Hi Michele,
Thanks for your reply.
On Mon, Feb 26, 2024 at 05:06:12PM +0100, Michele Perrone wrote:
> You are correct about the available input and outputs, but I'm not sure that
> I understand your routing scheme very well.
>
> * The currently active digital input can be selected with "Digital
> Input Capture Route", like this:
> o AES/EBU: amixer -c 0 cset numid=3,iface=MIXER,name='Digital
> Input Capture Route' 0
> o S/PDIF RCA: amixer -c 0 cset numid=3,iface=MIXER,name='Digital
> Input Capture Route' 1
> o S/PDIF toslink: amixer -c 0 cset
> numid=3,iface=MIXER,name='Digital Input Capture Route' 2
> When playing sound from stream input, it doesn't make any
> difference which one the above is currently selected.
> * The clock can be selected independently from the current input with
> AV/C [1].
> * The 2-channel output can be routed to all the digital and analog
> outputs simultaneously (but analog outputs are only available if the
> DAC is connected).
> One can choose which outputs are active or not with the "Output
> Playback Switch":
> o AES/EBU on: amixer -c 0 cset numid=6,iface=MIXER,name='XLR::XLR
> Output Playback Switch' 1
> AES/EBU off: amixer -c 0 cset numid=6,iface=MIXER,name='XLR::XLR
> Output Playback Switch' 0
> o S/PDIF RCA on: amixer -c 0 cset
> numid=7,iface=MIXER,name='RCA::RCA Output Playback Switch' 1
> S/PDIF RCA off: amixer -c 0 cset
> numid=7,iface=MIXER,name='RCA::RCA Output Playback Switch' 0
> o DAC on: amixer -c 0 cset numid=10,iface=MIXER,name='DAC::DAC
> Output Playback Switch' 1
> DAC off: amixer -c 0 cset numid=10,iface=MIXER,name='DAC::DAC
> Output Playback Switch' 0
> * There is no hardware mixer capability, i.e. the inputs cannot be
> routed directly to the outputs.
>
> Our preamp mode works like this:
>
> 1. A digital input source is selected with AV/C (Toslink, or RCA, or
> XLR), i.e. with "Digital Input Capture Route"
> 2. Clock input source is set to the same as audio input source with
> AV/C [1]
> 3. DICE clock rate is set to external clock rate [2]
> 4. A simple 2-channel ALSA loopback with digital volume control is
> created between input and output
>
> I hope this answers your questions, otherwise please let me know.
Hm. I'm sorry but it is still unclear that the destination of audio
signal in the IEEE 1394 isochronous packet arrived at your device
(precisely the source port in the router function of TCAT DICE chip).
It is Playback PCM channel in ALSA word, and depicted as
'stream-input-1/2' in my diagram for my convenience.
```
spdif-opt-input-1/2 ---+
spdif-coax-input-1/2 --(or)--> digital-input-1/2 -----------------> stream-output-1/2
aesebu-xlr-input-1/2 ---+ |
v
stream-input-1/2 --------------------(or)--+----------------------> spdif-coax-output-1/2
+----------------------> aesebu-xlr-output-1/2
+--analog-output-1/2 --> analog-xlr-output-1/2
+-----------> analog-coax-output-1/2
```
I assume that the actual source selection of 'digital-input-1/2' is done
in the router function of DICE chip as well as the selection between
'digital-input-1/2' and 'stream-input-1/2'. The mixer function of the
chip is not used as I expected, thus the selection should exist as the
source of audio signals for the outputs. However, in the above description,
I cannot find such selection.
Or the device has a fixed route between 'stream-input-1/2' and
'analog-{xlr,coax}-output-1/2'? The user can not hear the audio signal
of opt/coax/xlr digital input ports in the analog outputs?
Thanks
Takashi Sakamoto
