Electronics project notes/Audio notes - Digital sound communication
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This is mostly about hardware interconnects. For software media routing, see Local and network media routing notes
Typically external
S/PDIF
S/PDIF ("Sony/Philips Digital Interface") (a.k.a. IEC958) is purely the protocol, not a connector.
...but S/PDIF is often carried over either
- fiber, typically used with TOSLINK connectors
- a single RCA connector on (preferably) a coaxial cable
S/PDIF tends to carry either
- raw PCM
- 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)
See also:
Not to be confused with
- AES/EBU (next section)
AES3
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AES3, also marked AES/EBU, is a digital audio protocol very comparable to S/PDIF - the data format is actually largely the same,
yet not qutie compatible.
ADAT
| This article/section is a stub — probably a pile of half-sorted notes, is not well-checked so may have incorrect bits. (Feel free to ignore, or tell me) |
ADAT has referred to two distinct things
Historically, and now rarely, to the Alesis Digital Audio Tape, a way of storing eight tracks digitally onto Super VHS
And much more typically to the ADAT Optical Interface, more commonly known as ADAT Lightpipe or often just ADAT, 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 to two channels at 192kHz
See also:
Typically internal
I2S
(Note: no technical relation to I2C)
I2S (sometmimes IIS), Inter-IC Sound is meant to standardize PCM data between closeby chips.
It separates clock and data, means it can have somewhat lower jitter (and indirectly latency) than buses that don't.
It moves PCM data
- 16, 24, or 32-bit
- stereo
- 8kHz to 192kHz (verify)
The sender should be somewhat comfigurable, because not all receiver necessarily implement all of that(verify)
As I2S doesn't spec a plug, or how to deal with longer cables (impedance and such), it is indeed mostly used within devices.
Exceptions mainly being audiophile setups that want to choose their DACs. Because it wasn't made for that, this takes more care to do right, because impedance can cause synchronization issues, particularly at higher bitrates.
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))
- serial data
- ground
Some also add a master clock (MCLK).
This is not part of standard I2S, but can helps makes the DAC's timing a little more precise (DACs that don't need this may generate one internally).(verify)
BCLK pulses for each bit,
so should be samplerate * bitdepth * channelamount, e.g. 1411200 Hz for CD audio
LRCLK selects left/right channel
See also:
