Paul said ;''Is a compromise to convert '' . I think ... recording is always a compromise situation ,since the industry started. Now is more complicated y better. Thanks Paul for your knowledge .
Paul... it sounds like in many ways DSD is taking one back to the time when engineers recorded to lacquer disc. These were the engineers who had ears and knew how to mic properly and mix live. I've worked with those who were trained in the 50s and 60s, recording to mono, 2 track, or 3 track. I would love to come see Octave Studios as I'm an old 'studio rat.' I remember Brad Miller recording on the Colossus Digital recorder. He came to Portland, where I lived at the time, to record some jazz... what a wonderful session that was. Be well.
Good to see you and the PS Audio team in Ascot at the weekend. Interestingly I asked Tony Faulkner (well known and respected recording engineer) how he records digitally and his choice was high bit rate PCM giving his reasons all of the compromises in DSD! It's facinating that you take oposite approaches. Perhaps one day digital engineers will develop a format which resolves the issue. Until then it seems that anyone buying equipment needs to search out devices that support both PCM and DSD
Hi Paul, great channel - always informative ! I am seriously considering purchasing your PerfectWave DirectStream DAC with the network bridge as an upgrade to my Naim naim nd5 xs2. I plan to use Qobuz as an online source through the Bridge and output via XLR to my Yamaha A-s3200 amplifier. is this possible and does the PS Audio Connect app work with this setup?
@@JonAnderhub I guess I might be mistaken but I thought it was mentioned in previous videos that that was the one he used back in the day and purchased by Paul from him.
Thanks Paul and it wasn’t more than we wanted to know :). To clarify, what INPUT sources can cutting lathes accept (DSD Direct? PCM? I assume analog tape of course). What does Octave use?
I wonder what would work best, mixing to a cutter from analog, or mixing to DSD and throwing that '2 track' mix into the cutter. I know how good DSD is so I don't think I'd even hear the difference, but it's one conversion step less.
The DSD/Analog/DSD system is really very close to a tape workflow, right? So, a 24 track 2" tape you record into, then take the output through the console, then into a stereo 2-track tape. The DSD version would be 24 channels of DSD into the console and out to a 2 channel DSD recorder. Am I getting this right?
That is the system we use at Blue Coast Records.... I still have my 2" 24tracks tape machine (since 1986) and we do setup DSD much like we used our 2" machine
You can convert the DSD to mono analog and smash it with a cbs laboratories audimax and volumax from an old am radio station. Then convert back to DSD.
Hi Paul, after going into DXD for mixing, which option sounds better? 1 - DXD to analog to DSD (Please correct me if I am wrong, but going direct from DXD to DSD is not possible?) 2 - Stay in DXD and downsample to lower PCM resolutions?, I somehow had the feeling that not making another digital conversion from DXD back to DSD also sounds very good and hence DSD makes the most contribution at the first recording / capturing process from the microphones. Thanks in advance..
Going directly from DXD (PCM) to DSD (PDM) is possible and that is exactly what happens in the Pyramix system that Octave Records is using. The multi-channel 1-bit 11.2MHz sampling rate of DSD (PDM) has to have the noise filtered out of it and then it is converted to 32-bit floating point 352 kHz DXD (PCM) Contrary to the way it looks there is no loss in resolution and in fact there is an increase in resolution as DSD has less resolution than PCM because of its 1-bit nature. Once the necessary processes used to mix the recording are in place the 352 kHz DXD (PCM) is then converted back to 1-bit DSD (PDM) at whatever rate (64 fs for SACDs, higher for redistribution online, etc). There are in fact errors and problems with this conversion. First of all, DSD uses a large amount of noise recorded in the ultrasonic range in order to maintain quantization. This noise must be added to each individual track which creates HUGE files. This noise must be filtered out on each track in the conversion to PCM and then another noise at lower frequency ranges but at inaudible levels must be added to the DXD (PCM) to quantize it. Then more ultrasonic noise must be added to the DXD (PCM) to convert it back to DSD. Add in quantization errors for the uneven sample conversion and there is a whole lot of shakin goin on! So what is the right solution? Just record in high-resolution PCM.
i heard PCMs deeper than 16bit(namely 24bit ,32bit) are actually all converted back from original PDM inside modern ADC and DAC chips.. from Paul's "RTR ladder DACs" video.. if the PDMs in which are same with that of DSD's...so , by just using 24bit/32bit PCM ..there'd be less processes.. and hardware-wise already available in everywhere.. I begin to think that on current-day ADC-DAC chip design and the design of class-D amps.. PCM playbacks seem more closer to the original music performance than DSD's by having less PDM-PCM-Analog conversion processes...
Has Octave completely given-up on the Sonoma system? Has development been stopped & there will never been a new release of Sonoma? What’s happened with Gus? I notice that in several videos he isn’t mentioned?
There are always advantages and disadvantages in everything we do. Going to analog has the disadvantage of dealing with the sound of the DAC. Going to DXD is only a low pass filter that if done properly is almost perfect. That's the way we prefer but none of it is perfect.
I collected it for about 10-15 years an amassed a collection of rare punk vinyl that I was able to flip for $25,000.. this was over 10 years ago. I did it for the money.. I wasn't listening to the records, just collecting.. And why I didnt listen to them.. the CD's were far superior in sound quality.. and the collection was taking up too much space.. Hell my DVD/Bluray/CD collection was over 10,000 discs and I moved them all to books (tossing all the cases and inserts) years ago..
I love tape.. I've had my 2" tape machine since 1986. Sadly, finding good techs these days is hard... even harder is finding good tape. That's why we've moved to DSD as much as possible for Blue Coast Records
The TASCAM Hi-Res Editor claims it can be used to edit DSD without going to PCM, unclear what you can do though. Maybe just cut-and-paste and not really change the content?
I'm surprised Paul doesn't mention how all dacs on the market beside AKM convert the dsd signal from dsd to pcm to dsd. While AKM has the architecture to play dsd natively. R2R ladder dacs also convert dsd to pcm to dsd unless they are wired correctly and need a separate DSD pathway to play native dsd. You are right R2R is designed for PCM, but Holo Spring DAC has a separate hardware DSD section specially designed to process 1-bit data. The way R2R process data depends how it is wired to process PCM or DSD. This process is called Discrete DSD conversion, Chord, dCS and MSB all use this form to convert DSD to analogue.
With DSD, you’ll hear music in both the digital domain and the analog domain, but with PCM, digital = switching noises while analog = music. Am I right?
No, you are incorrect. DSD ( is really pulse density modulation) is a digital approximation of an analog signal created by sampling the analog signal, the same as PCM. The fundamental difference between PCM and PDM (DSD) is how the bits in the digital file are implemented. Both PCM and PDM are digital files containing information about an analog signal and must be converted to an analog signal to be heard.
@@drdelewded I know DSD is digital, but what I’m saying is that you’ll still hear music in the digital domain of DSD not because digital is analog, but because DSD is a bitstream, and bitstreams perfectly follow the analog waveform, allowing you to hear music. I read an article on the internet that said that. Also, does DSD sound more like a live performance than PCM ever could, whether in standard form or processed through MQA and fully decoded?
@@caleguillory5451 DSD stereo no mix would be the closest to precise source audio.. but for accurate live reproduction you'd want atmos.. and a mix and a multiple of sources that can be mixed. So DSD can never be as accurate as PCM for reproducing an accurate live performance.. no stereo mix can place you in the room. Now is live music "better" than mixed music? I'd say no.. I prefer studio recorded music over live. The live experience is dealing with shitty rooms, tired performers, audience making noise, shitty PA systems that don't resolve anywhere near as good as most high end studios..
I have heard a few decent "Live" board mixes.. Where the direct outs of the console are recorded.. these are isolated close miking off the drums, guitars, bass etc that are basically split to the PA and to a recorder.. you don't get much of the crowd and room.. just the sources close up.. mayvbe a few room mics.. then its mixed traditionally.. sometime the vocalist will come into the studio to suplement the live vocals.. or the guitarist will do the same for solos.. but is it really live then? and not just a sloppy studio recording?
so REAPER can actually mix DSD. turns out you can set the project sample rate to 2.8224 and a few plugins actually support those sample rates, like the EQ. i used SOX-dsd on linux to convert DSD 2.8224mhz to PCM .wav 8-bit at 2.8224mhz so it's a perfect translation. it works!
Ahhh no… and that guy Amir at ASR is an advocate of many sub $1k systems and DACs that measure and sound good enough to hear the differences. Look him up here on YT.
No sorry, it's not that simple. You can record live to 2 tracks, but you still need to edit for release. A very long, complicated story. We do it all the time at Blue Coast Records
@@soundconnex Thanks Cookie (I have purchased some of your recordings and they sound real nice 😊). To your point, under the right conditions, couldn’t you just record to 2-track (perhaps in a binaural fashion like Chesky does or with a single stereo microphone) and play the resulting DSD file as is without further processing? You seem to suggest that there is some mandatory post-production needed. Maybe I’ve been a bit too naive about this all along and need to educate myself a bit on this topic.
Hey Paul, the great Zig Ziegler once said "Perception is all their is", It's all about your personal experience, perfect reproduction should never come before the music
yeah I'm not sure about that you can't mix dsd, I think if you can take the signal split it up into one bit multiple 32 segments, multiply 32 bit simultaneously in parallel and those 32 individual bits can be individual manipulated and fed back to a 1 bit system. basically you're copying the 1 bit to a multiple of 32 individual segments, and putting them back together as one bit stream. can't see why that wouldn't work!
Unfortunately, it's not quite that simplistic due to other factors but what you are describing is somewhat similar to the process of converting DSD (PDM) to multibit PCM, and then converting the PCM back to DSD (PDM). The whole conversion process causes other problems as well.
DSD sounds great but some commercial popular music that weren't recorded in DSD originally are sold in DSD and they don't sound better than their CD counterpart but you're paying a premium for it.
Hey Paul, there's some very good information you've offered. One thing though, the use of the term "converting DSD to analog" .. maybe should be said, "taking the anlog outs of the DSD to the analog mixing board" (which you did say). My understanding of 'convertiing' has always been about staying in the digital domain and doesn't include any analog stage. It's been that way since the 80's as far as I remember. We would convert 4824 to 44.1/16 for making CDs for example. We'd use digital converters. Maybe we should all agree on what "conversion" means. "Transfer" might be a better word and less confusing when going from DSD to analog and back to DSD. When we are archiving multitrack tapes, we 'transfer' the audio from the analog outs and capture the sound in digital (we never call that converting). There's no converter involved. Happy to discuss it. I know what you mean, but I'm not sure the readers do. I think a lot of the decisions between mixing DSD through analog or going to DXD is a personal decision, depends on the gear available and up to the mixer. For Blue Coast Records, we find that going to DXD never sounds as good as going from DSD through an analog console to mix. Also, our outboard gear using the board sounds fantastic. I've never found the quality of effects in digital plugins. After mixing for 40 years, it's never been a problem to bring back a mix on analog. Not to contradict what you're saying, just offering your readers another point of view. Thanks for all you do. Cookie Marenco bluecoastmusic.com/
Going from DSD to analog or from analog to DSD is through a DAC (Digital to Analog CONVERTER) and then an ADC (Analog to Digital CONVERTER) to get back to DSD. As soon as you convert DSD to analog and route it through a bunch of analog op-amps, potentiometers in a mixer and so on, the quality drops dramatically e.g. the SNR gets reduced to the standard of the pre 80s analog audio (like
@@ThinkingBetter I understand your point, but all digital audio has to be converted at the output in order for the audio engineer (or anyone) to hear it. I will ask the pro audio community if they consider DSD or PCM output (which of course is a DA) a 'conversion'. What I can see is that the audiophile community becomes very confused by the processes. How would you descrbe a 'render' ?
@@soundconnex There is no need to get caught up in the semantics on this topic. Technically, any use of a DAC or ADC is about signal conversion between a flow of digital data and analog voltage variations no matter if it’s in your home or a studio.
If putting a capacitor on a dsd signal turns it into analog, seems like a simple computer program could act like a capacitor and now you can do the math on it inside the computer, and then computer converts back to dsd and sends it on its way
@@aubiepiper5699 I know that I want to like Paul because well , he is likable , but I would think at this stage in his life he would stop with the B.S. Marketing... Let his son do it
@Douglas Blake I'm sure Paul is NOT sitting at the Board doing the mixing and mastering. But that has nothing to do with his marketing claim he can easily hear the difference between high rate PCM vs DSD
@Douglas Blake LOL - You keep defending things I never mentioned or inferred: "old people are useless" and "not everyone my age is stone deaf" -- Later
0:51 "DSD is the very best by a large magnitude." Can you prove that Paul? It seems that the facts show that to not be true. DSD is not even close to the perfect capture given the need to overdrive the Delta-Sigma converter, the need to add a large quantity of noise to keep DSD quantized, and then the need to attempt to filter out that noise although the noise extends into the audible range. This is just the first disaster to occur! Now add in the harmonic distortion and quantization errors that occur in the DSD to PCM conversion, the fact more noise, but different noise has to be added to the PCM for quantization purposes and the Nyquist filtering required for PCM. If that isn't enough, yet more distortion, more quantization errors, and even more noise has to be added to the signal, and yet another Delta-Sigma converter distortion due to being overdriven again. So surely the DSD to analog, to DSD conversion must be better right? Sorry all you Audiophiles that pretend DSD is so much better sounding, it is not. Besides the already mentioned distortion caused by the overdriven Delta Sigma converter, the high levels of noise added for dithering, and the subsequent need to use a low pass filter in the digital to the analog conversion will either leave residual noise in the upper audible frequencies or will filter out the upper frequencies. Now add in the considerably higher noise level of an analog signal path, along with analog phase shift at every part of the "editing" process, the increased harmonic distortion, and loss of 30 dB or more of dynamic range. That loss of dynamic range and increased noise floor typically requires the addition of compression, read additional distortion, phase shift, and noise. If all that distortion and noise isn't enough then consider this. To go back to DSD (PDM) there is yet more Delta-Sigma converter overdrive distortion, the noise and phase shift from the analog source are recorded as well, and even more, large quantities of noise are added once again to quantize the DSD (PDM) signal. What you Audiophiles are mistaking for warmth is signal distortion, and what you are mistaking for "air", or "space" is the added noise. But scientifically a pure PCM signal path using quality converters is far superior to DSD in terms of noise, dynamic range, converter to converter distortion, and frequency response.
Tip: Your inability to split your thoughts into smaller, easily readable paragraphs betrays the chaos inside your mind, nevermind the lack of basic grammar skills.
In PCM you can process the audio much more easily and finally cheaper. But the resulting sound is very compromised if you compare that to analogue audio processing. I guess there is more evil in digital audio processing than in PCM vs DSD recording. AAD CD’s always sounded great, much better than ADD or DDD.)
Sound recorded in PCM is extremely less compromised than any analog process. First of all, PCM has a much better frequency response, with a lower noise floor and a significantly greater dynamic range than an analog source such as an analog mixing board or analog tape. Every step of an analog process adds more noise, phase shift, and distortion to the signal path. PCM can be edited without adding more noise, creating more signal phase shift, or adding more signal distortion.
@Douglas Blake I think people like the vinyl sound, not because of the physical format... or that it is analogue... but because of the mastering techniques involved. When mastering to vinyl... anything between 300Hz to 7Hz that may be out of phase, needs to be mono summed due to the way the cutter head cuts into the lacquer. You also have to limit the acceleration of the high end so that the stylus can still track the groove on playback without distortion. Also, you can't master with too much compression without the risk of burning out the cutting head coils due to the current involved in driving the torque tube and stylus. This makes music sound more natural.... after all,... although digital can produce stereo low end out of phase sounds quite easily, it is something we don't find in nature. Human perception is that: the lower the frequency, the more difficult it is to locate the direction of the sound source. For example: Thunder seems to come from all around.... as where we have a very good idea in which tree a singing bird is located. This is why we often only have one Sub-Woofer in a stereo speaker system. Point is... you can make digital sound like analogue vinyl without compromise if you know what you are doing. You cannot however, make analogue vinyl sound like digital. Sadly the trend in digital is just to brick-wall compress the hell out of everything and be totally ignorant to any low end stereo out of phase. Moreover, we EQ cut sounds below 20Hz and also cut anything above 20kHz under the excuse... "well that's the range of human hearing".... and then wonder why people say they prefer analogue ???? Don't know about you... but i like to feel the harmonics from the bass in the room.... not just hear it in the music! We may now have the technology... but i feel we have lost many of the skills involved in the mastering of music. And our customers know it!
@@Harald_Reindl From my experience, yes! They would on first listening. And that's what selling records is all about. That's why brick wall compression works... it's the monosodium glutamate of audio. But people soon tire of it. And then they want better... but they think it's their equipment not the recording.. so they long to upgrade... but they never are totally satisfied... which the equipment manufactures love... almost as though they were in cahoots with the mastering engineers and record labels. So few people are ever 100% satisfied. They always think... if i only get the next new gadget... the next new format,.. the higher res version. But go look through your own records and CDs. Look for the ones you really love, have had for years, will always have, and never get fed up of. Have a good look closely at the small number of recordings you don't tire of.... and more than likely... they were all mastered by the same small handful of mastering /cutting engineers. You may find all your favourite bands and artists all have the same thing in common. You may not find the info on the jacket / liner notes so you may need to do some research to look up who mastered them. Or look up the matrix number in the dead wax. But you will commonly find the names of: Bob Ludwing Bernie Grundman Bob Katz Ian Shepherd Steve Hoffman to name only a few.... But these people know the secrets i outlined in my previous post. They know how the magic works.
@@AccuphaseMan I listen to mostly CDs. Don't care to go any higher rez.. SACDs for surround is another thing.. music is interesting that way.. but really I just prefer it for film/TV
A lot of the music files you can download from octave records have a big problem! DSD has a lot of noise that they have put in the ultra sonic areas where humans cannot hear! the problem with them is that you get music files that are HUGE in size, take up a lot of space and take a long time to download! 1 bit has more noise than PCM! they should have taken away everything above 25khz
What you and others call 'noise' and claim that DSD has it over established PCM is actually more detailed and realistic sound. Give it a listen sometimes and be amazed of how DSD sounds. That's what Paul did and what took him on his DSD journey.
@@edfort5704 No DSD (really Pulse Density Modulation) requires a large amount of noise to be added in order to quantize the samples. This "noise" is supposedly shifted to upper "inaudible" frequencies, however, the noise level reaches the upper-frequency spectrum of hearing in DSD 64. This is reflected in the size of DSD files and this noise must be filtered or it will ruin amplifiers. PCM, on the other hand, also uses noise for quantization, but the noise is within the hearing spectrum but well below the threshold of hearing.
Nope, there is no such things as "do everything in DSD". Any digital studio is relying on PCM in the mastering process and all you can do with DSD is to do the initial track recording in the DSD format. As soon as you need to mix, EQ, compress or whatever, PCM is the format and the master output will ALSO be PCM. Any distribution of the output in DSD will therefore be a lossy transcoding of the PCM format used in the studio. I can't see how that lossy transcoded DSD version can be any better than the actual original PCM version. Now, there is a quality factor of your DAC in how well it translates high resolution PCM to analog vs. how well it does it for DSD. If your DAC is doing a poor job with PCM, the lossy DSD might sound better.
@Douglas Blake Digital audio can be moved with zero losses from a digital master to your playback device through the internet over 1000s of miles of cables and lots of routing gear. For analog audio, I agree.
@Douglas Blake I have a degree in computer science and of course understand the issue of a protocol stack that ensures the application layer is getting reliable data through lower layers of routing, retransmission, check sums etc. ensuring that your bank data, e-mails or audio data are 100% coming out as they came in. Anyone thinking digital audio via the internet is some sort of hard case is just plain ignorant on data communication.
@Douglas Blake At some point in my career I was leading a team of 50 software engineers developing the core CDMA protocol stack for a mobile chipset and at that time reaching 3.1 megabits per second error free over the air data rate (1xEVDO standard). Few people understand the complexity of mobile data protocols allowing many people to share the RF spectrum to stream music or 4K video with zero errors while on the move in a car where the data connections seamlessly hands over between base stations. Only a deep understand if the real science of how radio waves behave can make it possible.
@@Pedro-0839 Of course there are errors in digital communication on what is called the "physical layer" but any modern protocol stack will detect the errors and if necessary the data will be resent so the "application layer" doesn't see any errors (check the 7 layer OSI = Open System Interconnection model). This means you can stream a music file across the world and you can bet the data will be 100% as the original source file. I can reliably with ZERO errors stream 192kHz 24 bits FLAC lossless master quality music e.g. via Amazon Music HD across the world into a laptop connected on WiFi and a high quality DAC while your 44.1kHz CD is not only hugely inferior in the audio quality itself but also error prone as the CD media can get scratches causing audible issues.
@Douglas Blake The reason you have layers in a protocol stack is to ensure that the data gets delivered with output = input. The "application layer" is where your application sees the data and it will not not need to know or care what errors happened on lower levels that got fixed by those layers. You can create 100% reliable communication even the "physical layer" (e.g. transatlantic internet optical cable) is with occasional errors.The "data link layer" is where you will fix the errors in a given physical connection between nodes. On top of that you have the "network layer" that allows you to get the data to move through a complex network involving multiple paths of routers. When you stream music you might have some of the music data going through an entirely different path and you won't even notice it thanks to the "network layer". If you could show a map of the data you would be amazed how your music data could be spread globally from a far-away server before it hits your DAC flawlessly in correct sequence.
Wouldn't be easier just to record in analog? Can you imagine all the great studios like Abbey Road recording in digital? Me neither. Music is meant to be played analog.
You again. what makes you think that you are the arbiter of how music "is meant to be played"? How about just letting people choose how the want to listen to music instead of playing authority on the topic. Keep in mind you're not merely saying that you prefer listening to music in that way but that you prefer other people to listen to music in that way, which doesn't make sense in my book. You're clearly implying that if you listen to music in digital as opposed to analog you're doing something wrong, which is incorrect. Music is art if you didn't get the memo and there is nothing about the way that you enjoy or create music that could be classified as wrong or right in any objective manner at all. If I enjoy listening to music by compressing it to a 32 kbps mp3 and playing it over a mono subwoofer placed 180° behind me then that's the way it's going to be and saying that "that's not the way it's meant to be played" doesn't change anything about the fact that that's just the way I enjoy listening to my music. So I sincerely ask you to stop saying that "analog is the way music is meant to be played". Because that is just not true. And btw who meant for music to be played in that certain way? The artist? God? Or just you?
Sorry to tell you this.. Abbey Road records in Digital.. Pro Tools.. As does almost every major studio.. though some might provide the option for analog
Octave studios appears to go to great lengths to avoid using any analog procedure in the recording process as if it is some kind of sin. Very stupid IMO. Use common sense instead of bs.
@@el_zirb I believe he said they convert DSD signals to PCM because they get more data to play with compared to analog. What a dog's breakfast digital audio is!
I love your videos Paul. Thanks. I want All analog. In my opinion digital is and will always be inferior to analog. Can it sound OK or even good, sure. Is it something i will pay money for, NO. I want the waveform in its natural state with out it being converted into bits!
Paul said ;''Is a compromise to convert '' . I think ... recording is always a compromise situation ,since the industry started. Now is more complicated y better. Thanks Paul for your knowledge .
yes, agreed... every stage is a compromise.. including use of headphones :) Sometimes, a necessary evil.
Paul... it sounds like in many ways DSD is taking one back to the time when engineers recorded to lacquer disc. These were the engineers who had ears and knew how to mic properly and mix live. I've worked with those who were trained in the 50s and 60s, recording to mono, 2 track, or 3 track. I would love to come see Octave Studios as I'm an old 'studio rat.' I remember Brad Miller recording on the Colossus Digital recorder. He came to Portland, where I lived at the time, to record some jazz... what a wonderful session that was. Be well.
Great explanation. Thanks Paul
Thanks for the explanation. That was exactly what I wanted to understand. Not too much detail.
Good to see you and the PS Audio team in Ascot at the weekend. Interestingly I asked Tony Faulkner (well known and respected recording engineer) how he records digitally and his choice was high bit rate PCM giving his reasons all of the compromises in DSD! It's facinating that you take oposite approaches. Perhaps one day digital engineers will develop a format which resolves the issue. Until then it seems that anyone buying equipment needs to search out devices that support both PCM and DSD
Thanks Paul, I asked a similar question, its clear now!
This is a perfect topic given
the recent MoFi controversy.
🤗👍😎THANKS PAUL…FOR EASING US ALONG 😊💚💚💚
Hi Paul, great channel - always informative ! I am seriously considering purchasing your PerfectWave DirectStream DAC with the network bridge as an upgrade to my Naim naim nd5 xs2. I plan to use Qobuz as an online source through the Bridge and output via XLR to my Yamaha A-s3200 amplifier.
is this possible and does the PS Audio Connect app work with this setup?
Good video, with useful information.
Only your Mom would explain something to you like that. Dad would cut it short. What a nice man
I love DSD because to me it's basically the digital pulse density modulation equivalent of the magnetic flux laid down on a magnetic tape.
I would have to say the board behind you never failed Neil young, some fine sounding recordings.
There is absolutely no evidence that Neil Young ever owned that board or ever owned any Studer mixing board.
@@JonAnderhub I guess I might be mistaken but I thought it was mentioned in previous videos that that was the one he used back in the day and purchased by Paul from him.
@@koprcord5338 You are correct.
Thanks Paul and it wasn’t more than we wanted to know :). To clarify, what INPUT sources can cutting lathes accept (DSD Direct? PCM? I assume analog tape of course). What does Octave use?
I wonder what would work best, mixing to a cutter from analog, or mixing to DSD and throwing that '2 track' mix into the cutter. I know how good DSD is so I don't think I'd even hear the difference, but it's one conversion step less.
The DSD/Analog/DSD system is really very close to a tape workflow, right? So, a 24 track 2" tape you record into, then take the output through the console, then into a stereo 2-track tape. The DSD version would be 24 channels of DSD into the console and out to a 2 channel DSD recorder. Am I getting this right?
That is the system we use at Blue Coast Records.... I still have my 2" 24tracks tape machine (since 1986) and we do setup DSD much like we used our 2" machine
You can convert the DSD to mono analog and smash it with a cbs laboratories audimax and volumax from an old am radio station. Then convert back to DSD.
I think as time goes on analogue is going to become more special only because so much of the classics have been recorded in analogue
Hi Paul, after going into DXD for mixing, which option sounds better? 1 - DXD to analog to DSD (Please correct me if I am wrong, but going direct from DXD to DSD is not possible?) 2 - Stay in DXD and downsample to lower PCM resolutions?, I somehow had the feeling that not making another digital conversion from DXD back to DSD also sounds very good and hence DSD makes the most contribution at the first recording / capturing process from the microphones. Thanks in advance..
Going directly from DXD (PCM) to DSD (PDM) is possible and that is exactly what happens in the Pyramix system that Octave Records is using.
The multi-channel 1-bit 11.2MHz sampling rate of DSD (PDM) has to have the noise filtered out of it and then it is converted to 32-bit floating point 352 kHz DXD (PCM)
Contrary to the way it looks there is no loss in resolution and in fact there is an increase in resolution as DSD has less resolution than PCM because of its 1-bit nature.
Once the necessary processes used to mix the recording are in place the 352 kHz DXD (PCM) is then converted back to 1-bit DSD (PDM) at whatever rate (64 fs for SACDs, higher for redistribution online, etc).
There are in fact errors and problems with this conversion.
First of all, DSD uses a large amount of noise recorded in the ultrasonic range in order to maintain quantization.
This noise must be added to each individual track which creates HUGE files.
This noise must be filtered out on each track in the conversion to PCM and then another noise at lower frequency ranges but at inaudible levels must be added to the DXD (PCM) to quantize it.
Then more ultrasonic noise must be added to the DXD (PCM) to convert it back to DSD. Add in quantization errors for the uneven sample conversion and there is a whole lot of shakin goin on!
So what is the right solution?
Just record in high-resolution PCM.
What about analogue to DSD to analogue vinyl records?
i heard PCMs deeper than 16bit(namely 24bit ,32bit) are actually all converted back from original PDM inside modern ADC and DAC chips.. from Paul's "RTR ladder DACs" video..
if the PDMs in which are same with that of DSD's...so , by just using 24bit/32bit PCM ..there'd be less processes.. and hardware-wise already available in everywhere..
I begin to think that on current-day ADC-DAC chip design and the design of class-D amps.. PCM playbacks seem more closer to the original music performance than DSD's by having less PDM-PCM-Analog conversion processes...
Has Octave completely given-up on the Sonoma system? Has development been stopped & there will never been a new release of Sonoma?
What’s happened with Gus? I notice that in several videos he isn’t mentioned?
Is there any advantage or disadvantage in converting to dxd vs analog?
There are always advantages and disadvantages in everything we do. Going to analog has the disadvantage of dealing with the sound of the DAC. Going to DXD is only a low pass filter that if done properly is almost perfect. That's the way we prefer but none of it is perfect.
What DSD (DSDx1, DSDx4, etc) are you using in the process? Do you use your AD/DA or someone elses? Thx!
A very well times Question I must say :)
What do you do if you want to make vinyl pressings?
Send a master file to a record record plant.. then wait 8-12 months because of the backlog to get your records pressed.
@@drdelewded 😂That wouldn’t surprise me.
@@bikdav I looked into it when I thought I wanted to press vinyl.. back when I liked vinyl
@@drdelewded what made you to leave vinyl?
I collected it for about 10-15 years an amassed a collection of rare punk vinyl that I was able to flip for $25,000.. this was over 10 years ago. I did it for the money.. I wasn't listening to the records, just collecting.. And why I didnt listen to them.. the CD's were far superior in sound quality.. and the collection was taking up too much space.. Hell my DVD/Bluray/CD collection was over 10,000 discs and I moved them all to books (tossing all the cases and inserts) years ago..
Wouldn't it be better to record in analog, mix in analog, and then convert to DSD?
I love tape.. I've had my 2" tape machine since 1986. Sadly, finding good techs these days is hard... even harder is finding good tape. That's why we've moved to DSD as much as possible for Blue Coast Records
The TASCAM Hi-Res Editor claims it can be used to edit DSD without going to PCM, unclear what you can do though. Maybe just cut-and-paste and not really change the content?
I'm surprised Paul doesn't mention how all dacs on the market beside AKM convert the dsd signal from dsd to pcm to dsd.
While AKM has the architecture to play dsd natively.
R2R ladder dacs also convert dsd to pcm to dsd unless they are wired correctly and need a separate DSD pathway to play native dsd.
You are right R2R is designed for PCM, but Holo Spring DAC has a separate hardware DSD section specially designed to process 1-bit data. The way R2R process data depends how it is wired to process PCM or DSD. This process is called Discrete DSD conversion, Chord, dCS and MSB all use this form to convert DSD to analogue.
With DSD, you’ll hear music in both the digital domain and the analog domain, but with PCM, digital = switching noises while analog = music. Am I right?
No
DSD is digital
No, you are incorrect.
DSD ( is really pulse density modulation) is a digital approximation of an analog signal created by sampling the analog signal, the same as PCM.
The fundamental difference between PCM and PDM (DSD) is how the bits in the digital file are implemented.
Both PCM and PDM are digital files containing information about an analog signal and must be converted to an analog signal to be heard.
@@drdelewded I know DSD is digital, but what I’m saying is that you’ll still hear music in the digital domain of DSD not because digital is analog, but because DSD is a bitstream, and bitstreams perfectly follow the analog waveform, allowing you to hear music. I read an article on the internet that said that. Also, does DSD sound more like a live performance than PCM ever could, whether in standard form or processed through MQA and fully decoded?
@@caleguillory5451
DSD stereo no mix would be the closest to precise source audio.. but for accurate live reproduction you'd want atmos.. and a mix and a multiple of sources that can be mixed. So DSD can never be as accurate as PCM for reproducing an accurate live performance.. no stereo mix can place you in the room.
Now is live music "better" than mixed music? I'd say no.. I prefer studio recorded music over live. The live experience is dealing with shitty rooms, tired performers, audience making noise, shitty PA systems that don't resolve anywhere near as good as most high end studios..
I have heard a few decent "Live" board mixes.. Where the direct outs of the console are recorded.. these are isolated close miking off the drums, guitars, bass etc that are basically split to the PA and to a recorder.. you don't get much of the crowd and room.. just the sources close up.. mayvbe a few room mics.. then its mixed traditionally.. sometime the vocalist will come into the studio to suplement the live vocals.. or the guitarist will do the same for solos.. but is it really live then? and not just a sloppy studio recording?
So much for using the FR30's as monitors...
Haha! Yep.
Why no take a look at the latest Absolute Sound issue Golden Ear Awards. The FR 30’s just made the list. The review was spectacular. Not surprising.
If volume is the only thing you need, there ought to be DSP algorithms processing the DSD file without having to convert to analog and back. Isn’t it?
They need volume, eq, compression and various other inline or parallel effects.
so REAPER can actually mix DSD. turns out you can set the project sample rate to 2.8224 and a few plugins actually support those sample rates, like the EQ. i used SOX-dsd on linux to convert DSD 2.8224mhz to PCM .wav 8-bit at 2.8224mhz so it's a perfect translation. it works!
And unless you have a God awful expensive, highly resolving system, you'll never hear a lick of difference
Not true. Room treatment and sub $1000 DACs and speakers can reveal an obvious difference. I've done it.
Ahhh no… and that guy Amir at ASR is an advocate of many sub $1k systems and DACs that measure and sound good enough to hear the differences. Look him up here on YT.
MOFIs did it that way ? Now I understand the scam better Thank you Paul
Could you not record straight to DSD256 and be done (i.e., no mixing). It would be like direct to disk in the analog days.
No sorry, it's not that simple. You can record live to 2 tracks, but you still need to edit for release. A very long, complicated story. We do it all the time at Blue Coast Records
@@soundconnex Thanks Cookie (I have purchased some of your recordings and they sound real nice 😊). To your point, under the right conditions, couldn’t you just record to 2-track (perhaps in a binaural fashion like Chesky does or with a single stereo microphone) and play the resulting DSD file as is without further processing? You seem to suggest that there is some mandatory post-production needed. Maybe I’ve been a bit too naive about this all along and need to educate myself a bit on this topic.
Hey Paul, the great Zig Ziegler once said "Perception is all their is", It's all about your personal experience, perfect reproduction should never come before the music
yeah I'm not sure about that you can't mix dsd, I think if you can take the signal split it up into one bit multiple 32 segments, multiply 32 bit simultaneously in parallel and those 32 individual bits can be individual manipulated and fed back to a 1 bit system.
basically you're copying the 1 bit to a multiple of 32 individual segments, and putting them back together as one bit stream. can't see why that wouldn't work!
Unfortunately, it's not quite that simplistic due to other factors but what you are describing is somewhat similar to the process of converting DSD (PDM) to multibit PCM, and then converting the PCM back to DSD (PDM).
The whole conversion process causes other problems as well.
DSD sounds great but some commercial popular music that weren't recorded in DSD originally are sold in DSD and they don't sound better than their CD counterpart but you're paying a premium for it.
Whatever Paul says sounds good - but does it have to be so expensive !???
yes
Hey Paul, there's some very good information you've offered. One thing though, the use of the term "converting DSD to analog" .. maybe should be said, "taking the anlog outs of the DSD to the analog mixing board" (which you did say).
My understanding of 'convertiing' has always been about staying in the digital domain and doesn't include any analog stage. It's been that way since the 80's as far as I remember. We would convert 4824 to 44.1/16 for making CDs for example. We'd use digital converters. Maybe we should all agree on what "conversion" means.
"Transfer" might be a better word and less confusing when going from DSD to analog and back to DSD. When we are archiving multitrack tapes, we 'transfer' the audio from the analog outs and capture the sound in digital (we never call that converting). There's no converter involved. Happy to discuss it. I know what you mean, but I'm not sure the readers do.
I think a lot of the decisions between mixing DSD through analog or going to DXD is a personal decision, depends on the gear available and up to the mixer. For Blue Coast Records, we find that going to DXD never sounds as good as going from DSD through an analog console to mix. Also, our outboard gear using the board sounds fantastic. I've never found the quality of effects in digital plugins. After mixing for 40 years, it's never been a problem to bring back a mix on analog.
Not to contradict what you're saying, just offering your readers another point of view. Thanks for all you do.
Cookie Marenco bluecoastmusic.com/
Going from DSD to analog or from analog to DSD is through a DAC (Digital to Analog CONVERTER) and then an ADC (Analog to Digital CONVERTER) to get back to DSD. As soon as you convert DSD to analog and route it through a bunch of analog op-amps, potentiometers in a mixer and so on, the quality drops dramatically e.g. the SNR gets reduced to the standard of the pre 80s analog audio (like
@@ThinkingBetter I understand your point, but all digital audio has to be converted at the output in order for the audio engineer (or anyone) to hear it. I will ask the pro audio community if they consider DSD or PCM output (which of course is a DA) a 'conversion'. What I can see is that the audiophile community becomes very confused by the processes. How would you descrbe a 'render' ?
@@soundconnex There is no need to get caught up in the semantics on this topic. Technically, any use of a DAC or ADC is about signal conversion between a flow of digital data and analog voltage variations no matter if it’s in your home or a studio.
Hi ps audio. I am from India. Little former please help me🙏🙏. Give old system
If putting a capacitor on a dsd signal turns it into analog, seems like a simple computer program could act like a capacitor and now you can do the math on it inside the computer, and then computer converts back to dsd and sends it on its way
Wow I'm lucky to be one of the first in the first hour
8 tracks rule !!!!
This rigmarole is more about Marketing than fact - Paul is how old? No one believes his hearing is so good he can distinguish 24/96 vs DSD in a DBT
Sadly that is true.
@@aubiepiper5699 I know that I want to like Paul because well , he is likable , but I would think at this stage in his life he would stop with the B.S. Marketing... Let his son do it
@Douglas Blake I'm sure Paul is NOT sitting at the Board doing the mixing and mastering. But that has nothing to do with his marketing claim he can easily hear the difference between high rate PCM vs DSD
@Douglas Blake Some things cannot be avoided with age...
@Douglas Blake LOL - You keep defending things I never mentioned or inferred: "old people are useless" and "not everyone my age is stone deaf" -- Later
I think the marketing aspect is a big problem. This is good because it helps with translation.... comprised vs pure.. whatever
0:51 "DSD is the very best by a large magnitude."
Can you prove that Paul?
It seems that the facts show that to not be true.
DSD is not even close to the perfect capture given the need to overdrive the Delta-Sigma converter, the need to add a large quantity of noise to keep DSD quantized, and then the need to attempt to filter out that noise although the noise extends into the audible range. This is just the first disaster to occur! Now add in the harmonic distortion and quantization errors that occur in the DSD to PCM conversion, the fact more noise, but different noise has to be added to the PCM for quantization purposes and the Nyquist filtering required for PCM. If that isn't enough, yet more distortion, more quantization errors, and even more noise has to be added to the signal, and yet another Delta-Sigma converter distortion due to being overdriven again. So surely the DSD to analog, to DSD conversion must be better right? Sorry all you Audiophiles that pretend DSD is so much better sounding, it is not. Besides the already mentioned distortion caused by the overdriven Delta Sigma converter, the high levels of noise added for dithering, and the subsequent need to use a low pass filter in the digital to the analog conversion will either leave residual noise in the upper audible frequencies or will filter out the upper frequencies. Now add in the considerably higher noise level of an analog signal path, along with analog phase shift at every part of the "editing" process, the increased harmonic distortion, and loss of 30 dB or more of dynamic range. That loss of dynamic range and increased noise floor typically requires the addition of compression, read additional distortion, phase shift, and noise. If all that distortion and noise isn't enough then consider this. To go back to DSD (PDM) there is yet more Delta-Sigma converter overdrive distortion, the noise and phase shift from the analog source are recorded as well, and even more, large quantities of noise are added once again to quantize the DSD (PDM) signal. What you Audiophiles are mistaking for warmth is signal distortion, and what you are mistaking for "air", or "space" is the added noise. But scientifically a pure PCM signal path using quality converters is far superior to DSD in terms of noise, dynamic range, converter to converter distortion, and frequency response.
Great response, that sadly is sure to be ignored. People want the fast food content that Paul provides.
You don't need to prove anything for audiofools
Tip: Your inability to split your thoughts into smaller, easily readable paragraphs betrays the chaos inside your mind, nevermind the lack of basic grammar skills.
@@edfort5704 How would you have written this?
@@JonAnderhub More coherently and less discouraging to read at first sight.
This how I'm comenting on mofi-gate without commenting on mofi-gate.
You should have been invited to Michael 45s round table. You could have rebutted Bernie Grundman’s comments on DSD, and digital storage.
In PCM you can process the audio much more easily and finally cheaper. But the resulting sound is very compromised if you compare that to analogue audio processing. I guess there is more evil in digital audio processing than in PCM vs DSD recording. AAD CD’s always sounded great, much better than ADD or DDD.)
Sound recorded in PCM is extremely less compromised than any analog process.
First of all, PCM has a much better frequency response, with a lower noise floor and a significantly greater dynamic range than an analog source such as an analog mixing board or analog tape.
Every step of an analog process adds more noise, phase shift, and distortion to the signal path.
PCM can be edited without adding more noise, creating more signal phase shift, or adding more signal distortion.
@Douglas Blake I think people like the vinyl sound, not because of the physical format... or that it is analogue... but because of the mastering techniques involved.
When mastering to vinyl... anything between 300Hz to 7Hz that may be out of phase, needs to be mono summed due to the way the cutter head cuts into the lacquer.
You also have to limit the acceleration of the high end so that the stylus can still track the groove on playback without distortion.
Also, you can't master with too much compression without the risk of burning out the cutting head coils due to the current involved in driving the torque tube and stylus.
This makes music sound more natural.... after all,... although digital can produce stereo low end out of phase sounds quite easily, it is something we don't find in nature.
Human perception is that: the lower the frequency, the more difficult it is to locate the direction of the sound source.
For example: Thunder seems to come from all around.... as where we have a very good idea in which tree a singing bird is located.
This is why we often only have one Sub-Woofer in a stereo speaker system.
Point is... you can make digital sound like analogue vinyl without compromise if you know what you are doing.
You cannot however, make analogue vinyl sound like digital.
Sadly the trend in digital is just to brick-wall compress the hell out of everything and be totally ignorant to any low end stereo out of phase. Moreover, we EQ cut sounds below 20Hz and also cut anything above 20kHz under the excuse... "well that's the range of human hearing"....
and then wonder why people say they prefer analogue ????
Don't know about you... but i like to feel the harmonics from the bass in the room.... not just hear it in the music!
We may now have the technology... but i feel we have lost many of the skills involved in the mastering of music. And our customers know it!
Lol analog is crap
@@janedoe6350 people would like the digital sound exactly the same if someone tells them it's vinyl
@@Harald_Reindl From my experience, yes! They would on first listening. And that's what selling records is all about. That's why brick wall compression works... it's the monosodium glutamate of audio.
But people soon tire of it. And then they want better... but they think it's their equipment not the recording.. so they long to upgrade... but they never are totally satisfied... which the equipment manufactures love... almost as though they were in cahoots with the mastering engineers and record labels.
So few people are ever 100% satisfied. They always think... if i only get the next new gadget... the next new format,.. the higher res version.
But go look through your own records and CDs. Look for the ones you really love, have had for years, will always have, and never get fed up of. Have a good look closely at the small number of recordings you don't tire of....
and more than likely... they were all mastered by the same small handful of mastering /cutting engineers.
You may find all your favourite bands and artists all have the same thing in common.
You may not find the info on the jacket / liner notes so you may need to do some research to look up who mastered them.
Or look up the matrix number in the dead wax.
But you will commonly find the names of:
Bob Ludwing
Bernie Grundman
Bob Katz
Ian Shepherd
Steve Hoffman
to name only a few....
But these people know the secrets i outlined in my previous post.
They know how the magic works.
O o
Imagine paying a premium for a SACD only for it PCM
SACD even in PCM can be a higher resolution than CD if you use hi rez
IE track in 24 196.. mix in the same then master to DSD SACD.
@@drdelewded With an upsampling DAC there's no point to go above CD quality
@@AccuphaseMan I listen to mostly CDs.
Don't care to go any higher rez.. SACDs for surround is another thing.. music is interesting that way.. but really I just prefer it for film/TV
A lot of the music files you can download from octave records have a big problem! DSD has a lot of noise that they have put in the ultra sonic areas where humans cannot hear! the problem with them is that you get music files that are HUGE in size, take up a lot of space and take a long time to download! 1 bit has more noise than PCM! they should have taken away everything above 25khz
What you and others call 'noise' and claim that DSD has it over established PCM is actually more detailed and realistic sound. Give it a listen sometimes and be amazed of how DSD sounds. That's what Paul did and what took him on his DSD journey.
@@edfort5704 No DSD (really Pulse Density Modulation) requires a large amount of noise to be added in order to quantize the samples.
This "noise" is supposedly shifted to upper "inaudible" frequencies, however, the noise level reaches the upper-frequency spectrum of hearing in DSD 64.
This is reflected in the size of DSD files and this noise must be filtered or it will ruin amplifiers.
PCM, on the other hand, also uses noise for quantization, but the noise is within the hearing spectrum but well below the threshold of hearing.
Nope, there is no such things as "do everything in DSD". Any digital studio is relying on PCM in the mastering process and all you can do with DSD is to do the initial track recording in the DSD format. As soon as you need to mix, EQ, compress or whatever, PCM is the format and the master output will ALSO be PCM. Any distribution of the output in DSD will therefore be a lossy transcoding of the PCM format used in the studio. I can't see how that lossy transcoded DSD version can be any better than the actual original PCM version. Now, there is a quality factor of your DAC in how well it translates high resolution PCM to analog vs. how well it does it for DSD. If your DAC is doing a poor job with PCM, the lossy DSD might sound better.
@Douglas Blake Digital audio can be moved with zero losses from a digital master to your playback device through the internet over 1000s of miles of cables and lots of routing gear. For analog audio, I agree.
@Douglas Blake I have a degree in computer science and of course understand the issue of a protocol stack that ensures the application layer is getting reliable data through lower layers of routing, retransmission, check sums etc. ensuring that your bank data, e-mails or audio data are 100% coming out as they came in. Anyone thinking digital audio via the internet is some sort of hard case is just plain ignorant on data communication.
@Douglas Blake At some point in my career I was leading a team of 50 software engineers developing the core CDMA protocol stack for a mobile chipset and at that time reaching 3.1 megabits per second error free over the air data rate (1xEVDO standard). Few people understand the complexity of mobile data protocols allowing many people to share the RF spectrum to stream music or 4K video with zero errors while on the move in a car where the data connections seamlessly hands over between base stations. Only a deep understand if the real science of how radio waves behave can make it possible.
@@Pedro-0839 Of course there are errors in digital communication on what is called the "physical layer" but any modern protocol stack will detect the errors and if necessary the data will be resent so the "application layer" doesn't see any errors (check the 7 layer OSI = Open System Interconnection model). This means you can stream a music file across the world and you can bet the data will be 100% as the original source file. I can reliably with ZERO errors stream 192kHz 24 bits FLAC lossless master quality music e.g. via Amazon Music HD across the world into a laptop connected on WiFi and a high quality DAC while your 44.1kHz CD is not only hugely inferior in the audio quality itself but also error prone as the CD media can get scratches causing audible issues.
@Douglas Blake The reason you have layers in a protocol stack is to ensure that the data gets delivered with output = input. The "application layer" is where your application sees the data and it will not not need to know or care what errors happened on lower levels that got fixed by those layers. You can create 100% reliable communication even the "physical layer" (e.g. transatlantic internet optical cable) is with occasional errors.The "data link layer" is where you will fix the errors in a given physical connection between nodes. On top of that you have the "network layer" that allows you to get the data to move through a complex network involving multiple paths of routers. When you stream music you might have some of the music data going through an entirely different path and you won't even notice it thanks to the "network layer". If you could show a map of the data you would be amazed how your music data could be spread globally from a far-away server before it hits your DAC flawlessly in correct sequence.
Wouldn't be easier just to record in analog? Can you imagine all the great studios like Abbey Road recording in digital? Me neither. Music is meant to be played analog.
🤔 Are you sure about that 🤷♂️
You again. what makes you think that you are the arbiter of how music "is meant to be played"? How about just letting people choose how the want to listen to music instead of playing authority on the topic. Keep in mind you're not merely saying that you prefer listening to music in that way but that you prefer other people to listen to music in that way, which doesn't make sense in my book. You're clearly implying that if you listen to music in digital as opposed to analog you're doing something wrong, which is incorrect. Music is art if you didn't get the memo and there is nothing about the way that you enjoy or create music that could be classified as wrong or right in any objective manner at all. If I enjoy listening to music by compressing it to a 32 kbps mp3 and playing it over a mono subwoofer placed 180° behind me then that's the way it's going to be and saying that "that's not the way it's meant to be played" doesn't change anything about the fact that that's just the way I enjoy listening to my music.
So I sincerely ask you to stop saying that "analog is the way music is meant to be played". Because that is just not true.
And btw who meant for music to be played in that certain way? The artist? God? Or just you?
Sorry to tell you this.. Abbey Road records in Digital.. Pro Tools..
As does almost every major studio.. though some might provide the option for analog
Abbey Road records in digital (PCM) as do most major recording studios.
Sorry to burst your bubble.😶🌫
@@drdelewded 🤗👍 THANKS FOR SHARING THIS..WE ARE DELUDED NO MORE…
Some might be disappointed ☹️🤷♂️💚💚💚
Octave studios appears to go to great lengths to avoid using any analog procedure in the recording process as if it is some kind of sin. Very stupid IMO. Use common sense instead of bs.
And what's it to you? Is it your bread and butter?
Didn’t he just say the opposite (mixing panel)?
@@el_zirb I believe he said they convert DSD signals to PCM because they get more data to play with compared to analog. What a dog's breakfast digital audio is!
PCM can be dealt with on a conventional computer for manipulation.
DSD must be dealt with on a quantum computer for manipulation.
I love your videos Paul. Thanks. I want All analog.
In my opinion digital is and will always be inferior to analog. Can it sound OK or even good, sure. Is it something i will pay money for, NO. I want the waveform in its natural state with out it being converted into bits!