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DSF File Converter Software

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AuI ConverteR 48x44 is DSF converter to/from WAV, FLAC, AIFF, DSF. Conversion ISO, DFF to DSF is supported. Sample rates 2.8(D64) / 5.6(D128) / 11.2(D256) / 22.5(D512) / 45.2(D1024) MHz. Intended for audiophiles and music production. Operation systems: Mac OS X, Windows.


Audio Converter DSF



Download HD audio converter AuI ConverteR 48x44 FREE for high resolution music
WAV, FLAC, AIFF, ALAC, ISO, DFF, DSF, WMA, mp3,...

WARNING: Input DSF, DFF, ISO supported in Audiophil-E, PROduce-R and configurable Modula-R
Output DSF supported in maximal PROduce-RD and configurable Modula-R


 

Video: Studio Audio Converter DSF to FLAC, WAV, AIFF, mp3, ...

 


General issues

Aims of music conversion:

adjusting format to hardware / software abilitied

adjusting resolution (sample rate, bith depth, PCM/DSD) to hardware / software abilitied

using audio system resolution, that cause lesser distortions

 

DSF is audio format (.dsf extension) with high resolution lossless audio stream 1-bit sigma delta modulation (well known as DSD). DFF file is alternative DSD container. Conversion between the containers may be performed lossless way: as repacking binary identical musical content.

Simplified DSF file structure:

  • header
  • DSD data
  • id3 metadata block.

Format specification link see below.

The DSD data divided by blocks. Last of block may be filled partitially.

Stereo 2.0 and multichannel 5.1 formats are supported. Also other channel number may be supported.

 

Music players able playback 1-bit stuff straight on DSD DAC (native 1-bit DAC) or convert in real-time ("on fly") to PCM format so that PCM DAC feeding (multibit DAC).

Conversion 1-bit sigma delta modulation to PCM is lossy. Physically the conversion is filtering high frequency noise. The filtering band may have different width. It define music signal band width. Other side, it cause more high frequency noise energy, that can cause audible intermodulation distortions at some playback systems.

DSD resampling, gain altering is lossy anyway.

All losses noted in this article part is not significant for professional conversion software.

 


Real-Time vs. Non-Real-Time Conversion

PCM to DSD, DSD to PCM conversion audio have quality loses. It defined by digital-signal-processing precision.

As rule, offline (non-realtime) converter DSF to PCM or PCM to DSF implemented on computer have fewer time computing restrictions, comparing with real-time processing. Offline processing may allow higher calculation precision, that consume additional time. Qualitative offline processing may be performed at old computers even.

If processing is performed once, it save electric energy that computer consume in intensive calculations.

 

 


DSD vs. WAV, FLAC, AIFF

Conversion quality of the conversion software close theoretical limits.

Approximate Audio Format Quality by Results These Tests
(the best at first place)

1. WAV PCM 32-bit float, DSD 256 and above

2. DSD 128

3. WAV PCM 24-bit

4. DSD 64

5. WAV PCM 16-bit

Comparing of quality of PCM and DSF formats here. In the article compared WAV vs. DSF. This comparing matched ISO (1-bit hi-res audio), DFF, AIFF, FLAC and any lossless formats.

PCM 32-bit, DSD 256 and above in the tests have measurement tool limitation. Therefore, all these formats were placed at first position.

DSD sample rate is matter of noise level inside useful band. Reducing the noise level can cause unstability of sigma-delta modulator. However higher sample rate allow to reduce noise and get wider musical signal band.

 

 


What edition is DSF converter

The conversion software support DSF, DFF, ISO (D64, D128) as input (Audiophil-E, PROduce-R, PROduce-RD, configurable Modula-R editions).

Only maximal PROduce-RD, configurable Modula-R editions support:

1. Convert WAV, FLAC, AIFF, ISO, DSF, DFF to DSF

2. DFS sample rates 11.2 (D256),  22.5 (D512), 45.2 (D1024) MHz.

 


Sound quality settings

If input and output sample rates are different, resampling signal processing is used. Resampling base on filtering with settings described below.

At Settings > General tab > select one of Filter mode's options:

  • - "Optimized resampling filter" cut frequencies upper 20 kHz (eliminating intermodulation distortions by ultrasound);
  • - "Non-Optimized resampling filter" remove all upper 20...24 kHz (depend on input/output sample rates) - eliminating intermodulation distortions by ultrasound, wider signal band;
  • - "Non-Optimized wide resampling filter" suppress all upper 20...100 kHz (depend on input/output sample rates).
  • At some equipment sotware "Non-Optimized..." modes can cause audible noise (see video).
  • Generally, "Optimized..." mode is recommended. But you can prefer other modes.
  •  
  • Use minimum phase filter switch checked/unchecked minimum/linear phase filtering when "Optimized resampling filter" is selected. Other modes use linear filtering only.
  •  
  • Dithering application have sense in 16 bit PCM target format only.
    If output bit depth 24 bit and above (including 1-bit, float point PCM formats), dithering auto turn off.

 

 


Read more

How to work DSD converter >

DSF specification >

ID3 metadata standard >




Download HD audio converter AuI ConverteR 48x44 FREE for high resolution music
WAV, FLAC, AIFF, ALAC, ISO, DFF, DSF, WMA, mp3,...

WARNING: Input DSF, DFF, ISO supported in Audiophil-E, PROduce-R and configurable Modula-R
Output DSF supported in maximal PROduce-RD and configurable Modula-R

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Last News

Where is the Limit of Audio Quality?

I read many endless discussions about audio equipment: cables, lossless vs. lossy, compression algorithms, jitter, ultrasound, playback files from different places of hard disk, power conditioners, etc.

I suppose, reason of the discussions is small perceived sound difference, that is not obvious for everybody, or no difference.

It is the end of high-end?

Below I want to discuss: can we get sound improvement, obvious for almost everyone, in future audio equipment?

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Tutorials

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