The three Static.slb cues and the menu bed now export to Ogg Vorbis and play.
`sylpheed_formats::media` does the assembly; nothing in port/ has heard of XMA.
Three things this milestone got wrong before it got right, all recorded in
docs/port/DECISIONS.md because the corrections are the useful part:
1. The cue offsets were a Rust `const` in the exporter. They are MEASURED, not
decoded -- a measured value compiled into the exporter is a measurement
wearing the costume of a decoded field, and nobody deletes it because nobody
can see it. They are authored/audio.json now.
2. I picked BGM_001 and wrote a careful `why` calling the choice arbitrary. The
menu's music is BGM_103, and it is in HANDOFF at 0fd8e69 -- the exact commit
BLOCKED.md says that row was reconciled against. Not stale: wrong when
written. I had summarised a negative without its reach, so "the TABLES cannot
say which BGM a screen plays" became "it is not on the disc". One word of
scope was the whole answer, and the export failed only because BGM_001
without its .slb extension hashes to nothing. That is luck, not design.
3. The comment above the BGM sum argued for unity gain "because halving is a mix
decision nobody made". It clipped at +1.8 dBFS. 1/n is the smallest constant
that provably cannot clip -- the same reasoning video.rs already carried for
its 5.1 downmix, in this repository, unread.
Unsettled and shipped as such: media::sound_bank_riffs returns THREE sub-waves
for BGM_103.slb where HANDOFF Q10's census says exactly two (the third is the
leading headerless region slb.rs emits for the voice path). The exporter sums all
three and writes a manifest warning, because which bytes belong together is the
decoders' question, not this exporter's -- and dropping one would destroy the
evidence, since a corrected export looks exactly like a correct one. Raised with
the Decoder; row in BLOCKED.md.
The gate is a null control, not a peak reading. A master-bus WAV that is
non-silent proves nothing -- the bed alone would look identical. So the same
scripted walk was run with <- in place of <v>, which fires no cue (Q5, measured),
and the difference is one 0.55 s burst at t=1.10 s and silence everywhere else.
The first attempt at that control returned bit-identical zero and I nearly filed
it as "cues never reach the bus": both runs ended at 1.115 s and the first press
lands at 1.17 s. A null result from an instrument that was not running is not a
null result.
Refutation attempt: HANDOFF Q8's three cue durations. They looked attackable --
0.133/0.172/0.169 s per packet, no shared rate -- but an XMA1 packet carries a
variable number of 512-sample frames, and the three come to 50.0/32.3/95.3
frames. Measured off the decoded Ogg: 0.533, 0.344, 1.016 s, every published
digit. SURVIVES, with its reach stated -- it confirms the assembly path and my
transcription, not the event bindings, which only an oracle can retake.
Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01WM5XL4HfrHuxz8RiMWdCMC
6.5 KiB
Verifying audio without an audio device
Neither container has a sound card, so "does it actually play?" cannot be answered by listening. It can be answered by measurement, and the two things usually meant by that question need different measurements.
Separate them before reaching for a tool:
| question | needs Godot? | needs a device? |
|---|---|---|
| Is the transcoded file faithful to the source? | no | no |
| Does Godot actually route it to an output? | yes | no |
| What does the game play on a menu move? | no (Canary) | a virtual one |
1. Transcode fidelity — file against file
This is the question P4 actually raised, and it needs neither an engine nor a device. Decode both, subtract, and measure what is left.
# Source, for a reference level
ffmpeg -hide_banner -t 25 -i ADV.wmv \
-af "aformat=channel_layouts=stereo,astats=measure_perchannel=none" -f null - 2>&1 \
| grep "RMS level"
# The difference signal: source minus transcode
ffmpeg -hide_banner -t 25 -i ADV.wmv -t 25 -i ADV.ogv -filter_complex \
"[0:a]aformat=channel_layouts=stereo[a];\
[1:a]aformat=channel_layouts=stereo,volume=-1[b];\
[a][b]amix=inputs=2:normalize=0,astats=measure_perchannel=none" -f null - 2>&1 \
| grep "RMS level"
A faithful transcode puts the difference 40 dB or more below the source.
Three ways this measurement lies
Run it wrong and it reports a disaster that is not there. All three of these were hit on the first attempt:
- Alignment. A one-sample offset makes the difference nearly as loud as the source. Cross-correlate and compensate before subtracting, or the number is meaningless. A first run gave source −25.3 dB against difference −34.2 dB — only 9 dB down, which looks catastrophic and proves nothing.
- Channel layout. The source and the transcode do not have the same channel
count. You are not comparing like with like unless both sides are downmixed
the same way, and
astatswill give you a confident number regardless. Seemovie-audio-channelsfor which profile a given movie is in — that is a disc fact and lives in the RE corpus, not here. - A file still being written.
ffprobereported the.ogvas 33 s against the source's 137 s — apparent catastrophic truncation, actually a transcode in progress. Checkmtimeand packet count before believing a duration, and write to a temp name and rename on completion so a reader cannot see a partial file at all.
⚠️ The downmix is an unrecorded decision, and it is not ours to make quietly. Nothing in the manifest says a fold happened or on what weighting; it is whatever ffmpeg defaulted to, and that default can change between versions. Centre-channel dialogue folds into L/R, so this changes how speech sits against music — an aesthetic judgement, not a container detail. Pin it explicitly and record it, exactly as MISSION §6 requires of the transcode command itself.
2. Engine routing — Godot writes a WAV instead of a device
Godot does not need a sound card to produce audio you can inspect. Put an
AudioEffectRecord on the Master bus and it captures the mixed output from
inside a headless run:
var bus := AudioServer.get_bus_index("Master")
var rec := AudioEffectRecord.new()
AudioServer.add_bus_effect(bus, rec)
rec.set_recording_active(true)
# ... play the scene ...
rec.set_recording_active(false)
rec.get_recording().save_to_wav("user://master.wav")
This is implemented. godot --path port -- --menu … --audio=/tmp/p6.wav
installs the effect, records for the whole run, and saves on exit — in
_exit_tree rather than beside each quit(), because there are eight of those
and the one that would get missed is an error path, i.e. exactly the run whose
audio somebody wants to look at. The run prints the driver name beside the file
it wrote.
Then feed that WAV through §1 against the source. That closes the loop: it proves the asset is right and that the engine reached it, which no amount of file comparison can show on its own.
Confirm the dummy driver is what is actually in use rather than assuming it —
AudioServer.get_driver_name() — and say so in the write-up, because "recorded
under a dummy driver" is a weaker claim than "heard", and the difference matters.
3. A virtual device, when something insists on a real one
For anything that opens a device rather than a bus — the emulator, most
obviously — a PulseAudio null sink is a real device that records to a file.
pulseaudio-utils is in both images, and audio-capture wraps it:
audio-capture run /tmp/menu.wav -- run-canary # start sink, run, record
audio-capture start # or drive it by hand
PULSE_SINK=cap godot --path port
audio-capture record /tmp/out.wav &
This is the route to capturing what the game plays — the menu move and confirm cues behind HANDOFF Q8 — rather than what we believe it should play. Those bindings are currently a name match against the authors' own identifiers; a capture turns them into a measurement.
⚠️ audio-capture run reports the peak level and warns when the result is
silent, because silence is the failure that looks like success: a WAV of
exactly the right duration, full of zeroes, because the application opened a
different sink. A duration check alone would pass it.
What none of this establishes
That it sounds right. Every method here shows correspondence to a source, not that the source is the audio the game plays at that moment, and not that levels are sane in a mix. A ten-second human listen still answers something no measurement above does — so when a result rests on one of these, say which one.
4. What the exporter checks, so nobody has to remember to
sylpheed-export measures peak level and duration of every audio file it
writes and records both in manifest.json; sylpheed-export check refuses a
tree whose peak is ≤ −90 dBFS (silent) or ≥ 0 dBFS (clipping).
Those are content checks in a format validator on purpose. Silence is the failure this page opens by naming — right duration, right channel count, right size, full of zeroes — and every structural check passes it. Clipping is the other one, and the BGM can produce it, because a music bank is two stems summed at unity gain (HANDOFF Q10).
⚠️ Neither says the audio is the right audio. docs/port/BLOCKED.md says
which bindings are measured and which are still authored, and no measurement on
this page can move a row there.