mansr Posted May 2, 2017 Share Posted May 2, 2017 8 hours ago, tailspn said: As formats, DSD has +6dB greater dynamic range than PCM. DSD and PCM are digitally storable formats to record and transmit analog signals, mostly audio for our interest. 0dB is a definition in recording to signify a maximum level not to be exceeded. Below 0dB, both PCM and DSD are capable of the same signal level range, better known as dynamic range. Both can express an infinitely small signal, far below the practical minimal signal transferable by practical analog electronic circuits. In the case of PCM, 0dB represents the maximum binary range that the the format can represent; there's no bits left. It's a 2's compliment binary word for each sample that's either all 1's, or all 0's at 0dB (full scale). It's 2's compliment to support both positive and negative values. In DSD, there are no values represented. It's the density modulation of a bit clock, who's percent of modulation is proportional to the signal level, not a numerical value of a signal level (at an instant in time). All of that is background to the fact that from the lowest signal level deliverable to a A/D converter to the maximum that produces 0dB, the same dynamic range is expressible in both a PCM or DSD format up to 0dB. The DSD format however has an additional +6dB of headroom above 0dB that is not achievable with PCM. That's because DSD 0dB is specified as 50% modulation, allowing an additional 6dB of signal level to be represented. DSD requires the extra headroom to prevent the sigma-delta modulators destabilising, so it's not actually available to encode the signal. Moreover, to compare dynamic range of formats, you need to look at both the high and low end. With PCM, the range between the highest and lowest level representable is trivial to calculate, and it increases with 6 dB for every added bit. Since the quantization noise in PCM is evenly distributed over all frequencies, doubling the sample rate also lowers the noise floor by 6 dB. With DSD it's not so simple. The noise level in the audio band depends primarily not on the digital format but on the quality of the noise shaping filter, and no matter how good it is, there will always be some residual noise as no filter has infinite stop band rejection. In practice, both PCM and DSD are capable of representing the full signal range between the analogue noise floor and the maximum analogue level. The 6 dB headroom of DSD is an implementation detail and should be ignored for the purposes of this discussion. Link to comment
Popular Post tailspn Posted May 2, 2017 Popular Post Share Posted May 2, 2017 4 hours ago, mansr said: In practice, both PCM and DSD are capable of representing the full signal range between the analogue noise floor and the maximum analogue level. The 6 dB headroom of DSD is an implementation detail and should be ignored for the purposes of this discussion. I agree about the capability of both representing the full signal range, but for me as a recording engineer, the practical application is the most important difference. The performance of currently available professional quality A/D converters yields better than 145dB dynamic range from its quantization noise floor to 0dB full scale. Since there are no direct PCM A/D converters (all are Sigma-Delta Modulator front ended), PCM is derived from the 1-bit, or more likely multi-bit Pulse Density Modulated bit streams from the modulator(s), and therefore can not possibly possess greater dynamic range. To your point though, today's recording systems, regardless of whether outputting DSD or PCM, they have a greater dynamic range that the analog signals being fed them. Actually, by 20 to 30dB. So the positioning of the signal dynamic range within the recorders dynamic range is a recording/mastering engineer choice. The practical difference between DSD and PCM affecting that signal placement is the distortion performance when approaching the loudest recorded signal. PCM hard clips at 0dB with minimal distortion. DSD soft clips, like tape, with the 0dB level specified halfway up the modulation scale (50%), and the distortion performance still very acceptable. But like tape saturation, the distortion artifacts of DSD increase rapidly when approaching 100% modulation (+6dB) The +6dB headroom offered by DSD versus PCM is not a measure of DSD superiority, or an "implementation detail", but a practical consideration when making music recordings. Recording and mastering engineers aim at crowding 0dB to obtain the maximum loudness. In acoustic music recording, the ability to overshoot in DSD by almost double the signal level on occasional music peaks is a big asset. It's like having the effect of compression without having to compress. The lowest level signals can be effectively presented at a higher level without clipping the highest level. The DSD +6dB headroom is very much available and used in wide dynamic range acoustic music recording, although most recording/mastering engineers will avoid exceeding +3dB. orresearch, jabbr and blue2 3 Link to comment
audiventory Posted May 3, 2017 Share Posted May 3, 2017 On 02.05.2017 at 6:06 AM, tailspn said: As formats, DSD has +6dB greater dynamic range than PCM. I reduced it to 0.4 dB. Though in the settings possibly add headroom value: AuI ConverteR 48x44 - HD audio converter/optimizer for DAC of high resolution files ISO, DSF, DFF (1-bit/D64/128/256/512/1024), wav, flac, aiff, alac, safe CD ripper to PCM/DSF, Seamless Album Conversion, AIFF, WAV, FLAC, DSF metadata editor, Mac & WindowsOffline conversion save energy and nature Link to comment
elcorso Posted May 3, 2017 Share Posted May 3, 2017 18 minutes ago, audiventory said: I reduced it to 0.4 dB. Though in the settings possibly add headroom value: The reason? In the posible settings, are those minus, like in -6 dB ? Thanks, Roch Link to comment
audiventory Posted May 4, 2017 Share Posted May 4, 2017 21 hours ago, elcorso said: The reason? In the posible settings, are those minus, like in -6 dB ? Thanks, Roch Hi Roch, -0.4 dB used for maximal amplitude using. It was accepted by number of test files (test signals and different musical ones). This headroom provide stability for known me audio stuff (that was converted since 3-rd generation of filters was released). If used wide filter band mode (Settings > General > Filter mode "non-optimized wide resampling filter"), there is higher probability of break stability due overload. Especially for DSD 64. Though I still don't got reports about break stability issue since wide mode was released. If output file contains silence or constant oscillation, need to increase attenuation. I suppose, 1 dB must cover all cases. For more fine tuning 0.1 dB may be used Main window > Levels > Output file volume slider (if it is available in used edition/configuration). If you want to convert DSF files with "standard" headroom, you can set attenuator to -6 dB. Best regards, Yuri P.S. Besides amplitude limitation, other efforts for modulator-stability improving are used in AuI ConverteR. PAP 1 AuI ConverteR 48x44 - HD audio converter/optimizer for DAC of high resolution files ISO, DSF, DFF (1-bit/D64/128/256/512/1024), wav, flac, aiff, alac, safe CD ripper to PCM/DSF, Seamless Album Conversion, AIFF, WAV, FLAC, DSF metadata editor, Mac & WindowsOffline conversion save energy and nature Link to comment
elcorso Posted May 4, 2017 Share Posted May 4, 2017 7 hours ago, audiventory said: Hi Roch, -0.4 dB used for maximal amplitude using. It was accepted by number of test files (test signals and different musical ones). This headroom provide stability for known me audio stuff (that was converted since 3-rd generation of filters was released). If used wide filter band mode (Settings > General > Filter mode "non-optimized wide resampling filter"), there is higher probability of break stability due overload. Especially for DSD 64. Though I still don't got reports about break stability issue since wide mode was released. If output file contains silence or constant oscillation, need to increase attenuation. I suppose, 1 dB must cover all cases. For more fine tuning 0.1 dB may be used Main window > Levels > Output file volume slider (if it is available in used edition/configuration). If you want to convert DSF files with "standard" headroom, you can set attenuator to -6 dB. Best regards, Yuri P.S. Besides amplitude limitation, other efforts for modulator-stability improving are used in AuI ConverteR. Many thanks Yuri for the detailed explanation ! Best, Roch audiventory 1 Link to comment
The_K-Man Posted September 26, 2017 Share Posted September 26, 2017 On 4/25/2017 at 4:38 PM, The Computer Audiophile said: I didn't mean to come across as not nice. Perhaps it was read that way. I just wanted to state the facts :~) You were absolutely right, and none too hurtful! Mastering(and to a big extent mixing) plays a far more significant role in how the finished product sounds than does format. Comparing 8-Track Cartridge to DSD? Then that poster of "DSD wins" might have a case! lol But PCM(red book) vs DSD? Splitting electrons. ? PAP 1 Link to comment
PAP Posted January 1, 2018 Share Posted January 1, 2018 On 25-4-2017 at 9:41 PM, The Computer Audiophile said: Glad I took a screenshot. My comment has been removed. why would they do that Link to comment
barrows Posted April 3, 2018 Share Posted April 3, 2018 On 5/2/2017 at 6:25 AM, mansr said: In practice, both PCM and DSD are capable of representing the full signal range between the analogue noise floor and the maximum analogue level This is the key point for me. At best, in room during playback we might have 100 dB of DR. And that is a very, very quiet room with a superb system. So the original idea that 16 bits is enough (~96 dB) is mostly correct: of course it is nice to have a bit of headroom, and for recording, more bits are necessary for mixing without loss, making recording at 24 bits totally sensible. Everyone who thinks about this should play a -90 dB signal in their system (with their volume control set to their normal "loud" critical listening level) and get an idea of how low in level -90 dB is. Stereophile Test CDs have a -90 dB test tone which makes this easy. I love DSD, and convert all PCM to DSD 128 or 256 during playback with Audirvana Plus (I feel my ESS based DACs sound better with DSD input). But my love for DSD has nothing to do with DR... SO/ROON/HQPe: DSD 512-Sonore opticalModuleDeluxe-Signature Rendu optical with Well Tempered Clock--DIY DSC-2 DAC with SC Pure Clock--DIY Purifi Amplifier-Focus Audio FS888 speakers-JL E 112 sub-Nordost Tyr USB, DIY EventHorizon AC cables, Iconoclast XLR & speaker cables, Synergistic Purple Fuses, Spacetime system clarifiers. ISOAcoustics Oreas footers. SONORE computer audio | opticalRendu | ultraRendu | microRendu | Signature Rendu SE | Accessories | Software | Link to comment
DavidFaik Posted January 3, 2019 Share Posted January 3, 2019 Would love to hear more about the subjective or experienced side of DSD to compliment the technical comparisons in this thread. In order to get an idea of what others are finding. I'm new to high res music, having enjoyed Qobuz for about two months on their high rez subscription. I have a Chord Qutest DAC that is DSD capable, so wanted to try some DSD content top evaluate if there are sonic differences that I can perceive. I have downloaded a trial of Audirvana. I then found via search engine that Blue Coast Music offers a free download of a well recorded track in multiple formats for evaluation, and I hope I am not breaking forum rules by including that link ( https://bluecoastmusic.com/free-downloads#.XC31CPx7mmk ). I did not do blind testing, but I found it hard to tell much difference between the high rez FLAC file and the WAV file, but think that the WAV may have been a little more airy, that said I'd happily accept that this could be placebo as I know that WAV isn't compressed. I would say categorically that that the two DSD formats did *NOT* sound better at all. After playing with all the PCM upsampling in Audivarna I found in my system and with my ears that all degraded the sound quality with the exception of the setting 2X upsampling Izo filter (note not power of 2, but flat 2X). Upsampling PCM to DSD that my DAC is most capable of receiving made the sound quality worse IMHO. For straight DSD file playback I tried both the 1.0 and 1.1 DSD transfer setting, but not the ones converting DSD to PCM. To expand on what I mean by stating "did *NOT* sound better" with DSD: 1. There was less width 2. There was less volume difference between the peaks and the lows 3. Oddly enough, my volume control on the pre-amp had less apparent effect (I did check that my Mac volume was at Max and that Audirvana was at 0db) My full main system details are in my profile. While far from the last word in audiophile I think the rig is OK, and IMHO musically revealing. If any have time to conduct a similar subjective test then I'd be grateful to read about your findings. Likewise any thoughts very welcome on whether this might be the limitations of the Audirvana player. Further suggestions of other music tracks in multiple formats that could be used to test in a similar fashion, preferably samplers / free. Link to comment
barrows Posted January 3, 2019 Share Posted January 3, 2019 @DavidFaik, The Qutest converts incoming DSD to PCM (although at a still high rate, I believe 705.6) so it is probably not the best converter to use for DSD vs. PCM comparisons. Just something to consider... DSD may sound better with a DAC which does not decimate (convert to lower sample rate) the incoming DSD. SO/ROON/HQPe: DSD 512-Sonore opticalModuleDeluxe-Signature Rendu optical with Well Tempered Clock--DIY DSC-2 DAC with SC Pure Clock--DIY Purifi Amplifier-Focus Audio FS888 speakers-JL E 112 sub-Nordost Tyr USB, DIY EventHorizon AC cables, Iconoclast XLR & speaker cables, Synergistic Purple Fuses, Spacetime system clarifiers. ISOAcoustics Oreas footers. SONORE computer audio | opticalRendu | ultraRendu | microRendu | Signature Rendu SE | Accessories | Software | Link to comment
DavidFaik Posted January 3, 2019 Share Posted January 3, 2019 @barrows ... that's a bit of a shocker! When I researched the Qutest I relied on Chord's features statement: "DSD support: Native playback supported. DSD64 (Single) to DSD512 (Octa-DSD)". I cannot see how one can interpret the English word "native" to mean that the DAC down-samples DSD. While I am HIGHLY impressed with the sonic qualities of the DAC and most likely would have bought it any way, I feel that if this is the case it is very underhand. Would you mind sharing with me where you read this? On your system would you how would you describe the DSD versus FLAC on the same track then? Link to comment
barrows Posted January 3, 2019 Share Posted January 3, 2019 I never use any compressed formats here, all my files are either .dsf or .aiff. I have two (DIY) DACs, one is based on the ESS 9038, and the other on a balanced version of Jussi's discrete DSC-1 approach. I now oversample everything to DSD 256 in my system, but both my DACs have been specifically tailored to work best with high rate DSD, so comparisons are not really fair anyway. I prefer this approach, but it is for my system. I would not be dismayed by the Qutest's approach though, it is a very, very good DAC. With the Qutest I would consider comparing sending native rates vs. oversampling to high rate PCM. You can oversample in software to 705.6, for example, and compare that to sending native PCM rates and see which you prefer. Every system is different, so do not expect your preferences to mate up with others. I would leave DSD alone and just send it native to the Qutest and let the Qutest do what it does. SO/ROON/HQPe: DSD 512-Sonore opticalModuleDeluxe-Signature Rendu optical with Well Tempered Clock--DIY DSC-2 DAC with SC Pure Clock--DIY Purifi Amplifier-Focus Audio FS888 speakers-JL E 112 sub-Nordost Tyr USB, DIY EventHorizon AC cables, Iconoclast XLR & speaker cables, Synergistic Purple Fuses, Spacetime system clarifiers. ISOAcoustics Oreas footers. SONORE computer audio | opticalRendu | ultraRendu | microRendu | Signature Rendu SE | Accessories | Software | Link to comment
DavidFaik Posted January 3, 2019 Share Posted January 3, 2019 @barrows Thanks, I'm very happy with the Qutest but that doesn't detract from my dissatisfaction at Chord playing loose with the truth and claiming native playback. If the DAC is down sampling (more difficult than upsampling I'm told) then it is no wonder that PCM sounds far better than feeding DSD. On Audirvana the "full throttle" upsampling sounds worse than the "only do 2X upsampling". 2X does sound darned good, mind. I will try some other players to see if they have better upsampling capabilities, but if not will probably buy the Audirvana license. I find the interface a little basic, but could live with it. Link to comment
barrows Posted January 3, 2019 Share Posted January 3, 2019 1 hour ago, DavidFaik said: @barrows Thanks, I'm very happy with the Qutest but that doesn't detract from my dissatisfaction at Chord playing loose with the truth and claiming native playback. I do not really think this is the case. When Chord claims "native" DSD playback I am sure they are referring to the fact that the Qutest can accept a native DSD stream on its input (as opposed to only accepting DoP). What would you consider "native DSD"? How about the ESS chip, which does not decimate incoming DSD, but does re-modulate it into a very high rate multi bit format (6 or 7 bits), similar to what dCS does as well. Not many DACs convert DSD without some changes to it. In fact, not many DACs are true NOS with PCM either... SO/ROON/HQPe: DSD 512-Sonore opticalModuleDeluxe-Signature Rendu optical with Well Tempered Clock--DIY DSC-2 DAC with SC Pure Clock--DIY Purifi Amplifier-Focus Audio FS888 speakers-JL E 112 sub-Nordost Tyr USB, DIY EventHorizon AC cables, Iconoclast XLR & speaker cables, Synergistic Purple Fuses, Spacetime system clarifiers. ISOAcoustics Oreas footers. SONORE computer audio | opticalRendu | ultraRendu | microRendu | Signature Rendu SE | Accessories | Software | Link to comment
DavidFaik Posted January 4, 2019 Share Posted January 4, 2019 @barrows thanks for the suggestions to try using the forced upsampling power of 2. After a great deal of time I established that I had to make some quite extreme changes to the default "advanced settings" for Izoptope in Audirvana but THEN I was able to hit a complete sweet spot for the Chord Qutest. Now the upsampling power of 2 sounds better than the 2X, very wide sound stage remains (breadth of left to right beyond the speakers in case I got the audio vocab wrong), the "redbook" or 16 bit CD level files sound very much closer to high res 24 bit files and MP3s are much easier on the ear. Keeps the Chord signature but just "better". Especially with music where there is a lot going on at once. I've yet to research more on what the Nyquest Cuttoff Freq means / does and so have left that as the default (from reading I think I could perhaps damage speakers if I screw that around?) As I could find no one posting ideal Chord Qutest Audirvana settings online, I will add mine below in case it saves others the marathon fiddling, tweaking and listening that I did until the early hours (!) WRT to Chord comments on "native DSD playback", that remains a complete fib in my eyes. In a job interview if I state that I can "communicate with a native Dutch speaker" it means something totally different to saying "I can speak native Dutch". DSD is marketed as a "higher / better" music format than PCM. If the Qutest takes DSD and downsamples it to PCM that's not playing it "natively". Had I not met a smart person like yourself on this forum, I could well have wasted days trying to get DSD playback to sound better than the default settings. So THANK YOU VERY MUCH!!!, I am really most grateful. Caveat to the above comment: I remain completely blown away by this Qutest DAC, one of the best audio purchases that I have ever made (along with the discontinued Chord Prime pre-amp that I scored 2nd hand). Serving Qobuz high rez / 24 bit files to it was a revelation. Very modest priced tweaks have then upped the DAC to levels that I think years ago may have been in the realms of exotica hifi costing the equivalent of a car: namely a €50 usb cable (viablue), some isolation feet (€20), a second hand sbooster linear PSU that was being used for Squeezebox (€70) and now a music player that will upscale 2 million taps (2X taps of a hugo m scaler) for €74. Most things in audio are incremental and very few things in audio are really game changing. This IMHO is truly the latter :-))))) Best Chord Qutest Audirvana Audio Filter Settings (that I have found) Converter: Izotope 64 bit (note- didn't try the rest very long after reading the fora comments) Forced upsampling: power of 2 only (second on drop down. after advice I only played with this and the X2. maybe the "max" option can also be configured well. dsd options messed up SQ) Advanced Parameters Steepness 0db (slider totally left) Filter Max Length 2 million (slider totally right. my mac mini 8gb ram Intel i5 2012 does not break a sweat with this) Cutoff Freq 1.00 x nyquest (I left this as default as I didn't understand it, so maybe changes improve further) Anti-Aliasing 50 (slider totally right) Pre-ringing 0 (slider totally left / min phase side) Link to comment
ferenc Posted June 6, 2020 Share Posted June 6, 2020 On 1/3/2019 at 1:29 PM, DavidFaik said: Would love to hear more about the subjective or experienced side of DSD to compliment the technical comparisons in this thread. In order to get an idea of what others are finding. I'm new to high res music, having enjoyed Qobuz for about two months on their high rez subscription. I have a Chord Qutest DAC that is DSD capable, so wanted to try some DSD content top evaluate if there are sonic differences that I can perceive. I have downloaded a trial of Audirvana. I then found via search engine that Blue Coast Music offers a free download of a well recorded track in multiple formats for evaluation, and I hope I am not breaking forum rules by including that link ( https://bluecoastmusic.com/free-downloads#.XC31CPx7mmk ). ....... If any have time to conduct a similar subjective test then I'd be grateful to read about your findings. Likewise any thoughts very welcome on whether this might be the limitations of the Audirvana player. Further suggestions of other music tracks in multiple formats that could be used to test in a similar fashion, preferably samplers / free. If you want you can try our "My Reel Club" files for comparison here: https://we.tl/t-NUZsotqFSm This is one track from one of our latest live (with audience) studio recordings in the original DSD256 format (with no mastering, post-production or dynamics processing at all, only some very simple mixing in analog using an expensive SSL console). The original recording was converted to DSD128, DSD64 then Flac 192k, 96 / 24 bit and 44.1k / 16 bit. The conversion was done by Jussi at Sygnalist with his HQPlayer Pro software and algorithms. You can see the details of the dynamics of the complete live event (not the file I shared, but the complete event, measured with an iOS app and an extra mic roughly 3-4 m far from the musicians), in one of the corners. We did a tape recording as well, and copied a master tape from the original, the same track was digitized in 96k/24 bit too from this tape copy. The tape output was mixed a bit differently and on a Neve console, so it is not for comparing it to the digital recordings, only for fun. Link to comment
matthias Posted June 7, 2020 Share Posted June 7, 2020 20 hours ago, ferenc said: If you want you can try our "My Reel Club" files for comparison here: https://we.tl/t-NUZsotqFSm Nice recordings, thank you 🙂 I loaded them into Audirvana and could playback DSD256 via DoP into my DAC. Matt ferenc 1 "I want to know why the musicians are on stage, not where". (John Farlowe) Link to comment
ASRMichael Posted June 10, 2020 Share Posted June 10, 2020 @Miska I would like to hear Jussi's view on this? Link to comment
Popular Post Chris A Posted August 28, 2020 Popular Post Share Posted August 28, 2020 On 4/25/2017 at 2:39 PM, The Computer Audiophile said: I just read this post on DSD Guide about DSD and dynamic range. The post is a bit misleading, especially the sentence, "For me this analysis clearly shows that DSD has the best DR values." On 9/26/2017 at 10:47 AM, The_K-Man said: Mastering (and to a big extent mixing) plays a far more significant role in how the finished product sounds than does format. This thread, to me, exemplifies the anti-pattern of the drunk looking for his car keys under the streetlight because the light is better there. Engineers (like me) tend to do that a lot in order to stay away from those subjects that they know little about. I think The_K-Man has the right answer as to why, in practice, early DSD music tracks typically have measurably more dynamic range, but no one here has seemingly paid much attention. In this case, we're talking both about greater crest factor and/or EBU R 128 loudness levels. Does anyone here understand that DSD files cannot be edited (i.e., "mastered") in DSD without first converting them to some form of PCM? (Even Sonoma converts to PCM internally before editing DSD files). So the format itself--DSD--is one that forces those who put out music albums to either: 1) put them out without change from the mix-down tracks, or 2) to lie to the consumer and in fact convert to PCM so that they can be "mastered" (i.e., a euphemism to re-EQ and edit using other nice little tricks), then converted back to DSD. This is the chief reason why, in practice, early DSD tracks before Sonoma workstations hit the scene sounded better and had higher dynamic range: they weren't edited into corned beef hash. Sonoma actually lies about not converting to PCM. They do it--but you can't see when they do it (as Mark Waldrep pointed out in his article linked above). Nowadays, the record companies most often simply take the PCM master tracks and convert them to DSD, and call them "DSD" so they don't have to spend the money on Sonoma workstations. They lie by omission. Audiophiles typically have a blind spot about this portion of the music making business, that is, what the record companies do to the music before the consumer buys them. If there was a law that said that mixing and mastering people had to list the exact sequences and parameters of commands used for every music track that they produce, some/many of those people that call themselves "mastering engineers" would go to jail for falsification of records by lying to their customers about what they're actually doing. And most audiophiles I find don't want to know what's happening (...sort of like taking a tour through an abattoir...most people don't want to know). When you ask those who are in the business of "making the sausage" what they actually do to their product, I find that they rationalize their actions away...but not before you catch them with a smoking gun in hand, i.e., hard data to prove what they've done. Then they say something like "I have to make a living...", and appeal to their jury (the audiophiles) that "everybody does it", etc. Don't look too closely or you might find something that might disturb you. Chris One and a half and sandyk 2 "Those professional loudspeakers with dedicated electronics have a huge advantage over passive loudspeakers. Consumers in general, especially high-end audiophiles, have not caught up with the advantages that technology has to offer. Good loudspeakers and amplifiers can deliver good sound, but merging them with dedicated digital crossovers, equalizers and amplifiers designed for those specific loudspeaker components, in their specific enclosure, can yield even better sound." F. Toole, 2018, Sound Reproduction the Acoustics and Psychoacoustics of Loudspeakers and Rooms, 3rd ed., chap. 12.5, pg 356. Link to comment
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