Follow along with the video below to see how to install our site as a web app on your home screen.
Note: This feature may not be available in some browsers.
Welcome to ASR. There are many reviews of audio hardware and expert members to help answer your questions.
Click here to have your audio equipment measured for free!
I'd like to make a quick comment regarding SACD mastering compared to red book masters of the same material - they're NOT always the same and reviewer/engineer Martin Colloms once found noticeable eq differences in the spectra of supposedly the same recordings red book vs. SACD..
I seem to recall that SACD players have noise output between 20 and 40kHz. If still true, the N801s would no doubt be peaking this up rather too easily and this may or may not make for an audio-band subjective (intermodulation?) difference. Maybe the Oppo isn't like this, but I seem to recall this ehf noise is part of the SACD process...
Years ago I used to use my ps3 as my sound source. I then switched over to a macbook pro (both connected via hdmi) and I recall thinking the ps3 sounded better. I also recall posting this on another audio website and people saying it might be in my head. Coulda been. I don't recall. But I think I may have had the output volume on the Mac turned down or something making it lower in volume. Probably doesn't explain the OP's case though since he said he level-matched.
Over the weekend, we had a chance to rip a few of our own SACD discs into DSF file.
We then compare the playback of SACD Disc vs the Ripped DSF File on the same audio system.
Player: Oppo 105 SACD player (has USB input to playback the DSF file from original SACD Disc) - using XLR output
Pre-Amp: Denafrips Athena
Amplifier: Two Mark Levinson 333 Bi-Amp
Speakers: A Pair of B&W N801
Room: Acoustically treated
Playback is level match with pink noise track so both SACD Disc and DSF file playback are the same volume SPL.
The ripped DSF file sounds good, but when compare to SACD Disc playback of the same album/song, the Disc playback has slightly better dynamic and transparency.
We could tell with our eyes closed when one of us switches between Disc Playback vs DSF file playback from the same Oppo 105.
Why is that?
Is it because the Disc playback are clocked signal inside the SACD player circuit so it has less digital jitter than the USB file interface playback?
Hello Again,
As mentioned in my first post, I have found this to be an interesting thread. This is because I have been ripping my sacd collection to DSF files for both archiving and later on perhaps playback purposes. And for whatever reason I had yet to do a comparison between the disc and the file playback. So now I have done that. And I didn't really notice anything different. However, it was a simple comparison and not real time or ABX, or anything like that.
I use a Marantz Ruby player and use JRiver for file playback. JRiver allows bitstreaming of DSF files so there is no conversion either with the player or file playback.
Perhaps the best source of information on your situation here at ASR would be Kal Rubinson. Kal rips multichannel sacds to DSF and plays them on his Oppo 105. He has been doing this for years, so would be a very knowledgeable source of information for you. Perhaps you could PM him for his opinion.
Perhaps the best source of information on your situation here at ASR would be Kal Rubinson. Kal rips multichannel sacds to DSF and plays them on his Oppo 105. He has been doing this for years, so would be a very knowledgeable source of information for you. Perhaps you could PM him for his opinion.
Hi dualazmak,
I agree that even within the same SACD player (in this case oppo 105), there is difference between direct SACD disc play circuit vs the USB file play circuit.
I was sharing the observation that the SACD Disc playback sounds better than USB DSF file (ripped from the equivalent Disc) playback with the same exact SACD player/DAC....
so checking if someone have observed the comparison difference and understood why.
The audible difference is not huge but is consistently identifiable ......just like comparing a playback of Hybrid SACD Disc , between the SACD layer vs CD layer, with resolving sound system, one can hear the improved sound of SACD layer of the same album/song over the CD layer.
Sorry, but I do not have OPPO 105. Your observation or hearing experience is unique to your OPPO 105, I cannot, therefore, share my comment on this issue.
Incase if would you use another (brand) SACD player which has USB DSD file playback capability, your experience could be different, I believe.
I assume above two lines are the only response from myself (and highly possibly from all of us) to your top inquiry...
Other points and discussions on this thread, i.e. SACD vs. CD, SACD's extracted DSF=DSD64(1x) vs. SACD disk play, DSD64(1x) vs. DSD128(2x) vs. DSD256 (x4) vs. DSD512(x8), and so on, have been already intensively shared and discussed so many times in ASR Forum and other places. These are not only "format specific issues", but also quality-control (QC) matters; please also refer to my post here.
Your selection of digital file format(s) and their playback audio chain/system should/would be always dependent on your own stance, policy, practice(s) in your own unique audio setup/system, and for your own ears and brain.
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
Welcome to this project thread! Your above inquiry is nice and important point, indeed.
My present answer for you is "It is quite feasible enough and even ""needed"" to feed all the audio digital signals in 88.2 kHz or 96 kHz PCM (or 192 kHz, if you like) by JRiver's on-the-fly format conversion to be sent into DSP (XO/EQ) software EKIO. "
Various background and justifications for this answer are as follows;
Before starting this project, I had been enjoying music with ordinary PC audio setup with one DAC (OPPO Sonica DAC)) and one HiFi integrated amplifier (ACCUPHASE E-460) driving all the SPs through passive LC (inductors capacitors resistors) network. And I had been sticking to "native format feed" into OPPO Sonica DAC up to 1-bit/DSD256(4x), as you kindly pointed.
When I started considering possible multichannel multi-driver multi-way multi-amplifier project with software DSP (XO/EQ), I did intensive search and desk evaluations on various DSP software solutions, and I found the maximum PCM processing format is 192 kHz 24 bit in these DSP software solutions. (Even with the extraordinary expensive TRINNOV ALTITUDE 32 DSP processor, actually having PC in it, the internal DSP processing is up to 192 kHz).
I carefully considered the pros and cons of "DSP processing all tracks in 192 kHz or 96kHz" instead of "native format feed", and concluded that multichannel multi-amplifier approach would surpass the cons, at least in my system setup with still amazingly wonderful Yamaha SP drivers and cabinet.
Consequently, I decided to go into "multichannel multi-amplifier" world of "max. 192 kHz 24 bit processing", as you kindly have read through this project, including the "all in max. 192 kHz ASIO I/O within PC".
Then, rather recently, I (we) fully discussed and evaluated the UHF (ultra-high frequency) noise issue in poorly QC-ed HiRes music tracks including DSD formats, as you clearly noticed; - "Near ultrasound - ultrasound" ultra-high frequency (UHF) noises in improperly engineered/processed HiRes music tracks, and EKIO's XO-EQ configuration to cut-off such noises:#362-#386, #518
I wrote that such a high amount of UHF noises would be "possibly" harmful (and useless, meaningless) for our tweeters and super tweeters. I also pointed they would be highly possibly harmful for our beloved pets including dogs, cats, birds.
Having my intensive objective measurements of these "poorly QC-ed" HiRes tracks, and having so many intensive discussions on "enough PCM sampling rate in HiFi audio", now I conclude that 88.2 kHz or 96 kHz processing (i.e. up to 44.1 kHz or 48 kHz in L and R channels) would be just enough and feasible in my setup (and I believe so also in your setup) since I decided always having high-cut (low-pass) -48 dB/Oct filters at 25 kHz in my EKIO configuration to cut-off any of the possible UHF noises very frequently existing in HiRes tracks.
This means that I have finally landed on agreement with @mikessi's "enlightenment and belief" of "There is really no audible benefit to playback beyond 24/96 sampling, especially with any recordings other that those done with the most advanced high res gear and high fidelity values."
and now JRiver MC feeds all of the tracks usually (mainly) in 88.2 kHz 24 bit (i.e. max. 44.1 kHz Fq window in 2-ch stereo) by on-the-fly conversion into EKIO for crossover/EQ processing. As I have high-cut (low-pass) -48 dB/Oct LR filters at 25 kHz, max. 44.1 kHz in L & R channels are more than enough.
You would please note that I have "several" reasons and rationales for my above policy and practice, and they are again unique to my audio setup, listening acoustic environments, and importantly also they are unique to my ears and brain.
I am interested in, and curious about, after having discussions on this thread plus your possible setup advancements (e.g. new SACD player, etc.), what will be your general stance policy and practice(s) for digital music track listening enjoyment.
Hello Again,
As mentioned in my first post, I have found this to be an interesting thread. This is because I have been ripping my sacd collection to DSF files for both archiving and later on perhaps playback purposes. And for whatever reason I had yet to do a comparison between the disc and the file playback. So now I have done that. And I didn't really notice anything different. However, it was a simple comparison and not real time or ABX, or anything like that.
I use a Marantz Ruby player and use JRiver for file playback. JRiver allows bitstreaming of DSF files so there is no conversion either with the player or file playback.
Perhaps the best source of information on your situation here at ASR would be Kal Rubinson. Kal rips multichannel sacds to DSF and plays them on his Oppo 105. He has been doing this for years, so would be a very knowledgeable source of information for you. Perhaps you could PM him for his opinion.
He did but I am not of any assistance in this matter. I played files through my Oppos only rarely in the past and not at all now. The Oppos are only for ripping.
I seem to recall that SACD players have noise output between 20 and 40kHz. If still true, the N801s would no doubt be peaking this up rather too easily and this may or may not make for an audio-band subjective (intermodulation?) difference. Maybe the Oppo isn't like this, but I seem to recall this ehf noise is part of the SACD process...
Noise-shaped DSD quantisation noise usually begin to be seen from about 25 kHz to higher above noise floor of SA-CD players, but that noise isn't that significant. And, most importantly, since 1999 (launch year of SA-CD) nobody anywhere in the world has ever demonstrated that that noise have any effect, neither by properly conducted listening test nor by measurement. In fact, to the best of my knowledge, nobody in the world has ever published a single measurement in order to investigate the possible effect of noise-shaped DSD quantisation noise on downstream electronics or loudspeakers. Incredible but true, until someone point out to a relevant experiment !
I wish to insist on a point made by Doodski and Blumlein 88 from the very start of this thread : level matching two playback chains by measuring sound pressure levels with an SPL meter when playing pink noise, or other type of noise, is A TOTALY UNRELIABLE METHOD OF LEVEL MATCHING.
It cannot been dismissed that what you have heard is just plain level differences you were left unaware of because you tried to level match your playback chains through an irrelevant method.
This hybrid SA-CD does contain not only pink noise tracks, but sine wave tracks also.
If you get this SA-CD and rip the SA-CD layer, you then will have at your disposal sine signals. By playing them, you will be able to check the level matching of your two playback chains (disc playing and ripped file playing) at the analogue outputs of your Oppo player with a voltmeter. This method is much much more reliable to check levels.
I’m a firm believer in fair use when it comes to audio media — if you bought it, you own it, and you’re free to do with it as you please — as long as it’s for your personal enjoyment and you’re not trying to sell it illegally for profit. Apparently the code writers/hackers in the world agree...
Noise-shaped DSD quantisation noise usually begin to be seen from about 25 kHz to higher above noise floor of SA-CD players, but that noise isn't that significant. And, most importantly, since 1999 (launch year of SA-CD) nobody anywhere in the world has ever demonstrated that that noise have any effect, neither by properly conducted listening test nor by measurement. In fact, to the best of my knowledge, nobody in the world has ever published a single measurement in order to investigate the possible effect of noise-shaped DSD quantisation noise on downstream electronics or loudspeakers. Incredible but true, until someone point out to a relevant experiment !
I wish to insist on a point made by Doodski and Blumlein 88 from the very start of this thread : level matching two playback chains by measuring sound pressure levels with an SPL meter when playing pink noise, or other type of noise, is A TOTALY UNRELIABLE METHOD OF LEVEL MATCHING.
It cannot been dismissed that what you have heard is just plain level differences you were left unaware of because you tried to level match your playback chains through an irrelevant method.
This hybrid SA-CD does contain not only pink noise tracks, but sine wave tracks also.
If you get this SA-CD and rip the SA-CD layer, you then will have at your disposal sine signals. By playing them, you will be able to check the level matching of your two playback chains (disc playing and ripped file playing) at the analogue outputs of your Oppo player with a voltmeter. This method is much much more reliable to check levels.
Thank You for useful feedback.
I just ordered the disc from eBay , the link you send did not have it for sale .
Which track do i use to measure the volt at speaker terminal ?
volume control is 0.5dB increment .
If our ears as sensitive to 0.5dB increment , our ears are certainly as sensitive to hear disc playback difference between CD layer vs SACD layer or in this case SACD disc vs its equivalent file playback.
I also wonder if speakers quality makes a difference , because we could not tell the difference when playback with Martin Logan speakers or headphones.
I also ordered a used oppo 205 so we can repeat the experiments another time .
One time payed a download of SACD for e.s.t. London concert to compare to CD-rip.
The result: skipped both and listening joyfully every time again to Hamburg concert (ripped from CD).
I just ordered the disc from eBay , the link you send did not have it for sale .
Which track do i use to measure the volt at speaker terminal ?
volume control is 0.5dB increment .
If our ears as sensitive to 0.5dB increment , our ears are certainly as sensitive to hear disc playback difference between CD layer vs SACD layer or in this case SACD disc vs its equivalent file playback.
CD layer : tracks 12 to 15, respectively 1 kHz, 100 Hz, 10 kHz and 15 kHz
SA-CD layer, stereo area : tracks 12 to 16, respectively 1 kHz, 100 Hz, 10 kHz, 15 kHz and 30 kHz
SA-CD layer, multichannel area : tracks 28 to 32, respectively 1 kHz, 100 Hz, 10 kHz, 15 kHz and 30 kHz
Depending on the bandwidth of the voltmeter you have at hand, it is preferable to measure the output voltage with the 100 Hz or the 1 kHz signal.
Sine waves on the Denon test disc are at -16 dBFS (CD layer) and -16 dB SA-CD (SA-CD layer) and are level aligned : the -16 dBFS and -16 dB SA-CD signals have the exact same amplitude.
I would begin to measure the output level at source terminal, not at amplifier terminal. As the Oppo BDP-205 XLR output you will probably use have a nominal level of about 4.25 V at full scale, you should measure a voltage of about 0.55 V between pin 2 and pin 3 of the XLR with the -16 dB sine signals on the Denon disc. At unbalanced terminals, those voltages should be roughly divided by 2.
If the output levels will be different between disc playback and file playback, there is a risk that 0.5 dB increment may be to broad to accurately compensate the difference with the preamplifier volume control, but we will see.
Beware ! Do not try to listen to the high frequency sine test signals with your loudspeakers by increasing the volume: you may damage the tweeters !
For 2-Ch stereo system check and measurements, "Super Audio Check CD 48DG3 by CBS/Sony" too is really useful, I believe. Attached herewith, please find the PDF booklet of this CD (English translation by myself).
Just for your reference, I have analyzed all the tracks of this CD using ADOBE Audition 3.0.1 and MusicScope 2.1.0; please refer to my post here #651 on my project thread.
If you and/or any people would be seriously interested in using the tracks of this "Super Audio Check CD", please simply let me know through PM communication writing your wish.
Edit:
I have also shared this post;
- Five "real air-recorded transient-sound tracks" of Sony Super Audio Check CD played and analyzed by MusicScope 2.1.0:#760
Attachments
SONY Super Audio Check CD_ Booklet_English by dualazmak_rev03.pdf
For 2-Ch stereo system check and measurements, "Super Audio Check CD 48DG3 by CBS/Sony" too is really useful, I believe. Attached herewith, Please find the PDF booklet of this CD (English translation by myself).
Just for your reference, I have analyzed all the tracks of this CD using ADOBE Audition 3.0.1 and MusicScope 2.1.0; please refer to my post here #651 on my project thread.
If you and/or any people would be seriously interested in using the tracks of this "Super Audio Check CD", please simply let me know through PM communication writing your wish.
Edit:
I have also shared this post;
- Five "real air-recorded transient-sound tracks" of Sony Super Audio Check CD played and analyzed by MusicScope 2.1.0:#760
It looks the CD is no more available, except for if you would find it in used market.
This is why I wrote in my above post #76 "If you and/or any people would be seriously interested in using the tracks of this "Super Audio Check CD", please simply let me know through PM communication writing your wish."
I think a disambiguation is necessary : this Super Audio Check CD 48DG3 by CBS/Sony is not an SA-CD but an ordinary CD Audio !
So, this disc is pointless in case of SA-CD/DSD playback.
Yes, you are right, therefore I wrote in my above post #76 "For 2-Ch stereo system check and measurements, "Super Audio Check CD 48DG3 by CBS/Sony"too is really useful, I believe."
In fact, to the best of my knowledge, nobody in the world has ever published a single measurement in order to investigate the possible effect of noise-shaped DSD quantisation noise on downstream electronics or loudspeakers. Incredible but true, until someone point out to a relevant experiment !
What do you think about my typical experience shared in my post #362 on my project thread?
I suspect that such a high amount of UHF quantization noises, even almost inaudible, could be somewhat harmful to our tweeters and super-tweeters if we have no LP filter to cut them off.
And furthermore, such high amount of UHF noises, if transparently reproduced by our excellent super-tweeters, should be highly possibly audible and maybe harmful to our beloved pets, I mean dogs, cats, birds, etc. I wondered that the trouble case with his dog described by @Mutsuin his post here and thereafter would be related to this point, even this is still just my speculation though.
Furthermore, the UHF quantitation noises in DSD format is also definitely one of the main QC (Quality Control) issues of the company producing/distributing SACD and DSD discs/tracks; you would please refer to my recent post here on this point.
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
Welcome to this project thread! Your above inquiry is nice and important point, indeed.
My present answer for you is "It is quite feasible enough and even ""needed"" to feed all the audio digital signals in 88.2 kHz or 96 kHz PCM (or 192 kHz, if you like) by JRiver's on-the-fly format conversion to be sent into DSP (XO/EQ) software EKIO. "
Various background and justifications for this answer are as follows;
Before starting this project, I had been enjoying music with ordinary PC audio setup with one DAC (OPPO Sonica DAC)) and one HiFi integrated amplifier (ACCUPHASE E-460) driving all the SPs through passive LC (inductors capacitors resistors) network. And I had been sticking to "native format feed" into OPPO Sonica DAC up to 1-bit/DSD256(4x), as you kindly pointed.
When I started considering possible multichannel multi-driver multi-way multi-amplifier project with software DSP (XO/EQ), I did intensive search and desk evaluations on various DSP software solutions, and I found the maximum PCM processing format is 192 kHz 24 bit in these DSP software solutions. (Even with the extraordinary expensive TRINNOV ALTITUDE 32 DSP processor, actually having PC in it, the internal DSP processing is up to 192 kHz).
I carefully considered the pros and cons of "DSP processing all tracks in 192 kHz or 96kHz" instead of "native format feed", and concluded that multichannel multi-amplifier approach would surpass the cons, at least in my system setup with still amazingly wonderful Yamaha SP drivers and cabinet.
Consequently, I decided to go into "multichannel multi-amplifier" world of "max. 192 kHz 24 bit processing", as you kindly have read through this project, including the "all in max. 192 kHz ASIO I/O within PC".
Then, rather recently, I (we) fully discussed and evaluated the UHF (ultra-high frequency) noise issue in poorly QC-ed HiRes music tracks including DSD formats, as you clearly noticed; - "Near ultrasound - ultrasound" ultra-high frequency (UHF) noises in improperly engineered/processed HiRes music tracks, and EKIO's XO-EQ configuration to cut-off such noises:#362-#386, #518
I wrote that such a high amount of UHF noises would be "possibly" harmful (and useless, meaningless) for our tweeters and super tweeters. I also pointed they would be highly possibly harmful for our beloved pets including dogs, cats, birds.
Having my intensive objective measurements of these "poorly QC-ed" HiRes tracks, and having so many intensive discussions on "enough PCM sampling rate in HiFi audio", now I conclude that 88.2 kHz or 96 kHz processing (i.e. up to 44.1 kHz or 48 kHz in L and R channels) would be just enough and feasible in my setup (and I believe so also in your setup) since I decided always having high-cut (low-pass) -48 dB/Oct filters at 25 kHz in my EKIO configuration to cut-off any of the possible UHF noises very frequently existing in HiRes tracks.
This means that I have finally landed on agreement with @mikessi's "enlightenment and belief" of "There is really no audible benefit to playback beyond 24/96 sampling, especially with any recordings other that those done with the most advanced high res gear and high fidelity values."
and now JRiver MC feeds all of the tracks usually (mainly) in 88.2 kHz 24 bit (i.e. max. 44.1 kHz Fq window in 2-ch stereo) by on-the-fly conversion into EKIO for crossover/EQ processing. As I have high-cut (low-pass) -48 dB/Oct LR filters at 25 kHz, max. 44.1 kHz in L & R channels are more than enough.
You would please note that I have "several" reasons and rationales for my above policy and practice, and they are again unique to my audio setup, listening acoustic environments, and importantly also they are unique to my ears and brain.
I am interested in, and curious about, after having discussions on this thread plus your possible setup advancements (e.g. new SACD player, etc.), what will be your general stance policy and practice(s) for digital music track listening enjoyment.
