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Two reasonable DACs with a large PK Metric when tested with real music...

GXAlan

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I've previously shown that my SA-11s2 and Panasonic UB9000 sound similarly and measure similarly with DeltaWave.

In this post, I'm sharing some preliminary data showing
a) a measurable PK Metric difference between two reasonable DACs with real music
b) the DAC with a ~10 dB worse 1 kHz SINAD unit seems to be more accurate to the recording using PK Metric!

Please consider this preliminary data, since I am hoping that others will try to reproduce the results with their own gear and own music too. There is an excess of data presented to "show my work" since this is an unusual result.

The reasonable DACs? The Topping D90 MQA and Marantz SA-10. Both have been characterized by @amirm. The original D90 is still reference class by today's standards except for XLR phase inversion from convention, and the SA-10 does not measure as well, but to be fair, is a 2016-era product which was in development since 2014 or so and still "should" be good enough to threshold of audibility.

My 2018-production SA-10 does give slightly better PCM 1 kHz SINAD results than Amir's unit, though the test environment is different, and I got these results for DSD64. My Topping D90 gets to the -120dB 1kHz SINAD, so it's not defective either. I kept the D90 away from other transformers.

1679974293644.png



1679988364998.png



Test Track / Methodology
Concerto symphonique No. 4 in D minor, Op. 102: Scherzo from SFS0060 - Masterpieces in Miniature (Physical SACD) & DSD extract

I measured the DSD playback on the Topping D90 versus physical disc playback on the Marantz SA-10.

E1DA Cosmos ADC was running 32/176 to capture everything the SACD should reproduce. XLR balanced cables were used and L/R channel was verified each time before recording. For the SA-10, I played the disc, and then pressed the previous track button on the remote and then recorded using Audacity. For the D90, I used Foobar/ASIO4All to send a DSD stream to the D90. I confirmed that the LCD display indicated a 2.8MHz sampling rate, ensuring that my software configuration was valid for transmitting DSD. I pressed play in Foobar and then quickly used ALT-TAB and SHIFT-R to start the recording. When the 1 minute mark on the SACD player or the Foobar status bar was reached, I stopped the recording.

Control Experiments
This shows how repeatable the source devices are, and how repeatable the test environment is. The E1DA Cosmos ADC itself is subject to thermal effect. The PK Metric is super low, indicating the testing conditions are pretty good.

1679974847039.png
1679985899570.png



Comparison
Now I compare the Topping D90 as reference against the Marantz SA-10
- the phase is inverted on the D90; DeltaWave corrects for these
- DeltaWave will treat the "reference" as reference and then modify the comparison file. Since the SA-10 outputs a higher voltage than the D90 (2dB worth), DeltaWave will reduce the volume of the SA-10 recording to the D90. This might help the SA-10 by helping to reduce its inherent noise. I'm not sure. (@pkane?)

DeltaWave v2.0.8, 2023-03-27T20:43:12.1452921-07:00
Reference: D90 Repeat2.wav[?] 7892736 samples 176400Hz 32bits, stereo, MD5=00
Comparison: SA10 repeat 2.wav[?] 7905362 samples 176400Hz 32bits, stereo, MD5=00
Settings:
Gain:True, Remove DC:True
Non-linear Gain EQ:False Non-linear Phase EQ: False
EQ FFT Size:65536, EQ Frequency Cut: 0Hz - 0Hz, EQ Threshold: -500dB
Correct Non-linearity: False
Correct Drift:True, Precision:30, Subsample Align:True
Non-Linear drift Correction:False
Upsample:False, Window:Kaiser
Spectrum Window:Kaiser, Spectrum Size:32768
Spectrogram Window:Hann, Spectrogram Size:4096, Spectrogram Steps:2048
Filter Type:FIR, window:Kaiser, taps:262144, minimum phase=False
Dither:False bits=0
Trim Silence:True
Enable Simple Waveform Measurement: False

Discarding Reference: Start=6s, End=10s
Discarding Comparison: Start=6s, End=10s

Initial peak values Reference: -2.747dB Comparison: -0.791dB
Initial RMS values Reference: -25.778dB Comparison: -23.844dB

Null Depth=7.747dB
Trimming 0 samples at start and 0 samples at the end that are below -90.31dB level

Phase inverted
X-Correlation offset: 17670 samples
Trimming 0 samples at start and 0 samples at the end that are below -90.31dB level

Drift computation quality, #1: Excellent (1.01μs)


Trimmed 17963 samples ( 101.831066ms) front, 6976 samples ( 39.546485ms end)


Final peak values Reference: -2.747dB Comparison: -2.764dB
Final RMS values Reference: -25.83dB Comparison: -25.884dB

Gain= 2.049dB (1.2661x) DC=0 Phase offset=100.162876ms (17668.731 samples)
Difference (rms) = -44.88dB [-65.5dBA]
Correlated Null Depth=43.02dB [47.78dBA]
Clock drift: 12.64 ppm


Files are NOT a bit-perfect match (match=0.11%) at 16 bits
Files are NOT a bit-perfect match (match=0%) at 32 bits
Files match @ 50.0059% when reduced to 6.95 bits


---- Phase difference (full bandwidth): 15.2586814087706°
0-10kHz: 0.94°
0-20kHz: 0.81°
0-24kHz: 0.75°
Timing error (rms jitter): 2.2μs
PK Metric (step=400ms, overlap=50%):
RMS=-56.9dBr
Median=-59.2
Max=-49.2

99%: -50.56
75%: -55.79
50%: -59.16
25%: -61.21
1%: -66.19

gn=0.789857158565091, dc=0, dr=1.26390395685614E-05, of=17668.7312619653

DONE!

Signature: 5f180304410d4c7fb8d5a1775e8f2b5f

RMS of the difference of spectra: -103.878891018632dB
DF Metric (step=400ms, overlap=0%):
Median=-20.1dB
Max=-15.9dB Min=-31dB

1% > -29.41dB
10% > -25.18dB
25% > -22.79dB
50% > -20.13dB
75% > -18.97dB
90% > -17.65dB
99% > -6.78dB

Linearity 15.9bits @ 0.5dB error


Quality of the Match
I crop out the first 6 seconds and last 10 seconds of the recording since it throws DeltaWave off. Here are the files as-is, showing correct phase matching between the two recordings.

1679981565307.png


and the final match
1679981604413.png


We can zoom in and see how tight/good DeltaWave's level matching and alignment are. Blue is the Topping D90 and White is the Marantz SA-10. The increased noise and increased ultrasonics of the SA-10 are clearly seen in these ultra-zoomed-in views.

1679981664640.png

1679982339448.png


The aligned spectrum shows a slight roll-off in the bass for the SA-10 below 20Hz and the higher noise level in the high frequencies which does drop down into the audible range. The big difference is the higher ultrasonic noise of the Marantz SA-10.

1679981768898.png


Looked at a different way, we can see the delta of the Spectra. Here, you can see that the ultrasonics are really the main difference between the two files. No surprises yet. The spectra looks at the recording as a whole, not for specific portions of the music.

1679981820674.png


I don't know how best to interpret the phase, but I've shown it here. It seems like everything is pretty close to 0 degrees in the audible range, ignoring the bass I mentioned.

1679981879590.png



The important slide: PK Metric of -56.9 dBr rms and -48.4 dBFS for the transients that generate the biggest difference. -48.4 dBFS meets the PK metric threshold for audible.

This suggests that both sound similar but during that transient, it should be possible to hear a difference between the two.

(and a close-up of the spike at ~16 seconds shows that there's a lot going on in the audible frequencies. Note that I clip the first 6 seconds of the recording of you want to listen to the music yourself.

1679982841345.png


The PK Metric is a perceptually weighted value that tries to capture if a null comparison should be or should not be audible. Greater than -50 dB means that it's clearly audible. At -56.9 dBr, it's probably still really hard to hear the difference, but if you use the cool split window feature that @pkane put in, and hold CTRL and then move your mouse on the left to reach the area of the peak spikes around 15 seconds, you can see that the level there is -48.4 dBFS and in the audible frequency range.

So, here, we have a flagship DAC and a flagship SACD player that *should* get into the range of indistinguishable, but the PK Metric is showing us that there may actually be a difference.

Different, yes. But which is correct? Maybe, the Marantz!

For true digital comparisons, I took the DSD source file and then used the TASCAM Hi-Res Editor to convert the DSD 2.8MHz file into a 32-bit FP / 176 kHz PCM version. 176 kHz is how I made my E1DA recordings. I then redid the DeltaWave tests, using that Digital Source as the reference.

Referenced to the digital source file
Topping D90 gets a PK Metric of -73.3 dBr (rms)
Marantz SA-10 gets a PK Metric of -80.3 dBr (rms)

Let's look at the details because if anything, the Marantz performance is under-estimated.

DeltaWave v2.0.8, 2023-03-27T23:02:26.6131642-07:00
Reference: Master-Digital.wav[?] 7811807 samples 176400Hz 32bits, stereo, MD5=00
Comparison: SA10 repeat 2.wav[?] 7905362 samples 176400Hz 32bits, stereo, MD5=00
Settings:
Gain:True, Remove DC:True
Non-linear Gain EQ:False Non-linear Phase EQ: False
EQ FFT Size:65536, EQ Frequency Cut: 0Hz - 0Hz, EQ Threshold: -500dB
Correct Non-linearity: False
Correct Drift:True, Precision:30, Subsample Align:True
Non-Linear drift Correction:False
Upsample:False, Window:Kaiser
Spectrum Window:Kaiser, Spectrum Size:32768
Spectrogram Window:Hann, Spectrogram Size:4096, Spectrogram Steps:2048
Filter Type:FIR, window:Kaiser, taps:262144, minimum phase=False
Dither:False bits=0
Trim Silence:True
Enable Simple Waveform Measurement: False

Discarding Reference: Start=6s, End=10s
Discarding Comparison: Start=6s, End=10s

Initial peak values Reference: -0.862dB Comparison: -0.791dB
Initial RMS values Reference: -23.929dB Comparison: -23.844dB

Null Depth=13.749dB
Trimming 0 samples at start and 0 samples at the end that are below -90.31dB level

X-Correlation offset: -10531 samples
Trimming 0 samples at start and 0 samples at the end that are below -90.31dB level

Drift computation quality, #1: Excellent (0.81μs)


Trimmed 42353 samples ( 240.096372ms) front, 35254 samples ( 199.852608ms end)


Final peak values Reference: -0.862dB Comparison: -0.776dB
Final RMS values Reference: -23.902dB Comparison: -23.832dB

Gain= 0.0259dB (1.003x) DC=0 Phase offset=-59.704791ms (-10531.925 samples)
Difference (rms) = -40.55dB [-69.79dBA]
Correlated Null Depth=37.87dB [68.06dBA]
Clock drift: 15.35 ppm


Files are NOT a bit-perfect match (match=0.07%) at 16 bits
Files are NOT a bit-perfect match (match=0%) at 32 bits
Files match @ 50.0024% when reduced to 6.23 bits


---- Phase difference (full bandwidth): 5.48353378963904°
0-10kHz: 3.00°
0-20kHz: 3.84°
0-24kHz: 4.25°
Timing error (rms jitter): 15.8μs
PK Metric (step=400ms, overlap=50%):
RMS=-80.3dBr
Median=-83.7
Max=-64.5

99%: -70.77
75%: -80.38
50%: -83.73
25%: -86.47
1%: -93.91

gn=0.997023879532064, dc=-7.47465367171731E-08, dr=1.5353617707155E-05, of=-10531.9251125162

DONE!

Signature: 27331bba9f6fa02bd5743a3125da2397

RMS of the difference of spectra: -99.5984246212413dB
DF Metric (step=400ms, overlap=0%):
Median=-17.7dB
Max=-13.9dB Min=-28.3dB

1% > -27.01dB
10% > -23.16dB
25% > -20.04dB
50% > -17.68dB
75% > -16.28dB
90% > -15.09dB
99% > -5.65dB

Linearity 14.5bits @ 0.5dB error

Here are the RAW waveforms before matching
1679983447103.png


and then after matching super zoomed in
1679983749750.png


and the PK Metric is -80.3 dBr and you can see a good amount of time is spent below -80 dB.
1679983651331.png


The big spike at 35 seconds? It's obvious in the delta, but not obvious in the source recording.
1679983860461.png

1679983941742.png


The matched spectra shows the effect of the digital filter used by TASCAM. The slope in the 30 to 40 kHz range looks similar.
1679984309523.png


And here is the Topping D90

DeltaWave v2.0.8, 2023-03-27T23:19:19.0023470-07:00
Reference: Master-Digital.wav[?] 7811807 samples 176400Hz 32bits, stereo, MD5=00
Comparison: D90 Repeat2.wav[?] 7892736 samples 176400Hz 32bits, stereo, MD5=00
Settings:
Gain:True, Remove DC:True
Non-linear Gain EQ:False Non-linear Phase EQ: False
EQ FFT Size:65536, EQ Frequency Cut: 0Hz - 0Hz, EQ Threshold: -500dB
Correct Non-linearity: False
Correct Drift:True, Precision:30, Subsample Align:True
Non-Linear drift Correction:False
Upsample:False, Window:Kaiser
Spectrum Window:Kaiser, Spectrum Size:32768
Spectrogram Window:Hann, Spectrogram Size:4096, Spectrogram Steps:2048
Filter Type:FIR, window:Kaiser, taps:262144, minimum phase=False
Dither:False bits=0
Trim Silence:True
Enable Simple Waveform Measurement: False

Discarding Reference: Start=6s, End=10s
Discarding Comparison: Start=6s, End=10s

Initial peak values Reference: -0.862dB Comparison: -2.747dB
Initial RMS values Reference: -23.929dB Comparison: -25.778dB

Null Depth=14.57dB
Trimming 0 samples at start and 0 samples at the end that are below -90.31dB level

Phase inverted
X-Correlation offset: -28201 samples
Trimming 0 samples at start and 0 samples at the end that are below -90.31dB level

Drift computation quality, #1: Excellent (1.23μs)


Trimmed 24996 samples ( 141.70068ms) front, 15742 samples ( 89.240363ms end)


Final peak values Reference: -0.862dB Comparison: -0.877dB
Final RMS values Reference: -23.916dB Comparison: -23.925dB

Gain= -1.8904dB (0.8044x) DC=0 Phase offset=-159.868703ms (-28200.839 samples)
Difference (rms) = -51.06dB [-67.26dBA]
Correlated Null Depth=46.36dB [64.76dBA]
Clock drift: 2.78 ppm


Files are NOT a bit-perfect match (match=0.23%) at 16 bits
Files are NOT a bit-perfect match (match=0%) at 32 bits
Files match @ 50.0072% when reduced to 7.96 bits


---- Phase difference (full bandwidth): 5.84587571327181°
0-10kHz: 2.62°
0-20kHz: 4.38°
0-24kHz: 4.74°
Timing error (rms jitter): 9.4μs
PK Metric (step=400ms, overlap=50%):
RMS=-73.3dBr
Median=-76.2
Max=-63.3

99%: -65.62
75%: -73.15
50%: -76.16
25%: -78.12
1%: -81.09

gn=1.24313312684506, dc=1.46502821251829E-06, dr=2.78400992464591E-06, of=-28200.8391638284

DONE!

Signature: 06bd2487c490ccfe122634a8bd61ea15

RMS of the difference of spectra: -103.52586412889dB
DF Metric (step=400ms, overlap=0%):
Median=-28.5dB
Max=-24.4dB Min=-39.1dB

1% > -37.84dB
10% > -33.32dB
25% > -31.18dB
50% > -28.45dB
75% > -26.97dB
90% > -25.73dB
99% > -10.17dB

Linearity 18.6bits @ 0.5dB error

Original unmatched waveforms show that the Topping D90 has a phase inversion
1679984390231.png


Thankfully, DeltaWave identifies and corrects that. After matching, we see a well aligned waveform:
1679984741204.png


And silly zooming in gives us this, confirming a great match:
1679984854941.png


But the PK Metric is not as good at -73.3 dBr. In fact, it never really gets below -80 dB.
1679984881943.png


And you can see that the way that the Topping D90 processes DSD (white) results in a slightly lower ultrasonic noise than even the digitally source (blue).
1679985005962.png


Discussion
The biggest differences are in the ultrasonics. PK Metric will take this into account, but if the ultrasonics induce IMD in the audible range, it could make a difference. The Marantz does better when compared against the reference DSD digital source, so the Marantz is overall more accurate to the recording when using the perceptually weighted PK Metric. This argues against IMD being the reason for differences. The PK Metric really does show a lot of differences in the audible frequency range.

The E1DA is a great 1 kHz SINAD measurement device, but it's not so great of an ADC for recording music. It has a huge amount of DC, which DeltaWave corrects. It's also very thermally sensitive, so I may very well be seeing a glitch/defect in the way that the E1DA records things and variability in temperature. The E1DA has no buffer, so the input impedance doesn't reflect what you see with an actual pre-amp/integrated amp. I don't know how to control for this without spending more money than I really want to.

You can get false positives with null testing (suggesting that there is a difference between two recordings when there is none). I have seen this when DeltaWave makes an error in matching the waveforms and it's possible that trimming the audio files more precisely improve the match. That said, the silly zooming into the waveform shows that the matching is pretty good in my opinion.

I haven't really done listening tests. No good way to do ABX testing since I cannot level match closely enough. I could try a nearfield measurement of a speaker playing back the music, but then I'd be introducing the amplifier into the chain and I'd need a UMIK-2. All of this may still be academic, but it is interesting to see how the measurements and null testing surprises.
 
Last edited:

Koeitje

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I haven't read your post because I got too distracted by all the comparison graphs between the two devices having a different scale.
 
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GXAlan

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I haven't read your post because I got too distracted by all the comparison graphs between the two devices having a different scale.

That's intentional. This isn't ready for the general audience yet with a TLDR summary. I'm sure there's an error in my work somewhere, but I cannot find it yet.

If you don't read it carefully, you're bound to get distracted or misinterpret results. This isn't a standardized test like a 1 kHz SINAD or MultiTone 32. This is a rare case where the x-y of the graphs are not all that important -- it's just supporting data to show the quality of the DeltaWave matching along with a few other nuggets of data that may be interesting for the detectives trying to find an issue.

PK Metric is a mathematical, perceptually weighted way to compare two recordings.

Reference Digital File vs. recording
SA-10: -80.3 dBr (RMS)
D90: -73.3 dBr (RMS)

Conclusion: A recording from the SA-10 is closer to the digital source than a recording from the D90 (despite a worse 1 kHz SINAD).

Conclusion 2: Both of these would be audibly similar to a direct feed of the digital source to your brain.
---------------
SA-10 vs D90 with real music
-56.9 dBr (RMS)
-48.3 dBFS for the specific area where you see the peak.

Conclusion: The difference in sound signature between these two sources should be audible based upon PK Metric in certain portions of the music.

Conclusion: Even though the two units are inaudibly different from the source file, their difference must be in opposite “directions” (call it dawn versus dusk, East versus west) such that the two compared to each other sounds different.

TLDR conclusions:
1) Two DACs, one with 110 dB 1 kHz SINAD and one with 120 dB SINAD have different sound signatures based upon PK Metric
2) The DAC with the lower 1 kHz SINAD actually is closer to the source recording.

Both of those TLDR conclusions should be wrong. This should be a false positive for the comparison.

Potential for false positive all lies in how well DeltaWave is able to match the two waveforms. The graphs with ultra-zoomed in waveforms is to show you the quanta at 176 kHz. I have zoomed in to get to the point of jagged edges and visible differences between the blue/white. It's pretty darn good to me.

Potential for false positive is a limitation in the validity of the PK Metric calculation and ISO 226:2003.
It's possible that the PK Metric isn't downweighting something enough, leading to a false positive of differences.

But if we cannot find the error in testing, then it should convince another reader to try to reproduce the results. And if this is reproduced using this methodology, it's a game changer.

Again,
a) something that measures well on a 1 kHz SINAD is less accurate when null tested to music compared to something that measures less well. This gets into the, "how does it sound" vs. "how does it measure" debate. You can still measure it -- it just shows that we need more than the dashboard.

b) Two devices with 110 and 120 dB 1 kHz SINAD (rounded to 2 sig figs) have audible differences with music. In this comparison, the worse SINAD unit did better -- but it's more common that we see comparisons where the better measuring one is in fact better.

If so, 120 dB SINAD isn't enough to guarantee "end-game" electronics. We genuinely need even better electronics if we want to take transparency of music out of the question.

c) If ultrasonics are imparting a form of dither somehow, it gets interesting too when it comes to ultrasonics or signal processing

I picked the Marantz SA-10 because of this interview (in Japanese) which really talked about the energy spent in development as well as the comments about not reducing noise if it resulted in worse sound quality. D&M have APx555's and they clearly have the ability to make high SINAD products when they want. Here, they went not for measurements but sound.


At least based upon these tests, there is validity to their process. It's closer to the recording and the difference is audible.
 
Last edited:

Koeitje

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That's intentional. This isn't ready for the general audience yet with a TLDR summary.
So you just make it harder for everyone to read? Not really a good way to publish your findings.
 

JktHifi

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I wonder why sometimes the 44.1KHz or 48KHz sample rate music come from my DAC is better sounding than 96KHz or above. Specifically on loud volume music listening and I always listen to loudspeaker, not headphone. Seperation between different musical instruments can be heard clearly.
Is this because of the recording or the DAC/amp/speakers? It’s not a high-end system, normal budget.
 

MaxwellsEq

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Interesting. Thank you. Initially I wondered if there were very small variations in the mastering of the DSD layer than the PCM and that you are actually measuring mastering technology rather than DAC performance, but the matching is very close, so perhaps not.

I am naturally suspicious of ultrasonic noise and IMD in the audio band caused by it. You seem to have adjusted for this, but perhaps they is an effect which causes an artificially "better" result when its present?
 

pkane

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Reference Digital File vs. recording
SA-10: -80.3 dBr (RMS)
D90: -73.3 dBr (RMS)

Either one of these should produce a mostly inaudible difference judging by the numbers.

The dBr value is reporting the level of the perception-adjusted error metric relative to the actual signal level. In other words, the error is at -73dB below the music signal in one case, and -80dB in the other. Both are extremely unlikely to be audible.
 
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GXAlan

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Either one of these should produce a mostly inaudible difference judging by the numbers.

The dBr value is reporting the level of the perception-adjusted error metric relative to the actual signal level. In other words, the error is at -73dB below the music signal in one case, and -80dB in the other. Both are extremely unlikely to be audible.

Yeah, which makes sense because both are high-performance DACs.

But, the academic part is that if my measurements are right, the SA-10 is imperceptibly "east" of target and the D90 is imperceptibly "west" of target and when you compare the SA-10 against the D90, the difference between the two DACs is detectable (-56.9 dBR with the spikes at -48.4 dBFS?).


index.php



Your match is so good that you can get to the silly levels of zoom to see the matching.
index.php


The white has more noise, but reference to the DSD file, it actually matches overall better?
 

Snoopy

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How I hate DSD and the plague I think it is on the audio world. Can we not just ignore DSD? If not, then why not?


what's your problem with DSD exactly? there are some pretty good masters that are only available on SACD. sure if you don't care for certain masters you might just listen to what's currently on streaming services. but I would rather see MQA never getting used again.
 

pkane

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what's your problem with DSD exactly? there are some pretty good masters that are only available on SACD. sure if you don't care for certain masters you might just listen to what's currently on streaming services. but I would rather see MQA never getting used again.

Not to speak for him, but I think @Blumlein 88 has a philosophical objection to the format, not the content that's already created/produced in DSD. I generally agree with this. While DSD has a few minor benefits over PCM, it also has some significant downsides that are well known and have been discussed many times before on ASR.
 

Snoopy

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Not to speak for him, but I think @Blumlein 88 has a philosophical objection to the format, not the content that's already created/produced in DSD. I generally agree with this. While DSD has a few minor benefits over PCM, it also has some significant downsides that are well known and have been discussed many times before on ASR.

But the downsides are not really a "problem" Pretty much every half-decent DAC / streamer supports DoP or is able to play DSD natively.

Ripping and playing SACDs or DSF files is not rocket since either and even freeware players support it.

And for people that don't care about certain masters there is probably for everything a pcm version. :)

It's just another physical Format that failed but is still around because enough people seem to like it.

Much worse are MQA-CDS that are only available in that format.
 
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GXAlan

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Not to speak for him, but I think @Blumlein 88 has a philosophical objection to the format, not the content that's already created/produced in DSD. I generally agree with this. While DSD has a few minor benefits over PCM, it also has some significant downsides that are well known and have been discussed many times before on ASR.

1) the biggest one, which is easily seen here, is the high ultrasonic noise.
2) the second biggest thing is all of the limitations in processing in digital audio workstation software

The strength, which could be corrected by Blu-Ray, is that SACD is still the best way to have physical media with multichannel (which is rarely used today) and physical media that generally assures attention has been put toward the mastering quality.

To my knowledge, you pretty just have new classical music on SACD.

And to be fair, at the time DSD was invented (not the 1954 development of delta sigma but DSD itself) in 1993/1994, the PCM DACs were nowhere as good as the tech we had today.
It's just another physical Format that failed but is still around because enough people seem to like it.
No different from LP vinyls…

Or maybe even more accurate, there is enough inertia. Deutsche Grammophon has been putting out pure audio Blu-Rays. 2L as well. Blu-Ray makes much more sense as a physical media for video and audio. You can have longer tracks than CD and you can do multi channel and there are far more Blu-Ray players out there. You just haven’t gotten enough critical mass like you do with SACD. I wonder if SACD production is cheaper than Blu-Ray since it’s really just DVD based.
 

Snoopy

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1) the biggest one, which is easily seen here, is the high ultrasonic noise.
2) the second biggest thing is all of the limitations in processing in digital audio workstation software

The strength, which could be corrected by Blu-Ray, is that SACD is still the best way to have physical media with multichannel (which is rarely used today) and physical media that generally assures attention has been put toward the mastering quality.

To my knowledge, you pretty just have new classical music on SACD.

And to be fair, at the time DSD was invented (not the 1954 development of delta sigma but DSD itself) in 1993/1994, the PCM DACs were nowhere as good as the tech we had today.

No different from LP vinyls…

Or maybe even more accurate, there is enough inertia. Deutsche Grammophon has been putting out pure audio Blu-Rays. 2L as well. Blu-Ray makes much more sense as a physical media for video and audio. You can have longer tracks than CD and you can do multi channel and there are far more Blu-Ray players out there. You just haven’t gotten enough critical mass like you do with SACD. I wonder if SACD production is cheaper than Blu-Ray since it’s really just DVD based.

There is so much more on SACD .. sure there is lots of classical stuff but there is also really good and modern jazz recordings.

And obviously rolling stones, Bob Dylan, Santana and so many more artists.

Blu-ray audio is great as well but some good releases are already OOP or are limited editions to begin with at questionable prices.

DVD-Audio could have replaced CD and SACD before Blu-ray audio was a thing but SACD is still kicking around:)

The ultra sonic noise gets cut-off anyway by pretty much every SACD player ever made.
 
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GXAlan

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There is so much more on SACD .. sure there is lots of classical stuff but there is also really good and modern jazz recordings.
Blu-ray audio is great as well but some good releases are already OOP or are limited editions to begin with at questionable prices.

That’s another great point. SACDs are still more reasonably priced which may be the fact that it is built on DVD pressing tools that probably don’t see a lot of use nowadays.


The ultra sonic noise gets cut-off anyway by pretty much every SACD player ever made.

As you can see in the SA-10 measurements by me and Amir, the SA-10 still leaks a lot of ultrasonic noise. That said, it sounds great likely because the ultrasonic noise is cut off by the speaker too.
 

pkane

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But the downsides are not really a "problem" Pretty much every half-decent DAC / streamer supports DoP or is able to play DSD natively.

Ripping and playing SACDs or DSF files is not rocket since either and even freeware players support it.

And for people that don't care about certain masters there is probably for everything a pcm version. :)

It's just another physical Format that failed but is still around because enough people seem to like it.

Much worse are MQA-CDS that are only available in that format.
There is so much more on SACD .. sure there is lots of classical stuff but there is also really good and modern jazz recordings.

And obviously rolling stones, Bob Dylan, Santana and so many more artists.

Blu-ray audio is great as well but some good releases are already OOP or are limited editions to begin with at questionable prices.

DVD-Audio could have replaced CD and SACD before Blu-ray audio was a thing but SACD is still kicking around:)

The ultra sonic noise gets cut-off anyway by pretty much every SACD player ever made.
Other than being bulky, inefficient, noisy and not subject to EQ or any sort of mixing, it’s otherwise almost as good as PCM ;)
 
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GXAlan

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Other than being bulky, inefficient, noisy and not subject to EQ or any sort of mixing, it’s otherwise almost as good as PCM ;)

Here’s what I wonder though… for the actually DSD recorded and mastered music, maybe the the INABILITY to throw dynamic range compressors and all sorts of fancy processing onto the mix makes the resulting sound better. Many photographers get better results on analog film compared to digital *because* they are more thoughtful and deliberate in their photo making when each exposure costs money.

The assumption is that you have a skilled recording engineer — but an unskilled recording engineer may be protected against bad decisions when working in DSD.
 

Blumlein 88

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what's your problem with DSD exactly? there are some pretty good masters that are only available on SACD. sure if you don't care for certain masters you might just listen to what's currently on streaming services. but I would rather see MQA never getting used again.
I don't see DSD as much better than MQA. 96/24 can do everything DSD can do and better without the downsides. I am not against music released in the format. Nor anyway we can get better masters. However just having DSD as an alternative complicates that for no real benefit. The discs that hold SACD can hold multi-channel 96/24 with much less data. I even understand DSD for a very brief period had some capability beyond redbook CD, but that period didn't last long before DVD audio discs were available and now with downloads and streams there is even less reason because it is so inefficient with the data. I don't see the need for more than 96 khz myself, but at least even DXD is a high rate PCM variant.
 

Blumlein 88

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Here’s what I wonder though… for the actually DSD recorded and mastered music, maybe the the INABILITY to throw dynamic range compressors and all sorts of fancy processing onto the mix makes the resulting sound better. Many photographers get better results on analog film compared to digital *because* they are more thoughtful and deliberate in their photo making when each exposure costs money.

The assumption is that you have a skilled recording engineer — but an unskilled recording engineer may be protected against bad decisions when working in DSD.
Yes and no. All it takes is discipline or a different recording philosophy to do the same with any PCM. The other part is some processing can occur over analog devices prior to putting it to DSD. That happens quite a lot and I think even 2L does a little of that on their recordings if my memory isn't wrong. I don't equate it with film photography or even recording with reel to reel tape prior to making it digital.
 
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