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DSD and PCM. Real competitors?

Educational line

Introduction

Many disputes risen around DSD vs. PCM competition. DSD and PCM is not so different as seems. Here we consider some myths and its technical explanation.
 

DSD and PCM. Real competitors?

 

If resume some wide spread opinions:

1. DSD can considered as format superior in audio quality than PCM.

2. DSD can’t be "native" edited without intermediate conversion to PCM.

3. Point 2 give significant loss of quality due decimation.

Below we will consider the validity of this assertions. 


What is digital audio format quality


First we can define: what is "audio quality" that will discussed below.

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

There are no any esoterics. It fine detected via spectral-time analysis in different forms.

Sometimes try detect identity degree via measuring of simple difference of original and restored signal.

Spectral method is more informative and sensitive to possible distortions.
 


Physical base DSD and PCM

Differences PCM and DSD (sigma delta modulation) not so strong , as seems.

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

DSD and PCM physical principles comparison
 

There are difference in quantization noise distribution and behavior.

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

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

PCM quantization noise correlate with useful signal: more signal - more noise.

DSD noise don’t depend on signal and present during silence too. DSD DAC eliminate this noise.

 

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ADC

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

There are many types of ADC.

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

Input PCM DAC


Otherwise the stuff will shifted/mirrored into range of low half of sample rate.

For DSD DAC enough suppress all what above sample rate, theoretically.

Practically recommended suppress all above transmitted audio band 0 … 20 kHz (may be slightly more). For avoiding transmitting excess energy, what consume resources of dynamical 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 a difference of frequencies in 2 times.

More recorded sample rate - more suppression - more quality of captured sound.

DSD ADC have significantly higher sample rate than PCM DAC. It provide better suppression in forbidden frequency range.

There all excess stuff can be further filtered in digital form.

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

Simpler decision is using of fast-growing saw voltage for measuring input analog value.

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

 

DAC

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

DSD DAC is simple low frequency filter (that pass low frequency - music stuff - only).

Higher sample rate than for PCM, simplify using of analog filter. No need so steep transient to suppression area as for PCM.

No need so many precise components.

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

If use DSD as end-user format need 1 precise reference voltage and simple analog filtering only.

I.e. same result with less efforts than "native" PCM.
 


Problems with restoring some signals

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

There more steep front and less ringing in front/end sides of square impulse showed as DSD advantage.

Let consider how ideally digitize/restore the square wave.

Square wave have infinite spectrum. I.e. for ideal digitizing/restoring need infinite sample rate.

Sample rate DSD significantly higher sample rate of PCM. It is reason of steeper front/end of the square impulse.

Square wave DSD vs. PCM


Lower ringing for DSD is result less steep filter than used for PCM, that have lower sample rate.

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

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

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


With increasing PCM sample rate possibly achieve steeper front/end of the square impulse too.

I.e. no difference between DSD and PCM for restoring square. There are values of sample rate and filter steepness only.


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

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

Sometimes refers to the article.

However the article consider brain analysis audio environment via principle dissimilar by Furie.
That allow discriminate short time quants of audio content.
However there no word about new in mechanical capabilities of human ears: to listen up to 20 kHz.

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

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

So need provide maximal fidelity in 0 … 20 kHz range.

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

Inside our head we have same less steepness of front level for ideally played back on speakers anyway.

It is feature of our ears.

So, why need restore square form better than can receive our ears?

Upper range after restoring can be shifted to audible range due non-linear distortions.

And will analyzed via "principle dissimilar by Furie" :)

 


DSD edit vs. PCM edit

Almost everybody know what impossibly "native" edit DSD.

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

Need consider two things:

1. What is PCM?

2. What bad in intermediate converting to PCM?

PCM is format where each sample is multi bit value.

PCM have quantization noise that damage "pure" DSD during conversion.

Problem of quantization noise simple solved via increasing bit depth.


Also often possibly listen "scary" word "decimation".

Decimation is simplest removing excess samples for decreasing sample rate.

Before decimation need filter all frequencies upper half output sample rate due avoiding distortions in audible range.

This filtering can cause ringing artifacts.

However often forgotten that qualitative filtration in total have significantly less artifacts than mixing and effect processing.


Why need conversion to PCM? PCM have no noise in upper part of spectrum. It allow successfully apply non-linear processing (as example, overdrive/distortion effects) to musical stuff.


As alternative suggested for using miltibit DSD. However "multibitness" allow solve only mixing and simplest multiply volume changing.

Elementary changing level to 1 dB becomes trouble.

Multibit DSD have noise in upper part of spectrum too. Thus for apply non-linear processing need converting DSD to "PCM".

Also need remember that computer can apply multibit math processing only.

End-user of editing system no need worry about intermediate conversion(s). It is engineer's troubles how to find "hidden" possibilities and what "tricks" need apply.

Need consider editing system as ready decision with certain features at input and output.
 


How compare quality DSD and PCM

Main trouble, what DSD and PCM technically impossibly compare as digital formats.

Possibly compare only systems that use either PCM or DSD or both.

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

Final result depend on used components, adjusting, technical decisions.
 


Conclusion


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

1.  Technically impossibly compare DSD and PCM as pure digital formats. Need compare released systems, that use these formats.

2. Editing DSD with qualitative intermediate conversion (decimation) to PCM give less distortions than mixing and effects. So here is not a bottle neck.
    For minimizing loss recommended use pro-quality algorithms and "hidden" technical possibilities (field for inventors).

3. "Native» DSD processing can be performed via precise miltibit-DSD format. However, possible elementary operations only.

4. DSD is perfect as recording and end-user format due simpler (cheaper) apparatus than «native» (voltage matrix) PCM ADC/DAC.

5. Master recommended save in music production project format (precise PCM). After possibly convert in 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.

Need use strong sides of each format in music eco system.

 

 

Yuri Korzunov,
Audiophile Inventory's founder,
2015 September 09
 

Read als Why We Can't Compare Different Audio Formats

 

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