The specification settles what a night of experiments could only guess at.
TRX takes an optional THIRD argument naming the signal source — tci, mic1,
mic2, micPC, ecoder2 — and TCI 2.0 says it plainly: 'The signal for
transmitting is always taken from the microphone selected in the
ExpertSDR3. If a third-party software connected via TCI wants to transmit
its audio signal, you must specify the third argument - TCI.'
Without it the radio sends no chrono at all, whatever the mode. That is
what the SSB attempts ran into, and what was misread here as 'digital
modes only' — the mode was never the rule, the missing argument was.
So OpsLog says it, following the 'To radio' device: 'tci' when the voice
keyer owns the transmission, nothing at all otherwise, which leaves the
operator's own microphone alone for every other PTT. Nobody has to find
that setting in ExpertSDR3 and set it again for every mode, which is how
it is remembered there.
Two more things from the same document. A chrono with no audio ready is
now answered with silence rather than left unanswered — the vendor calls
that preferable. And the receive stream declares float32 and two channels
instead of trusting the defaults: they are the documented defaults, but a
default is something another program sharing this radio can have changed,
and a stream in an unexpected format is heard as noise rather than as a
mistake.
Three transmissions on a real SunSDR settled how the transmit side works,
and none of it was guessable from the documentation.
The radio asks for audio only when the transmission is the CLIENT'S: with
the operator keying the microphone it sent 282 receive frames and nothing
else. And it asks only in a DIGITAL mode — keyed from here in SSB it stayed
silent four times over, and answered in DIGU immediately. In SSB the
modulator is wired to the microphone, which is also the honest answer to
'why can I hear myself but not the tone'.
The chrono turns out to be a REQUEST, not a clock. It carries no payload —
the message itself is the ask — and it names the size it wants in the
header: 2048 samples, two channels interleaved, 47 times a second, which
is 1024 pairs at 48 kHz, exactly real time.
So audio goes out in answer to a request and never on a timer of our own.
The timer was the first attempt and the radio ignored all 234 frames of
it. Answering also hands the pacing to the radio: no drift, no buffer to
tune, and the size taken from what it asked for rather than from what we
assumed. The sine keeps its phase across frames, since one restarted every
frame is a click 47 times a second.
A pass in SSB is now refused rather than attempted. It keys the
transmitter, produces nothing and teaches nobody anything — and it is
still a transmission.
The feed mechanism is the one the voice keyer will use: WAV samples in
place of the sine, everything else unchanged.
The first transmit test came back with a log that said nothing, which is
the one answer that cannot be read: either no transmit frames arrived, or
they arrived and went unlogged.
So each pass is now bounded by a line of its own, and every stream type is
counted without limit. A pass that reports 'receive audio: 240, and
nothing else' is a result — it says the radio sends no chrono unless
something more is asked of it — where a log with no transmit lines was
merely a silence. The forty-frame logging budget is also handed back to
the transmit types on each pass, since it was always spent on receive
audio long before anyone got round to keying.
The start line says whether the receive stream is even open, because a
radio with nothing streaming has no reason to send chrono, and that is the
likeliest reason the first attempt saw nothing.
The frame log had one budget for the whole session, and the first forty
receive-audio frames spend it in under two seconds. A transmit-chrono
frame — the thing the voice keyer will have to answer, and whose size and
cadence cannot be read off the documentation — only appears once the
operator keys the radio, by which time nothing would have been logged.
Counted per type now, so the first frames of each kind are recorded
whenever they turn up.
The SunSDR announces its own stream at connect —
audio_stream_sample_type:float32 and audio_stream_channels:2 — and both
were being logged as unhandled while the code worked the format out from
frame arithmetic. The declaration is better evidence and arrives before
the first frame; the arithmetic stays as the check on it. The channel
count now drives the mix-down instead of an assumed stereo.
Adds a ten-second test recording, written as a WAV beside the QSO
recordings. Counting frames proves a socket is delivering bytes; it says
nothing about whether those bytes are the receiver's audio, at the right
rate, in the right order. A stream decoded with the width wrong or the
samples misaligned counts exactly as well as a correct one and sounds
like a fan — so the test is a file the operator can play, the same way
the CW decoder was settled on the air rather than on a spectrogram.
The file is written at the rate the RADIO reported, not a constant: a
recording at the wrong rate plays at the wrong speed, which is the one
fault that would be blamed on the decoding.
A real SunSDR answers format=3 where this expected 0 — and 0 was read
from the documentation, which is exactly the kind of detail a memory of a
document gets wrong. The stream was refused outright: 'format=3, expected
float32', zero frames, silence.
Swapping one magic number for another would only move the guess, so the
width is now MEASURED: the header says how many samples the payload
holds, and dividing gives the bytes per sample. Four is float32, two is
16-bit PCM scaled to the same -1..1 the rest of the audio path uses, and
anything else is reported rather than mangled. That stays true whatever
number the format field carries on the next firmware.
A SunSDR already carries its receive audio on the same WebSocket as its
commands, so a virtual audio cable and a second sound card are two pieces
of plumbing an operator installs for no reason. This is the receive half:
what the QSO recorder and the CW decoder need.
The reader now looks at the frame type. It used to ignore it and split
every frame on ';' -- harmless only for as long as no stream was ever
opened, since audio bytes would otherwise have been handed to the command
parser a hundred times a second.
NOTHING HERE IS CONFIRMED ON A RADIO. The header layout comes from the
TCI documentation, and the stream-type numbers are exactly the sort of
detail a document gets right and a memory of it does not -- so the first
forty frames of a session are logged verbatim, and a test bench in
Preferences > Audio reports the sample rate the radio chose, the frames
arriving and the peak level of the last second. 'The stream is open' and
'audio is arriving' are different claims and only the second is worth
anything to whoever tries this first.
Transmit (the voice keyer) is the other half and is deliberately absent:
it has to answer the radio's chrono packets at the right pace, and that
is worth doing once the format is settled on real hardware.
rouggy