DSD comparison to PCM.

LynnOlson

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Feb 22, 2013
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Well, it's the yardstick that matters, doesn't it? If the criterion is movie soundtracks, that's a completely different thing. Since soundtracks are completely artificial confections, there's really no reference at all, unless you work at Skywalker Ranch in California, and have ready access to their theater.

My yardstick is music coming from physical instruments recorded in a real acoustic space - I listen to plenty of German techno and various styles of world-music electronica, but that only seems to benefit from a system tuned to acoustical music. But that puts me in a small minority of audiophiles these days, and even further from the HT enthusiasts. Right off the bat I have to warn PFO readers that they probably won't agree with me, on musical tastes alone.

A musically tuned HT system has an interesting effect on movie soundtracks: no, it doesn't sound like a movie theater, but the symphonic parts of the soundtrack come to the fore, which intensifies the emotional parts of the movie. The explosions and other completely artificial parts of the soundtrack aren't as harsh, raucous, and irritating as the theater experience, but that's a plus, not a negative. I don't need to hear the breakup of a titanium diaphragm that's been digitally boosted by 10 dB in the 8~15 kHz region and pumped through a 1-kilowatt Class AB/G amplifier; no thank you.

Background explanation: Large-format (4") aluminum compression-driver diaphragms have their first breakup in the 7~9 kHz region; however, aluminum went out of use in the late Seventies due to much higher playback levels creating problems with fatigue cracking and a high rate of failures in the field. Aluminum was replaced by titanium, which is fatigue-resistant but unfortunately has the first breakup an octave lower, around 4 kHz (for large-format diaphragms). Around this time, Altec multicell horns were replaced by constant-directivity horns like the Altec Mantaray and the JBL Bi-Radial, which require substantial (around 10 to 15 dB) of HF boost to deliver flat response at the listener position. Amplifier powers increased from the 60 to 100 watts in the vacuum-tube days to 300 watts by the Seventies to multi-kilowatt levels today. (During the same time period, the source changed from discrete six-track magnetic to mono optical, a brief period of stereo optical with matrixed surround, to the discrete multichannel lossy-compression digital used today.)

Net result: When you go the theater, you are listening to titanium-diaphragm compression drivers that are driven very deep into the breakup region by heavy equalization and powerful amplifiers. The ideal solution is replacing titanium with beryllium, which is free of breakup until you reach 15 kHz or higher, and more durable as well. But ... large-format beryllium diaphragms cost around $700 to $1500 each, while titanium is around $150 each. Digital projectors and 3D are something you can see on the screen and advertise, but nobody cares which diaphragms are used. The conversion to beryllium, if it happens at all, will probably take a while.

Further note: No, digital equalization cannot undo diaphragm breakup, except in the gross sense of offsetting power loss through the addition of more driving power. The active-feedback systems used in advanced subwoofers only work as long as the loudspeaker diaphragm moves as single piston; once it starts to break up, the feedback cannot correct for it, and may instead go unstable. (You don't want to be around when that happens.)
 
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opus111

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I'm not a fan of $1 opamps and 30-cent electrolytics in a product that sells for more than $1500; the Marantz AV-8003 (and MM8003) were built in China, so it's not like the labor was expensive.

Your pricing is a tad quaint - at retail I get NE5532s for about $0.15 so you can be sure the manufacturers pay substantially less than this. As for electrolytics, the price of them just gets lost in the noise, a 100uF is certainly under $0.01, again retail. China's no longer the go-to place for cheap labour since in recent years the workers have had fairly large pay hikes. Manufacturers I guess will be looking at going (if they've not moved already) to Vietnam, Indonesia - perhaps in future Burma will become the new 'sweatshop'.
 

NorthStar

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...

P.S. If you are listening to DSD through Audyssey, it ain't DSD no more. Audyssey is a PCM signal processing system. Whether it runs at 24/88.2, 24/96, or 24/192 depends on how much processing power is available in the HT pre-pro. Conversion of DSD to non-integer speeds like 96 or 192 kHz, although simplifying the signal path for movies (which are based on 48 and 96 kHz sample rates), does no favors for DSD, which is based on multiples of 44.1 kHz.

Same story for bass management and user-selectable loudspeaker delay. These are PCM processes, and in a HT product, quite probably running at 96 or 192 kHz speeds. You'd have to look for HF artifacts on a spectrum analyzer to find out the effective speeds of the internal PCM signal processing system.

---- Ah, you added a postscript. :b

I am not an expert on the domain, but I can add Audyssey on top of DSD from SACDs (through the digital HDMI connections).

And Audyssey is also available for digital bass management, and when playing hi-res audio (lossless) soundtracks from DTS-HD MA and Dolby TrueHD (again, from the digital HDMI connections).

I don't know exactly the computational transformation that is taking place digitally, but what I got is what my pre/pro shows and what my ears hear.

I believe you. ...I bet it's full of digital artifacts somewhere down the audio signal paths.
...Are we slave to the PCM world?
 

NorthStar

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Music recordings, or Movie audio soundtracks; none better none worst. ...Both prefabricated, or both live with the full orchestra.
Some music multichannel Hybrid SACDs sound magnificent, so are some Blu-ray movies with multichannel music from the world's best classical orchestras.
...All hi-res audio anyway, and in both cases, of immense beauty (auditory).
DSD (SACD), DTS-HD MA, Dolby TrueHD, and LPCM uncompressed.
...High bits (20 to 32, and even 64 and 128 Bits), and high frequencies (48 to 768 kHz - with 48, 88, 96, 176, and 192 kHz being the average).

* In reply to post #541.
 

LynnOlson

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Feb 22, 2013
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Based on the in-thread comments posted by professionals who use DSD, at the highest level where almost unlimited amounts of money and processing power are available, signal processing involving delays, equalization, reverb synthesis, compression, etc. involves a conversion to DSD-wide (Sonoma) or DXD (Pyramix). From a signal processing perspective, both are PCM.

We probably won't see this level of performance in home theater equipment for a while yet, although who knows, with enough demand from multichannel enthusiasts, it could happen. I'm a fan of multichannel music myself, but doing it at the highest level (a stack of three Playback Designs, Phasure, TotalDAC, etc. converters) is insanely expensive. You could probably buy a pretty nice house for what a top-level multichannel system would cost. Unlike the house, the digital portions of the system would not retain its value for long ... give it ten years, and most of your money would be gone.

Thanks, Opus111, for the update on parts pricing. You're a lot closer to the manufacturing powerhouse than the little guys in the high-end business in North America ... we get the privilege of paying high prices from distributors who have almost no interest in fulfilling small orders. I suspect that European manufacturers probably pay the highest parts-cost pricing of all.
 
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opus111

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Thanks, Opus111, for the update on parts pricing. You're a lot closer to the manufacturing powerhouse than the little guys in the high-end business in North America ... we get the privilege of paying high prices from distributors who have almost no interest in fulfilling small orders. I suspect that European manufacturers probably pay the highest parts-cost pricing of all.

You're welcome - it was a real eye-opener for me coming to China from working as a design engineer in UK to see what the real prices were for electronics parts. Up until then, I only had catalogues like Farnell's to go on, at the highest price breaks. It was a source of frustration to me when designing that I had no real way to ensure I was doing the most cost-effective job. There was one time when doing active speakers where I asked a Taiwanese colleague to buy tubes of chipamps and ship them in for us because we could not get anything approaching a decent quote from the UK distributors. The business of electronics component distribution in Europe would indeed be something worthy of study, from an economic perspective. I do rather suspect widespread price-fixing would be uncovered.
 

LL21

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My yardstick is music coming from physical instruments recorded in a real acoustic space - I listen to plenty of German techno and various styles of world-music electronica, but that only seems to benefit from a system tuned to acoustical music. But that puts me in a small minority of audiophiles these days...

Totally agree...use Clapton Unplugged as one of my references, Amos Lee, a mix of several classical recordings from Reference Recordings, soundtracks...but also German deep house (Ame, for example)...and i find Ame takes on a very very different human quality that is absolutely great. [when there are voices], the voices no longer are electronica words, but it is clear a human being stood in front of a mike to do some funky recording.
 

LynnOlson

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Feb 22, 2013
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----
...Are we slave to the PCM world?

In the time domain, DSD looks like little clouds of noise, with an average cloud-size about 714 nanoseconds. Signal modulation is carried out by varying the size of these clouds; on any given group of samples, it's going to be very difficult to discern how this clump of noise is any different than another clump of noise.

PCM, by contrast, is convertible to an exact number for every single sample. The magnitude of that sample, unlike DSD, does not need to look to its neighbors to figure out how big it is; it's already precisely specified in the PCM code itself.

DSD is somewhat akin to Frequency Modulation, or its close cousin, Phase Modulation. FM, of course, is used for FM radio, and the sound carrier for analog television. FM is also notorious for not tolerating signal manipulation; ripples in the frequency response, or more importantly, variations in group delay vs frequency, translate into outright distortion when demodulated back to analog. SECAM color television, which uses an FM carrier for the chroma information, is very difficult to work with in the studio; my understanding is that SECAM territories ended up using PAL in the studio, and converting to SECAM at the transmitter.

Sony has plenty of experience working with NTSC, PAL, and SECAM broadcast regions, since they made a big push into professional broadcasting gear as far back as the early Seventies, when they challenged Ampex for domination in the VTR world ... and beat Ampex at their own game. One side effect of their broadcast work is they made some of the earliest time-base correctors, which allows a helical VTR with an inherently jittery time-base to be accurately phase-locked to network timing standards.

A TBC is basically a FIFO buffer that uses precise external sync (from the network) to clock out the contents of the buffer and an over/under sensor to speed up or slow down the motors of the VTR. The digital converter for the TBC is also interesting: NTSC analog video is converted to 8-bit digital at 14.32Mhz (4x the 3.58MHz chroma frequency used in the Americas and Japan). Hey ... that sounds like DSD-wide! Yup. Pretty much like digital video, just not as fast.

Which gets us back to DSD-narrow. It's an OK way to capture analog in the digital domain, but actually doing anything with the digital signal (aside from storing it) is close to impossible. Convert to a form of PCM, which DSD-wide is, and you can do what you want. Although reducing from 8-bits back to 1-bit (discarding all of the smaller bits) is intellectually unattractive, nothing has been lost compared to the DSD-narrow 1-bit original.

You do have to watch for buildup of ultrasonic noise, though. Remember, for DSD-narrow the ultrasonic noise is the carrier. Every conversion in and out of the 1-bit domain carries a noise penalty; that's probably the reason that Scarlet Book specifies no more than 50% modulation for the analog signal going into DSD-narrow, in order to prevent clipping from added noise over several conversion steps.
 
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LL21

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Gents,

I would appreciate your technical guidance on my Zanden digital. I often speak with Zanden Audio about their equipment...and recently suggested they return to their digital after issuing the new 3100 preamp, 1300 phono stage and 6000 and i think 6100 amps. Since their last upgrade was a few years ago, i asked them if they thought enough progress had been made they might consider designing a 'Zanden Ultimate' to push the digital boundaries further.

Can anyone take a quick peek at the current Zanden 4-box given the techno reviews herein on PCM and NOS...and provide any guidance about what YOU might consider ways to improve. The Zanden currently uses TDA1541A, NOS via 7308 output with 2 6CA4 and 6X4 tubes, and is connected to its Transport via i2s connection...where the Transport also has its own separate PSU with 6 separate transformers, and the Transport itself is built with alternating layers of thick acrylic then aluminum, then acrylic...etc. All components inside have a form of RFI/EMI shielding around them via this white shielding which is apparently similar to very thick aluminum.

Would love to get some thoughts on where the current Zanden 4-box could possibly be further improved from the clearly knowledable people here...happy to forward these on as I (and a few other owners) encourage them to revisit their digital. I have, for example, suggetsed they consider 2 separate topologies inside...one for redbook where i strongly suspect they will stick with TDA 1541A, and one for DSD/hi-res. Thanks for any guidance.
 

LynnOlson

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Feb 22, 2013
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Based as it is on the TDA1541 architecture, I can't see how anything could be much different. You can quibble about the 7308 versus other tubes, but that's pretty much a subjective decision.

I've had good luck with Pure Music converting 128fs DSD to 24/88.2 PCM ... what I heard sounded like very, very good DSD, despite the fact I what was actually listening to was a pure ladder converter with no noise-shaping at all. If the Zanden accepts an external USB input, or you're OK with S/PDIF -> USB adapters (I've heard the JKenny unit is one of the best, and he's right here on the forum), you can always try software conversion from DSD to high-quality PCM and draw your own conclusions.

You might be surprised how good it sounds; I now think one of the reasons DSD -> PCM conversion has gotten a bad reputation is that the cheaper delta-sigma converters don't do a very good job of it. Same story for upsampling; high-quality software conversion seems to sound a lot better than old-school hardware-based oversampling, which was a pretty crude process.
 

opus111

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Would love to get some thoughts on where the current Zanden 4-box could possibly be further improved from the clearly knowledable people here...happy to forward these on as I (and a few other owners) encourage them to revisit their digital.

Based on what JA measured with his 2nd review sample there could be considerable improvements to jitter (though I'm not sure it would be particularly audible) and, more significantly they could flatten out the NOS HF droop which I've heard as a definitive improvement on my NOS DACs.

http://www.stereophile.com/content/...emium-cd-transport-second-sample-measurements

The image filtering could be substantially improved - eyeballing the IMD spectrum (fig6), the first image of the 20kHz tone at 24.1k looks to have practically zero attenuation. Removing these ultrasonic frequencies makes the amp's job easier, reducing the IMD generated.

I have, for example, suggetsed they consider 2 separate topologies inside...one for redbook where i strongly suspect they will stick with TDA 1541A, and one for DSD/hi-res. Thanks for any guidance.

They could employ the PCM1704 for hi-res, along with switchable steep passive filtering according to sample rate.
 

Orb

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A just as important factor IMO Opus is the 0dBFS figure, which if one listens to modern pop/some jazz/etc recordings can hit that limit due to loudness-compression.
This is why I feel having a 0dBFS and say -10dBFS to -20dBFS is useful as it can throw up some quirks in the hardware/architecture-implementation.

Cheers
Orb
 

opus111

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I used to think that but these days down play the 0dBfs figure, its heavily overrated. Music spends approaching zero percent of its time up there. -30 to -40dBfs is where the measurements matter for me nowadays :) Interestingly that's exactly where modern S-D converters suck most, as they have 6bit DACs.
 

Orb

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You would be surprised Opus how much music has content at -10dBFS to 0dBFS.
But this depends to what one listens to, although I have noticed it for some modern jazz music as well.
Average music-recordings (not talking about a small selection of excellent) has it anywhere from 0dBFS to -20dBFS for quite a fair bit of a recording.

In the past and for older recordings this was never a problem, but the "loudness wars" syndrome really does exist and I am seeing it even for "classical lite" type recordings and as mentioned can be seen for some modern jazz and others.
It is a subject I have followed closely for awhile and blogs from various studio engineers.
Cheers
Orb
 

Orb

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Yeah I take your point but its not what I call 'music' when its spending most of its time above -10dBfs:) My diet consists mainly of classical, a little jazz and world, plus the odd movie soundtrack.

Yeah I can understand the context and where you were coming from, but hey my secret guilty pleasure is Bat out of Hell (seriously compressed) along with some rock-pop albums :)
Movie soundtracks seem to be the best-most consistent modern "general" recordings these days and interestingly outside the loudness wars.
And like you I also listen to a lot of classical, and classical lite that seems to now be suffering a little of the loudness wars syndrome.

Cheers
Orb
 

LL21

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Based as it is on the TDA1541 architecture, I can't see how anything could be much different. You can quibble about the 7308 versus other tubes, but that's pretty much a subjective decision.

I've had good luck with Pure Music converting 128fs DSD to 24/88.2 PCM ... what I heard sounded like very, very good DSD, despite the fact I what was actually listening to was a pure ladder converter with no noise-shaping at all. If the Zanden accepts an external USB input, or you're OK with S/PDIF -> USB adapters (I've heard the JKenny unit is one of the best, and he's right here on the forum), you can always try software conversion from DSD to high-quality PCM and draw your own conclusions.

You might be surprised how good it sounds; I now think one of the reasons DSD -> PCM conversion has gotten a bad reputation is that the cheaper delta-sigma converters don't do a very good job of it. Same story for upsampling; high-quality software conversion seems to sound a lot better than old-school hardware-based oversampling, which was a pretty crude process.

Based on what JA measured with his 2nd review sample there could be considerable improvements to jitter (though I'm not sure it would be particularly audible) and, more significantly they could flatten out the NOS HF droop which I've heard as a definitive improvement on my NOS DACs.

http://www.stereophile.com/content/...emium-cd-transport-second-sample-measurements

The image filtering could be substantially improved - eyeballing the IMD spectrum (fig6), the first image of the 20kHz tone at 24.1k looks to have practically zero attenuation. Removing these ultrasonic frequencies makes the amp's job easier, reducing the IMD generated. They could employ the PCM1704 for hi-res, along with switchable steep passive filtering according to sample rate.

Thanks, Gents. Earlier, Lynn you had talked about Op/Amp or IV or something you felt you'd not seen before...i could not quite follow. Opus - i take your point on jitter, and most people always felt the i2s was supposed to help...and those measurements are a surprise. As for droop, yes, i am sure Yamada San is aware of that. I wonder if that will be addressed in the future.
 

MarinJim

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Hi Kal:

Thanks very much for the input. I too have no issues with solid state, even though I lean toward tubes.

I just combine the two.
 

Orb

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Yeah I take your point but its not what I call 'music' when its spending most of its time above -10dBfs:) My diet consists mainly of classical, a little jazz and world, plus the odd movie soundtrack.

Just to add a modern example of a "lite" genre recording.
High rez version of Diana Krall's Glad Rag Doll has most of its spectra content up to 1khz between -10dBFS and -30dBFS, this is from an artist who is recognised to having great talent and fits closer to the audiophile niche.

I did emphasise 0dBFS, but should be noted my concern in that post also for up to -20dBFS, which shows some performance considerations for certain DAC architectures and implementation.
Cheers
Orb
 

Kal Rubinson

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I am not an expert on the domain, but I can add Audyssey on top of DSD from SACDs (through the digital HDMI connections).
...............................
...Are we slave to the PCM world?
Lynn is correct and this is, or should be, common knowledge. It does not mean that one cannot choose either to listen to DSD uncompromised (except by room effects) or with bass management/EQ via a PCM conversion. Your move, of course.
 

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