which can be converted to analog using something as simple as a low-pass filter to remove the ultra sonics.

The way it works is you have a sawtooth wave generator that is capable of running at an insanely fast rate; and every time you trigger the generator, it switches "directions". The idea is that you basically trigger this sawtooth wave generator (which is outputting a voltage) so that the sawtooth stays "right around" the area of your analog wave form. The output would be, from an analog stand point; a representation of your audio with a super-sonic waveform superimposed on it. IT's also worth mentioning that filters can be built easier for higher rates than lower rates; so dealing with a 2.8mhz output rate means you get to use a relatively simple filter on the output. Remove the ultrasonic sawtooth from the signal; and you're left with a relatively clean analog waveform.

DSD is basically storing this 1-bit audio format on the disc rather than enumerating it to PCM like we do for CD. So what advantage is this?

Well..the main advantage is if you're storing your audio in a 1-bit format already; you don't have to convert it to a 1-bit stream; you can just send it directly to your DAC. As I mentioned; 99% of DACs on the market (basically every dac capable of more than 16-bit) use an oversampling method; storing your audio in 1-bit actually eliminates a conversion step. Part of the perceived enhanced quality is because of that; converting PCM to DSD and vise-versa is not an easy or accurate task. While you do have this advantage; there are a number of disadvantages.

For starters, even with the insanely high sampling rate; the 1-bit suffers from horrible dynamic range in general. SACD overcomes this by using noise-shaping; a chunk of inaudible ultra-sonic spectrum contains nothing but noise...basically dithering. This allows you to increase the apparent dynamic range of the recording. It's used in GIF's to make the color downsampling look better; it's also used in better PCM Editors to convert 24-bit audio to 16-bit audio without encountering quantization noise...you put enough noise somewhere where it won't be heard so that you maintain an amplitude over the quantization level.

This is why it has been said "SACD is only as efficient as 20-bit 88.2khz". Basically, just about everything above 40khz in the audio spectrum is noise. It's not that the format itself cannot respond to something that high; in fact I've seen sweeps well over 100khz in a DSD file; it's just you have way more noise than signal up at that end. Most SACD players, in reality, have a 30 to 40khz lowpass filter.

The other issue is that a lot of people seem to want to play back DSD files on their PC without having a DSD capable DAC. So you're going though a DSD->PCM step..just to go from PCM->1-bit in your DAC. It's a pretty wasteful process; and it's one of the reasons people who attempt this start to say "DSD is stupid". It is not a horrible process as long as you realize what you are doing and use good filtering; I myself have a couple of DSD recorded tracks that have been converted to high-rate PCM and enjoy them just fine.

Some players, especially multi-format players; have a lousy habit of converting the DSD to 24/88.2PCM before decoding them; adding to the mess.

Now...another small truth about SACD...things hardly ever say true DSD. One of the problems with 1-bit files is that you cannot edit them; you can do some simple stuff to them; but the usual amount of post-production done to a CD during the mastering phase cannot be done in the 1-bit domain; likewise, you cannot record sessions in 1-bit, mix in 1-bit, and export in 1-bit. There are two ways they've worked around this; one of them is using a DSD format with a higher bit-depth. However, those stations are very expensive; and there is still the issue of we live in a largely PCM world. So...they came up with something else; DXD.

So what is DXD? DXD is basically high-rate PCM; specifically, it's 24(maybe 32) bit PCM audio at a sampling rate of 352khz! Whenever you are dealing with stuff recorded in DSD; 99% of times it gets converted to DXD PCM for production. In fact, I was told by one guy that just about *every* SACD on the market; at some point in the chain; was PCM.

Whether any of this is enough to turn one off of the format depends on your personal preferences. I happen to like DSD as a format; it, in a way..is future proof. You can take one DSD source and convert it to a higher rate PCM while..in theory..enjoying some of the benefits of being a higher rate. There is also a slight fact that most ADC's actually capture audio to a 1-bit pulse stream before that is converted down to a PCM rate. I have seen *many* 24/192khz files that show dithering noise in the upper end of the spectrum; a tell-tale sign of conversion.

Basically, it all boils down to how many times do you want to push the audio through format conversions; and whether or not you think what the professionals are using, or even what you using; do an adequate job of it. Of course, there are higher rate 1-bit formats out there; like 5.6mhz DSD128 and I've even seen some 11.2mhz DSD256 devices floating around out there.

Now...funky facts about SACD.

SACD medium is, physically, a DVD. The main difference is that the discs incorporate pit modulation as a form of copy protection; and as a result...cannot be read by anything except a SACD player. That's why he only way to extract DSD from an SACD is using an older hacked PS3; it not only contains the drive capable of reading SACD...but the software hacks let you get the DSD audio out. But they are 4.7GB/layer, just like a DVD.

The format is actually 2.8mhz *per channel*; this means you need 5.6mbps transfer rate for a 2-channel stereo content. If you do the math, this means you need close to 15mbps to support a 5.1 stream; much higher than a DVD itself can physically transfer per second. To overcome this; there is a special lossless codec for DSD called DST (digital stream transfer); it manages to get really good compression (2.5:1 to 3:1) on the content. It's this DST stream that is decompressed in the player to obtain a 5.1 DSD stream. It is mandatory on 5.1 audio; it is optional on 2.0 mixes. Some hybrid discs use DST on stereo content, just for space constraints.

Of course, by that math; you only need 5.6mbps to store this audio; where as 24/192khz requires 9.2mbps...then again it's not really all that better than 24/96 when you take in to account the noise-shaping; which only requires 4.6mbps.

Unlike the audio CD, which is a set length; the amount of audio you can put on a SACD is only limited by how much space you have. When using DST compression; you can actually cram about 4 or 5 hours worth of music on a 4.7GB disc. As an example; Hotel California in 2.0 DSD consumes about 1.3 GB of data; after conversion to DST, it's under 600MB. But don't try to burn that to a CD-R and expect it to work; the SACD players know the difference in the mediums (or at least my Oppo does). Even when you burn your own SACD (after obtaining the software to convert to DSD, merging your DSD files in to one large one; and figuring out the authoring software), it still shows up as an invalid disc on your PC.

I think I've said just about everything I can on the subject.
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 Post subject: Re: DSD vs PCM - and a small truth about SACD
PostPosted: 28 Oct 2015, 03:46 
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To my understanding, some newer Oppos can read DSD 512 from usb and some older could read selfburnt sacd-rs
There is iFy which converts dsd from your pc to analogue (over a pcm containter as usb can't handle dsd directly).
Or Hdmi to a dsd direct dac. Hard to exactly find out which player and or receiver truly uses direct dsd dac.

For the moment i can drag an sacd iso onto foobar or audirvana on my mac mini via pcm conversion.
Or play normal Sacds on my Sony BDP S 790 or play it on the Marantz DV 6500.
The Sony can bitstream Sacd via hdmi. The Marantz only outputs via 6ch analogue. No clue if it converts to pcm first, but on the service manual i saw it uses one dac chip for both pcm as dsd :( .

As for the soundquality i rather prefer vinyl.
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 Post subject: Re: DSD vs PCM - and a small truth about SACD
PostPosted: 28 Oct 2015, 04:42 
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Also keep in mind that most if not all AV reciever do dsp in pcm. Even if you bitstream dsd into your reciever, it applies bass management, speaker delay, room correction/eq in pcm. You have to put the receiver into pure direct mode to bypass all processing(only certain models allow this) then the dsd bitstream is natively converted to analog. But now you are dealing with full range signal sent to your speakers and they aren't time aligned.

I think hdmi isn't very good for unpacked/uncompressed audio. The jitter is significantly higher than coax and optical. But coax and optical can't pass dsd.

I think it is best to track down one of those now defunct sony sacd players, 777es comes to mind. Afaik these have dsp that can work in dsd.

I use an oppo bdp-105, convert dsd internally to analog, connect to mcintosh mx-121 via balanced xlr, mx-121 in pure direct/dsp bypass mode which basically only does volume control in balanced analog domain, balanced xlr output from mx-121 into pass labs xvr1 active xover which is configured for linkwitz riley 24db at 75Hz, low pass outs goes to a pair of martin logan descent i subwoofers and high pass outputs goes to a McIntosh power amp then martin logan theos towers. I get the lowest jitter, least processed/colored sound this way.
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 Post subject: Re: DSD vs PCM - and a small truth about SACD
PostPosted: 01 Nov 2015, 03:56 
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substance wrote:
Also keep in mind that most if not all AV reciever do dsp in pcm. Even if you bitstream dsd into your reciever, it applies bass management, speaker delay, room correction/eq in pcm. You have to put the receiver into pure direct mode to bypass all processing(only certain models allow this) then the dsd bitstream is natively converted to analog. But now you are dealing with full range signal sent to your speakers and they aren't time aligned.


The Onkyo TX-NR515 has pure 1-bit. The last firmware update I applied (which is likely way outdated by now) added DSD support; I think it supports both DFF and DSF files either over network or off USB. It was a pain because my UPNP server would only accept DFF files...and a request to add DSF support got shot down. But it automatically turns off all sound processing when playing back 1-bit.

As far as signals not being time-aligned...that is not an issue if your speakers are placed properly in a room. Room correction is largely there to fix incorrect/bad room placements. Face it, have you ever seen someone who's not 100% serious about sound have the speakers set up properly? If you have proper placement, then the lack of time alignment from the receiver isn't an issue.

I did run Audyssey on mine, even for the crazy 3-channel setup I used to have...as it did do a nice job of "pushing the speakers back".

substance wrote:
I use an oppo bdp-105, convert dsd internally to analog, connect to mcintosh mx-121 via balanced xlr, mx-121 in pure direct/dsp bypass mode which basically only does volume control in balanced analog domain, balanced xlr output from mx-121 into pass labs xvr1 active xover which is configured for linkwitz riley 24db at 75Hz, low pass outs goes to a pair of martin logan descent i subwoofers and high pass outputs goes to a McIntosh power amp then martin logan theos towers. I get the lowest jitter, least processed/colored sound this way.


Yeah, you lost me at McIntosh. I worked on a number of McIntosh pieces. I failed to see why they would come with the price-tag they did; especially for later models. I worked on a power-amp that on the used market was going for $1200; it sounded pretty dead to me. Very sterile; it actually sucked some life out of the music. There was also nothing to speak of component wise in there; it was like you were paying $1000 for transformers and autoformers. I'm not a big fan of autoformers; bit of a waste of power. I mean..that sounds like an impressive setup; but I'll say this. I once put my Moto tube amps with my Yaqin tube preamp and USB DAC on a pair of DCM TimeFrame 600s up against a McIntosh system that was using tri-amped Mangepans. Double blind couldn't tell which was which; and my system cost a whole lot less than just a pair of Mangepans.

svwees wrote:
To my understanding, some newer Oppos can read DSD 512 from usb and some older could read selfburnt sacd-rs
There is iFy which converts dsd from your pc to analogue (over a pcm containter as usb can't handle dsd directly).
Or Hdmi to a dsd direct dac. Hard to exactly find out which player and or receiver truly uses direct dsd dac.


Some of the newer DACs that support 32/384 actually support direct DSD. You can push it over USB if you've got the right software/drivers, at least stereo; that'll fit fine in a USB 2.0 stream. I've yet to see a dac use HDMI input. But I have an older pre-blu-ray oppo with the modified firmware that plays SACD-R just fine. Before I got my Onkyo and playing 24/192 directly on it; I was authoring my 24/192 content to SACD to play it back.

svwees wrote:
For the moment i can drag an sacd iso onto foobar or audirvana on my mac mini via pcm conversion.
Or play normal Sacds on my Sony BDP S 790 or play it on the Marantz DV 6500.
The Sony can bitstream Sacd via hdmi. The Marantz only outputs via 6ch analogue. No clue if it converts to pcm first, but on the service manual i saw it uses one dac chip for both pcm as dsd :( .


Just because it uses the same DAC chip doesn't mean it doesn't do DSD. There are a number of systems that use a 1-bit DAC for everything and have extra chips for PCM conversion. Some of the Rockola jukeboxes I worked on worked this way; one chip converted to 1-bit, the other was the actual DAC.

svwees wrote:
As for the soundquality i rather prefer vinyl.


Vinyl is nice; but as I point out to people....the sound quality, as well as the sound you experience; depend heavily on too many variables. The LP barely has the frequency response and falls way short of the dynamic range. The only advantage is while vinyl has a more limited dynamic rage; the masters tend to have more dynamic range than loudness-war mastered stuff because you can't put very hard limited audio on an LP and expect it to work. As I've told people (and gotten flamed for by people who don't know); mastering is technically only required for making LPs; originally to cut the master disc prior to tape; and then to make the master discs when they started using tape. The idea that you have to master a release for any other medium is bogus; and partially done because mastering engineers somewhat convinced people it was required...plus a lot of mix engineers are deaf. When I mix stuff, I master it in the mix rather than mixing then mastering...you can often fix a lot more stuff a lot easier if you adjust it within the mix rather than post-mix.
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 Post subject: Re: DSD vs PCM - and a small truth about SACD
PostPosted: 01 Nov 2015, 04:41 
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You didn't understand me here. It's not the dac alone which needs to process the signal in native 1bit dsd. The entire pipeline needs to be dsd which doesn't exist. There is no dsp out there which will work in dsd. Soon as you turn on audyssey, your dsd is converted to pcm.

For dsd to hit your 1bit dacs without ever converting to pcm, you need to bypass %90 what's in your av processor. Bass management, delay time, audyssey, dts neo music and such will force your av processor to convert dsd to pcm because its dsp won't recognize dsd. You need to put your av reciever into pure direct mode which will pass dsd into dacs without any processing which means you lose your subwoofer, time alignment and audyssey. 2ch full range signal in, 2 ch full range signal out to your speakers but you get %100 dsd. Only older sony players have dsp which work in dsd, those can do bass management and time alignment.


You need all your speakers to be equally distanced from your seat in order to ignore time delay. Generally not the case for most home theaters.

Not liking McIntosh gear is your personal opinion and I respect that, also keep in mind that my McIntosh gear is several magnitudes of $1200. I wouldn't object if your blind testers found both systems sound equally impressive or choose the less expensive one over tbe more expensive but I find it surprising that they couldn't tell an electrostatic panel from a conventional cone driver.

To go back to sacds, I prefer them over vinyl in many cases. One thing I disagree, depending on the cartridge, vinyl can have greater frequency response than most digital media. Vinyl is a very expensive way of listening music, records are $25-35 a piece, and need a great setup to have good sound coming out of them.
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 Post subject: Re: DSD vs PCM - and a small truth about SACD
PostPosted: 01 Nov 2015, 05:18 
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dewdude wrote:
As I've told people (and gotten flamed for by people who don't know); mastering is technically only required for making LPs; originally to cut the master disc prior to tape; and then to make the master discs when they started using tape. The idea that you have to master a release for any other medium is bogus; and partially done because mastering engineers somewhat convinced people it was required...plus a lot of mix engineers are deaf. When I mix stuff, I master it in the mix rather than mixing then mastering...you can often fix a lot more stuff a lot easier if you adjust it within the mix rather than post-mix.


While you are right that mastering is technically needed for vinyl in order to get the pressing playable on every setup, there is a massive misunderstanding when it comes to mastering and mixing and what is needed and what is not.

Unless you're an audio engineer, or work very close to an artist in question, it's very likely that you haven't heard more than a single album or two that hasn't gone through a mastering stage. It's basically because pretty much every song ever released by a more or less established company (no matter if it's very small or in the same league as UMG) is mastered. I can guarantee that every single music album or song that every person on this forum has purchased has gone through the process, with no doubts whatsoever.

So why is mastering needed? The entire idea of mastering is to make the album, song or sound for that matter sound as good as possible in as many setups as possible. If you're using a format with extremely little to no influence on the stored audio, like CD, DVD-Audio, SACD etc. you don't have to observe the limitations of the format and for this reason, mastering a vinyl requires many years of experience.

And now I know some of you guys are thinking "But if the mix is good then you don't need to master it", but to tell you the truth, mastering and mixing are two very different occupations. Mixing is basically to create, while mastering is to refine a mix in order to make a finished product. Of course if the mix is very good, the changes won't need to be very drastic compared to the finished master but the process will still be done. And it is actually very common that even with today's extremely loud music that the final master is better than the original mix, even though many people may think it isn't. You can find examples though where the engineer in question has ruined the album completely, but it still doesn't make sense to think that mastering itself should be abandoned by the music industry. On the other hand I would like to se the EBU R.128 standard being used by more companies.

Then there's of course music that doesn't sound good no matter what you try to accomplish. That goes for nearly every album with Bruce Springsteen (with possible exception for Human Touch). But even so, a mastering process can make the bad sounding mixes sound somewhat more enjoyable.

So in conclusion, there's no doubt that an insane amount of our CDs have gone through a mastering process and sound in many cases "like the engineer wanted" even in an non optimal, simplistic setup. Saying that mastering isn't needed for music if the mix is decent is like saying that color correction isn't needed for films if they're shooting it with a good camera and good film stock.
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 Post subject: Re: DSD vs PCM - and a small truth about SACD
PostPosted: 02 Nov 2015, 04:20 
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Color correction sounds ok to me, color grading not. I hate that pruissian blue or green/sepiahue in many action movies.
As do i with mastering to death in music.
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 Post subject: Re: DSD vs PCM - and a small truth about SACD
PostPosted: 02 Nov 2015, 04:59 
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Do I have to repeat what I said? Mastering isn't only about compression of dynamic range. A whole lot more can be applied in the process. And of course color grading has to be done in films. The negative footage is pretty much never a reference for any feature film.

If anyone says that mastering should be improved then I have no objections whatsoever, but saying that mastering for post-vinyl formats would be "unnecessary" is just plain stupid.
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 Post subject: Re: DSD vs PCM - and a small truth about SACD
PostPosted: 02 Nov 2015, 05:33 
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Color grading is necessary because home video formats use inferior color spaces than film and digital cinema. Vast amount of colors wouldn't exist on the colorspace of home video medium. These need to be color graded to the nearest color or as in many cases, different colors are selected in their place. It is more difficult to color grade for LD and DVD as they have a much narrower color space. In theory, it is easier to color grade for HD and even easier to 4K. More tools are also available to the color grader and they can work in digital entirely. This obviously makes things easier but also encourages the color grader to be more enthusiastic which result in vastly altered look.
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 Post subject: Re: DSD vs PCM - and a small truth about SACD
PostPosted: 02 Nov 2015, 05:34 
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Well said Substance. :)
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