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What is optimization audio for DAC

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Here considered issues of digital analog conversion audio and ways for improving quality of conversion.

Optimization audio for DAC

Bases of analog signal restoring from digital form

Simplest PCM digital analog converter (DAC) contains:

1. Circuit conversion of digital sample (number, value of electrical voltage) to electrical voltage

2. Analog low frequency filter.

Still no robust proofs that we hear ultrasound. By Naquist theorem enough 40 kHz sample rate + band reserve for transient band of the low frequency filter.

So we have 44 kHz sample rate standard. However real implementation of restoring audio stuff (especially low resolution) rise some issues of quality.


Spectrum of audio sample sequence converted to voltage (stairs) have:

1. useful spectrum in low half of spectrum.

2. mirrored useful spectrum an high half of spectrum.

In this article is not real scales in graphs, but scales for more clear explanation.


PCM coded analog signal

Analog signal restored from digital form

 

Ideal analog filter should:
1. fully pass by useful spectrum {0 … 20 kHz}
2. fully eliminate mirrored spectrum {20 kHz…[sample rate]}.

Ideal analog filter should provide transient band (area between pass and stop frequency bands).

Ideal analog filter in DAC

Analog filter’s frequency response showed as grey line.

 

Issue of restoring to real analog signal

The DAC’s analog filter have enough sloping transient between pass and stop band.

And it can’t deep suppress mirrored spectrum.
 

Real analog filtering in DAC

 

For ideal analog part of playback apparatus mirrored spectrum is harmless.

But real apparatus have non-linear distortions in analog parts of DAC, amplifiers, speakers, etc.
It lead to shifting of mirrored spectrum to audible range.
Therefore audible distortions appear because apparatus have non-linear distortions.
 

Audible distortions by ultrasound aliases for digital analog converter

 

How to decrease the distortions in audible range?

Level of the mirrored distortions in audible range depend on amplitude/power of mirrored spectrum. All that placed at upper half of the spectrum of restored digital signal.

Therefore we should decrease the power of mirrored spectrum.

We can’t significantly improve the real analog low frequency filter of DAC. It will slope for modern hi-fi demands.

However we can apply oversampling
 

Oversampling into DAC

and steep digital filter before analog.
 

Analog filter vs. digital filter

Oversampling create additional periodical mirroring of {0…[sample rate]} area at frequencies above original sample rate.

But digital filter better (than analog) can suppress artifacts about higher limit of audible range.
 

Oversampled signal after digital filtration

 

Now analog filter should suppress area of highest frequencies in oversampled spectrum.
Where analog filter have maximal suppression in band {0 … [new (oversampled) sample rate]}.
 

Analog signal after digital oversampling-filtration and analog filtration

 

Hardware vs. software oversampling-filtration

Realtime-hardware-digital-oversampling filter can have computing limitation due restricted resources on chip.

So there may be slope transient band issues too.

As alternative can be used additional offline or inline oversampling filter released in personal computer software.
Computer have big calculation power, including floating point precision.
 

Analog vs. digital hard vs. digital soft

 

There DAC’s digital-oversampling and analog filters applied to preliminarily filtered spectrum.

Pre-filtration oversampled audio stuff allow to decrease the mirrored spectrum level even more.


AuI ConverteR have resampling filter modes:

1. Optimized (cutting all above 20 kHz)

2. Non-optimized (traditional resampling mode)

3. Non-optimized wide (band up to 100 kHz for ISO, DSF, DFF conversion)

Click Settings button in AuI ConverteR's main window. Appear Settings window.

AuI ConverteR optimized and non-optimized modes

There need select mode via Filter mode list.



 

Inline vs. offline software oversampling-filtration


Inline filter can be released in audio player software. Audio file played without any pre-processing.


Inline conversion has next issues:
 

1. possible lack of computer’s performance for realtime processing.

2. consuming of significant part of total performance.

3. consuming additional electric energy for hard work. There rise additional heating (and sometimes cooling fan noise) matter each time of playback.



Offline filter solve the issues:

1. For old computers playback of pre-oversampled-filtered files is easier.

2. Released computing resources for other tasks (graphics, music database, network, etc.).

3. Economy of electrical energy during playback with lower noise of cooling fans of the audio computer.

Offline filter can be applied in audio converter / resampler. Pre-oversampled-filtered files used for playback in bit-perfect mode (unmodified binary content sent to DAC).
 


Resume

1. Quality of restoring analog signal from digital depend of sharpness of DAC’s analog filter.

2. Sharpness of analog filter technically restricted.

3. It solved via digital pre-oversampling-and-filtration builtin DAC.

4. In some cases, for better result may be used additional software pre-oversampling-and-filtration.

5. Inline realtime oversampling-filtration don’t demand pre-converting audio files.

6. Offline conversion allow apply heaviest oversampling-filtration on old computers and save electrical energy during playback.
 


Yuri Korzunov,
Audiophile Inventory's developer
20 May 2016,
updated 08 May 2017
 


 

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