Hello everyone,
Disclaimer: I've never worked in the sound/ layer, and it is possible that
some of my questions are silly or obvious.
Basically, I'm trying to implement some form of eARC(*) on an arm64 SoC.
(*) enhanced Audio Return Channel (from HDMI 2.1)
The setup looks like this:
A = Some kind of audio source, typically a TV or game console
B = The arm64 SoC, equipped with some nice speakers
HDMI
A ------> B
If we look inside B, we actually have
B1 = an eARC receiver (input = HDMI, output = I2S)
B2 = an audio DSP (input = I2S, output = speakers)
I2S ?
B1 -----> B2 -----> speakers
If I read the standard right, B is supposed to advertise which audio formats
it supports, and A is supposed to pick "the best". For the sake of argument,
let's say A picks "PCM, 48 kHz, 8 channels, 16b".
At some point, B receives audio packets, parses the Channel Status, and
determines that A is sending "PCM, 48 kHz, 8 channels, 16b". The driver
then configures the I2S link, and forwards the audio stream over I2S to
the DSP.
QUESTION_1:
How is the DSP supposed to "learn" the properties of the audio stream?
(AFAIU, they're not embedded in the data, so there must be some side-channel?)
I assume the driver of B1 is supposed to propagate the info to the driver of B2?
(Via some call-backs? By calling a function in B2?)
QUESTION_2:
Does it ever make sense for B2 to ask B1 to change the audio properties?
(Not sure if B1 is even allowed to renegotiate.)
Regards.
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