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DSD vs PCM [Myth Debunking, Read Definitive Guide 2019]

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Audio Basis - educational articles

yuri korzunovDSD (DSF, DFF files, SACD) competition with PCM (FLAC, WAV, AIFF, mp3, others) generates many disputes around. But DSD and PCM formats have common features and principles. Read about sound quality comparison, myths and its technical explanation by audio software developer Yuri Korzunov.

 
 
WARNING:
• DSF, DFF, ISO (1-bit audio) is supported in maximal PROduce-RD and configurable Modula-R
• For ISO tracks, DSF, DFF with length more 3 minutes FREE demo mute 2 second silence in the output middle
• DVD ISO is NOT supported

 

Watch and share: DSD versus FLAC [Format Comparison]
 

 

When we think about new DAC, we ask: Is DSD worth it? What is better: DSD or PCM? What about dsd vs vinyl? Or DSD 2.8 versus 192 24 PCM?

DSD is implemented in:

PCM is implemented in WAV, AIFF, FLAC, ALAC, mp3, CD-audio, etc. (PCM files)

Vinyl, as the analog mechanical source, cause more distortions, than digital DSD. But some audio distortions may be considered as subjective sound advantages. So this matter is complex enough.

 

What about PCM versus DSD?

Both PCM and DSD are pure digital formats. By summing up some widespread myths:

  1. DSD can be considered as format superior in audio quality than PCM.
  2. DSD can be "native" edited without intermediate conversion to PCM.
  3. Point 2 give significant quality losses due decimation.


Below we will consider technical explanation and debunk these myths.

Watch and share: What is DSD audio?
 

 

Read also: DSD vs FLAC >

 

 

What is the sound quality of digital format

First, we can define: what is "sound quality" to discuss below.

Digital audio format quality is identity degree of restored (from the digital form) and original analog waveforms.

There is no mystery. It fine detected via spectral-time analysis in different forms.

Identity degree may be detected via the measurement of simple difference between original and restored waveforms (sample difference).

However, the spectral method is more informative and sensitive to distortions.

 

 

DSD and PCM physics

PCM (pulse code modulation) and DSD (sigma-delta modulation) difference is not so strong, as seems at first glance.

Both kinds of modulation contain carried (musical) signal in the most low part of spectrum.

DSD and PCM physical principles comparison
 

The difference in quantization noise distribution and behavior is there.

For PCM quantization noise is evenly distributed across range {0 … [sample rate]/2}.

For DSD noise is pushed to inaudible (high) part of the spectrum. To push (noise shaping) significant energy of noise out of audible range, the reserve of the total band is need. I.e. higher sample rate, than for PCM, is needed.

PCM quantization noise correlate with useful signal: no signal is no noise.

DSD noise doesn’t depend on signal and present during silence too. DSD DAC eliminates this noise.

 

 

DSD vs DXD

DXD is high-resolution PCM, that extracted from DSD with the keeping of "legacy" ultrasound noise. DXD compatible hardware must filter the noise. Otherwise, ultrasound noise can cause audible products due to non-linear distortions in playback audio system.

DXD is designed for DSD editing. However, some processing has non-linear distortions and the ultrasound noise can cause audible products.

 

 

Nyquist theorem (PCM vs DSD)

The Nyquist theorem is the same for both DSD and PCM. In both cases, the upper half of spectrum like to mirrored lower one. I.e. useful spectrum consumes lower half of the spectrum. Upper half should be filtered to restore analog signal.

For DSD high part of the spectrum into the lower half, should be filtered too. Because it contains high-frequency noise of sigma-delta modulation.

 

 

ADC

Each audio application begin from analog-to-digital converter (ADC).

There are many types of ADC.

ADC must provide suppression all stuff over [sample rate/2] before analog signal digitizing.

DAC input


Otherwise, the stuff will be shifted/mirrored into the range of low half of the sample rate. Read more here >

Practically, it is recommended to suppress all above transmitted audio band 0 … 20 kHz (maybe slightly more). It's necessary to avoid transmitting of energy excess, that consumes resources of dynamic range.

For PCM and DSD ADC suppression provided via analog low-frequency filter only. Filter have slope suppression characteristic by frequency (suppression about 20 … 48 dB per octave).

Octave is 2 times difference of frequencies.

More recording sample rate - more suppression - more quality (lesser distortions) of captured sound.

DSD ADC has significantly higher sample rate, than PCM DAC. It provides better suppression in the forbidden frequency range.

All excessive stuff may be filtered in digital form.

Using of resistor matrix in DAC demands very high precision of components and voltage.

The simpler decision is using of fast-growing saw voltage to measure input analog value.

This principle is the principle of DSD. I.e. DSD is simpler/cheaper format for sound capturing, than PCM.
 

 

 

DAC

Applying of DSD DAC allow maximally simplify scheme and DAC adjusting.

In the first approach, DSD DAC is a simple low-frequency filter (that pass low frequencies - music stuff - only).

Higher sample rate, than for PCM, simplify using of the analog filter.  So steep transient to suppression area, like PCM, is no need.

So many precise components is no need too.

Almost all modern DAC use internal PCM to DSD conversion "on fly" for digital-to-analog conversion.

If use DSD as end-user format,

  • 1 precise reference voltage and
  • simple analog filtering

are needed only.

I.e. same result with fewer efforts than "native" PCM is there.

 

Read more about PCM and DSD DAC comparison >

 

 

Restoring issues of some signals

Not once digitizing/restoring to the analog of the square wave for PCM and DSD was compared.

More steep front and lesser ringing in front/end sides of square impulse are considered as DSD advantage.

Let's consider "ideality" of digitizing/restoring of the square wave.

Square wave has the infinite spectrum. I.e. for ideal digitizing/restoring, infinite sample rate is needed.

The DSD sample rate is significantly higher, than sample rate of PCM. It is the reason for steeper front/end of the square impulse.

Square wave DSD vs. PCM


Lower ringing for DSD is the result of lesser steepness of the filter, than one used for PCM. Because PCM has lower sample rate.

Another side, using wider (more 20…24 kHz) bands for DSD give more noise energy, that fast growth upper 24 kHz.

I.e. price of a better form of square is higher noise level.

Lesser ringing due to lesser steepness of DSD DAC filter (less ringing) leads to worse filtration. Thus it causes a higher noise level.


With increasing of PCM sample rate, steeper front/end of the square impulse may be achieved too.

I.e. no difference between DSD and PCM for restoring square. Values of sample rates and filter steepness are there only.


Now let me ask: why us need to restore ideal square for audio applications?

While only one solid practically proven theory exists: humans listen up to 20 kHz.

Sometimes we can see references to the article.

However, the article considers brain analysis of audio environment via principle dissimilar by Furie.
It allows discriminating short time quants of audio content.
However, there no word about new in mechanical capabilities of human ears - listening up to 20 kHz.

Therefore, why we need ideally re-create form of square impulse?

In audio applications we listen via ears, don’t watch via eyes.

So it is necessary to provide maximal fidelity in 0 … 20 kHz range.

It leads to visible (by eyes!) lesser steepness of fronts.

Anyway, inside our head, we have the same lesser steepness of front level, that ideally (theoretically) played back on speakers.

It is a feature of our ears.

So, why restoring of square form better, than can receive our ears, is needed?

Upper range, after restoring, can produce audible products due to non-linear distortions.

And the range will be analyzed via "principle dissimilar by Furie" :)

 

 

 

PCM vs DSD editing

Almost everybody knows that it is impossibly editing DSD "natively".

Here «native» means editing without intermediate converting to PCM. Such converting is "very-very bad"!

It's necessary to consider two things:

  1. What is PCM?
  2. Why intermediate converting to PCM is bad?

PCM is format with multi-bit samples. Adding bits to DSD change nothing.

Probably you listened "scary" phrase "destructive decimation".

Decimation is the simplest removing of excess samples to decrease sample rate.

Before decimation, the filter of all frequencies upper half output sample rate is applied, to avoid distortions in the audible range.

This filtering can cause ringing artifacts.

However, qualitative filtration has significantly lesser artifacts, than mixing and effect processing in music production.

 

Why conversion to PCM is needed?

PCM has no noise in the upper part of the spectrum. It allows to successfully apply non-linear processing (as for an example, overdrive/distortion effects) to musical stuff.


As an alternative, multi-bit DSD is suggested. However, "multibitness" allow solving only mixing and simplest multiply volume changing.

Elementary changing level to 1 dB cause trouble.

Multi-bit DSD has noise in the upper part of the spectrum too. Thus, to apply non-linear processing, converting DSD to "PCM" is desirable.

Also, it needs to remember, that computer can apply multibit math processing only.

End-user of the editing system should not worry about intermediate conversion(s). It is an engineer issue, how to find "hidden" possibilities and what "tricks" are need to apply.

It is necessary to consider editing system as a ready decision with given features at input and output.

Read more about DSD editing >
 

 

 

How to compare DSD and PCM sound quality

 

The main issue is DSD and PCM are technically impossible (including blind test) to compare as digital formats.

It is possible to compare only real systems, that use either PCM or DSD or both.

The systems must be considered as "black boxes" with input and output analog signal. These signals can be compared via spectral methods or hearing tests.

Final result depends on used recording, components, settings, technical decisions.

 

 

Conclusions

  1. Digital audio format quality is identity degree of restored (from the digital form) and original analog waveforms.
  2. Comparison of DSD and PCM as pure digital formats is impossible by technical reasons. It needs to compare released systems, that use these formats.
  3. Editing of DSD files with qualitative intermediate conversion (decimation) to PCM give fewer distortions, than mixing and effects. So bottleneck is not there.
    For minimizing of losses, using of pro-quality algorithms and "hidden" technical possibilities (field for inventors) is recommended.
  4. "Native» DSD processing can be performed via multibit-DSD format. However, elementary operations are possible only.
  5. DSD is perfect as recording and end-user format due simpler (cheaper) apparatus than «native» (voltage matrix) PCM ADC/DAC.
  6. It is recommended to save master in music production project format (precise PCM). Further, it is possible to convert the master to any format: lossless PCM, DSD, lossy.

DSD PCM in music life cycle

 

Practical superiority of quality for each system is subject of engineer art, but not used format.

Use strong sides of each format in music ecosystem.

 

 

Yuri Korzunov,
Audiophile Inventory's founder
 

 

 


Read more about DSD
  1. What is DSD audio >
  2. DSD Decoder Audio >
  3. How work sigma delta modulation in audio >
  4. DSD Converter of Audio Files: What Inside? >
  5. What is Audio Formats DSD 2.8 DSD64 DSD5.6 DSD128 DSD256 DSD512 DSD1024 >
  6. DSD vs DSF vs DFF Files Audio. What is difference >
  7. DSF vs PCM. What is common base? >
  8. DSD versus FLAC (look to infographics) >

 

 

Read also

Why We Can't Compare Different Audio Formats >

DSD and PCM. Real competitors?

 

 


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