# 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. ```bash # 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 `astats` will give you a confident number regardless. See [`movie-audio-channels`][mac] for 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.** `ffprobe` reported the `.ogv` as 33 s against the source's 137 s — apparent catastrophic truncation, actually a transcode in progress. Check `mtime` and 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. [mac]: https://git.mc02.dev/fabi/Syplheed-Reborn/src/branch/main/docs/re/structures/movie-audio-channels.md ## 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: ```gdscript 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: ```bash 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.