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Audio Basis - articles about audio
Music lovers compare various digital audio formats by sound quality. Probably, you already heard explainations how DSD audio (Direct Stream Digital) works. But it maybe is not as simple to understand why 1-bit DSD may successfully compete with traditional multi-bit audio.
Read an easy explanation of what is DSD, how it works, why 1-bit provides hi-fi sound quality, what is better DSD or PCM, myth debunking, where to download, and other interesting things.
- What is DSD audio?
- DSD music file formats
- How can I play DSD?
- Where can I get DSD music in files and SACD?
- How to convert DSD?
- DSD vs PCM
- DSD music production: record, edit, mixing, mastering
- DSD history and development
- DSD specification
- How DSD works?
- Sample rates
- DSD and Nyquist Theorem
- Noise, Maximal Level and Overload Stability
- DSD in figures
- DSD myths
- DSD vs FLAC, WAV, PCM comparison
- PCM versus DSD ADC
- PCM versus DSD DAC
- What is DXD?
- What is DoP (DSD over PCM)?
- Frequently Asked Questions [general]
What is DSD audio?
DSD (Direct Stream Digital) is an audiophile high-resolution digital audio format. It is a representation method of acoustic waveform as digital data. The format is based on 1-bit wideband sigma-delta modulation. Each sample of this signal type contains 1 bit per channel.
Also, Direct Stream Digital (DSD) means a registered trademark by SONY and Philips.Back to top
DSD music file formats
DSD was originally intended for archiving analog recordings. Now, this audio format is popular among music lovers and can bring master studio quality.
Direct Stream Digital is distributed in SACD optical disk, .dsf, .dff and SACD ISO stereo, and multichannel computer files.
Also, CUE file + DSF/DFF audio files may contain 1-bit album.
DFF, SACD ISO files and optical discs may be either uncompressed or compressed with DST method.
DoP is an open protocol that allows packing 1-bit audio as multibit data for compatibility with software and hardware. DoP cannot be played as usual pulse-code modulation. See details below.
Also, Direct Stream Digital audio may be streamed via a network.
Uncompressed DSD64 demands capacity: 2.7 Mbit/sec = 44100 Hz * 64 / 1024 / 1024.
Learn more:Back to top
How can I play DSD?
To play DSD audio you are need mobile phone, computer, media server, SACD-players, dedicated digital audio player (DAP).
SACD optical disks may be played back at hardware player. The author knows nothing about available SACD drives for consumer computers to playback/record SACD optical disks.
A stereo player may downmix multichannel to stereo. As alternative multichannel files may be downmixed to 2.0. It allows for saving space on the hard disk of an audio player. Downmix is lossy audio processing. Its quality is defined by an implementation.
Some devices and playback software can support native feed of DSD DACs.
1-bit playback may be performed via special ASIO driver under Windows, including DoP (DSD over PCM) protocol. Below it's considered.
Sound quality is a complicated matter. It depends on recording, hardware and software.
DSD may be listened to on ordinary multibit digital-to-audio converter. DSD player may be capable to inline transformation of Direct Steram Digital to such DAC.
Alternatively you can convert DSD to PCM audio files via software.
See more:Back to top
Where can I get DSD music in files and SACD?
You can buy Direct Stream Digital music in online stores as DSD file downloads or SACD disks with physical delivery.
Back to top
How to convert DSD?
How rip DSD from SACD to SACD ISO read in the guide...
To listen DSD on iTunes, convert:
When convert with audio processing (including Direct Stream Digital to Pulse Code Modulation), it's not lossless technically. But final quality is a matter of used conversion software and playback setup. Read details
Yes. There is a different distortion level and tolerance to overload. Read details here.
Back to top
DSD vs PCM
An audio signal is an air oscillation (sound) converted to electrical oscillations via a microphone.
Digital form of an electrical audio signal is a number sequence that is converted in analog-t-digital converter (ADC).
The sequence allows the creation of electrical oscillations back via digital-to-analog converter (DAC).
An audio signal (the electrical oscillations) may be presented in Direct Stream Digital or Pulse Code Modulation.
It’s like we can assemble the same mosaic from small squares or circles.
Below we’ll return to the metaphor.
What is the difference between Direct Stream Digital and Pulse Code Modulation?
Both onebit and multibit streams are a sequence of measured values (samples) of an audio signal's oscillation.
Sound/air oscillation each moment has a certain position (instant pressure level). The pressure level after the microphone is converted to an electrical oscillation.
The electrical oscillation at each moment has a certain position that refers to the air pressure level.
Pulse Code Modulation consists of samples. The sample is a measurement (number value) of an instant position of the electrical oscillation (voltage) at the moment of the measurement.
1 sample = 1 instant measurement of the voltage.
The voltage level at the microphone input is measured over equal time.
In sigma-delta modulation (a.k.a. DSD) each 1-bit sample shows a positive  or negative  change of the pressure level relatively to its previous value.
And, as rule, the sigma-delta modulation is explained as a saw (level grows or falls) between samples. But, it does not give an easy understanding of the basic DSD principle.
Below we'll consider the easier explanation. Keep reading.
Study articles:Back to top
DSD music production: record, edit, mixing, mastering
Audio editing is cutting, merging, gain altering, equalizing, etc. of file(s).
"Native editing" (without conversion to multibit and DSD re-modulation) is possible for cutting and merging onebit records in bit-perfect mode only.
All other kinds of editing demand conversion to multi-bit values and re-modulation. In this case, losses of the editing are comparable with multibit resampling.
For DSD recording, dedicated audio interfaces are used.
Pulse Code Modulation here may be considered as "multibit DSD". Pulse Code Modulation is not obligatory to mean "24 bit / 352 kHz" or so on. Author recommends use 32- or 64-bit float point bit depth. This PCM contains high-frequency modulation products. They are excessive. But, for conversion of this "multibit DSD" to 1-bit one, re-modulation is required.
Losses of editing with 1-bit/multibit conversion are comparable with resampling.
Thus, audio tracks in 1-bit format should be converted to PCM audio (filtered or DXD). After it them are mixed and mastered. Final mix is modulated to DSD back.
Recording studios may distribute 1-bit records without editing. These recording are once captured from microphones.
Study details here >Back to top
DSD history and development
Audio CD optical disk (Pulse Code Modulation format) was one of the first digital formats. It was a source with low distortion level in comparison analog recordings. I think, now many people forgot what is "real analog hum".
From CD (PCM) to SACD (DSD)
Audio quality (sound quality) is a level of distortions, that format/software/equipment causes. The level may be normalized by psychoacoustic criteria.
However, Sony and Phillips decided to use another kind of modulation - 1-bit sigma-delta modulation. At the first glance, using 1-bit is impossible, because such bit depth (resolution) causes a huge noise level. But, if high sample rates are used, the noise energy may be pushed out of the audible frequency range (to ultrasound). That action is called "noise shaping".
Practically, 1-bit conception may be easier, than multi-bit, in a DAC implementation (read below).
When DSD music is converted to analog, ultrasound useless content is filtered.
Medium, which contains 1-bit DSD audio stream was called SACD (Super Audio CD).
It allows solving some technical issues that are described below.
Look at DSD file infographic >Back to top
|Direct Stream Digital
|Audio data coding method:
|sigma-delta modulation (details and video)
|1 bit or more (details about sound quality)
|DSD64 (2.8 MHz), DSD128 (5.6 MHz, double), DSD256 (11.2 MHz, quad), etc. (details)
|Stereo or multichannel
|SACD optical disk, including hybrid SACD (with CD layer), computer files [SACD ISO (ripped albums from SACD) and DSF, DFF, CUE+DSF/DFF] (details)
|"Scarlet book" (1999)
|Music production, home hi-fi/hi-end audio
Direct Stream Digital is one of high-resolution audio formats to improve CD-audio dynamic range in the audible band.
See below about sound quality issues (distortions, bit depth, band, sample rate, Direct Stream Digital versus Pulse Code Modulation).
How DSD works?
Multibit signal is like to a computer display. It consists of several pixels along both X and Y axis.
DSD looks like a 1-pixel screen.
And we can't understand: how a 1-pixel display is capable to show a qualitative picture?
Imagine that, this one pixel is put on a truck. And the truck moves the pixel sequentially through all locations of multiple pixels of the first display. The movement is cyclic.
X . . . .
. . . . .
. . . . .
. X . . .
. . . . .
. . . . .
. . X . .
. . . . .
. . . . .
And it moves very-very quickly. And so rapid movement allows us to see high qualitative picture.
As rule, DSD has been explained in the time domain: positive samples increase level, and, negative sample decreases. After it, analog filter smooths "saw" form of signal.
In DSD analog-to-digital converter (ADC), measuring the "saw" signal is compared with the input analog signal.
At the ADC output "1" is present. And measuring-signal level grows until it exceeds the current level of the input analog signal.
It causes the output signal to switch to "0". And measuring signal begins to reduce level.
When the measuring signal level is a lesser input signal of the ADC, its output is switched to "1". And measuring signal grows until the exceeding the input signal.
Learn how to sigma-delta modulator works...
Above we discussed onebit and multibit formats like a mosaic from squares and circles.
These figures may be significantly smaller than the assembled image. If our eye can't resolve the figures, there is no difference between the squares or circles we use.
So, qualitative onebit and multibit systems may bring us the same final result.
Such an explanation causes difficulties in understanding what is the actual difference between these formats and how to process Direct Stream Digital. Probably, it creates the myth about "native" DSD processing. About the processing see below in "DSD editing software" part.
Instead of the explanation, in the author's opinion, the easiest way to understand how DSD works is its spectrum consideration.
To estimate DSD and PCM sound quality, the quantization-noise level is the main criterion.
There is an explanation, that DSD has a super high sample rate and, thus, super low bit depth 1-bit does not matter for noise.
To decrease noise level, noise shaping is used.
1-bit quantization-error (noise) energy has a significant level.
However, there is, so-called noise shaping of the spectrum is used. And quantization-error energy is re-distributed.
Noise shaping in the simple words
Let's imagine noise energy as water in a pool.
If we blow strongly to the water surface at one poolside, water level of this side is decreased (noise is decreased in audible range).
It is the same adding more bits of audio resolution.
The water will be displaced to the other poolside (inaudible ultrasound range). And level there grow.
The quantization-distortion spectrum (at the left of the picture) has a level comparable with a musical signal. The sigma-delta modulator pushes a significant part of the energy from the low- to high-frequency range, out of the audible band (0 ... 20 kHz).
Noise shaping is implemented in a sigma-delta modulator (a.k.a. DSD modulator).
What is DSD decoding?
When the 1-bit record is played back, a low-frequency filter into a sigma-delta demodulator (a.k.a. DSD demodulator, DSD decoder) cut all excessive sound information.
DSD decoder (demodulator)
Spectrum error is noise level.
Therefore, DSD error into the audible range is comparable with the error of multi-bit pulse code modulation.
Also, sigma-delta modulation may have a multi-bit resolution. Read the details below and watch the video on this page.Back to top
The format uses standard range of sample rates based on 44100 Hz:
- DSD64 = 44100 * 64 = 2'822'400 Hz = 2.8 MHz
- DSD128 = 44100 * 128 = 5'644'800 Hz = 5.6 MHz
- DSD256 = 44100 * 256 = 11'289'600 Hz = 11.3 MHz
- DSD512 = 44100 * 512 = 22'579'200 Hz = 22.6 MHz
- DSD1024 = 44100 * 1024 = 45'158'400 Hz = 45.2 MHz
Also, 48000-kHz base is possible. There are no technical limitations to the sample rate value. But hardware/software compatibility issues are very probable.Back to top
DSD and Nyquist Theorem
For the Nyquist theorem no difference between Direct Stream Digital (sigma-delta modulation) and Pulse Code Modulation:
the spectrum above [sample rate]/2 is
the same as the spectrum below [sample rate]/2
that flipped along the frequency axis.
Noise, Maximal Level and Overload Stability
When a sigma-delta modulator is designed, the engineers pay attention to two main parameters:
- the distortion level in the audible range and
- tolerance to overload.
To solve these issues the engineers use:
- sample rate,
- noise shaping,
- set default output level -6 dB for the modulator.
All these parameters should be considered comprehensively.
Bit depth reduces the quantization error level itself.
NS "pushes" a significant part of the quantization error energy out of the audible range.
We can suggest that more energy may be pushed out of the range in the sigma-delta modulator. But it requires a steeper noise shaper.
Steeper NS is capable to cause a higher probability of glitch (broken stability) of a sigma-delta modulator when input overload happens.
After the glitch, silence or some oscillations are generated at the output of the sigma-delta modulator.
After broken stability, the modulator should be forcibly reset.
We remember that quantization error is noise spectrum level. It is a level of distortions.
A higher sample rate distributes the same noise-hump energy into a wider band.
It reduces the hump level or "quantization error" of Direct Stream Digital.
Also, a wider band allows using sloper noise shaping.
Energy is a square of the figure, concluded between the spectrum line and horizontal axis into band 0 ... [sample rate]/2.
In the left and right pictures, the squares of the noise energy figures are the same. But the figure, which is more expanded in the horizontal axis, gives a lower noise hump.
A higher sample rate allows for reducing the hump level in the audible frequency range. It allows for reducing noise shaping's steepness, which increases the overload tolerance of the modulator.
We can see that lower distortions and higher stability to overload may be achieved in different ways.
As an example, better quality is noise-shaping implementation matter for the same bit depth and sample rate. But, on the other hand, we can increase the sample rate and/or bit depth to decrease the error level for the same NS method. It allows for the improvement of the audio quality.
Back to top
DSD in figures
Pro audio modulators have noise levels in the audible frequency range for sample rates (read details):
- D64 about -125 ... -145 dB (comparable with 24-bit)
- D128 about -165 dB (better than 24-bit)
- D256 and higher about -170 ... -200 dB (comparable with 32-bit)
Noise hump in the audible frequency band almost doesn't depend on demodulator. And, the excessive products should be maximally suppressed out of the band. Ultrasound content can cause audible intermodulation distortions.
Learn more: DSD vs DSF vs DFF >Back to top
No. Both format quality is defined by implementation in a device and software.
Also played back record is a matter.
Direct Stream Digital's low noise level is a result of noise shaping.
DSD bit depth defines distortion level abilities. After the editing, sigma-delta re-modulation is needed.
Study DSD editing...
The editing with decimation is implemented with a low-frequency filter. The filter causes ringing audio.
However, the author knows no serious researche, that is studied the real impact of ringing on the listening perception.
Also, the ringing depends on filter implementation.
So, currently, we don't know the exact answer to the question.
Read about ringing audio...
Back to top
DSD vs FLAC, WAV, PCM comparison
Now we'll consider, which is better PCM or DSD?
Bit depth defines audible noise. More bit depth means lesser noise.
In the picture, we can see that bit-depth decreasing is achieved by using only part of the full signal frequency band. In other words, if we want to reduce bit depth, we must increase the sample rate to keep the audible noise level.
See details of the format comparison in the table: DSD vs WAV vs PCM vs FLAC vs DXD
|pulse code modula
|pulse code modula
|pulse code modula
|pulse code modula
|44100 x 64 x N
|up to 384'000 Hz
|up to 2'147'483'648 Hz
|352 / 705 kHz
16 ... 24 bit,
16 ... 64 bit inte
|Band for audio signal
|Low part of the band
|Low part of the band
|file, optical disk: SACD
|file, digital tape, optical disk: CD, DVD
Note: WAV and FLAC are PCM format implementations.
Read which is better DSD or FLAC (Infographic) >
Study more audio file formats >
Direct Stream Digital (sigma-delta modulation) is like to pulse code modulation, but quantization error spectrum is shaped for decreasing noise into the audible range.
We may apply the noise shaping for usual Pulse Code Modulation. But the difference here is band reserve for pushed noise out of audible frequency range.
PCM has a lesser band reserve (above audible range) than sigma-delta modulation format, but pulse code modulation has higher bit depth.
Noise shaping for pulse-code modulation is an implementation matter too, like sigma-delta modulation.
Therefore, no format advantages as itself. But implementation makes a difference.
Sigma-delta-modulation decoder (demodulator) is 2-position (1 / -1) voltage generator and low-frequency filter. It is simpler than pulse-code-modulation hardware demodulator. Because pulse-code demodulator contains multi-voltage matrix or 1-bit decoder(s). So we have more abilities to make cheaper 1-bit DAC better than multi-bit one.
There is no univocal answer. There are several reasons:
- Quality of a record
- Mixing/re-mixing of the record (including quality of the processing)
- Playback software/hardware implementation
- DAC implementation.
These reasons trigger infinite discussions about "what is better PCM or DSD".
Technically, Direct Stream Digital and Pulse Code Modulation are incomparable. Even as single multiformat digital-to-analog converter is used for the test, the difference may be caused by different mixing/post-production of records, and the DAC uses different circuits (see here).
General answer: "Each case is individual and depends on playback system implementation and listened to record".
There is no unambiguous answer to the question. It depends on record quality, your software and equipment.
There is no answer. Theoretically, DSD has better potential than FLAC and other multibit formats. Recording, certain equipment makes sound quality. See details...
1 minute of DSD64 stereo is about 40 Mbytes.
Yes. Hybrid SACD also may contain CD layer compatible with usual CD player.
Sound is defined by record quality, your playback program and apparatus.
Upsampling can optimize audio file resolution to optimal (minimal playback distortions) for your equipment. Read details >
PCM versus DSD ADC
Both kinds of DAC contain sigma-delta modulator. It converts the analog voltage to digital samples in a format like Direct Stream Digital.
Analog filter ["DSD" sample rate]/2 is applied before the modulator. Looks details here...
PCM ADC is more sophisticated and contains a multibit modulator.
To reduce the sample rate, the "multibit-DSD" signal should be filtered (low-frequency filter) and decimated (removing part of samples). The filter also eliminates the shaped noise of the "DSD signal".
Read more about the format difference...
Back to top
PCM versus DSD DAC
To easier providing of multibit DAC precision/linearity in this kind of DAC, a digital sigma-delta modulator is used. After it, sigma-delta de-modulator converts digital "DSD" to an analog signal.
In the general case, DSD DAC is simpler. It contains sigma-delta de-modulator only.
It is technically impossible to compare multibit and DSD DACs, because they use different processing / electrical circuits.
Study more: what is DSD and PCM DACs and their comparison >Back to top
What is DXD?
DXD is Pulse Code Modulation (as rule "24 bit / 352 kHz" or so on) with high sample rate and bit depth and legacy DSD noise hump. However, it can cause audible intermodulation distortions.
Only low part of the DXD spectrum contains useful musical signal. Noise level is growing to higher frequencies. Before non-linear processing, this high-frequency hump cutting is recommended.
As rule, the format has sample rates 352 800 Hz and above and 24-bit and above resolution.
Primarily, DXD format is intended to edit DSD.
However, there are PCM lossless formats that may be used like DXD.
Read DSD editing details...
However, not all PCM files with such sample rates are DXD. Traditional Pulse Code Modulation has no significant noise level at high frequencies.
Playback devices can filter high-frequency noise to save dynamic range and avoid intermodulation distortions.
More about DXD see here.
DXD is pulse-code-modulated sound format with noise hump in high frequencies. DSD DAC is simpler than PCM one in theory. From DAC point of view, DSD is ideal format. But, actual result is refer to digital-to-analog converter model. Read details here...Back to top
What is DoP (DSD over PCM)?
There are compatibility and other issues of tranfer 1-bit and multibit audio data from computer to PCM or DSD DAC.
All audio interfaces (USB, in instance) are designed for multibit data. And 1-bit data packing into ordinary multibit value was suggested.
DoP (DSD over PCM) is a format providing compatibility with some DSD hardware (USB DAC, for example). The format allows packing 1-bit DSD data in usual multi-bit words (like PCM).
These multibit packed DSD words are transferred as ordinary audio signal. Upper bits of the multi-bit word contain special "DSD over PCM" marker code.
DoP receiver should recognize the marker code and interpret lower part of the multi-bit word as raw DSD bits.
DoP format may be stored into usual lossless multibit file: WAV, FLAC, AIFF.
DoP coding may be chosen in audio player settings.
Back to top
- DSD (SACD) appear as improvement of CD-audio.
- Sound quality is not matter of bit number of format, but depend on implementation and quality of a musical record.
- "Native" DSD editing is possible for limited cases. For other cases, the own editing loses is comparable with PCM resampling.
DSD is onebit audio data (sigma-delta modulation) format for hi-fi audio.
For the same sample rate and bit depth, the format may have different quality, that depends on implementation.
DSD digital-to-analog converter is simpler than the multibit one. It allows getting the better quality simpler way.
DSD may be edited via 1-bit to multibit and back conversion. The edition is lossy, except merging/dividing.Back to top
Frequently Asked Questions [general]
SACD is optical disk that contains digital audio data in DSD format. Read more...
DoP format is protocol for audio unit interaction. The protocol contains DSD audio data. Read details...
DSD is well known audiophile format. It allows achieving high sound quality and often used for classic and jazz music recording.
The best audio quality has 2 definitions:
- most low distortions, or
- most"nice" sound.
When correct scientific approach is applied, both definitions may be used.
One-bit and multibit lossless files are the highest-quality audio files. Read details here...
Bit depth of Direct Stream Digital and Pulse Code Modulation may be compared by quantization distortion level in the audible band. However, equipment and software play big role there. See approximate comparison of the bit depths...
FLAC format supports high-resolution audio. It give technical abilities to better sound quality.
Read details here...
DSD is family of high-resolution audio formats. FLAC is one of multibit audio formats.
Technically, DSD is better than mp3.
You can play DSD music under Windows. Read more...
As the author know, VLC player software can't play DSD files. Read the discussion...
DSD streaming service you can find here...
Spotify doesn't support DSD audio. Study information about DSD streaming service...
Qobuz does not providing DSD streaming. Check out DSD streaming service...
- DSD vs. PCM. Real competitors? What is common base? >
- DSD Decoder Audio >
- How work sigma delta modulation in audio >
- DSD Converter of Audio Files: What Inside? >
- Where to get free DSD recordings? >
- Where to buy DSD music? >
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