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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 PCM.
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, and other interesting things.
- What is DSD
- DSD history
- DSD specification
- How DSD works?
- Sample rates
- DSD and Nyquist Theorem
- Noise, Maximal Level and Overload Stability
- DSD in figures
- DSD myths
- DSD is better PCM and vice versa
- DSD has low noise level due to high sample rate
- Multi-bit DSD may be edited natively
- DSD editing with decimation dramatically degrade sound
- DSD vs FLAC, WAV, PCM comparison
- PCM versus DSD ADC
- PCM versus DSD DAC
- File formats and compression method
- How to play DSD?
- How to convert DSD?
- DSD editing software
- What is DXD?
- What is DoP (DSD over PCM)?
- DSD audio samples
- Frequently Asked Questions
DSD (Direct Stream Digital) is an audiophile high-resolution audio format. It is a representation method of acoustic waveform as digital data. The format is based on 1-bit sigma-delta modulation. Each sample of this signal type contains 1 bit per channel.
Initially, DSD format was designed as an alternative to PCM (Pulse Code Modulaion), which is used in CD-audio, WAV, FLAC, mp3 and other formats. Direct Stream Digital is distributed in SACD optical disk, .dsf, .dff and SACD ISO stereo, and multichannel computer files.
DSD was originally intended for archiving analog recordings. Now, this audio format is popular among music lovers and can bring master studio quality.
DSD may be played back with computer software or digital-player devices.
Also, Direct Stream Digital (DSD) is a registered trademark by SONY and Philips.
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-to-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 as DSD or PCM.
It’s like we can assemble the same mosaic from small squares or circles.
Below we’ll return to the metaphor.
Both DSD and PCM 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.
PCM 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, DSD 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.
Audio CD optical disk (PCM modulation / PCM format) was one of the first digital formats. It was a source with low distortion and noise level comparing analog ones. I think, now many people forgot what is "real analog noise".
From CD (PCM) to SACD (DSD)
Audio quality (sound quality) is a level of noise and distortions, that cause format/software/equipment. 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, 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 noise is filtered.
Medium, which contains 1-bit DSD audio stream was called SACD (Super Audio CD).
Look at DSD file infographic >
|Abbreviation of:||Direct Stream Digital|
|Audio data coding method:||sigma-delta modulation (details and video)|
|Bit depth:||1 bit or more (details about sound quality)|
|Sample rate:||DSD64 (2.8 MHz), DSD128 (5.6 MHz, double), DSD256 (11.2 MHz, quad), etc. (details)|
|Channel number:||Stereo or multichannel|
|Medium:||SACD optical disk, including hybrid SACD (with CD layer), computer files [SACD ISO (ripped albums from SACD) and DSF, DFF, CUE+DSF/DFF] (details)|
|Specification:||"Scarlet book" (1999)|
|Applications:||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.
Read below about sound quality issues (noise, bit depth, band, sample rate, DSD versus PCM).
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 can 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.
Read how to sigma-delta modulator works...
Above we discussed DSD vs PCM 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 DSD and PCM systems can bring us the same final result.
Such an explanation causes difficulties in understanding what is the actual difference between PCM and DSD and how to process DSD. Probably, it creates the myth about "native" DSD processing. About the processing read 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-noise 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 the noise.
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.
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.
For the Nyquist theorem no difference between DSD and PCM:
the spectrum above [sample rate]/2 is
the same as the spectrum below [sample rate]/2
that flipped along the frequency axis.
When a sigma-delta modulator is designed, the engineers pay attention to two main parameters:
- the noise 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 can 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.
A higher sample rate distributes the same noise energy into a wider band.
It reduces noise level or "quantization error" of DSD.
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 lesser noise level.
A higher sample rate allows for reducing the noise 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 noise and higher stability to overload can 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.
Pro audio modulators have noise levels in the audible frequency range for sample rates (read details):
- D64 about -125 ... -145 dB (comparable with PCM 24-bit)
- D128 about -165 dB (better than pulse code modulation 24-bit)
- D256 and higher about -170 ... -200 dB (comparable with PCM 32-bit)
Noise level in the audible frequency band almost doesn't depend on demodulator. But the noise should be maximally suppressed out of the band. Ultrasound noise can cause audible intermodulation distortions.
Read more about DSD vs DSF vs DFF >
No. Both format quality is defined by implementation in a device and software.
Also played back record is a matter.
DSD's low noise level is a result of noise shaping.
DSD bit depth defines noise level abilities. After DSD editing sigma-delta re-modulation is needed.
Read more about 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...
Bit depth defines audible noise. More bit depth is 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.
Read 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||no limit||352 / 705 kHz|
16 ... 24 bit,
16 ... 64 bit inte
|Band for audio signal||Low part of the band||Full band||Full band||Full band||Low part of the band|
|Medium||file, optical disk: SACD||file||file||file, digital tape, optical disk: CD, DVD|
Note: WAV and FLAC are PCM implementations.
Read which is better DSD or FLAC (Infographic) >
Read more about 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 can apply NS for usual PCM. 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, DSD and PCM are incomparable. Even as single DSD/PCM compatible DAC 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 PCM formats. Recording, certain equipment makes sound quality. Read details...
1 minute of DSD64 stereo is about 40 Mbytes.
Yes. Hybrid SACD also may contain CD layer compatible with usual CD player.
.dsf, .dff, .iso (sacd iso), flac/wav (DoP). Read details...
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 >
Read DSD store site list here >
You can listen to .dsf (DSD audio files) here:
To listen to DSD files, you can download them for free or buy special audio player software or hardware.
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 PCM and DSD difference...
To easier providing of PCM 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 PCM and DSD DACs, because they use different processing / electrical circuits.
DSD 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 DSD records 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 PCM resampling.
Read details here >
- Sigma-delta modulation is applied on optical disks SACD (Super Audio CD).
- The modulation is supported by files DSF, DFF, SACD ISO (SACD disk image).
- WAV, FLAC as DoP (DSD over PCM) container.
SACD ISO image may be extracted to DSF and DFF files. Lossless extraction may be applied.
1-bit audio files (DSF, DFF, SACD ISO) and disks 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.
Also, 1-bit audio may be streamed via a network.
Uncompressed DSD64 demands capacity 2.7 Mbit/sec = 44100 Hz * 64 / 1024 / 1024.
Also, index file CUE + DSF/DFF audio files may contain 1-bit album.
DSD may be played on a mobile phone, portable players, music servers, SACD-players, and other devices.
Some devices can support native DSD feed of DSD DACs. Other ones demand pre-conversion to PCM or make conversion "on fly".
Sound quality is a complex matter. Read details about the resolution of audio files and sound quality.
Read details on how to open, play DSD on Mac, Windows, iOS, Android, Linux: player list >
Read how to download DSD music and how to get free DSD files here >
DSD audio players (hardware and software) can playback all or some of the 1-bit file formats.
1-bit files may be played back directly on DSD DAC/player or converted to PCM "on fly" to playback to PCM DAC/player. About SACD conversion read more here
1-bit playback may be performed via special ASIO driver under Windows, including DoP (DSD over PCM) digital audio packing format (example).
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 pre-converted to stereo. 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.
To play DSD audio, use DSD audio players...
Yes with additional software. Read details here
Yes. Read details here...
As the author knows, not. Read details...
Yes. Look at the player list...
As the author knows, Vox convert .dsf and .dff files to PCM on fly without native DSD playback.
As far as the author knows, Plex can playback DSF. Details
As the author knows, yes. Details
Yes. Look at the player list
As the author knows, there is conversion to PCM files is necessary.
Yes, Mac can play DSD...
Read details on how to convert DSD on Mac, Windows and other platforms: converter list >
How to create, edit DSD files read here >
How to rip DSD from SACD (to SACD ISO) read here
DSD converter is software that can perform (depend on implementation):
- conversion DSD to PCM,
- conversion PCM to DSD,
- extracting SACD ISO to DSD,
- extracting SACD ISO to PCM,
- resampling 1-bit audio,
- altering volume,
- volume normalization,
- click suppressing on track borders,
- other processing.
Read how the converters work...
Converter list (convert some or all of DSD files [DSF, DFF, SACD ISO])
- AuI ConverteR 48x44
- TASCAM Hi-Res Editor
- See more converters...
Look at DSD converter comparison list >
Conversion with processing audio (including DSD to PCM) adds distortions. But final distortions is a matter of used playback distortions. Read details here
It may be lossless when audio converter doesn't process audio stuff (resampling and other) in conversion.
Yes. There is noise level, frequency/phase distortions and tolerance to overload are estimated. Read details here.
Read main article about DSD editing >
DSD editing is sophisticated issue due modulation noise. Non-linear processing can cause audible intermodulation distortions by ultrasound noise.
Currently no information, that is known for the author, about "native" 1-bit processing (example: gain altering, resampling, etc.) without conversion 1-bit to multibit and back. Except, merging/dividing the audio file.
Read the article about DSD versus DSF versus DFF >
PCM 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 noise. But for conversion, this "multibit DSD" to 1-bit need re-modulation with NS.
Losses of editing with 1-bit/multibit conversion are comparable with resampling.
Recording studios may distribute 1-bit records without editing.
DXD is PCM (as rule "24 bit / 352 kHz" or so on) with high sample rate and bit depth and legacy DSD noise. However, the noise can cause audible intermodulation distortions. Before non-linear processing, the high frequency noise cutting is recommended.
DXD is PCM 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...
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 multi-bit words are transferred as usual PCM. 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 PCM file: WAV, FLAC, AIFF.
DoP coding may be chosen in audio player settings.
Audio samples may be downloaded here:
Check more DSD downloading resources >
- DSD (SACD) appear as improvement of CD-audio.
- Sound quality is not DSD/PCM format matter, 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 1-bit 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 PCM 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.
Read more about 1-bit format
- 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? >
DSD has multiple meanings. In audio, DSD means Direct Stream Digital format. Read details...
DSD audio format is way to store sound recordings into computer files. This format is intenged for music lovers.
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.
Both DSD and PCM have advantages and disadvantages. Its discussing is theoretical rather. Read details here...
No. DSD is alternative audio format to PCM one. Read more information here...
DSD and PCM lossless files are the highest-quality audio files. Read details here...
DSD and PCM bit depth may be compared by quantization noise level in the audible band. However, equipment and software play big role there. See approximate comparison DSD and PCM bit depth...
FLAC format supports high-resolution audio. It give technical abilities to better sound quality.
Read details here...
DSD streaming service you can find here...
DSD is family of high-resolution audio formats. FLAC is one of PCM audio formats.
Technically, DSD is better than mp3.
You can play DSD files with special hardware and/or software.
DSD music can be played on a computer via DSD player software...
To play DSD you can use DSD audio player or iOS/Android mobile phone.
Also, you can play DSD on computer with ordinary PCM audio card.
It's possible either:
- use playback software capable to inline DSD to PCM conversion, or
- convert DSD to PCM audio file with high sound quality.
If you have external DAC (DSD or PCM), you can connect it to the computer or player device.
DAC is required thing to playback music. However, you can play DSD at traditional PCM DAC.
- use player, that is capable to convert DSD to PCM inline, or
- preconvert DSD to PCM with high sound quality.
DSD over PCM is packing method of 1-bit audio to ordinary PCM format.
You can play DSD music under Windows. Read more...
As the author know, VLC player software can't play DSD files. Read the discussion...
Spotify doesn't support DSD audio. Read about DSD streaming service...
Qobuz does not providing DSD streaming. Check out DSD streaming service...
Here you can look for DSD music files: