Sylpheed/docs/port/AUDIO-VERIFICATION.md

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 astats will give you a confident number regardless. See movie-audio-channels 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.

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.

5. A multichannel capture must pass a provenance check BEFORE it is analysed

tools/port/check-capture FILE.wav — run it first, every time.

⚠️ This section exists because a capture of the game's own 6-channel output was analysed at length and the file was corrupt. It got three controls, a drift test and a written-up negative, and every one of those was sound; none of them could see that channels were missing, because the corruption was upstream of everything they tested.

PulseAudio was remapping between two mismatched channel maps, and a 6-channel remap silently drops and duplicates. The Decoder proved it with a control that needs no emulator and no disc — six channels each carrying a different tone, through the same sink and the same parec invocation (docs/re/audio-capture-channel-map-trap.md):

ch	played	recorded
0	400	400
1	800	3200
2	200	200
3	1600	800
4	3200	800
5	6400	200

Two source channels were gone entirely and two were duplicates. Setting the sink's channel_map to the guest's own (FL,FR,FC,LFE,RL,RR) and passing the same map to parec returns all six.

The signature is an exact duplicate pair, and only a hash finds it

Duration is right. Channel count is right. Corked: no. There is no error anywhere, and the per-channel levels look entirely reasonable — which is the whole difficulty. In the tool's own known-bad control, all six channels report a peak of −18.063656 dB, identical to six decimals, while containing three duplicate pairs. A level check cannot see this. Hashing each channel can.

Two channels of a real surround mix are never byte-identical over tens of seconds. On the corrupt game capture the tool reports:

ch2  peak -4.466272    ba497de78217c438a3e430c5ef6b951b
ch5  peak -4.466272    ba497de78217c438a3e430c5ef6b951b
🔴 ch2 and ch5 are BYTE-IDENTICAL

⚠️ It is a necessary check, not a sufficient one. Passing says the file has no duplicated channels. It says nothing about whether the right thing was recorded — that is what §1's correlation against a known source is for, and a capture should survive both before anything is concluded from it.

Two more conditions, learned the same way

• Start the recorder before the process you are capturing, so t = 0 precedes it and the window certainly contains the moment of interest.
• Log what was on screen, with timestamps keyed to the recording's own clock. A capture that matches nothing is then diagnosable rather than ambiguous; the corrupt one could not be told apart from "recorded the wrong phase of the boot" by any amount of analysis at this end.

And the failure this page already warns about, in a second costume: run-canary is silent twice over — SDL_AUDIODRIVER=dummy and --mute=true. Fix only the first and Canary attaches a healthy 6-channel stream at 100 % volume, reports Corked: no, and emits a 19 MB WAV of zeroes.

7. A capture can be starved — right duration, holes punched through it

check-capture tests this too, and it is the second way a recording looks perfect and carries nothing.

A monitor sink advances at wall-clock rate and substitutes silence whenever the producer is late. An emulator running below real time therefore yields a file of exactly the right duration, the right channel count, no duplicated channels — chopped into fragments with holes between them, thousands of times over.

Measured independently on the capture that prompted this (the Decoder's numbers on the untruncated original in brackets):

frames silent on all six channels	35.6 % [39.3 %]
alternating runs	10 482 [10 595]
median burst / gap	13.5 ms / 3.9 ms [13.6 / 3.9]
period	17.4 ms → 57 Hz [≈17.5 ms → 57 Hz]

⚠️ This destroys envelope correlation by construction. What dominates the envelope of such a file is the dropout schedule, not the content — so §6's method was working correctly on a file that could not carry the signal, and the negative it produced said nothing about the game.

Two thresholds I invented were wrong, and the controls caught both

1. Counting exact-zero frames. Real audio crosses zero constantly, so a clean voice track scored 5 947 "gaps" of median 0.0 ms and was called starved. A gap is a run, not a sample: only runs of ≥ 1 ms count.
2. Gap count and median length. A genuine music-and-effects bed has 454 gaps at a median of 1.4 ms — quiet 16-bit passages really are zero for milliseconds — so neither statistic separates it from a starved file.

What does separate them is the RATE, by an order of magnitude in both directions:

	gaps/s	median gap	all-channel silence
the starved capture	32.9	3.9 ms	35.6 %
a real music+SFX bed	3.3	1.4 ms	1.1 %
a voice track, 53 % pauses	0.03	—	—

The bar is 20 gaps/s — 1.6× below the bad case, 6× above the worst good one, and set from those controls rather than chosen and then justified.

🔴 The monitor-sink route cannot be fixed by configuration

parec reads a monitor that advances at wall-clock rate and substitutes silence, so every moment the emulator runs below real time is a hole, and the timebase is warped non-uniformly — deleting the silences compresses time unevenly rather than repairing it. The route that would work is an internal tap at SDLAudioDriver::SubmitFrame, which sees every frame the guest produces in guest order with no wall clock in the loop.

⚠️ That needs a Canary rebuild, and the Decoder has costed it: build-canary targets a source root that does not exist in that container, the warm build tree is configured against the same missing path, so any change is a full reconfigure plus a full compile on a box with ~700 MB free and a history of parallel builds OOM-killing the host. A whole session for one probe — the human's call, not an agent's.

And a header that never got patched

A streaming writer leaves data declaring 0 bytes. check-capture says so and tells you the duration is unverified — which is not pedantry: the file shared here was copied while it was still being written, and the provenance claim that came with it was wrong about both its length and what it contained.

6. Finding one component inside a mix — and why §1's method cannot

🔴 This section begins with a retraction. Two captures of the game's own output were analysed with sliding envelope cross-correlation and declared not to contain the intro's audio. The instrument was never controlled for the actual task, and when it finally was, it failed:

Can it find the movie's bed inside a synthetic mix of that bed plus the three voice streams? r = 0.415 — below the r > 0.8 bar those negatives were judged against.

The first negative happened to be right (the file was independently proved corrupt by a tone control). It was right by luck, and the reasoning behind it was not supported. A filter that fails its own known-positive is dead, not tuneable.

What was wrong: the threshold, not the idea

r > 0.8 was calibrated on clean-against-clean comparisons, where it is correct — a transcode against its source scores 1.000. A component inside a mix can never score that, because everything else in the mix is uncorrelated noise from the component's point of view. Judging one task by the other's bar guarantees a false negative.

Judge on the LAG and the MARGIN instead. A real match lands at the right lag with a clear gap to the runner-up; a false one is a plateau. And band-limit first, so the component you are hunting dominates what you measure.

The calibration, on a known-present and a known-absent pair

Both bands, both directions, envelope at 0.1 s, minimum 60 s overlap:

hunting	band	against	r	lag	margin
the movie bed	40–180 Hz	mix containing it	0.663	0.0 s ✓	+0.111
the movie bed	40–180 Hz	voice-only mix	0.262	−31.9 s ✗	+0.005
voice stream 2	300–3000 Hz	mix containing it	0.810	0.0 s ✓	+0.248
voice stream 2	300–3000 Hz	the bed alone	0.358	−58.4 s ✗	+0.005

A 20–50× separation in the margin, and the lag is right or absurd. That is a decision rule set by controls rather than by tuning until the data agreed — which is the distinction that matters, and the one the first version of this method skipped.

⚠️ Reach. The known-positive is a synthetic mix at equal gains. A real game mix weights its components differently, so this bounds the method rather than modelling the real case exactly. It is enough to separate present from absent; it is not a level measurement.

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.