Electronics project notes/Audio notes - Digital sound communication: Difference between revisions

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==AES3 and S/PDIF==
==AES3 and S/PDIF==


{{stub}}


Serial, one-directional, digital data.




'''AES3''' is a digital audio protocol from 1985{{verify}}.
<!-- apparently aimed at 44.1 kHz CD and 48 kHz DAT but ''could'' be run at any rate-->


===S/PDIF===
It was codeveloped by AES and EBU, and at the time was often marked 'AES/EBU' on devices.
S/PDIF ("Sony/Philips Digital Interface") (a.k.a. IEC958, or IEC60958 after its renumbering)
You can treat AES/EBU as meaning AES3.
is purely the protocol, not a connector.


...and is itself based on AES3 (a.k.a. AES/EBU);
 
S/PDIF can be seen as a consumer variant of AES3, ''simplifying'' the way it would be implemented, and appear to consumers{{verify}}
'''S/PDIF''' ("Sony/Philips Digital Interface")
is based on AES3, can be seen as a consumer variant of AES3, ''simplifying'' the way it would be implemented, and appear to consumers{{verify}}
 
 
From a practical standpoint, you mostly care about S/PDIF until you work with some older devices.
 
 
IEC 60958 (IEC958 before IEC's renumbering in 1998) seems to have absorbed both, which makes historical distinctions and incompatibilities a little harder to figure out.
 
This also confuses the connector part. Say, IEC60958 now defines:
: Over '''XLR3''' (IEC 60958 Type I), balanced
: Over '''RCA''' (IEC 60958 Type II), unbalanced
: Over '''TOSLINK''' (IEC 60958 Type III)
: Over '''BNC''', unbalanced, was used in broadcasting (probably in part because it could be used over existing BNC/Coax)
 
Broadly speaking, XLR3 and BNC are likely to be AES3 pedigree, and TOSLINK and BNC are likely to be S/PDIF flavour.
 
Footnotes:
* TOSLINK ''can'' now be largely assumed to be S/PDIF, but there was a time at which it could also be AES3.{{verify}}
<!--
 
* AES3 XLR3 was often for shorter distances (also because you never know what kind of cables are actually used), BNC for longer {{verify}}




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: a single [[RCA]] connector on (preferably) a coaxial cable
: a single [[RCA]] connector on (preferably) a coaxial cable


S/PDIF tends to often carry
* '''raw [[PCM]]'''
yet newer revisions also allow it to carry surround (compressed, because of bandwidth limitations), often either:
* '''[[DTS]]''' for 5.1/7.1 -- more specifically the [[DTS Coherent Acoustics]] (DCA) codec
* '''[[AC3]]''' (Dolby Surround)
-->


S/PDIF tends to carry either
 
* '''raw [[PCM]]'''
<!--
* surround (compressed, because of bandwidth limitations), often either:
It is ''not'' recommended to plug AES3 directly into S/PDIF without thought.
** '''[[DTS]]''' for 5.1/7.1 -- more specifically the [[DTS Coherent Acoustics]] (DCA) codec
 
** '''[[AC3]]''' (Dolby Surround)
Conversion at electrical level is often not hard, and some devices are engineered to accept both, but ''don't count on it''.
 
 
* both S/PDIF and AES3 transfer 24 bit words, ''but''
:: in AES3, the last 4 bits are reserved and not usable for audio
:: in S/PDIF those bits are unspecified
:: which is why the spec only guarantees 20 bits, but you can put audio in those four bits.
:: devices may send in 16-bit (even if they work in 24-bit)
 
 
 
Compatibility notes:
* Alterations to the ''protocol'' are minor, and in particular sending just stereo PCM is largely compatible.
 
* The subcode data is different between AES3 and S/PDIF ''but'' in practice not a lot of devices send those.
: so you can sometimes get away with AES3 to S/PDIF
 
* Some later devices made changes that earlier devices may not understand
: e.g. 24-bit TOSLINK
: this also matters to converters
 
* AES3 XLR3 is not electrically compatible with AES3 BNC
 
 
* AES3 should not be ''assumed'' to be electrically compatible with S/PDIF copper --
:: but: can AES3 BNC be connected to S/PDIF copper?{{verify}}{{verify}}
 
* CDROM drives might output S/PDIF data -- but often at 5V (which out of spec of direct connections, but fine internally) {{verify}}
-->
 
<!--
{{comment|(...and ''neither'' is plug-compatible or protocol-compatible with S/PDIF (RCA connectors or optical), which is probably what most of your modern digital-audio gear speaks)}} {{verify}}
 
 
AES-EBU balanced is differential with an up-to-10V voltage swing -- in fact sort of similar to [[RS-422]] and you ''can'' use RS-422 interconnects.
 
S/PDIF on copper is 0.5..1V
 
 
 
http://lampizator.eu/lampizator/transport/spdif.html
 
https://www.thewelltemperedcomputer.com/Intro/SQ/SPDIF_AES.htm
-->
 
<!--
AES3 1985 (revised in 1992, 2003)
 
 
IEC 60958 exists in five parts:
* IEC 60958-1: General [https://webstore.iec.ch/publication/71031]
:: linear PCM up to 24bit
:: 1999, 2004, 2008, 2014, 2021
 
* IEC 60958-2: Software Information Delivery Mode
::
 
* IEC 60958-3: Consumer applications
:: 1999, 2003, 2006, 2021 {{verify}}
 
* IEC 60958-4 (in three parts): Professional applications
:: "wider range of physical media", more sampling frequencies, deprecation of "minimum implementation" of channel status data.
:: 2016
 
* IEC 60958-5: Consumer application enhancement
:: multichannel, multi-stream, high-resolution, multimedia extension
:: 2021
 
 
https://webstore.iec.ch/publication/62827
 
 
Related:
* EIAJ CP-340 1987-9 seems to be equivalent to IEC958-3:1989 ?
 
 
* '''AES3id''' (a.k.a. AES-3id-1995, AES-75, AES-BNC) is specifically the unbalanced 75-ohm coax,
which is compatible with S/PDIF copper, but itself ''typically'' implemented over [[BNC]] connectors.
 
AES3id has a taste of being developed to carry audio over existing video coax on longer distances (order of 100m).
Converters between the balanced and AES3id forms exist.
 
 
IEC 61937 - sends things other than PCM[https://webstore.iec.ch/publication/6141], apparently AC-3, MPEG-1 (Layer 1 & 2), MPEG-3 (Layer 3), MPEG-2(multichannel), MPEG-2/4 AAC in ADTS, DTS, Dolby Digital Plus[https://learn.microsoft.com/en-us/windows/win32/coreaudio/representing-formats-for-iec-61937-transmissions]
 
 
There is a spec of how to carry it over network with low latency




-->




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See also:  
See also:  
* http://en.wikipedia.org/wiki/S/PDIF
* http://en.wikipedia.org/wiki/S/PDIF
* http://www.lampizator.eu/LAMPIZATOR/TRANSPORT/spdif.html





Revision as of 17:38, 2 April 2024

The physical and human spects dealing with audio, video, and images

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Video: format notes · encoding notes · On display speed · Screen tearing and vsync


Audio physics and physiology: Sound physics and some human psychoacoustics · Descriptions used for sound and music

Noise stuff: Stray signals and noise · sound-related noise names · electronic non-coupled noise names · electronic coupled noise · ground loop · strategies to avoid coupled noise · Sampling, reproduction, and transmission distortions · (tape) noise reduction


Digital sound and processing: capture, storage, reproduction · on APIs (and latency) · programming and codecs · some glossary · Audio and signal processing - unsorted stuff

Music electronics: device voltage and impedance, audio and otherwise · amps and speakers · basic audio hacks · Simple ADCs and DACs · digital audio · multichannel and surround
On the stage side: microphones · studio and stage notes · Effects · sync


Electronic music:

Electronic music - musical terms
MIDI · Some history, ways of making noises · Gaming synth · microcontroller synth
Modular synth (eurorack, mostly):
sync · power supply · formats (physical, interconnects)
DAW: Ableton notes · MuLab notes · Mainstage notes


Unsorted: Visuals DIY · Signal analysis, modeling, processing (some audio, some more generic) · Music fingerprinting and identification

For more, see Category:Audio, video, images

This article/section is a stub — some half-sorted notes, not necessarily checked, not necessarily correct. Feel free to ignore, or tell me about it.

This is mostly about hardware interconnects. For software media routing, see Local and network media routing notes

Typically external

AES3 and S/PDIF

This article/section is a stub — some half-sorted notes, not necessarily checked, not necessarily correct. Feel free to ignore, or tell me about it.

Serial, one-directional, digital data.


AES3 is a digital audio protocol from 1985(verify).

It was codeveloped by AES and EBU, and at the time was often marked 'AES/EBU' on devices. You can treat AES/EBU as meaning AES3.


S/PDIF ("Sony/Philips Digital Interface") is based on AES3, can be seen as a consumer variant of AES3, simplifying the way it would be implemented, and appear to consumers(verify)


From a practical standpoint, you mostly care about S/PDIF until you work with some older devices.


IEC 60958 (IEC958 before IEC's renumbering in 1998) seems to have absorbed both, which makes historical distinctions and incompatibilities a little harder to figure out.

This also confuses the connector part. Say, IEC60958 now defines:

Over XLR3 (IEC 60958 Type I), balanced
Over RCA (IEC 60958 Type II), unbalanced
Over TOSLINK (IEC 60958 Type III)
Over BNC, unbalanced, was used in broadcasting (probably in part because it could be used over existing BNC/Coax)

Broadly speaking, XLR3 and BNC are likely to be AES3 pedigree, and TOSLINK and BNC are likely to be S/PDIF flavour.

Footnotes:

  • TOSLINK can now be largely assumed to be S/PDIF, but there was a time at which it could also be AES3.(verify)





See also:


Not to be confused with

  • AES/EBU (next section)

ADAT

This article/section is a stub — some half-sorted notes, not necessarily checked, not necessarily correct. Feel free to ignore, or tell me about it.

ADAT has referred to two distinct things


Historically, and now rarely, to the Alesis Digital Audio Tape, a way of storing eight digital audio tracks onto Super VHS.


These days, so much more typically, it refers to the ADAT Optical Interface, more commonly known as ADAT Lightpipe or often just ADAT (or lightpipe), also from Alesis.

It looks the same as TOSLINK / S/PDIF, but speaks a different protocol, and somewhat faster.


It carries audio channels that are always 24 bit (devices that are 16-bit will effectively just use the 16 highest bits).

Its speed lets it carry

up to eight channels of those at 48kHz.


...or, with the common S/MUX extension

up to four channels at 96kHz
up to two channels at 192kHz


See also:



Typically internal

I2S

This article/section is a stub — some half-sorted notes, not necessarily checked, not necessarily correct. Feel free to ignore, or tell me about it.

(Note: no technical relation to I2C)

(has existed since the mid-eighties)

I2S (sometmimes IIS), Inter-IC Sound, is meant as an easy and standard way to transfer PCM data between closeby chips.

It separates clock and data, so it can have slightly lower jitter (and indirectly latency) than buses that don't.


As I2S doesn't spec a plug, or how to deal with longer cables (impedance and such), it is mostly used within devices.

Exceptions mainly being audiophile setups that want to choose their DACs separately. Because it wasn't made for that, this takes more care to do right, because impedance can cause synchronization issues, particularly at higher bitrates.


Lines and bits and interpretation

The lines are

  • bit clock (BCLK) (a.k.a. continuous serial clock (SCK))
  • left-right clock (LRCLK) (a.k.a. word clock, word select (WS), Frame sync (FS))
  • data
  • ground

BCLK pulses for each bit, so should be samplerate * bitdepth * channelamount, e.g. 1411200 Hz for CD audio (44100*16*2).

LRCLK selects left/right channel

Some also add a master clock (MCLK). This is not part of standard I2S


Note that:

  • The protocol is fundamentally 2-channel (in part due to LRCLK's function)
If you functionally want to send mono, you could send zero on the other.
but if you have that sample anyway, then it makes just as much sense to output it twice, i.e. in both channels, so that if a receiver decides to implement mono by picking one channel, it doesn't matter which one. And stereo playback will be double mono rather than seeming to miss one channel.
  • Sample rate is not configured, it is implicit from the sending speed(verify),
which is part of why software bit-banging I2S would probably never sound great
  • Bit depth is implied by when LRCLK switches (which it can do because the MSB goes first)
with some work left to the receiver


DIY abuse

Because I2S interfaces are fairly high-speed, and typically DMA-assisted, it has found other uses.


Because channels=2, sample rate is controlled by the clock, and bit depth is somewhat implied, you can vary some aspects of what it sends without negotiating it.


For example, when feeding in data into an I2S DAC, you do need to do the stereo interlacing as in the spec, and the bit depth as the DAC expects, but it doesn't need know the sample rate - it will do what you ask of it, at the rate you ask it to.


For example, the ESP8266 and ESP32's I2S is actually run from a more generic piece of hardware, roughly a glorified shift register, used to implement I2S as well as LCD and camera peripherals.

It happens to go at ~1.4MHz for audio, probably from a configured, but if you can control the output rate, then you can produce other sorts of signals, and DIYers have found it's fairly stable at 40MHz, which makes it possible to produce NTSC and VGA signals, and could even sample data at that rate.


Similarly, RP2040 has a Programmable I/O (PIO)[1] [2] [3]


You could probably send PDM over these - which would be an ironic use of something already intended for audio, but which might makes sense if the receiving side isn't an I2S DAC(verify).



See also:

On DACs