d6252481e1
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
141 lines
6.5 KiB
Markdown
141 lines
6.5 KiB
Markdown
# Verifying audio without an audio device
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Neither container has a sound card, so "does it actually play?" cannot be
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answered by listening. It can be answered by measurement, and the two things
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usually meant by that question need different measurements.
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**Separate them before reaching for a tool:**
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| question | needs Godot? | needs a device? |
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|---|---|---|
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| Is the transcoded file faithful to the source? | no | no |
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| Does Godot actually route it to an output? | yes | no |
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| What does the *game* play on a menu move? | no (Canary) | a virtual one |
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## 1. Transcode fidelity — file against file
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This is the question P4 actually raised, and it needs neither an engine nor a
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device. Decode both, subtract, and measure what is left.
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```bash
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# Source, for a reference level
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ffmpeg -hide_banner -t 25 -i ADV.wmv \
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-af "aformat=channel_layouts=stereo,astats=measure_perchannel=none" -f null - 2>&1 \
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| grep "RMS level"
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# The difference signal: source minus transcode
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ffmpeg -hide_banner -t 25 -i ADV.wmv -t 25 -i ADV.ogv -filter_complex \
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"[0:a]aformat=channel_layouts=stereo[a];\
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[1:a]aformat=channel_layouts=stereo,volume=-1[b];\
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[a][b]amix=inputs=2:normalize=0,astats=measure_perchannel=none" -f null - 2>&1 \
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| grep "RMS level"
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```
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A faithful transcode puts the difference **40 dB or more below** the source.
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### Three ways this measurement lies
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Run it wrong and it reports a disaster that is not there. All three of these
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were hit on the first attempt:
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* **Alignment.** A one-sample offset makes the difference nearly as loud as the
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source. Cross-correlate and compensate *before* subtracting, or the number is
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meaningless. A first run gave source −25.3 dB against difference −34.2 dB —
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only 9 dB down, which looks catastrophic and proves nothing.
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* **Channel layout.** The source and the transcode do not have the same channel
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count. You are not comparing like with like unless both sides are downmixed
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the same way, and `astats` will give you a confident number regardless. See
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[`movie-audio-channels`][mac] for which profile a given movie is in — that is
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a disc fact and lives in the RE corpus, not here.
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* **A file still being written.** `ffprobe` reported the `.ogv` as 33 s against
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the source's 137 s — apparent catastrophic truncation, actually a transcode in
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progress. Check `mtime` and packet count before believing a duration, and
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write to a temp name and rename on completion so a reader cannot see a partial
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file at all.
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⚠️ **The downmix is an unrecorded decision, and it is not ours to make quietly.**
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Nothing in the manifest says a fold happened or on what weighting; it is whatever
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ffmpeg defaulted to, and that default can change between versions. Centre-channel
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dialogue folds into L/R, so this changes how speech sits against music — an
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aesthetic judgement, not a container detail. Pin it explicitly and record it,
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exactly as MISSION §6 requires of the transcode command itself.
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[mac]: https://git.mc02.dev/fabi/Syplheed-Reborn/src/branch/main/docs/re/structures/movie-audio-channels.md
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## 2. Engine routing — Godot writes a WAV instead of a device
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Godot does not need a sound card to produce audio you can inspect. Put an
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`AudioEffectRecord` on the **Master** bus and it captures the mixed output from
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inside a headless run:
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```gdscript
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var bus := AudioServer.get_bus_index("Master")
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var rec := AudioEffectRecord.new()
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AudioServer.add_bus_effect(bus, rec)
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rec.set_recording_active(true)
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# ... play the scene ...
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rec.set_recording_active(false)
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rec.get_recording().save_to_wav("user://master.wav")
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```
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**This is implemented.** `godot --path port -- --menu … --audio=/tmp/p6.wav`
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installs the effect, records for the whole run, and saves on exit — in
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`_exit_tree` rather than beside each `quit()`, because there are eight of those
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and the one that would get missed is an error path, i.e. exactly the run whose
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audio somebody wants to look at. The run prints the driver name beside the file
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it wrote.
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Then feed that WAV through §1 against the source. That closes the loop: it
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proves the asset is right **and** that the engine reached it, which no amount of
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file comparison can show on its own.
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Confirm the dummy driver is what is actually in use rather than assuming it —
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`AudioServer.get_driver_name()` — and say so in the write-up, because "recorded
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under a dummy driver" is a weaker claim than "heard", and the difference matters.
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## 3. A virtual device, when something insists on a real one
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For anything that opens a device rather than a bus — the emulator, most
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obviously — a PulseAudio **null sink** is a real device that records to a file.
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`pulseaudio-utils` is in both images, and `audio-capture` wraps it:
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```bash
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audio-capture run /tmp/menu.wav -- run-canary # start sink, run, record
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audio-capture start # or drive it by hand
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PULSE_SINK=cap godot --path port
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audio-capture record /tmp/out.wav &
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```
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This is the route to capturing what the **game** plays — the menu move and
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confirm cues behind HANDOFF Q8 — rather than what we believe it should play.
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Those bindings are currently a name match against the authors' own identifiers;
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a capture turns them into a measurement.
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⚠️ `audio-capture run` reports the peak level and **warns when the result is
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silent**, because silence is the failure that looks like success: a WAV of
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exactly the right duration, full of zeroes, because the application opened a
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different sink. A duration check alone would pass it.
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## What none of this establishes
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That it *sounds right*. Every method here shows correspondence to a source, not
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that the source is the audio the game plays at that moment, and not that levels
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are sane in a mix. A ten-second human listen still answers something no
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measurement above does — so when a result rests on one of these, say which one.
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## 4. What the exporter checks, so nobody has to remember to
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`sylpheed-export` measures **peak level and duration** of every audio file it
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writes and records both in `manifest.json`; `sylpheed-export check` refuses a
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tree whose peak is ≤ −90 dBFS (silent) or ≥ 0 dBFS (clipping).
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Those are content checks in a format validator on purpose. Silence is the failure
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this page opens by naming — right duration, right channel count, right size, full
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of zeroes — and every structural check passes it. Clipping is the other one, and
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the BGM can produce it, because a music bank is two stems summed at unity gain
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(HANDOFF Q10).
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⚠️ Neither says the audio is the **right** audio. `docs/port/BLOCKED.md` says
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which bindings are measured and which are still authored, and no measurement on
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this page can move a row there.
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