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There are none. You can’t process DSD. For any operation other than a simple cut you must convert to multibit first. So any software for mastering DSD will be a lot more resource intensive. Why bother?
On my return home, I will order the SACD at HMV Japan.
I have my "secret" method of extracting the intact DSD layer into DSF (DSD64 2.8 MHz 1 bit) files using "specific version model" of Sony PlayStation. Of course I can rip the CD layer in very accurate manner by using dBpoweramp CD Ripper.
Interesting thread and conclusions. I have a question though - many professional studios use DSD for recording. Merging has an entire line of DSD capable processors. What is the advantage of a DSD workflow for professional studios? And if it's good for the pros, why isn't it good for us?
I don’t think DSD is used for recording that much. Most modern music is done with PCM or DXD which is PCM. There is probably no advantage to DSD in the present day (as opposed to when DSD was developed in 1995. At that time, the best PCM DAC was the PCM1702 which just had a THD+N of -96 dB.
So I think this is the first time I have seen someone in the modern day (2023) advocate for DSD recording with a rationale beyond “sounds good”.
Even being able to ABX the CD and DSD layer, if you want the ASMR effect, the CD is actually better!
I see. I actually have a contact from Merging (because I bought a Merging DAC from him). At the time, it was much earlier in my audio journey and I was sold on the idea of DSD because "it's what the pros use". Since then, I have discovered that convolution in DSD is possible but it brings my very powerful PC to its knees, and I couldn't hear much of a difference between DSD and PCM anyway. So I started wondering why the heck pros use it if it's so inconvenient and the differences are so hard to hear. Choosing DSD actually cost me money, and not just on the DAC. I had a CPU die prematurely from subjecting it to temperatures >85C due to the heavy duty cycle of DSD. I rebuilt the computer with a very powerful CPU capable of DSD convolution, but decided to do all my processing in PCM. I now have a ridiculously powerful PC with an expensive quiet cooling solution which is barely ticking along at 5% CPU usage when convolving in PCM!
Hence the question, whether there is any advantage to studios for using DSD. This thread really piqued my interest and made me wonder if I should go back to DSD.
However, the fact that Merging are still in business selling a lot of Anubis, Hapi, and Horus interfaces and have not gone under suggests that there is market demand for DSD.
Even though I am interested in spectral comparison between native DSD layer and the CD layer of that SACD, nowadays I dare seldom listen to native DSD sound with DSD-native capable DACs. You would please refer to my specific post on my project thread for the details; - Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing:#532
Audio engineers are as prone to remarkable claims about formats, the extent of human hearing, and such, as us idiots at the receiving end of their efforts. Rupert Neve believed human hearing is affected by 100kHz signal, and he has a following among engineers. We should not be surprised by such or expect rationality to exist in any part of the audio industry.
There might just be a case that the noise shaping in a DSD ADC could result in less noise when recording than a PCM ADC, but I wouldn't buy that without seeing proof.
Also in sense vice versa...DSD was literally conceived as an archival format designed to be downconverted to multiples of CD sample rate for consumer products.
Though I'm skeptical that's the reason any studios use it today. (Indeed, years ago Emil Berliner/DG compared DSD to hi rez PCM and chose the latter for its archiving)
Files are NOT a bit-perfect match (match=0.28%) at 16 bits
Files are NOT a bit-perfect match (match=0%) at 32 bits
Files match @ 49.7006% when reduced to 8.31 bits
So I decided to edit the recording of the DSD layer and amplify it by 4.06 dB to match the PEAKS.
Test #1: ABX Matched Peaks (p = 0.0021)
Because the E1DA has a minimum 1.7V sensitivity, I had to really boost my headphone amp. Previously I was -31.5 at low gain. I went to -0 dB at low gain and it still wasn't loud enough, so I switched to high gain and ended up at -8 dB on the volume.
At this point of the evening, the CD layer did not give me a very strong ASMR effect. Even when I knew the reference "B" mode was the CD layer, I could only occasionally generate the ASMR effect. So for this ABX test, I had to keep switching back and forth until I actually thought I felt something. I still got a p=0.0021 which is statistically significant.
Files are NOT a bit-perfect match (match=0.18%) at 16 bits
Files are NOT a bit-perfect match (match=0%) at 32 bits
Files match @ 49.7021% when reduced to 7.64 bits
Files are NOT a bit-perfect match (match=0.25%) at 16 bits
Files are NOT a bit-perfect match (match=0%) at 32 bits
Files match @ 49.7027% when reduced to 8.16 bits
With this comparison, the volume difference was notable. I could hear it across the entire track (including bass) whereas before there was just a subtle difference in the stringed instruments. Running my headphone amp as -31.5 dB low gain the first time and then having to go past 0 dB low gain to get into the -8 dB high gain mode, may alter the bass response too.
There is a difference in sound between playing the music normally versus recording the music and then boosting the amplification of the recording at the headphone amp stage. Because it is a copy of a copy, I am adding more THD+N.
The sound of DSD is ultrasonic noise
There are silly levels of ultrasonics when recording the DSD playback. Remember, I'm running this through a zero negative feedback headphone amplifier which only makes things worse. You cannot truly volume match this because the waveforms are fundamentally different. PK Metric remains in the -90 dB range because the spectral content is the same in the 20Hz - 20KHz.
Over the last few months, as I've done my measurements of different setups, I've recently made the statement that "if there is such a thing as a DSD sound, it’s ultrasonic noise.”
I also made a comment when measuring a tube amplifier: "I wonder if tubes produce the effect of analog dither"
So my opinions
1) Ultrasonic noise might be generating IMD into the audible band.
2) Ultrasonic noise may account for the “preferred” coloration of DSD or tube amplification.
3) There is no missing fundamental effect when listening exclusively to ultrasonics. Although I did describe a pressure sensation when trying it.
4) Both the CD and SACD layer are enjoyable but the SACD layer is preferred to me.
And my facts
1) Digital files at the same volume are not necessarily played back at the same actual volume
2) Ultrasonic noise is very high which makes waveform matching difficult
3) In my signal chain, with all sorts of a volume matching strategies, there remains a detectable difference in ABX testing
4) Listener fatigue is real. My ability to discern the ASMR effect in the late evening was worse than earlier in the day.
5) For this album, there are differences between the SACD and CD layer.
Those differences between peak and RMS values suggest extra peak limiting on the CD. And the difference file shows that also with the very short spikes, these are the ones left on the DSD file but limited on the CD. So are you really comparing 2 identical files?
Edit: Clipping the CD file will do almost the same thing, so maybe thats whats going on.
Right. This is especially true when mixing formats between DSD and PCM. DSD is usually recorded at -3 to -6dB relative to PCM to leave some headroom so as to not overload the S-D modulator. SACD specifies -6dB, for example, although most DSD content I've seen seems recorded or converted closer to -3dB. Some playback software automatically make adjustments for this difference, so if two files, DSD and PCM are produced for the same dynamic range the result will be a different playback level. This is why it's important to level match at the output, not at the input
Definitely not, identical. But that is part of the hypothesis or question.
We cannot confirm that DSD 11.2 is necessary for the recording, but it should be able to confirm that it was “worthwhile” to distribute in DSD 2.8 as opposed to plain CD.
My answer is that it is ABX’able. It doesn’t seem like a simple mastering choice, because the first set of tests show that the inputs are very similar. But it does show that the outputs are even more different.
The outputs still have a PK Metric of -90 dB or so, yet visually you can see visually very big differences.
I myself was also able to hear difference in blind tests between DSF and FLAC downloaded from one audiophile site. But after converting DSF to CD quality 44100 WAV I was suddenly not able to here any difference in blind tests.
Let's rekindle the old discussion about high resolution audio formats :) I have conducted my own tests which can be considered equivalent to double blinded and I find that I can determine correct file playback 10 times out of 10 between pairs of DSF64/256 and DSF64/FLAC192 files. Here are the...
www.audiosciencereview.com
Therefore most probably I was hearing difference not between formats but between masters.
Easy test which saved me expensive trips down the audiophile rabit holes.
The specific hybrid SACD arrived on my desk, and I could successfully extract the DSD layer into DSF files using old model of Sony PlayStation and "the" unofficial SACD ripping tool. Of course, I could rip the CD layer into AIFF PCM 16 bit 44.1 kHz files using dBpoweramp CD Ripper.
I am on my intensive process of analyzing the track-11 in DSF and AIFF using MusicScope.
My comparative listening with my audio setup will be performed very soon, as follows.
1.Native DSD vs. AIFF using JRiver MC --> single DAC (OKTO DAC8PRO in 2-Ch stereo mode, of course capable of DSD and PCM) --> single HiFi amplifier (Accuphase E-460) --> passive LCR-network --> five-way SP setup (with L&R sub-woofer, L&R super-tweeter) under objectively proved/determined exact level matching in 0.1 dB precision.
BTW, I have already converted the extracted DSF file of Track-11 into 16 bit 44.1 kHz AIFF using dBpoweramp Music Converter and did comparative objective analysis by MusicScope between the DSF->AIFF file and the CDLayer-ripped-AIFF file.
Therefore, intensive comparative listening will be also done soon; 3.DSF->AIFF filevs.CDLayer-ripped-AIFF using my DSP-based multichannel multi-SP-driver multi-amplifier time-aligned stereo 5-way 10-channel audio system; in this case both of the AIFF tracks would/should be converted into 88.2 or 96 kHz 24 bit PCM on-the-fly by JRiver for DSP processing with EKIO again under objectively proved/determined exact level matching in 0.1 dB precision.
Hopefully, the details of my above investigations will be posted here in this thread within a few weeks.
Even if you use same SACD player or same software audio player (like JRiver, Roon), there could be subtle difference(s) of internal DAC procedure on DAC processor chip or on the software for DSF feed vs. PCM feed.
"Difference between the masters" would of course too could cause audible difference(s).
The issue on audible/objective difference between DSF(DSD) and PCM, therefore, would be always so complicated and never to be perfect, I assume.
After DSF was converted to PCM I was not able to hear the difference in blind tests on the same hardware.
This along with scientific consensus gives me 99% assurance that the difference was audible because of master or transcoding parameters.