1. Dsd Playback Software
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3. Dsd Editing Software For Mac No Watermark
4. Dsd Editing Software

Mar 05, 2014  Download DSD Master for macOS 10.7 or later and enjoy it on your Mac. ‎DSD Master is a professional-grade tool for producing PCM versions of music files encoded in the DSD format. DSD Master also produces BitPerfect Hybrid-DSD files required for playing native DSD.

AuI ConverteR 48x44 is DSF converter to/from WAV, FLAC, AIFF, DSF files. Conversion (1-bit audio) ISO, DFF, DSF 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 OSX, Windows.
Look quality, format, configuration issues below. Check AuI ConverteR's minimal price options >

General issues

Aims of music conversion:

• format adjusting (DSF, DFF, FLAC, WAV, etc.) to hardware / software abilities,
• resolution adjusting [sample rate, bit depth, 1-bit sigma delta modulation (a.k.a. DSD) or PCM] to hardware / software abilities,
• using of audio system resolution, that cause lesser distortions.

DSF is audio file format (.dsf extension) with high resolution DSD audio stream. DFF file is alternative DSD sound format container. A DSD converter between the files (containers) may be implemented as lossless re-packer of binary identical musical content. Or the conversion tool may have appropriate mode to repacking without losses.

Simplified DSF file structure:

• header
• DSD sound 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 PCMconversion audio quality depend on signal-processing precision. But it is lossy action anyway.

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.

Dsd Playback Software

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

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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

Read about quality comparison of PCM and DSF formats. 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 instability of sigma-delta modulator. However higher sample rate allow to reduce noise and get wider musical signal band.

Price options of DSF converter

Minimal price options

Minimal configuration to convert DSF/DFF (D64/128) to FLAC, WAV and other PCM (up to 192 kHz, 24 bit, 32 bit float):
[ Mac OS X ][ Windows ]

Minimal configuration to convert FLAC, WAV and other PCM (up to 192 kHz, 24 bit, 32 bit float) to DSF/DFF (D64/128):
[ Mac OS X ][ Windows ]

Minimal configuration to convert DSF/DFF (D64/128) to DSF/DFF (D64/128):
[ Mac OS X ][ Windows ]

The conversion software support DSF, DFF, ISO (D64, D128) as input (maximal PROduce-RD, configurable Modula-R editions to save budget):

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

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

Read articles about different kinds of conversions below.

Configure and order Modula-R right now or request us appropriate configuration here >

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 '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 articles about DSF Converter

Read also

References

DSD editing software is intended to altering DSF and DFF audio file content: gain, cutting, merging, EQ, etc. DSD is 1-bit signal based on sigma-delta modulation and its editing is not obvious at first glance unlike PCM. There are many disputes about 'native' and 'non-native' editing, damage of original stuff. Read how to edit DSD and about real damage of the editing.

1. DSD processing types

DSD and PCM oscillogramms

It is necessary to remember, that digital signal isn't 'stairs', because the 'stairs' is filtered by low frequency filter. And analog signal, restored from digital one is smooth.

We can imagine how PCM editor work. It apply math on each sample or group of samples of musical signal.

As example, to boost level we can multiply each sample of PCM to gain value.

DSD is 1-bit signal and it is difficult to understand what is possibly to do with 1-bit sequence.

When DSD sample is multiplied to gain value, the result value is different from both 0 and 1. And it isn't possible to store the value as 0 or 1.

To easier introducing into DSD editor matters, it is need to find out total base for one- and multi-bit stuff.

In general, DSD and PCM are the same. But there are several differences:

At the picture, spectrums of a quantization noise are depicted.

PCM and DSD transformation. What is difference

At left part of the picture PCM quantization noise is depicted. At the middle picture bit depth is decreased. At right part of the picture noise shaping applied to the quantization noise. This technique is used for DSD to improve signal-noise ratio in the audible band.

Read more details about DSD vs PCM here >

Comparison DSD vs FLAC (infographic) read here >

DSD have bit depth 1 bit.

But:

1. Modern computers 'natively' perform multi-bit calculations only.
2. Adding of two 1-bit numbers demand 2 bits for result.
3. What about result bit depth of non-integer multiplication (dB scaling as example)?

Therefore, for editing of 1-bit DSD sample may be converted to multi-bit sample for compatibility with processor unit. Further different processing may be applied like PCM.

Edited stuff must be reduced back to 1-bit. It may be implemented as sigma-delta re-modulation.

Simple scheme of DSD editor.
Upper part of the picture: spectrums per each stage are shown

In this scheme high-frequency 'legacy' DSD noise come to editing. It can cause:

• overload, that lead to broken stability of output sigma delta modulator (re-modulator),
• audible products of intermodulation distortions (audible noise).

DXD PCM format is partitially same such 1-bit to multi-bit conversion. But, there is filtering with decimation is performed. And DXD have integer format of samples.

Decimation is multiple sample rate division by deleting samples. Filtering should be applied before decimation. Because high frequencies may become audible if decimated signal.

Sample format is matter of target signal/noise ratio. Any further processing should be 'transparent' for target resolution. I'd recommend use 3...10 times better precision, than final bit depth. Such figures are same to measurement recommendations. But it is not hard rule.

For DSD, target precision is defined by noise level, that depend on quality of sigma-delta modulator and target sample rate. Therefore, to choose internal bit depth of the editor software, need to refer to available re-modulator's noise level at target sample rate.

Read details below.

DSD editor with filtering of 'legacy' DSD noise at high frequencies

For such way, probability of overload and level of audible intermodulation distortion products are decreased. Because lesser ultrasound content generate lesser intermodulation audible products.

DSD editor with filtering of 'legacy' DSD noise and decimation

Decimation allow to save computing-resource consuming (time of processing). Because sample number, that used for calculations, is lesser. However, simple processing may have lesser computing profit, than resources, spent to decimation and upsampling.

As example, if you perform gain altering only, this operation don't consume many time. Upsampling time losses may be more, than profit by decimation. But filtering give advantages, described in the previous case. And decimation reduce filtering computation number.

Therefore, during desing need to acount these things.

What issues is caused by processing of multi-bit DSD audio?

When 1-bit audio sample is converted to multi-bit one, it keep noise energy comparable with useful audio signal at high frequencies. It is excess energy. It does not contains information.

But the energy can cause:

• overload;
• intermodulation distortions.

These 2 things are most dangerous for sound quality in DSD editing.

Overload can cause stability disrupting of the re-modulator.

Intermodulation distortions can give audible products despite the DSD-noise is ultrasound. Especially need to account, that the noise level is comparabnle with useful audio signal.

Mixer (sum)

Mixing is a sum of original 1-bit samples, transformed to multi-bit values. Сonversion to 1-bit value is applied via sigma-delta re-modulation after summation.

DSD mixing audio

As example, 2 signals are mixed. Such operation sum full frequency bands with width [sample rate]/2 of both original signals.

Maximal level may growth up to 2 times. Also noise energy is growth up to 2 times.

'Up to' because sum level depend on phase of the original signals. If 2 identical signals are summarized with zero time shifting, total level 2 times more than each of input signals.

Noise is stochastic process. So two noise sum give stochastic value too. Double level value also probable.

If original signal have 0 dB (maximal) level, summation the signal with any other one cause input overload of the sigma-delta re-modulator.

Even if no overload then summing useful audio signals, summation of the high-frequency noise of both signals can cause overload.

To avoid the overload need use filtering.

Gain altering (multiplication)

Gain altering is multiplication. Multiplication may be either integer or float point. Gain altering step of audio applications in range 0.1 ... 2 dB, as rule.

DSD gain altering

To keep output signal precision, we can try to use only integer calculations with expanding bit resolution. But float point formats are significantly easier for programming. However, integer formats may be preferable for processing implementation on field-programmable gate array (FPGA) chips.

Overload isn't defined by level of separate spectral component, like music. But overload is defined by total energy of the spectrum, that include not only useful audio signal, but 'legacy' DSD noise.

Even if useful audio signal is lesser than maximal level, the total energy can cause overload of the re-modulator. Because there is the 'legacy' noise at high frequencies.

Gain reducing can some decrease probability of overload, because the 'legacy' noise is reduced too. But level boosting give higher overload probability.

Dsd Editing Software For Mac No Watermark

When the noise is filtered, overload is solely defined by useful audio signal.

There is opinion, that digital filtering is destructive, because it generate ringing artefacts after filter. But it is balance between the artifacts and maximal available level after DSD editing.

However, author don't know about researches of impact ringing to perceived audio quality. And ringing is matter of steep level transients rather. But musical signal is smooth, as rule.

If we use filter with 'deep' suppression in stop-band, we can apply decimation of samples to decrease sample rate and faster processing. Smooth filter can cause additional audible products transferred from high frequencies of original spectrum.

From point of view of my development experience, these matters is reference points only. And we can look for hidden opportunities around these points to achieve our aims.

Overdrive, tube and tape emulators, other enhancers, compressor (non-linear processing)

These kinds of processing are pure non-linear processing. When signal with high frequency noise pass thru non-linear circuit, the noise can generate audible products due intermodulation distortions. Main trouble here is noise energy that comparable with useful signal. Thus the audible intermodulation products may have significant level.

DSD enhancers

Therefore, non-linear processing can cause audible noise. Similar example we can watch in the video.

Compressing can not be considered as pure non-linear processing. Because in defined level range compressor is linear for momentary level value. However, linearity is changed during time of compressor gain switching. When the gain is switched and constant at new value, compressor work as linear device.

Video: Removing of audible products of intermodulation distortions

For correct non-linear processing, 'legacy' DSD noise should be suppressed before the processing. Otherwise audible intermodulation products of high frequency noise may have significant value.

Dsd Editing Software

Equalizer (amplitude-frequency correction of spectrum)

Equalizer (EQ) is linear processing. However, it can cause overload when equalization band is boosted. The overload can cause broken stability.

DSD equalizer audio

Reverberation (sum)

Reverberation is linear processing in general case. But there summation and multiplication are used. Also EQ may be applied. Convolution algorithm may have different effect to level and spectral energy. So overload at input of sigma-delta remodulator may happen.

DSD reverberator

Merging and spliting

These operations are lossless. Because it is only manipulations under samples without changing.

DSD splitting

DSD merging

There are 2 ways:

1. Using of integrated editor (1-bit/multi-bit converter [optionally: filtering and decimation] + processing + [optionally: upsampling] + re-modulator).
2. Using conversion DSD to PCM > editing PCM in audio editor > conversion PCM back to DSD.

DSD editing ways

When converter is used, it is recommended to use PCM formats with 32- or 64-bit integer or float bit depth to 'transparent' (for music) processing. Float point format have better overload abilities comparing integer ones.

Decimation in the converter can improve processing performance and compatibility with PCM editor software.

Steeper filter cause more ringing artefacts. But decimation after sloping filter can cause additional audible products by high frequency components.

Sample rate must provide 0 ... 20 kHz useful band.

Depending on converter implementation, higher sample rates can cause additional 'legacy' DSD noise due wider band. But some conversion software allow to choose passed band of input 1-bit signal, to reduce impact the modulation noise.

For PCM sample frequencies multiple 44.1 kHz, time of conversion (DSD to PCM) may be lesser, than for sample frequencies based on 48 kHz.

General recommendation to choose the conversion software is comparing by level difference between overload and noise/aliases (maximal dynamic range).

Read more about multiple vs non-multiple resampling here >

1. Before processing 1-bit samples should be converted to multi-bit ones.
2. After processing multi-bit samples need to re-modulate back to 1-bit ones.
3. Overload after processing is dangerous for the re-modulator.
4. Non-filtered DSD 'legacy' noise (after conversion 1-bit to multibit samples) can cause input overload of the re-modulator.
5. For non-linear processing the non-filtered noise can cause audible products (intermodulation distortions).
6. Losses of DSD editing without processing is similar to PCM-resampling.
7. Decimation can save time of processing, but cause additional noise for sloping filters before decimation.
8. DSD editing may be performed via either integrated DSD editor or DSD to PCM conversion, usual PCM editor and PCM to DSD back conversion.

References:

1. Josh Reiss, Mark Sandler. Digital audio effects applied directly on a DSD bitstream: link