Something that needs to be understood is that the file is exactly what it should look like for DSD. The whole point of DSD is to use an extremely high sample rate - way above what even dogs hear - to allow for all of the noise to be shaped in such a way that the audio in and around the audible range can be extremely accurate while filters remove the noise in the ultrasonic band before it is amplified. This allows for very shallow filters near the audible band of frequencies, and thus very low distortion in what is sent to the speakers. DSD dacs filter all of the noise above about 50khz, whereas for, say, 44.1khz pcm, the filtering starts at 22khz. This results in a pcm equivalent (for dsd 64) of about 100khz sample rate with a bit depth of around 20bits. If you look at the spectrum the video, you will see that the noise shaping begins at 50khz, right where the DSD DAC filters start filtering higher frequencies.
Thanks for educating people here including Amir it seams, Kinda sad to read all this and they dont know how DSD works. Its also the timing that is so fantastic with DSD. Nothing can compare with it I think I even make DSD files of my vinyls with an ADC and it sounds exactly like the vinyl.
@@tweakknob I believe Amir said in the video that this is expected from a DSD file so to me it seems he understands this : ) Although yes I agree that Amir does sometimes skip or misunderstand a few things. To me it also seems like a fun (weird) thing to capture old music on DSD since what capture is already being restricted to a certain frequency range. So what you add to the music is something that's not there. But if you do enjoy it-I'm happy you do.
I explained that noise shaping was used as a result of DSD encoding so not sure why you say it was not understood. What you are not understanding is that this is NOT SACD. This is a DSD256 file at much faster sampling rate. The content is only played on a computer and as such, you are at the mercy of analog filtering in the specific DAC. I showed in the forum post that AKM AK4493 for example, has whopping 400 kHz bandwidth for DSD256! Not 50 kHz as you state. See: www.audiosciencereview.com/forum/index.php?threads/inside-high-res-music-jazz-at-the-pawnshop-naxos-dsd.23137/post-771685 It will even have respond as high as 600 kHz. So please don't use your assumptions about DSD here. These are digitally downloaded files and you are paying for the bits literally and in download file size. This one file alone is nearly 900 Megabytes! Yes, this single file is 1.5 times an entire CD! This is due to high DSD rate which encodes even more noise than standard DSD.
@@tweakknob You are mistaken about both unfortunately. DSD has no advantage in timing. Just because it samples faster doesn't mean it has better timing. It has extremely shallow bit depth which sets the format's timing resolution. As to how it sounds, it has 100% to do with the mastering, not anything to do with the format which by itself is faulty compared to high-rate PCM. I would also not chime in after the above poster said everything is filtered above 50 kHz. If that is so, then it has no bandwidth advantage either! Nor is there value in pumping up noise above 20 kHz lest that is your idea of high fidelity. It not mine. :)
@@AudioScienceReview Thanks for the clarification. I thought I would chime in a little about DSD since many comments did not seem to understand the purpose of the loud noise above 50khz. If indeed DACs are not filtering frequencies above 50khz or so, then that is worrisome since it can negatively affect the speakers. I agree though that DSD’s benefits compared to PCM are tenuous at best, and that the file sizes are unreasonably large.
DSD is a natural format for delta-sigma DAC. So majority of DS will filter these high frequency noise, since they also have the same signature. This is due to how delta-sigma work.
@@bltzcstrnx actually most speakers can't reproduce frequencies that high anyway. but like he said in the video, if you have a wide band amp and its still pumping those frequencies into them, what's that doing to the tweeters? can't be good for them? especially at high volumes.
@@XX-121 speakers are passive device, it will reproduce whatever that is sent to them, at least it will try. Of course this won't be good and can be damaging. Hence competent amplifier will have both DC protection on the low-end of spectrum and low pass filter on the high-end of spectrum.
You are a hero of the audio comunity Amir! It would be super cool to see a comparison with lossy convertions as well, perhaps also a comparison to a vinyl hooked up to your Apx555 ? :)
Hi, I am the mastering engineer at 2xHD that did the transfer and mastering. The point of buying the DSD256 is to play it in DSD256. The DSD256 have no noise compare to lower DSD so the filter in the DAC that remove all that noise is very gentle. Important to know that if you analyze the analog output of a DSD DAC that noise that you taking about is not there because of the low pass filter. The resolution of 11.2.MHz sample every second really sound more sweet and close to the analog master compare to any PCM if you play it that right way. I did release also Jazz at the Pawnshop on Vinyl in all analog all tubes also on 2xHD. Rene Laflamme
Amir. Thank you for doing this. I think you are uncovering a huge issue with ‘Hi-Res’ recordings. For the average consumer there is no effective method for them to determine that they are getting what they pay for. And without such a method, there is zero incentives for the recording industry to clean up their act. Given that I can’t identify a single difference between CD or hi-Res (and not for the lack of trying), you have finally given me the confidence that I am not missing out if I stick with 16bit/44.1kHz for music.
Yeh I feel bad myself since I spent thousands of dollars buying all these albums only to see so many issues in them. A quick analysis like I did is within the abilities of anyone producing such content yet it is not done. It is like shipping software without ever testing it. Or serving a dish in a restaurant without ever tasting it.
I believe DSD dacs have analog low pass filters on the outputs; I doubt any of the noise would make it through to the subsequent stages, like you imply. Thank you for your video, either way. Your expertise is appreciated.
It is usually an optional filter. That doesn't solve the problem of getting more noise in the form of larger file size to download and in some instances, paying extra for higher sample rates.
Yes, the idea of DSD is that with the extreme sample rate the noise can be pushed way into the ultrasonic band, which as you can see increases in amplitude with increase of frequency, which allows for comparatively shallow filters to be used in the frequencies closer to what humans can hear, avoiding a pitfall of PCM recordings (steep filters), and increasing the steepness of the filters as the frequencies get farther and farther from human hearing.
@@LordAus123 Well, the notion of a "high-sample rate" format which fills the ultrasonics with noise and then proceeds to filter some of it, doesn't sit well with me. What possible benefit comes out of pumping noise into your system? And as I noted elsewhere, your assumptions about this file are incorrect. This is not SACD/DSD. These are higher rate DSD256 played on a computer, not SACD player. As such, there is no typical filter for them. I showed in the ASR thread for example that the AKM AK4493 DSD filter goes to 400+ kHz and above! So what you see in my analysis, is all pushed into your audio system. There is no need at all to do that with 24-bit, PCM format.
This album was recorded on a Nagra IV in 1976 and first transferred to digital in 1996. It has since been subjected to countless reissues and different iterations including a UHD 32 bit version and a remaster by Doug Sax. Somehow I doubt that the analog master was used.
Dream Theater - A Dramatic Turn of Events is a very well recorded 96kHz album with audio signals up to the 30kHz range, but doubtful I can hear any difference from the CD.
I can't yet since I prepared these a couple of years ago and am just posting them. Once I post all of them, I will create some new ones that show the format in better light. :)
This video reminds me a story from early 1990's. I read an article in a newspaper, where an audiophile claimed that he heard 'wow and flutter' from CD. It turned out the CD was mastered from analog tape recording. THe digital audio CD able to reproduces 'everything' including flaws and garbage of the original analog tape recording. It is wise to find a proper full digital recording, mixing and mastering. Thank you.
Thanks for in depth analysis Amir! To me, you are like Audiophile Mythbuster!! If possible, can you please analyze a FLAC from Reference Recordings, highly praised by PSAudio / Paul McGowan ?
His quality comes more from excellent mastering, loads of dynamic range (I don't think he even uses compression at all), and the use of a Pacific Microsonis Model 2 as the A/D converter. Nothing to do with this hi-res non-sense, even though he gives you such downloads.
pretty much. one that's properly mastered. actually i wish HDCD would've caught on all those years ago. Not only do they sound superb when properly decoded, but they were also the answer to the loudness war with extra, up to 6db of headroom they had at their disposal. i think the only time it was used like that was on a Green Day album tho.
Hey Amir! I am a frequent viewer of your channel and enjoy much of your content. I think you provide a valuable resource for the audiophile community by providing measurements for DACs/AMPs that we can’t listen to or demo ourselves before making a buying decision. So I really appreciate what you do. Having said that, I have to push back on everything you said on this video. First of all this video isn’t really about this specific recording but about the DSD format in general. I don’t think you really understand how DSD works. It was designed by Sony/Phillips to replace the original PCM CD format which they also invented. You have the same group of engineers who made PCM make DSD. Also DSD unlike PCM is a 1 bit format. What DSD does is it basically takes all the best aspects of PCM and all the best aspects of analogue and combines them in a digital format. You get the clarity and dynamic range of digital with the smoothness of analogue without all the detractors of both formats.It really is a FAR superior format over any other available today. Also in this specific case, your analysis is total garbage. You take your DSD track and import it into Adobe and say “look at all this garbage above the listening range” etc... This shows your ignorance that you really don’t have any idea of what you are talking about. DSD is designed to spread itself over a very high frequency band because it is a 1 bit format. Furthermore you go on to say “I don’t have a DAC that I can use to listen to this so I will just convert this back into PCM and do my analysis on that instead...” Well that defeats the whole purpose of this video does it not? We are talking about a high res DSD file format recording here and you are poo-pooing it when you can’t even listen to it and you can’t properly analyze it because you clearly don’t understand how it works. Normally when you do your measurements and analysis on this channel on say a DAC or something it makes scientific sense, but in this case it’s total garbage and you are misleading your viewers and pandering to the naysayers who seem to frequent your channel. I suggest you do a deep dive and learn more about the engineering specifics of DSD and get a DAC like say the Kitsume Holo May DAC that can actually play back DSD before making a judgement and misleading your viewers and fans. Cheers!
The hubris of this guy is astounding. He somehow believes that thousands of other engineers that actually make high end audio gear are all crooks or stupid! His claim to fame was doing telephone systems with a cord - not even wireless. So, if others want to listen to how awesome his phone boards are then I suggest they buy a high end phone with speaker. As for me, I will continue to enjoy my hifi without listening to imposters like this joker.
If every Dsd compatibile dac has filter (or filters to choose) somewhere between 30-50khz how this noise could be transfered into system ????????? Main profits of new digitalizations is that they are made on modern equipment which has much better specs then previous attempts , mostly feom the 80s.
There is no such thing as "compatible DAC." You have no idea what the filtering is in the DAC you are using. Indeed as I showed in an example I looked up of AKM's AK4493, the bandwidth for DSD256 which this file is, exceeds *400 kHz* !. Not 30 to 50 kHz. See this post: www.audiosciencereview.com/forum/index.php?threads/inside-high-res-music-jazz-at-the-pawnshop-naxos-dsd.23137/ I didn't realize there is this much confusion between SACD as a physical format played on SACD player of years back, and today's high-rate DSD downloads. They are completely different animals. The latter is played on a computer and at very high bandwidth. Indeed, customers of such formats demand the high bandwidth or they would not chase bigger and bigger DSD numbers. We are now up to DSD1024 with the original SACD version being just DSD64. But yes, we can do a bitter job digitizing tapes now but the proper format for that would be PCM with objective analysis like i am doing to make sure we are not just manufacturing noise and shipping it to people. As I noted elsewhere, this one file is 900 Megabytes! Vast amount of what is there is ultrasonic noise with no value.
@@AudioScienceReview i guess you are using Rme Adi2 . Look at user manual. You can choose between 30khz or 50khz LPF during DSD playback. Also you can check datasheet for any DAC chip with DSD support , it's always shows schematics for the output stage for dsd playback that cuts high freqs at some point . So non of this noise will ever get into amp or speakers. I can't understand high bandwith dsd download, especially then it's done by "upsampling". Nativedsd web store is full of such downloads.
I find it funny when they offer the remasters of older music in SACD/DSD or 24bit/96khz. From what I can find most of the great music from the 60s and 70s was recorded on 2" tape that was capable of about 70db dynamic range and 18 to 20khz. The mics used back then also were not capable of capturing this. Sometimes these remasters do sound better to me, BUT my gut tells me its because of the remaster and a competent engineer and not so much the hi-res format.
I hear you. :) We certainly are limited by the dynamic range of the tape format. There is more spectrum to be had though as the analog formats were not brickwalled so to the extent someone wants them, there should be a way to delivery them. Personally I like to see high-res format be about non-loudness-compression where higher bit depth and sample rates come as bonus.
Thanks Amir - I'm kind of confused with what you're showing us - - I've always had the "layman" kind of idea that yes, while these higher sample rates go beyond our normal hearing, I thought the primary idea was to sample the sounds more frequently during any given moment in time - in order to achieve a higher resolution of sound, like more pixels per square inch in a picture - so - I guess I'm really not understanding everything involved with these higher resolution recordings - - that is, in that all that noise content that come with these higher sample rates, at least with the ones you've shown us so far, would seem to do more harm than good. I hope you show us more, I would like to understand it all a bit better.
If you just use higher sample rate, you get wider bandwidth. NOT higher resolution. If you limit the bandwidth to say, 20 kHz, and then keep increasing the bandwidth, you do gain some resolution in the way you intuit. Indeed DSD works this way by sampling at very high frequency and just one bit of depth. PCM downloads though just keep increasing the bandwidth with higher sample rate since bit depth is 24 bits anyway so doesn't need help in that front. And sorry about these videos not having as much tutorial value as my newer ones. I created these a couple of years ago with a different intent and style.
High sample rates mean you can potentially build a more linear analogue output stage on your dac that will result in better quality sound. But a dac can do this internally as well with digital algorithms.
The snake oil SLAYER strikes again. Please, keep them coming. I know nothing about audio, half of the things they say does not mean anything to someone as me "noob' Thanks for this awesome channel.
This is even more fascinating than face value. Unless there was no bias tone on the recording tape deck, you would see a very large spike at an ultrasonic frequency. Maybe 75 or 100 kHz. Unless if course it's just covered by that horrendous hf noise of the dsd. Still, even a simple first order low pass filter would do a lot here. Yikes.
Your analysis is based on digital domain of the file. On the analog domain all the high frequency should be filtered by a lowpass, on a medíocre designed dac, this should not happen .
No such thing happens in "analog domain." As I showed in the ASR thread, the playback bandwidth of today's modern DACs is 100s of kHz for these high rate DSD files. The whole point of formats like DSD256 is to give audiophiles more spectrum so filtering them like SACD did as you think, would go against the very value proposition of these formats.
@@AudioScienceReview Well these DACs are not correctly designed for DSD. The DAC *IS* part of the playback chain in DSD and a correctly designed one will attenuate the HF noise with FIR DSD filter and at least 3rd order analog (linear phase) LPF.
@@AudioScienceReview Im sorry, but I did not read the thread, but I will. How can a noise shapped signal not be filtered? So. supposing we have a 70KHz usable signal on a DSD detected signal by the DAC, lets say 256fs, you are stating that the engineers went loose and all the cut off frequency, -30db/octave FIIR lowpass do not work and they are not complying with the scarlett book. Thats reassuring to know a highly noise shaped PDM HF´s product is making all the region bats nuts and generating a hell of instability on my Amplifier
Only a small amount of that has to do with analog tape capture. All that ultrasonic noise shaping is due to the digital format in use, not analog capture.
I'm confused. You said that Audition will not play the file (I believe that to be correct) and yet you read the file into Audition. So what is Audition showing you? I think it's just showing you the DSD data but with some spurious mapping to PCM. Or did you set up Audition to somehow understand the 1-bit nature of DSD, in which case the filtering that is necessarily applied to DSD decoding is missing? Either way, I don't believe Audition is showing what the DSD stream is meant to represent. Or have I misunderstood this completely? That is also a distinct possibility.
Audition can open and and analyze the file but it lacks the output capability to play it to the DAC. Playing DSD requires a pipeline to either encapsulate it over PCM or through ASIO. Audition doesn't support either.
This should be clearly labelled as NOT a high resolution recording but a DSD encoding of “what exactly was the original ? “ Amir mentioned that he’s assuming that this is of the master tapes....
Amir might debunk this, but from what I understand it's better to convert DSD to 24/88 files because they are natural multiples and converting it to 24/96 can introduce quantization errors or noise or something.
This video is misleading. If the DAC is designed correctly, most of that HF noise won't be there at it's output. A properly designed DAC will have internal DSD (FIR) and analog low pass filters. That's the way DSD works, the DAC as part of the 'reconstruction' chain attenuates it's HF noise as part of the process. Amir shouldn't be missing this stuff it's pretty basic digital audio design.
Am I mis-understanding the implementation of hi-res audio? I thought higher sample frequency meant more data points per second. Does higher sample rate only mean larger frequency spectrum at the same sample rate? I think I need a primer and example on how PCM sample frequency is implemented for 44.1 vs. 48, 96, 192 etc.
You have it mostly right. The thing is, you only need a sample rate 2x of a given frequency to capture it faithfully (that’s nyquist-shannon), so while higher sample rates do have more data points, there’s not necessarily more information there in the audible range. Bit depth is a different (and more straightforward) story however.
DSD is not PCM, it is PDM. Sample rate is directly related to the ability to correctly encode a given frequency range, which is half of the sample rate. Bit depth is how many levels of loudness that can be captured for each sample. The benefit of sample rates above 44.1khz os not that they capture the source sound more faithfully, but that when the digital file is played back and converted to analog the necessary filters used to band-limit the audio are both farther from the hearing range of humans and less steep. When the filters are close to our hearing range and very steep they can create artifacts that affect the reproduction of the audio in the system, though technically the correct digital information was always there. Very high sample rates are most useful when the music is being made, since it allows for mixing and mastering manipulations of the sound with minimal artifacts in the audible range. The benefit of high res audio for only playback is debated. Almost always 48khz sample rate is not noticeably worse than anything higher than it.
Good suggestion but we jeed a high-bandwidth microphone to do that which I don't have (tells you my priority as far as ultrasonics :) ). We would also have to assure the same master was used for both.
@@AudioScienceReview I am in no way an electrical engineer so forgive if it’s nonsense. If you ran the analyser in series with the drivers wouldn’t you see also higher order harmonics generated by the tweeter and the woofer?
Looks like all DSD does is just increase the file sizes with ultrasonic noise. Does it provide anything cleaner in the audible band range? Thanks for doing all the analysis on these types of downloads!!!
All nowadays digitized studio recordings of original analog tapes are DSD (sigma delta). 16 bit R2R DACs are dead now (resistors precision problems above 16bit)!!! DSD (sigma delta) does not know any 16, 20, 24 integer nor 32 flat bit quantisation, it is stupi 1-bit(or more) pwm. DSD (pwm) is for editing/commercial purposes latter converted to PCM 16/20/24-bit (32 float) stream. This (DSD) is the max digitized quality which you can get from music studios, there is no other better. Latter it is up to you which DAC with which (digital filter and analog filter) you will use. Some DACs have multiple filters and you can switch between them. That is why Audiophiles have PS Audio products. So I do not understand what is Amir complaining about?
So, you pay for downloaded hi res by file size and 2/3'rds of the file is noise out of the audible range? Could they filter it out before you pay for it? I realize it would reduce the profit margin considerably. I understand this stuff ok but 95% of what I listen to at home when I just sit and listen is LP's and maybe 5% CD.
almost any DAC anti aliasing filter is going to filter that ultra high out anyway and if it doesn't you bought a DAC filled with Voodoo and horsesh*t because there is 1. No reason to not filter those ultra high frequencies out since it if far beyond the detectable hearing range, and 2. Even if your tweeters could reliably reproduce those frequencies, you wouldn't want them to because... You cant hear or detect it anyway. It is wasted energy. In the air, humidity would dampen most of that above 30-40k to almost nothing after about 10 feet as well. not to mention how absurdly easy to scatter wavelengths that small are. It's just not there to human ears at all and does not affect what we can actually hear in any practical way.
I'm confused, are you analysing the digital samples of the music or a filtered analogue output? Digital samples will always have high spectrum components because they are level measurements of a very short time interval
These are digital analysis of the bits as delivered to you. In this case, the single track is 900 megabytes! I am showing what came in that file. You can do the analysis across the whole file if you like and get similar results. I am showing the real-time analysis which uses a window in the order of 1/10th of second which is till tens of thousands of samples but small enough to show the dynamic behavior. Spectrum leakage is very low in that regard.
Yes although as I have been explaining in other comments, the problem has been amplified greatly due to these high-rate DSD formats as used in this track.
Excuse me, but this is a big ripoff. You cannot convert an analog recording into hi-rez. You cannot gain any more resolution from an analog tape than what is on the tape. Amir, how many bits of information can be gotten from analog tape? 8bits? 10bits? 12bits? 14bits? To create hi def/hi rez, where are all these additional bits coming from? What bit rate do you get when you convert analog to digital? It falls short of even Red Book standard. To convert it to higher bit rates, there is no information in these bit buckets. You're taking up more space, allowing for info that's not there. Where are all these non existent bits coming from. Look at the MoFi class action lawsuit. They are selling analog recordings at hi rez prices. All people are getting is analog tape resolution. They're not magicians, can't pull bits that ain't there out of a hat. Selling analog tape @ hi rez prices. Selling you a 1960s Volkswagen @ Porsche prices, and taking huge file sizes with empty bits.
Unfortunately that was only true of SACD players playing SACD discs. These are digital downloads and today's DACs have far higher bandwidth and usually no control of filtering for DSD for end user.
Ahh yes, look at all that garbage. All that garbage that is pushed out of the audible range, which is what 1-bit audio is supposed to do. Odds are most don't have a tweeter that can reproduce 30-40kHz anyway and a well implemented DSD DAC starts filtering out ultrasonic information right around 50kHz. Not to mention these frequencies were the cutoff occurs by the filter on playback being somewhere in the order of -120 to -100dB unless your speaker is a piece of crap it's not gonna be a problem. If it was, people would be blowing and destroying tweeters left and right. Besides that's not even considering by the time we hit our 30th birthday's we've already lost a good 2kHz of hearing ability at the top end, 19/19.5kHz on a good day. Our ultrasonic hearing starts to degrade not long after you stop growing, it's called presbycusis and is natural and irreversible. Luckily our brain's internal EQ adjusts for this slowly overtime so we don't notice it. But by the time you are in your late 50s or 60s, most people can no longer hear above 16kHz. By 80 or 90 if you're lucky enough to live that long, 12-13kHz.
Out of curiosity I bought the lp (reissue, not expensive). Honestly? Mediocre small scale jazz with a public that prefers to make their own sounds with wine glasses and cutlery.
Statistically, the exponential drop should continue. Some content though may not follow that though. For this reason -- other than this one video -- I play the content so we can see if the waveform is dancing up and down. If it is, then it is "music," if not, it is not. Or at least is is buried in the noise floor.
How exactly is this a lie, this is exactly what is promised. You get a bit more that twice the frequency range of CD quality pcm and a higher bit depth. This is a very good example of what is possible in DSD. All the noise in the upper band is by design a feature of dsd, so that the noise is not in the audible band, it is meant to be filtered out.
After all of your wonderful high end DAC reviews, you don't have one capable of playing DSD's? I'm not questioning your conclusions but I'd like to see your analysis using track appropriate equipment. I have a couple of versions of Pawnshop in DSD256 that I play through a DX7 Pro with Burson Vivids in the pre-amp stage. I quite enjoy the sense of being in attendance at the live recording.
These are videos that I produced 2+ years ago. At that time, I was still using my favorite Mark Levinson DAC which had come out back in year 2000 and naturally didn't play DSD. I did have DSD capable DACs then but just didn't want to let go of my beloved ML DAC. :) I have since replaced it with a Matrix Audio DAC which of course plays all formats including DSD.
Amir, I appreciate the sentiment of what you are trying to accomplish but this seems an incomplete way to present what is going on not least the absence of a baseline to compare against, i.e. the same recording in Redbook. The danger here is that this is just going to further polarise an already increasingly toxic environment, not least as there will be other recordings that are better and best in class for the respective formats. People will take absolute positions based on these videos and I just don’t see how that can be helpful in the wider context. I am quite comfortable with the notion that there is little to no information pertaining to the music above 48 khz, at least MQA is quite open about this notion of the ‘golden triangle’ (I am not commenting about MQA itself but just the assertion that 0 - 48 khz is where the most attention should be applied). I understand these are older videos currently but hope that a new approach might be the way in future 🙏
@@xiaobaozha Most modern amplifiers have a power bandwidth that extends to 100 Khz. Left unfiltered this high frequency garbage will be amplified and fed to the speakers. Tweeters are not designed to handle this much energy, generally the amount of energy above 20 Khz is less than a few watts. If this garbage were fed to a tweeter at the power levels shown it would quickly overheat and fail.
@@xiaobaozha Yes it would be. This is a DSD256 file and your DAC will most likely output way into hundreds of kilohertz. We are not talking about SACD players.
@@AudioScienceReview my DAC is a Mytek Brooklyn that automatically selects the appropriate DSD filter according to the incoming DSD rate. What is that filter doing? According to the manual it is applying an IIR filter, low (47,44 khz), medium (60 khz) or high (70 khz).
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Something that needs to be understood is that the file is exactly what it should look like for DSD. The whole point of DSD is to use an extremely high sample rate - way above what even dogs hear - to allow for all of the noise to be shaped in such a way that the audio in and around the audible range can be extremely accurate while filters remove the noise in the ultrasonic band before it is amplified. This allows for very shallow filters near the audible band of frequencies, and thus very low distortion in what is sent to the speakers. DSD dacs filter all of the noise above about 50khz, whereas for, say, 44.1khz pcm, the filtering starts at 22khz. This results in a pcm equivalent (for dsd 64) of about 100khz sample rate with a bit depth of around 20bits. If you look at the spectrum the video, you will see that the noise shaping begins at 50khz, right where the DSD DAC filters start filtering higher frequencies.
Thanks for educating people here including Amir it seams, Kinda sad to read all this and they dont know how DSD works. Its also the timing that is so fantastic with DSD. Nothing can compare with it I think I even make DSD files of my vinyls with an ADC and it sounds exactly like the vinyl.
@@tweakknob I believe Amir said in the video that this is expected from a DSD file so to me it seems he understands this : ) Although yes I agree that Amir does sometimes skip or misunderstand a few things.
To me it also seems like a fun (weird) thing to capture old music on DSD since what capture is already being restricted to a certain frequency range. So what you add to the music is something that's not there.
But if you do enjoy it-I'm happy you do.
I explained that noise shaping was used as a result of DSD encoding so not sure why you say it was not understood. What you are not understanding is that this is NOT SACD. This is a DSD256 file at much faster sampling rate. The content is only played on a computer and as such, you are at the mercy of analog filtering in the specific DAC. I showed in the forum post that AKM AK4493 for example, has whopping 400 kHz bandwidth for DSD256! Not 50 kHz as you state. See: www.audiosciencereview.com/forum/index.php?threads/inside-high-res-music-jazz-at-the-pawnshop-naxos-dsd.23137/post-771685
It will even have respond as high as 600 kHz.
So please don't use your assumptions about DSD here. These are digitally downloaded files and you are paying for the bits literally and in download file size. This one file alone is nearly 900 Megabytes! Yes, this single file is 1.5 times an entire CD! This is due to high DSD rate which encodes even more noise than standard DSD.
@@tweakknob You are mistaken about both unfortunately. DSD has no advantage in timing. Just because it samples faster doesn't mean it has better timing. It has extremely shallow bit depth which sets the format's timing resolution. As to how it sounds, it has 100% to do with the mastering, not anything to do with the format which by itself is faulty compared to high-rate PCM. I would also not chime in after the above poster said everything is filtered above 50 kHz. If that is so, then it has no bandwidth advantage either! Nor is there value in pumping up noise above 20 kHz lest that is your idea of high fidelity. It not mine. :)
@@AudioScienceReview Thanks for the clarification. I thought I would chime in a little about DSD since many comments did not seem to understand the purpose of the loud noise above 50khz. If indeed DACs are not filtering frequencies above 50khz or so, then that is worrisome since it can negatively affect the speakers. I agree though that DSD’s benefits compared to PCM are tenuous at best, and that the file sizes are unreasonably large.
I just love the way you call then noise "pure garbage" haha! I don't know why but I find that so funny, even though it's true.
It's scientific terminology
If you have a group of bats picking on your window when listening to DSD files, now you know why : )
DSD is a natural format for delta-sigma DAC. So majority of DS will filter these high frequency noise, since they also have the same signature. This is due to how delta-sigma work.
@@bltzcstrnx actually most speakers can't reproduce frequencies that high anyway. but like he said in the video, if you have a wide band amp and its still pumping those frequencies into them, what's that doing to the tweeters? can't be good for them? especially at high volumes.
@@XX-121 speakers are passive device, it will reproduce whatever that is sent to them, at least it will try. Of course this won't be good and can be damaging. Hence competent amplifier will have both DC protection on the low-end of spectrum and low pass filter on the high-end of spectrum.
You are a hero of the audio comunity Amir! It would be super cool to see a comparison with lossy convertions as well, perhaps also a comparison to a vinyl hooked up to your Apx555 ? :)
Hi, I am the mastering engineer at 2xHD that did the transfer and mastering. The point of buying the DSD256 is to play it in DSD256. The DSD256 have no noise compare to lower DSD so the filter in the DAC that remove all that noise is very gentle. Important to know that if you analyze the analog output of a DSD DAC that noise that you taking about is not there because of the low pass filter. The resolution of 11.2.MHz sample every second really sound more sweet and close to the analog master compare to any PCM if you play it that right way. I did release also Jazz at the Pawnshop on Vinyl in all analog all tubes also on 2xHD. Rene Laflamme
Amir. Thank you for doing this.
I think you are uncovering a huge issue with ‘Hi-Res’ recordings.
For the average consumer there is no effective method for them to determine that they are getting what they pay for. And without such a method, there is zero incentives for the recording industry to clean up their act.
Given that I can’t identify a single difference between CD or hi-Res (and not for the lack of trying), you have finally given me the confidence that I am not missing out if I stick with 16bit/44.1kHz for music.
Yeh I feel bad myself since I spent thousands of dollars buying all these albums only to see so many issues in them. A quick analysis like I did is within the abilities of anyone producing such content yet it is not done. It is like shipping software without ever testing it. Or serving a dish in a restaurant without ever tasting it.
I believe DSD dacs have analog low pass filters on the outputs; I doubt any of the noise would make it through to the subsequent stages, like you imply. Thank you for your video, either way. Your expertise is appreciated.
It is usually an optional filter. That doesn't solve the problem of getting more noise in the form of larger file size to download and in some instances, paying extra for higher sample rates.
@@AudioScienceReview Fair enough--point taken!
Yes, the idea of DSD is that with the extreme sample rate the noise can be pushed way into the ultrasonic band, which as you can see increases in amplitude with increase of frequency, which allows for comparatively shallow filters to be used in the frequencies closer to what humans can hear, avoiding a pitfall of PCM recordings (steep filters), and increasing the steepness of the filters as the frequencies get farther and farther from human hearing.
@@LordAus123 Well, the notion of a "high-sample rate" format which fills the ultrasonics with noise and then proceeds to filter some of it, doesn't sit well with me. What possible benefit comes out of pumping noise into your system? And as I noted elsewhere, your assumptions about this file are incorrect. This is not SACD/DSD. These are higher rate DSD256 played on a computer, not SACD player. As such, there is no typical filter for them. I showed in the ASR thread for example that the AKM AK4493 DSD filter goes to 400+ kHz and above! So what you see in my analysis, is all pushed into your audio system. There is no need at all to do that with 24-bit, PCM format.
So why force people to download 80% more data than needed? It's pure garbage, both in the file and in the business ethics imo.
This album was recorded on a Nagra IV in 1976 and first transferred to digital in 1996. It has since been subjected to countless reissues and different iterations including a UHD 32 bit version and a remaster by Doug Sax. Somehow I doubt that the analog master was used.
Indeed with no provenance for these high-res releases, one can never know how the bits came about.
The recording itself is epic anywya
Thanks Amir! Show us some positive examples... 😜
yeah It would be interesting to know a "HiRes" version actually better than its redbook conterpart.
Dream Theater - A Dramatic Turn of Events is a very well recorded 96kHz album with audio signals up to the 30kHz range, but doubtful I can hear any difference from the CD.
I can't yet since I prepared these a couple of years ago and am just posting them. Once I post all of them, I will create some new ones that show the format in better light. :)
You can only listen to DSOTM so many times before it gets tiresome.
I hope you know that any half decent DSD capable DAC (and SACD Player) will cut away the ultrasonic noise you are showing in you computer...
This video reminds me a story from early 1990's. I read an article in a newspaper, where an audiophile claimed that he heard 'wow and flutter' from CD. It turned out the CD was mastered from analog tape recording. THe digital audio CD able to reproduces 'everything' including flaws and garbage of the original analog tape recording.
It is wise to find a proper full digital recording, mixing and mastering.
Thank you.
Great story! Indeed digital captures of LP are as good as the LP itself.
In the early days most cds were just remastered analog tapes with often a huge treble boost added. Sometimes they were painful to listen to.
@@dingdong2103
Agreed, and there were people thought the CD/digital audio was bad.
@@AudioScienceReview Have you ever experienced the laser tracking LP player where no physical needle is used? :)
Thanks for in depth analysis Amir! To me, you are like Audiophile Mythbuster!!
If possible, can you please analyze a FLAC from Reference Recordings, highly praised by PSAudio / Paul McGowan ?
I was just looking at one of RR's album last night as Keith Johnson is a friend. I will do a video on that.
His quality comes more from excellent mastering, loads of dynamic range (I don't think he even uses compression at all), and the use of a Pacific Microsonis Model 2 as the A/D converter. Nothing to do with this hi-res non-sense, even though he gives you such downloads.
@@AudioScienceReview Have you done the Reference Recordings analysis yet?:)
It's honestly disappointing. From what I'm seeing here with most of the measurements, a really good CD, for the most part, is all we need.
Ok, so blended whiskey is all you need....
Alright, than ill take the original single malt whiskey which tastes far better.
pretty much. one that's properly mastered. actually i wish HDCD would've caught on all those years ago. Not only do they sound superb when properly decoded, but they were also the answer to the loudness war with extra, up to 6db of headroom they had at their disposal. i think the only time it was used like that was on a Green Day album tho.
Hey Amir! I am a frequent viewer of your channel and enjoy much of your content. I think you provide a valuable resource for the audiophile community by providing measurements for DACs/AMPs that we can’t listen to or demo ourselves before making a buying decision. So I really appreciate what you do. Having said that, I have to push back on everything you said on this video. First of all this video isn’t really about this specific recording but about the DSD format in general. I don’t think you really understand how DSD works. It was designed by Sony/Phillips to replace the original PCM CD format which they also invented. You have the same group of engineers who made PCM make DSD. Also DSD unlike PCM is a 1 bit format. What DSD does is it basically takes all the best aspects of PCM and all the best aspects of analogue and combines them in a digital format. You get the clarity and dynamic range of digital with the smoothness of analogue without all the detractors of both formats.It really is a FAR superior format over any other available today. Also in this specific case, your analysis is total garbage. You take your DSD track and import it into Adobe and say “look at all this garbage above the listening range” etc... This shows your ignorance that you really don’t have any idea of what you are talking about. DSD is designed to spread itself over a very high frequency band because it is a 1 bit format. Furthermore you go on to say “I don’t have a DAC that I can use to listen to this so I will just convert this back into PCM and do my analysis on that instead...” Well that defeats the whole purpose of this video does it not? We are talking about a high res DSD file format recording here and you are poo-pooing it when you can’t even listen to it and you can’t properly analyze it because you clearly don’t understand how it works. Normally when you do your measurements and analysis on this channel on say a DAC or something it makes scientific sense, but in this case it’s total garbage and you are misleading your viewers and pandering to the naysayers who seem to frequent your channel. I suggest you do a deep dive and learn more about the engineering specifics of DSD and get a DAC like say the Kitsume Holo May DAC that can actually play back DSD before making a judgement and misleading your viewers and fans. Cheers!
The hubris of this guy is astounding. He somehow believes that thousands of other engineers that actually make high end audio gear are all crooks or stupid! His claim to fame was doing telephone systems with a cord - not even wireless. So, if others want to listen to how awesome his phone boards are then I suggest they buy a high end phone with speaker. As for me, I will continue to enjoy my hifi without listening to imposters like this joker.
Totally agree. Well said Brent.
Just enjoy the music. Thanks for the analysis as usual.
Enjoy the music but maybe just pick up the CD for a fiver, not this 31 Euro nonense with 95% inaudible random numbers.
If every Dsd compatibile dac has filter (or filters to choose) somewhere between 30-50khz how this noise could be transfered into system ?????????
Main profits of new digitalizations is that they are made on modern equipment which has much better specs then previous attempts , mostly feom the 80s.
There is no such thing as "compatible DAC." You have no idea what the filtering is in the DAC you are using. Indeed as I showed in an example I looked up of AKM's AK4493, the bandwidth for DSD256 which this file is, exceeds *400 kHz* !. Not 30 to 50 kHz. See this post: www.audiosciencereview.com/forum/index.php?threads/inside-high-res-music-jazz-at-the-pawnshop-naxos-dsd.23137/
I didn't realize there is this much confusion between SACD as a physical format played on SACD player of years back, and today's high-rate DSD downloads. They are completely different animals. The latter is played on a computer and at very high bandwidth. Indeed, customers of such formats demand the high bandwidth or they would not chase bigger and bigger DSD numbers. We are now up to DSD1024 with the original SACD version being just DSD64.
But yes, we can do a bitter job digitizing tapes now but the proper format for that would be PCM with objective analysis like i am doing to make sure we are not just manufacturing noise and shipping it to people. As I noted elsewhere, this one file is 900 Megabytes! Vast amount of what is there is ultrasonic noise with no value.
@@AudioScienceReview i guess you are using Rme Adi2 . Look at user manual. You can choose between 30khz or 50khz LPF during DSD playback. Also you can check datasheet for any DAC chip with DSD support , it's always shows schematics for the output stage for dsd playback that cuts high freqs at some point . So non of this noise will ever get into amp or speakers.
I can't understand high bandwith dsd download, especially then it's done by "upsampling". Nativedsd web store is full of such downloads.
I find it funny when they offer the remasters of older music in SACD/DSD or 24bit/96khz. From what I can find most of the great music from the 60s and 70s was recorded on 2" tape that was capable of about 70db dynamic range and 18 to 20khz. The mics used back then also were not capable of capturing this. Sometimes these remasters do sound better to me, BUT my gut tells me its because of the remaster and a competent engineer and not so much the hi-res format.
I hear you. :) We certainly are limited by the dynamic range of the tape format. There is more spectrum to be had though as the analog formats were not brickwalled so to the extent someone wants them, there should be a way to delivery them. Personally I like to see high-res format be about non-loudness-compression where higher bit depth and sample rates come as bonus.
Thanks Amir - I'm kind of confused with what you're showing us - - I've always had the "layman" kind of idea that yes, while these higher sample rates go beyond our normal hearing, I thought the primary idea was to sample the sounds more frequently during any given moment in time - in order to achieve a higher resolution of sound, like more pixels per square inch in a picture - so - I guess I'm really not understanding everything involved with these higher resolution recordings - - that is, in that all that noise content that come with these higher sample rates, at least with the ones you've shown us so far, would seem to do more harm than good. I hope you show us more, I would like to understand it all a bit better.
If you just use higher sample rate, you get wider bandwidth. NOT higher resolution. If you limit the bandwidth to say, 20 kHz, and then keep increasing the bandwidth, you do gain some resolution in the way you intuit. Indeed DSD works this way by sampling at very high frequency and just one bit of depth. PCM downloads though just keep increasing the bandwidth with higher sample rate since bit depth is 24 bits anyway so doesn't need help in that front.
And sorry about these videos not having as much tutorial value as my newer ones. I created these a couple of years ago with a different intent and style.
High sample rates mean you can potentially build a more linear analogue output stage on your dac that will result in better quality sound. But a dac can do this internally as well with digital algorithms.
@@AudioScienceReview Thanks Amir - I appreciate you taking the time to explain a bit more - Skip
Do you know anyone that actually coocked tweeters or made any damage by playing high sample rate DSD files?
Well, I've heard of a noise floor. Now we have a noise ceiling. Our house of noise wouldn't be complete without it ;)
what you call garbage is what others calls a live atmosphere
The snake oil SLAYER strikes again. Please, keep them coming. I know nothing about audio, half of the things they say does not mean anything to someone as me "noob' Thanks for this awesome channel.
Will do. Thanks Beni.
This is even more fascinating than face value. Unless there was no bias tone on the recording tape deck, you would see a very large spike at an ultrasonic frequency. Maybe 75 or 100 kHz. Unless if course it's just covered by that horrendous hf noise of the dsd. Still, even a simple first order low pass filter would do a lot here. Yikes.
That's a very good point Sam! I wonder if this file then comes from another capture first that had band limiting.
Your analysis is based on digital domain of the file. On the analog domain all the high frequency should be filtered by a lowpass, on a medíocre designed dac, this should not happen .
No such thing happens in "analog domain." As I showed in the ASR thread, the playback bandwidth of today's modern DACs is 100s of kHz for these high rate DSD files. The whole point of formats like DSD256 is to give audiophiles more spectrum so filtering them like SACD did as you think, would go against the very value proposition of these formats.
@@AudioScienceReview Well these DACs are not correctly designed for DSD. The DAC *IS* part of the playback chain in DSD and a correctly designed one
will attenuate the HF noise with FIR DSD filter and at least 3rd order analog (linear phase) LPF.
@@AudioScienceReview Im sorry, but I did not read the thread, but I will. How can a noise shapped signal not be filtered? So. supposing we have a 70KHz usable signal on a DSD detected signal by the DAC, lets say 256fs, you are stating that the engineers went loose and all the cut off frequency, -30db/octave FIIR lowpass do not work and they are not complying with the scarlett book. Thats reassuring to know a highly noise shaped PDM HF´s product is making all the region bats nuts and generating a hell of instability on my Amplifier
Thank you.
BIG DSD [SA-CD] fan [not as much as DVD-A/though that doesn't matter now]. Always told to avoid DSD to PCM conversion; now will do it.
Thanks for showing this. I was just smiling through the whole video.
Not surprised, this is what you get when you digitize studio tape
Only a small amount of that has to do with analog tape capture. All that ultrasonic noise shaping is due to the digital format in use, not analog capture.
I'm confused. You said that Audition will not play the file (I believe that to be correct) and yet you read the file into Audition. So what is Audition showing you? I think it's just showing you the DSD data but with some spurious mapping to PCM. Or did you set up Audition to somehow understand the 1-bit nature of DSD, in which case the filtering that is necessarily applied to DSD decoding is missing? Either way, I don't believe Audition is showing what the DSD stream is meant to represent. Or have I misunderstood this completely? That is also a distinct possibility.
Audition can open and and analyze the file but it lacks the output capability to play it to the DAC. Playing DSD requires a pipeline to either encapsulate it over PCM or through ASIO. Audition doesn't support either.
@@AudioScienceReview OK - thanks.
This should be clearly labelled as NOT a high resolution recording but a DSD encoding of “what exactly was the original ? “ Amir mentioned that he’s assuming that this is of the master tapes....
Yes, I would like to have some music with my high-resolution noise.
Best comment thus far. 😊
Last week I have converted all my DSD files ( about 50 albums ) to WAV 24/96 , 🙂
Were you able to just rip it over or did you have to do it in real time?
What sense does it make?
@@kopczas yes, one more transcode step, a lot of software can do it on the fly
Amir might debunk this, but from what I understand it's better to convert DSD to 24/88 files because they are natural multiples and converting it to 24/96 can introduce quantization errors or noise or something.
Amir, it would be interesting to see these tracks compared to an example where the unwanted noise is indeed not present.
This video is misleading. If the DAC is designed correctly, most of that HF noise won't be there at it's output. A properly designed DAC
will have internal DSD (FIR) and analog low pass filters. That's the way DSD works, the DAC as part of the 'reconstruction' chain attenuates it's
HF noise as part of the process. Amir shouldn't be missing this stuff it's pretty basic digital audio design.
Very interesting! I have 192/24 pcm soundgarden superunknown -album. It shows exactly similar noise in the upper region. DSD source? Hmm...
Yes, often the capture is DSD and then a PCM conversion is done with who knows what ultrasonic filtering.
Am I mis-understanding the implementation of hi-res audio? I thought higher sample frequency meant more data points per second. Does higher sample rate only mean larger frequency spectrum at the same sample rate? I think I need a primer and example on how PCM sample frequency is implemented for 44.1 vs. 48, 96, 192 etc.
You have it mostly right. The thing is, you only need a sample rate 2x of a given frequency to capture it faithfully (that’s nyquist-shannon), so while higher sample rates do have more data points, there’s not necessarily more information there in the audible range. Bit depth is a different (and more straightforward) story however.
DSD is not PCM, it is PDM. Sample rate is directly related to the ability to correctly encode a given frequency range, which is half of the sample rate. Bit depth is how many levels of loudness that can be captured for each sample. The benefit of sample rates above 44.1khz os not that they capture the source sound more faithfully, but that when the digital file is played back and converted to analog the necessary filters used to band-limit the audio are both farther from the hearing range of humans and less steep. When the filters are close to our hearing range and very steep they can create artifacts that affect the reproduction of the audio in the system, though technically the correct digital information was always there. Very high sample rates are most useful when the music is being made, since it allows for mixing and mastering manipulations of the sound with minimal artifacts in the audible range. The benefit of high res audio for only playback is debated. Almost always 48khz sample rate is not noticeably worse than anything higher than it.
It would be nice to run a track, DSD and RedBook, through a whole system and analyse the spectrum coming from the tweeter and woofer.
Good suggestion but we jeed a high-bandwidth microphone to do that which I don't have (tells you my priority as far as ultrasonics :) ). We would also have to assure the same master was used for both.
@@AudioScienceReview could you attach your spectrum analyser in series to each driver?
@@volpedo2000 You mean post crossover but before the driver? If so, yes, I can do that but requires more effort than my lazy self allows. :)
@@AudioScienceReview I am in no way an electrical engineer so forgive if it’s nonsense. If you ran the analyser in series with the drivers wouldn’t you see also higher order harmonics generated by the tweeter and the woofer?
@@AudioScienceReview I appreciate it’s a faff! 😊
Looks like all DSD does is just increase the file sizes with ultrasonic noise. Does it provide anything cleaner in the audible band range? Thanks for doing all the analysis on these types of downloads!!!
It has the potential to provide > 16 bits of dynamic range in the audio band. Alas, this is very difficult to measure.
All nowadays digitized studio recordings of original analog tapes are DSD (sigma delta). 16 bit R2R DACs are dead now (resistors precision problems above 16bit)!!!
DSD (sigma delta) does not know any 16, 20, 24 integer nor 32 flat bit quantisation, it is stupi 1-bit(or more) pwm.
DSD (pwm) is for editing/commercial purposes latter converted to PCM 16/20/24-bit (32 float) stream.
This (DSD) is the max digitized quality which you can get from music studios, there is no other better.
Latter it is up to you which DAC with which (digital filter and analog filter) you will use. Some DACs have multiple filters and you can switch between them.
That is why Audiophiles have PS Audio products.
So I do not understand what is Amir complaining about?
So, you pay for downloaded hi res by file size and 2/3'rds of the file is noise out of the audible range? Could they filter it out before you pay for it? I realize it would reduce the profit margin considerably. I understand this stuff ok but 95% of what I listen to at home when I just sit and listen is LP's and maybe 5% CD.
this is wild
thanks Amir AWESOME video!, any other (cheaper) software to analyze audio like this?
The free Audacity software can also show spectrum although a while ago I found some oddities in it.
There is free tool called "Spek". It will build you a similar graph, give it a try if you feel
Oh, just realized that you may be asking about DSD analysis in which case, your choices are far more limited.
Watch the film "The Menu" and guess which character you are
almost any DAC anti aliasing filter is going to filter that ultra high out anyway and if it doesn't you bought a DAC filled with Voodoo and horsesh*t because there is 1. No reason to not filter those ultra high frequencies out since it if far beyond the detectable hearing range, and 2. Even if your tweeters could reliably reproduce those frequencies, you wouldn't want them to because... You cant hear or detect it anyway. It is wasted energy. In the air, humidity would dampen most of that above 30-40k to almost nothing after about 10 feet as well. not to mention how absurdly easy to scatter wavelengths that small are. It's just not there to human ears at all and does not affect what we can actually hear in any practical way.
Looks like bitstream, I think all that filtering and noise is why many prefer R2R.
I love my R2R because I love looking at the wheels spin. :) Nothing matches that pleasure. And it does sound vey good despite its flaws.
I'm confused, are you analysing the digital samples of the music or a filtered analogue output? Digital samples will always have high spectrum components because they are level measurements of a very short time interval
These are digital analysis of the bits as delivered to you. In this case, the single track is 900 megabytes! I am showing what came in that file. You can do the analysis across the whole file if you like and get similar results. I am showing the real-time analysis which uses a window in the order of 1/10th of second which is till tens of thousands of samples but small enough to show the dynamic behavior. Spectrum leakage is very low in that regard.
Would the same issues apply to SACD?
Yes, SACD has DSD.
Yes although as I have been explaining in other comments, the problem has been amplified greatly due to these high-rate DSD formats as used in this track.
Excuse me, but this is a big ripoff. You cannot convert an analog recording into hi-rez. You cannot gain any more resolution from an analog tape than what is on the tape. Amir, how many bits of information can be gotten from analog tape? 8bits? 10bits? 12bits?
14bits? To create hi def/hi rez, where are all these additional bits coming from? What bit rate do you get when you convert analog to digital? It falls short of even Red Book standard. To convert it to higher bit rates, there is no information in these bit buckets. You're taking up more space, allowing for info that's not there. Where are all these non existent bits coming from.
Look at the MoFi class action lawsuit. They are selling analog recordings at hi rez prices. All people are getting is analog tape resolution. They're not magicians, can't pull bits that ain't there out of a hat. Selling analog tape @ hi rez prices. Selling you a 1960s Volkswagen @ Porsche prices, and taking huge file sizes with empty bits.
I would like to see a dsd that was originally recorded in dsd and not a transfer.
PS Audio's Octive Records does just that.
Never understood Audiophile label of this album...
Most players cuts above 30-50 kHz, or should IMHO.
Unfortunately that was only true of SACD players playing SACD discs. These are digital downloads and today's DACs have far higher bandwidth and usually no control of filtering for DSD for end user.
AK4490 has a filter for DSD at 50khz by default and normally there is a LPF after the dac chip.
But typically there is no choice by end user indeed.
Ahh yes, look at all that garbage. All that garbage that is pushed out of the audible range, which is what 1-bit audio is supposed to do. Odds are most don't have a tweeter that can reproduce 30-40kHz anyway and a well implemented DSD DAC starts filtering out ultrasonic information right around 50kHz. Not to mention these frequencies were the cutoff occurs by the filter on playback being somewhere in the order of -120 to -100dB unless your speaker is a piece of crap it's not gonna be a problem. If it was, people would be blowing and destroying tweeters left and right. Besides that's not even considering by the time we hit our 30th birthday's we've already lost a good 2kHz of hearing ability at the top end, 19/19.5kHz on a good day. Our ultrasonic hearing starts to degrade not long after you stop growing, it's called presbycusis and is natural and irreversible. Luckily our brain's internal EQ adjusts for this slowly overtime so we don't notice it. But by the time you are in your late 50s or 60s, most people can no longer hear above 16kHz. By 80 or 90 if you're lucky enough to live that long, 12-13kHz.
Out of curiosity I bought the lp (reissue, not expensive). Honestly? Mediocre small scale jazz with a public that prefers to make their own sounds with wine glasses and cutlery.
So what should we be seeing on those charts after 25khz? A drop off?
Statistically, the exponential drop should continue. Some content though may not follow that though. For this reason -- other than this one video -- I play the content so we can see if the waveform is dancing up and down. If it is, then it is "music," if not, it is not. Or at least is is buried in the noise floor.
@@AudioScienceReview thank you so much
Are we being sold a lie with DSD or is it just this case?
How exactly is this a lie, this is exactly what is promised. You get a bit more that twice the frequency range of CD quality pcm and a higher bit depth. This is a very good example of what is possible in DSD. All the noise in the upper band is by design a feature of dsd, so that the noise is not in the audible band, it is meant to be filtered out.
It is not a lie, but selling fake dsd, fake pcm HD is a big lie
@@Lesterandsons yes but this isn't fake.
After all of your wonderful high end DAC reviews, you don't have one capable of playing DSD's? I'm not questioning your conclusions but I'd like to see your analysis using track appropriate equipment. I have a couple of versions of Pawnshop in DSD256 that I play through a DX7 Pro with Burson Vivids in the pre-amp stage. I quite enjoy the sense of being in attendance at the live recording.
These are videos that I produced 2+ years ago. At that time, I was still using my favorite Mark Levinson DAC which had come out back in year 2000 and naturally didn't play DSD. I did have DSD capable DACs then but just didn't want to let go of my beloved ML DAC. :) I have since replaced it with a Matrix Audio DAC which of course plays all formats including DSD.
Amir, I appreciate the sentiment of what you are trying to accomplish but this seems an incomplete way to present what is going on not least the absence of a baseline to compare against, i.e. the same recording in Redbook. The danger here is that this is just going to further polarise an already increasingly toxic environment, not least as there will be other recordings that are better and best in class for the respective formats. People will take absolute positions based on these videos and I just don’t see how that can be helpful in the wider context. I am quite comfortable with the notion that there is little to no information pertaining to the music above 48 khz, at least MQA is quite open about this notion of the ‘golden triangle’ (I am not commenting about MQA itself but just the assertion that 0 - 48 khz is where the most attention should be applied). I understand these are older videos currently but hope that a new approach might be the way in future 🙏
I'm going to respectfully suggest to you that the objective presentation has tainted your subjective listening experience. This album is pure magic.
This would actually be dangerous to play through your amplifier and speakers unless bandwidth limited as Amir suggested.
No, it would not be dangerous as this is not the analogue output your amplifier would see.
@@xiaobaozha depends on software and dac filters
@@xiaobaozha Most modern amplifiers have a power bandwidth that extends to 100 Khz. Left unfiltered this high frequency garbage will be amplified and fed to the speakers. Tweeters are not designed to handle this much energy, generally the amount of energy above 20 Khz is less than a few watts. If this garbage were fed to a tweeter at the power levels shown it would quickly overheat and fail.
@@xiaobaozha Yes it would be. This is a DSD256 file and your DAC will most likely output way into hundreds of kilohertz. We are not talking about SACD players.
@@AudioScienceReview my DAC is a Mytek Brooklyn that automatically selects the appropriate DSD filter according to the incoming DSD rate. What is that filter doing? According to the manual it is applying an IIR filter, low (47,44 khz), medium (60 khz) or high (70 khz).