Agreed. It all depends on what the recording technician did regardless of what was used to do it.

The way I see it is that the overwhelming majority of music out there is PCM (or analog) and there’s no way a rational person is going to ditch all that just to have a handful of DSD recordings. And yes, the recording is everything, perhaps even more so than the format itself. I’d rather listen to a well-recorded track on 16/44.1 (or even 320 kbps MP3) than a poorly recorded track on DSD.

Listening to music is a subjective experience and the competence of the engineering is critical. That said, the idea that red book is "good enough" assumes that DSD is a marketing conspiracy (mqa for example). I was shocked at the detail in the mofi dsd64 transferred vinyl of Bill Evans & Jim Hall "Undercurrent" when I upgraded my turntable with a Grado Platinum Tibre. I record my own music at 192Khz 32bit and listen to vinyl rips all the time. CDs sound amazing on high end gear but HR sounds good on just about anything worth listening too. I believe Paul and Kevin Grey when they say DSD256 is better than CD quality.

@@housepianist That’s an excellent point.

@@housepianist My thoughts exactly.

Why don't you back up your statements with facts and give us links to prove your points? I know, it's just easier your way...

its totally different on a even half decent system

You have a world of differences to discover then. :) Does not mean you cannot enjoy both systems though.

when you decide your not really happy i was pretty dissapointed try the mola mola tambaqui it eats it for dinner without breaking sweat but they are unicorns to find secondhand x

Thanks, Jim. I never knew him but knew of him. And I assume it was PCM. The first commercial DAW was the Sonoma System Sony designed, but the first standalone recorders were from Sony, Tascam, and Korg (best I know)

That’s because fremer confuses DSD and DXD

Resolution increase.

"The digital signal can be directly used as an analog signal" is the bit I find confusing, or obfuscated is that what is happening in the DAC? and i mean the appliance DAC rather than strictly the converter circuit process, that is the amplifier down stream scaling the filtered digital output to desired on off voltages for listening? and the reactance/capacitance of the analog circuits and drivers doing the smoothing of the pulse width into a waveform? Similar to PWM motor control?

Hush! Don't spill the beans! There is a whole industry and associated marketing industry built on general misconceptions around PCM vs DSD. In the digital domain both are virtually interchangeable as long as the information density of both encodings are comparable. The sensitivity to bit corruption and associated ability for error correction does differ between the two encodings.

I only have a Denon 1520ae, a Fiio K7 dac and Dali Zensor 7 and i can hear the difference clearly on this cheap system.

Even the most basic entry hi-fi system can clearly show the differences, so don't get bogged down in hardware tiers.

A decent dongle DAC and IEMS will reveal the difference. Headphones like the Sennheiser HD560s, which is usually under $200, will reveal the difference. You can even upsample PCM and hear the difference with something like Audirvana, HQPlayer, or Neutron player, though it won't be as great as music which originated as DSD.

Most of them use analog compressors and pre-amps in recording, tube or solid state. That being said, in mixing, there has been a lot of progress in digital plugins that sound like analog outboard gear, and some of the most famous mixers use mixing in the box these days. Pure analog on tape is almost gone in recording, even with the biggest bands. Did you know that The River album by Springsteen was recorded on 16bit digital equipment.

Looks like there is 2 from bubu land here to, why write something like that?

More La, La Land, Paul! Rocky Mountain High dreams of capturing AND releasing great sound is a noble pursuit!

Because using DSD as a consumer is a big hustle. As simple as that. (+ in order to edit DSD, as just a random consumer, you have to convert it to PCM anyway)

I'm guessing for the same reason SACD isn't mainstream

@@djlolerkoster What editing? I want to archive my albums. Previously, employed a Teac 2300SD, then B 77, both biased for Grandmaster 456 tape, which I bought from Fred Locke Studios in Berlin, CT back in the mid '70's. That turned out to be a disaster for the long term.

@@xaviermontalban717My guess is that SACD was way overpriced. Had they used a modest up charge the market response would have been far different.

One of the reasons is the high amount of resources required for it to become a mainstream standard. We've gotten closer to it becoming a reality, but apparently we're not there yet.

I've found that PCM sounds better than DSD on Chord devices. I think it has to do with how their FPGAs upsample all audio. Seems to be a better synergy for PCM than DSD.

@glenncurry3041: Regarding "DSD should be editable": The problem is that the DSD signal's value at any point in time (i.e., any sample) does NOT represent the analog signal's actual (absolute) value at ANY point in time (as IS the case for PCM). It only encodes the DIFFERENCE of the current sample from the previous sample. For PCM, each sample answers the question "What is the analog signal's voltage now?", over and over and over again, 44,100 times per second (or whatever the sampling rate is), with 16 bits of resolution (65,536 different levels) or whatever the bit depth (resolution) is. You can "cut" the signal stream at any sample point and do whatever you want with it - make it louder, softer, add (mix) it with another stream, etc., all in that same digital signal processing mathematics domain. With DSD, each sample only answers the question "Is this sample louder or softer than the previous one?", over and over and over, 1.5 million times per second (or whatever the sampling rate is), with only 1 bit per sample. It's all only relative - no absolute reference level - except for the very beginning of the stream. You have to have the entire stream to reconstruct the original analog signal. You can't just "cut" (edit) the digital stream at any random point (sample) and edit it in any way (louder, softer, mix with another signal, etc.). I hope that helps.

@@boblehman1726 I am well aware of the technology behind DSD and PCM. And: "With DSD, each sample only answers the question "Is this sample louder or softer than the previous one?", is wrong. DSD is ONE bit. There is no "softer" only "louder". Each bit is determined by a comparator to determine if the voltage is higher than the existing level, YES/NO. So no you confuse rather than help. In DSD playback bitstream each bit has one task. Turn on Yes/no. If 8 streams are being mixed the output is either turn on that bit Yes/No. Not turn on 8 bits higher or 6 bits higher. Just ONE bit YES/NO. By time domain multiplexing as is done in video to remove zizzies, edge crawl, movement artifacts and such, calculations can be done outside of instantaneous voltage values. One of the early devices to do so and Ampex ADO (which I sold) had dedicated CPU power equivalent to a Super Computer of the time to do that processing. But in the end each bit of the new DSD data stream is either 1 or 0. Either that data sample is saying turn on or not.

@@glenncurry3041 Explaining how DSD works immediately gets very complex, too much so for this forum and most of its audience to be helpful for conceptual understanding (without getting bogged down into circuit modules, algorithms, noise/distortion generation, noise shaping, etc.). I only wanted to focus on the concept for why typical simple editing functions (so easily done in analog or PCM) are so difficult with DSD. Here are some simple excerpts from web sources such as Wikipedia, WhatHiFi, Motorola, Audiophile Inventory, etc. that should be helpful to interested readers: "Direct Stream Digital (DSD) is a high-resolution audio format that uses a single bit to sample an analog waveform and determine if it's higher or lower than the previous sample. ... DSD is a 1-bit Sigma Delta Modulated (SDM) audio format developed by Sony and Philips. ... DSD uses a single bit of information, and all [that] this information tells us is whether the current sample of the analogue waveform is higher or lower than in the previous one. Compared with the over 65,000 different values 16-bit PCM has, the two values (0 if the new sample [o]f the signal is lower or 1 if it’s higher) of DSD appear mighty limiting. That resolution shortfall is made up by the very high sampling rate of over 2.8 million times a second - that’s 64 times the speed of CD. ... If you could look at a DSD digital stream, it’s possible to draw the analogue waveform simply by looking at the density of 0s and 1s. The more 0s there are, the lower the waveform goes, and it’s the opposite for ones. Where there is a balance of the two values, we’re at, or close to, the zero signal point. ... That all looks good, but there are also issues with DSD. It’s not very practical to manipulate a DSD recording, for starters. All the things that are required post-recording such as equalisation, editing, dynamic range control and adding reverb usually involve the DSD stream being converted to PCM to do the processing and then switched back. That’s hardly a pure way of doing things, right? Just about every studio recording made with DSD goes through this process. It’s down to a lack of suitable equipment and processing software. You may come across the term DXD on a recording. This is where the original DSD signal has been converted to 24-bit/352kHz PCM and processed in that form throughout. While the name (intentionally or not) sounds like DSD, it is simply very high-resolution PCM. ... DSD uses delta-sigma modulation, a form of pulse-density modulation encoding, a technique to represent audio signals in digital format, a sequence of single-bit values at a sampling rate of 2.8224 MHz. This is 64 times the CD audio sampling rate of 44.1 kHz, but with 1-bit samples instead of 16-bit samples. ... DSD is simply a format for storing a delta-sigma signal without applying a ... process that converts the signal to a PCM signal. ... DSD differs from the PCM format used by compact disc or typical computer audio systems: while PCM uses a multi-bit value (representing a large range of amplitudes) at a low sample rate, DSD instead uses a single-bit value (representing an increase or decrease in amplitude) at a sample rate much higher than the signal's bandwidth. The process of creating a DSD signal is the same as ... a 1-bit delta-sigma analog-to-digital converter (ADC) ... . The short-term average of the 1-bit DSD bitstream signal is proportional to the original signal amplitude. DSD music mixing and mastering for SACD or Internet download presents challenges due to the difficulty of performing digital signal processing (DSP) operations (such as equalization, balance, panning) in a one-bit environment. Older analog recordings were processed using analog equipment and then digitized to DSD. It is also possible to avoid processing by using only the available adjustments in the studio equipment while recording to DSD. One DSP technique available is to convert the DSD to PCM and use standard PCM equipment such as Pro Tools, useful for rock and contemporary music which rely on multitrack techniques, then digitally convert back to DSD format. Some DSD proponents dislike this technique claiming that the PCM conversion to a lower sample rate reduces the sound quality of DSD. A format and set of tools for PCM processing of DSD has been developed under the name Digital eXtreme Definition (DXD). This is a PCM format with 24-bit resolution sampled at 352.8 kHz. ... DXD was initially developed for the Merging Technologies Pyramix workstation ... in 2004. This combination meant that it was possible to record and edit directly in DXD, and that the sample only converts to DSD once before publishing. This offers an advantage to the user as the noise created by converting DSD rises dramatically above 20 kHz, and more noise is added each time a signal is converted back to DSD during editing. The Pyramix Virtual Studio Digital Audio Workstation allows for recording, editing and mastering all DSD formats up to DSD256. ... The coarsely-quantized output of a delta-sigma ADC is occasionally used directly in signal processing or as a representation for signal storage (e.g., Super Audio CD stores the raw output of a 1-bit delta-sigma modulator). ... Delta modulation is an earlier related low-bit oversampling method that also uses negative feedback, but only encodes the derivative of the signal (its delta) rather than its amplitude. The result is a stream of marks and spaces representing up or down of the signal's movement, which must be integrated to reconstruct the signal's amplitude. ... 1-bit delta-sigma modulation is pulse-density modulation: In the specific case of a single-bit synchronous ΔΣ ADC, an analog voltage signal is effectively converted into a pulse frequency, or pulse density, which can be understood as pulse-density modulation (PDM). A sequence of positive and negative pulses, representing bits at a known fixed rate, is very easy to generate, transmit, and accurately regenerate at the receiver, given only that the timing and sign of the pulses can be recovered. Given such a sequence of pulses from a delta-sigma modulator, the original waveform can be reconstructed with adequate precision. ... [S]igma-delta modulation was developed as an extension to the well established delta modulation. ... Delta modulation is based on quantizing the change in the signal from sample to sample rather than the absolute value of the signal at each sample. ... In each clock cycle, the value of the output of the modulator is either plus or minus full scale, according to the results of the 1-bit A/D conversion. When the sinusoidal input ... is close to a plus full scale, the output is positive during most clock cycles. A similar statement holds for the case when the sinusoid is close to minus full scale. In both cases, the local average of the modulator output tracks the analog input. When the input is near zero, the value of the modulator output varies rapidly between a plus and a minus full scale with approximately zero mean. ... Pulse-density modulation, or PDM, is a form of modulation used to represent an analog signal with a binary signal. In a PDM signal, specific amplitude values are not encoded into codewords of pulses of different weight as they would be in pulse-code modulation (PCM); rather, the relative density of the pulses corresponds to the analog signal's amplitude. The output of a 1-bit DAC is the same as the PDM encoding of the signal. In a pulse-density modulation bitstream, a 1 corresponds to a pulse of positive polarity ... and a 0 corresponds to a pulse of negative polarity ... . A run consisting of all 1s would correspond to the maximum (positive) amplitude value, all 0s would correspond to the minimum (negative) amplitude value, and alternating 1s and 0s would correspond to a zero amplitude value. The continuous amplitude waveform is recovered by low-pass filtering the bipolar PDM bitstream. ... In pulse-density modulation, a high density of 1s occurs at the peaks of the sine wave, while a low density of 1s occurs at the troughs of the sine wave. ... Sound/air oscillation each moment has a certain position (instant pressure level). The pressure level after the microphone is converted to an electrical oscillation. The electrical oscillation at each moment has a certain position that refers to the air pressure level. Pulse Code Modulation consists of samples. The sample is a measurement (number value) of an instant position of the electrical oscillation (voltage) at the moment of the measurement. In sigma-delta modulation (a.k.a. DSD) each 1-bit sample shows a positive [1] or negative [0] change of the pressure level relatively to its previous value. And, as rule, the sigma-delta modulation is explained as a saw (level grows or falls) between samples. ... Is native DSD editing possible? Audio editing is cutting, merging, gain altering, equalizing, etc. of file(s). "Native editing" (without conversion to multibit and DSD re-modulation) is possible for cutting and merging onebit records in bit-perfect mode only. All other kinds of editing demand conversion to multi-bit values and re-modulation. In this case, losses of the editing are comparable with multibit resampling. ... Thus, audio tracks in 1-bit format should be converted to PCM audio (filtered or DXD). After it the[y] are mixed and mastered. Final mix is modulated to DSD back.

@@boblehman1726 DSD is as simple as it gets. Compare current signal level to the next sample. Is it higher? Yes/No? But to you this is "very complex, too much so for this forum and most of its audience to be helpful for conceptual understanding"? So you obfuscate as a red herring? SAD! But the length of the reply seems to make you feel better? Good for you!

there's nothing really about dsd that you couldn't do operations to it as dsd like mixing and cutting, keeping track of that current values supposed to be. as for why distribute in dsd.. well you could mix it in pcm at 4x(or whatever) quality/rates and then retain some of that in the dsd. not that it matters. it certainly doesn't matter if your dac takes 16/24/32 bits at a time in or such. you know what the dsd myths resemble? early mp3 myths, such that an mp3 would be impossible fundamentally to seek etc.

@@lasskinn474 I think of MQA when DSD gets hyped too much. Underlying commercial interests are skewing the message. For calculations to happen on a DSD track, you need to convert the track to PCM.

If the DAW supports rendering in PCM and DSD, the DSD version will be closer to the lossless format, and it'll sound more like what you hear when playing back the multitrack. Even if you're using plugins which require PCM conversion, there's another step of PCM conversion on the output rendering. On a DSD output rendering, you're skipping that PCM conversion and simply writing the analog output waveform onto a digital medium. At least that's my experience with trying it both ways.

@@arthurkillen396 No, it doesn’t work like that at all. You can’t mix tracks of DSD without first running a transcoding into PCM streams. So first your DSD data will have to be converted to PCM, then you can process it, and afterwards you have a PCM master. If your DAW outputs DSD, it still means it ended up with a lossy transcoding from an internal PCM stream to an external DSD stream. It might make you feel it produced DSD, but really it had to make a PCM version first and the output will only be less than that actual master PCM version. If you think there is some gear that can output DSD as the genuine master in a digital music production, name me the gear or software exactly. Theoretically, you could run a DSD-256 into PCM at 256x44.1kHz=11.2896MHz PCM 32 bits and do the math with such insane PCM frequency and convert it to DSD-256 again and claim you maintained the original time resolution. But I don’t believe anyone does that???

This is quite simply incorrect. Though, a good try. All modern DACs use a form of PDM to work with PCM or DSD. It's a Sigma Delta Modulator at the heart of modern DACs (essentially multibit DSD). If you run single rate DSD then yes, the noise shifting requires a fairly steep filter to lower it. However, running 2X to 4X pushes the noise way out into easy territory. And in a direct A/B DSD sounds remarkably better than PCM.

Not all Dacs use sigma delta conveniently forgetting R2r dacs ?

@@Paulmcgowanpsaudio High-resolution PCM and DSD formats of comparable resolution are statistically indistinguishable from one another in blind listening tests. Loudspeakers and room acoustics are sar, sar more important.

@@geoff37s57 Well we can certainly agree on the importance of rooms and loudspeakers. Be careful swallowing the blind listening test bit. You (likely) don't know the experience level of the listeners nor the resolving power of the system, nor the environment and circumstances of the test. That all matters.

​@@pascalne2414you hit the nail there have a Sony cdp-x7esd with burr brown Comparing it with the latest dacs or players with delta sigma it simply can not be beat details are missing with delta sigma

And ears, and chest, and the fun 'gland'. :)

I am not selling any DSD products, rather listening to them, and I hear immediate and huge differences between DSD and classic PCM standards (CD, FLAC, nevermind MP3). A lot of PCM stuff is very nice and enjoyable, but pretending it is perfect and unimprovable is dishonest.

This is not true

@edfort5704 Have you done these comparisons on the same music fone for both formats?

@@craigellsworth3952 If you mean the same songs or the same music types, yes, both.

So then why don’t you make a video on your channel about PCM, DSD etc? and spread your absolute vast knowledge of the subject so others can know the truth rather than waste it on bashing paul. I’m sure many could benefit since you have all the correct answers.

@@Audiofreak71 Because the Digital Show & Tell video already exists. Pointing out that Paul is wrong is not bashing him.

@@ClaytonMacleod well i already know it’s out there and i know the answer as well , but what’s the point of pointed out he’s wrong? Makes no sense to me unless it’s self gratifying to you? Either way I find it strange when people find it necessary to point out a mistake or wrong answer a professional has made as if it discredits their whole life’s work due to one mistake.

@@Audiofreak71 Dude, he's telling others, and others believe him even if he's wrong, because they don't know any better. So if Paul instead had the correct information and shared that instead that means not only does he now possess the correct answer, everyone he tells that believes it is the correct answer will also have the correct answer. Spreading correct information rather than bogus information is a Good Thing. TM It is better for everyone, including him, if he both has and shares correct information. Has absolutely nothing to do with me or how I feel about myself. Smarten up.

Paul should block you for that comment. He is the most supreme audiophile there ever was. He uses his time to make these videos and tries to teach feeble minded people stuff for free and this is the thanks he gets? If I was Paul, I'd send everyone to your channel to harass you for a few weeks but he is far too nice of a guy to do that.