RP412/docs/SOUND.md

Red Planet — the sound system, from two AWE32s to OpenAL

How a 1996 arcade pod produced true quadraphonic positional audio out of two consumer sound cards, what the modern port kept, what it silently dropped, and exactly where the original assets are.

Sources. The surviving engine in MUNGA_L4/ (the L4AUD* family) and the preserved hardware layer in MUNGA_L4/sos/; the complete original RP 4.10 C++ source and shipping assets in ../TeslaRel410/; and the BattleTech sibling tree ../BT411/, which shares this engine verbatim and has already fixed most of what's described here.

Companion docs: docs/audionotes.rtf (Stephen Baynham, 2007) covers the renderer's control flow — sources, sockets, the mix/running/dormant plugs. ../BT411/docs/AUDIO_FIDELITY.md is the 685-line fidelity audit this document maps onto RP. This doc covers the hardware model underneath both.

Two headlines:

1. The quadraphonic engine is still in the tree, still runs every frame, and its output is discarded. Nothing was deleted in the port. Four channel gains and four time-delay offsets are computed for every sound in the world, then dropped, because the OpenAL back-end that replaced the sound cards never reads them.
2. Red Planet's original soundbanks, authored sequences, source code and hardware configuration all survive in ../TeslaRel410/. Nothing about the original audio is lost. RP412 simply ships without them.

     source azimuth
          │
          ▼
 CalculateSpatialization()      ← quadrant pan + ITD, L4AUDIO.cpp:60
          │
          ├──► frontLeftScale / frontRightScale ─┐
          ├──► rearLeftScale  / rearRightScale   │   1996: CC7 volume to
          ├──► 4 × ITD delay targets             ├── 4 MIDI channels across
          └──► 4 × ITD pitch offsets (cents)     │   2 AWE32 cards
                                                 │
                                                 └── today: /* ... */ dead code,
                                                     OpenAL pans from AL_POSITION

---

1. Why two cards

AudioHardware held exactly two, named for what they drove (MUNGA_L4/L4AUDHDW.h:345-346):

AudioCard frontCard;
AudioCard rearCard;

The second card was not for extra voices. Each AWE32 gives you one stereo pair, and a four-corner speaker layout needs two. The allocator makes this explicit (MUNGA_L4/L4AUDRND.cpp:1309-1352): front-left and front-right channels are requested from front_card, rear-left and rear-right from rear_card. Either card refusing kills the whole allocation and the sound doesn't play.

Per card the engine assumed a stock EMU8000 (MUNGA_L4/L4AUDHDW.h:80-82):

Constant	Value
AWE_VOICE_COUNT	32
AWE_CHANNEL_COUNT	16
AWE_PERCUSSIVE_CHANNEL	9

64 hardware voices total, 32 MIDI channels, two independent stereo outputs.

2. The hardware layer: HMI SOS

Everything went through Human Machine Interfaces' Sound Operating System, selected at compile time (MUNGA_L4/L4AUDHDW.h:87):

#define _MIDI_DRIVER_TYPE _MIDI_AWE32

MUNGA_L4/sos/ still carries the complete driver headers in both flavours the build needed — bc4/ for Borland C++ 4 and wc/ for Watcom — alongside SOSMAWE.C, whose header comment reads "Module to handle AWE32 .SBK file uploads." That file is the bank loader: sosMIDIAWE32SetSBKFile, sosMIDIAWE32ReleaseSBKFiles, sosMIDIAWE32NoteOn/NoteOff.

AudioCard also poked the hardware directly for MPU-401 UART setup — the _inp/_outp port macros and MPU_RESET_CMD/MPU_ENTER_UART at MUNGA_L4/L4AUDHDW.cpp:14-25 are still there.

2.1 The actual pod hardware configuration

Card addresses were parsed by GetEnvironmentSettings (MUNGA_L4/L4AUDHDW.cpp:269-350) out of a BLASTER-format string. The standard single BLASTER= variable can only describe one card, so each got its own (MUNGA_L4/L4AUDHDW.cpp:935-936):

frontCard.GetEnvironmentSettings(FRONT_CARD_ENV_VAR);
rearCard.GetEnvironmentSettings(REAR_CARD_ENV_VAR);

Those two macros are referenced in four places and defined nowhere in this tree. The values survive in the shipping release — ../TeslaRel410/ALPHA_1/REL410/RP/SETENV.BAT:

set AWE_FRONT=A220 I5 D1 H5 P330 T6
set AWE_REAR=A240 I7 D3 H6 P300 T6

	Front card	Rear card
Base I/O	0x220	0x240
IRQ	5	7
DMA (8-bit)	1	3
DMA (16-bit)	5	6
MPU-401	0x330	0x300
Type	6	6

Two fully independent SB16/AWE32s, non-conflicting across every resource — a genuinely awkward ISA configuration to get stable, which is presumably why SETENV.BAT hardcodes it rather than probing.

Two cards were mandatory, not optional. L4Application::MakeAudioRenderer returned NULL — no audio renderer at all — unless both variables were present (MUNGA_L4/L4APP.cpp:505-514, now commented out). There was no one-card or stereo fallback in the shipping build.

SETENV.BAT also drove the SB16 mixer per card via sb16set, and carries three details worth recording:

• Master volume defaults to AWE_MASTER_VOLUME=200, overridable by an operator file c:\setvol.bat — the per-cabinet volume trim.
• Intercom mode (L4INTERCOM=ON) swaps audio\ctmix.cfg for audio\icom.cfg and adds sb16set /li:220;0 on the front card only. Diffing the two configs (ALPHA_1/REL410/RP/AUDIO/), the only change is line-in routing: LIL+/LIR+ into the input and output paths. The intercom fed the front card's line input.
• There are two :SOUNDCOMMON labels. DOS batch jumps to the first, so the second block — which trims the cards differently from each other (bass 245 vs 240, treble 110 vs 135) — is unreachable dead code. Someone tuned front and rear separately and it never shipped.

3. Nothing streamed — the game was a MIDI sequencer

There is no mixer and no audio thread in the original design. Sound effects were SoundFont samples resident in each card's onboard sample RAM, and playing a sound meant allocating a MIDI channel and sending note-on plus CC7 volume. The game drove two samplers in real time.

That is why both cards were loaded with identical banks (dist/AUDIO/AUDIO.INI):

[AudioResources]
front_audio_resource=audio\audio1.res
front_audio_resource=audio\audio2.res
rear_audio_resource=audio\audio1.res
rear_audio_resource=audio\audio2.res

Same content in both cards' RAM, so any sound could be placed anywhere in the ring without a reload. The cost is that the entire sound set had to fit twice over in AWE32 sample memory.

It also explains the shape of the whole audio API. AudioChannel exposes SendNoteOn, SendProgramChange, SendPitchBend, SendNRPN, SelectBank — a MIDI abstraction, not a sample-playback abstraction. Every positional and DSP decision the engine makes has to be expressed as a MIDI controller value.

4. The quad panner

L4AudioSpatialization::CalculateSpatialization(azimuth) (MUNGA_L4/L4AUDIO.cpp:60-290) is the whole positional model. It is character-for-character identical to the 1995 original at ../TeslaRel410/CODE/RP/MUNGA_L4/L4AUDIO.CPP:150-290, down to the // HACK comments.

Azimuth is rewrapped so 0° is dead ahead and the range is ±180°, then split into four 90° quadrants around azimuth_max = 45°:

                    front
              FL ─────┬───── FR
               │  Q1  │
               │      │
        Q2     │  ▲   │     Q4
      (left)   │  │   │   (right)
               │ +az  │
              RL ─────┴───── RR
                     Q3
                    rear

        +azimuth → left        −azimuth → right

Within a quadrant it constant-power pans between the two bracketing speakers only — a source never feeds more than two of the four, which is correct for a four-corner layout:

tangent_ratio = (tan(azimuthOfSource) / tan(azimuth_max)) * 0.5f;

frontLeftScale  = Sqrt(0.5f + tangent_ratio);
frontRightScale = Sqrt(0.5f - tangent_ratio);

tangent_ratio runs ±0.5, so the gains trace sqrt(0.5±t) — sum of squares constant at 1.0, i.e. constant acoustic power across the sweep, no hole in the middle.

Quadrant	Arc	Active pair
Q1	−45° … +45°	front-left / front-right
Q2	+45° … +135°	rear-left / front-left
Q3	±135° … 180°	rear-right / rear-left
Q4	−135° … −45°	front-right / rear-right

Q1/Q3 use tan(azimuth_max) as the half-width while Q2/Q4 use tan(DEG_90 - azimuth_max). With azimuth_max = 45° these are equal and all four arcs are 90°, but the code is written so front/rear arcs could be widened against the side arcs independently. azimuth_max is hardcoded with a // HACK - should come from audio.ini comment at L4AUDIO.cpp:103.

Q3 relies on tan having period 180° to handle the wrap at ±180° — for az < −135 the expression azimuthOfSource - DEG_180 goes below −315°, and the result is only correct because tangent is periodic. It works; it is not obvious.

5. The ITD trick

Amplitude panning alone gives direction but not much externalization. The engine also modelled interaural time difference — the sub-millisecond arrival-time gap between your ears that the brain actually uses to localize. AUDIO.INI:

distance_between_ears=12.0
itd_difference=0.0015

The problem: an EMU8000 has no delay line. You cannot ask an AWE32 to play a voice 1.5 ms late. There is no such MIDI message and no such hardware path.

The solution: don't delay the voice — detune it. To make a voice arrive progressively earlier or later, momentarily shift its pitch, which shifts its playback rate, which slides it through time. Return the pitch to normal and the voice stays there, phase-shifted. Doppler used as a phase-steering primitive.

CalculateSpatialization sets a delay target per channel; the caller converts the rate of change of that target into a cents offset (MUNGA_L4/L4AUDIO.cpp:414-438):

const Scalar itd_pitch_offset_constant =
    0.003831f / 0.000002f;              // period / delay

frontLeftITDPitchOffset =
    itd_pitch_offset_constant *
        (spatialization.frontLeftDelay - currentFrontLeftDelay) /
            (Scalar)itd_delta_time;
currentFrontLeftDelay = spatialization.frontLeftDelay;

Only one of the two active channels gets a nonzero delay target, scaled by the same tangent_ratio as the gain, so maximum offset at full pan is exactly itd_difference — 1.5 ms. The rear quadrant negates the sign (rearLeftDelay = -(itd_delay * tangent_ratio * 2.0f), L4AUDIO.cpp:214), flipping the lead/lag relationship behind the listener.

On the magic constant — a derivation, not something the source states. 0.003831 / 0.000002 = 1915.5 cents per unit of delay slew. The exact small-signal value for a Doppler-style rate-to-pitch conversion is 1200 / ln 2 ≈ 1731 cents. They agree within about 10%, which confirms the mechanism: a hand-tuned first-order approximation, presumably trimmed by ear on the pod.

Three further details:

• Computed against real elapsed frame time (Now() - lastITDFrameTime), so the slew is framerate-independent.
• Applied only while the target is moving. A stationary source contributes zero pitch offset and sits at whatever phase it reached.
• distance_between_ears=12.0 is commented "-> average size of cockpit". BT uses 2.0. This is the clearest surviving fingerprint of the pod build: the head model was scaled to the physical cabinet, because the speakers really were in the corners around the player. Confirmed authentic — RP412's AUDIO.INI is byte-identical to the shipping 4.10 file dated 31 August 1995 (../TeslaRel410/ALPHA_1/REL410/RP/AUDIO/AUDIO.INI). Every tuning constant in this repo is the original; there has been zero config drift in thirty years.

6. The rest of the per-frame model

All of it expressed as MIDI, all driven from AUDIO.INI:

Effect	Mechanism	INI keys
Distance attenuation	CC7 volume, knee + rolloff curve	amplitude_rolloff, _knee, _distance_scale
Distance muffling	AWE initial-filter-cutoff NRPN 21 (100–8000 Hz)	high_frequency_rolloff, _knee, _distance_scale
Doppler	pitch bend in cents	doppler_range, speed_of_sound
Reverb	CC91 send, wet exterior / dry cockpit	global_reverb_scale
Source compression	gain curve on the summed mix	compression_scale, compression_exponent
Clipping	hard cull sphere	clipping_radius

NRPN constants are still declared at MUNGA_L4/L4AUDHDW.h:63-68 (AWE_FILTER_CUTOFF_NRPN 21, AWE_VOL_ATTACK_TIME_NRPN 11, AWE_PITCH_NRPN 16).

AUDIOMR.INI is a shipped variant differing from AUDIO.INI in exactly one respect — compression is far more aggressive (compression_scale=0.1, compression_exponent=9.0 vs 0.92/8.5). Everything else is identical.

7. What the port did

Both trees replaced the AWE32/SOS back-end with OpenAL Soft, by commenting out rather than deleting. MUNGA_L4/L4AUDHDW.h is 530 lines of which the great majority is preserved-in-amber AWE code: AudioChannel, AudioCard and AudioHardware are entirely inside /* */. The quad CC7 volume switch survives the same way from MUNGA_L4/L4AUDIO.cpp:1964.

The replacement is SourceSet (MUNGA_L4/L4AUDHDW.h:9-13):

struct SourceSet
{
    int count;
    ALuint sources[5];
};

Four MIDI channels-per-sound became up to five OpenAL sources — one per sample zone in the preset, not one per speaker. Placement is handed to OpenAL via alSource3f(..., AL_POSITION, ...) (MUNGA_L4/L4AUDIO.cpp:1413).

The consequence is the first headline. CalculateSpatialization is still called every frame from UpdateSpatialModelImplementation, still computes four gains and four ITD pitch offsets. Every consumer of those values is commented out.

8. Where the original assets actually are

Update (2026-08-05): both banks now live in assets/RP411/AUDIO/, and the WAV set and preset table are generated from them by tools/rp_sf2extract.py. The history below is kept because it is what made that possible, and because the .SCP sources are still only in TeslaRel410.

RP412 used to ship 223 loose .wav files loaded through libsndfile, a hand-maintained preset table in RP_L4/WTPresets.cpp, and a 1-byte stub AUDIO.RES. The front_audio_resource/rear_audio_resource lines in AUDIO.INI were never stale leftovers — they are the original authored configuration, and the banks they name existed all along, simply never carried into the port.

Everything below is verified present in ../TeslaRel410/:

Asset	Location	Detail
RP soundbanks	ALPHA_1/REL410/RP/AUDIO/AUDIO1.RES, AUDIO2.RES	Genuine SoundFonts (RIFF…sfbk), 3,781,754 B (Oct 1996) and 3,708,348 B (May 1996)
Earlier bank revision	sda4/RPLIVE/AUDIO/	Nov 1995 / Oct 1995; AUDIO1 differs by 4 bytes
Authored sequences	sda4/RPLIVE/AUDIO/*.SCP	70 files including STATIC.SCP
Sequences (partial)	CONTENT/RP/AUDIO/*.SCP	62 files, no STATIC.SCP
Original C++ source	CODE/RP/MUNGA_L4/L4AUD*.CPP	Complete pre-port DOS source
Hardware config	ALPHA_1/REL410/RP/SETENV.BAT	The AWE_FRONT/AWE_REAR strings in §2.1
Mixer configs	ALPHA_1/REL410/RP/AUDIO/CTMIX.CFG, ICOM.CFG	Normal and intercom routing

Three things this settles:

1. RP's banks are its own. MD5s differ from BT's, which are byte-identical between TeslaRel410/ALPHA_1/REL410/BT/AUDIO/ and BT411/content/AUDIO/ — so the provenance chain is proven on the BT side, and RP's distinct content is sitting unused.
2. The .SCP files are build-time sources, not runtime assets. CreateStaticAudioStreamResource (MUNGA_L4/L4AUDRES.cpp:769) is called only from the asset tool (MUNGA/TOOL.cpp:100), which compiles them into RPL4.RES. RP412 ships a working RPL4.RES, so the authored audio objects are present — what's missing is the editable source form, now recovered.
3. RP has a reference BT lacks. BT411's audit had to Ghidra-decompile BTL4OPT.EXE to confirm F4, F9, F10, F11 and F12. For Red Planet the actual C++ source exists, so every one of those can be verified directly rather than inferred.

9. Fidelity gaps — the BT411 audit mapped onto RP412

BT411's audit graded its OpenAL port across 23 findings and has since fixed most of them. Its sections C and D (dead attribute bindings, ReportLeak, torso-twist servos) are BattleTech-entity-specific and do not transfer. Its synthesis and spatial findings do.

Every gap below was re-verified against RP412's own code, not assumed. The comment-block state of each cited line was checked programmatically.

Engine-side — asset-independent

Status: all fixed (2026-08-05). Line references are to the pre-fix tree.

#	Gap	Evidence found in RP412	What landed
F3	Authored distance curve computed then discarded; AL_LINEAR_DISTANCE used instead	volume_scale *= GetDistanceVolumeScale() commented at L4AUDIO.cpp:1449; alDistanceModel(AL_LINEAR_DISTANCE) live at MUNGA/AUDIO.cpp:97; AL_MAX_DISTANCE written at :1081,1416,1941	alDistanceModel(AL_NONE); multiply restored on Dynamic3D; new Static3DPatchSource::CalculateSourceVolumeScale override; the three AL_MAX_DISTANCE writes dropped
F4	Volume written linearly where the original used the CC7 squared law	three live alSourcef(..., AL_GAIN, volume_scale) at L4AUDIO.cpp:1082,1414,1939	AL_GAIN, volume_scale * volume_scale at all three
F9	Brightness / HF-rolloff chain dead	GetHighFreqCutoffScale() had zero callers	new L4AUDEFX lowpass: Dynamic3D takes HF-rolloff × brightness, Static3D and Direct take brightness alone
F10	Doppler wrong constants and wrong sign	alDopplerFactor(0.3f); GetDopplerCents() zero callers	alDopplerFactor(0.0f) + pitch_offset += GetDopplerCents() on the dynamic path only
F11	Reverb wet-exterior / dry-cockpit split dead	CC91 sends commented at L4AUDIO.cpp:1227,1717	EFX EAXReverb aux slot at global_reverb_scale; sends attached on Dynamic3D/Static3D, Direct left dry
F12	Direct placement dead — everything dead-centre	all three switch (audioPosition) blocks commented	AL_POSITION written per the authored enum after SetupPatch
P1	AL_PITCH never called anywhere in the tree	relativePitch computed at :1034,1408,1922 and discarded at all three	pitch applied at all three sites
F22	Quad + ITD model dead	§4–5 above	still open — needs multichannel output (§10 step 4)

P1 is an RP-specific find with no BT counterpart, and it is larger than F10 alone. RP412 had no AL_PITCH call at all, so the entire pitch chain was inert — not just doppler but pitch_mix_offset / PitchAudioControlID, which RP's own sequences author 97 times. Fixing F10 without this would have changed nothing audible.

A note on note-pitch: BT411 applies 2^((note-60)/12), because its SF2-derived presets carry authored key-splits. RP is different — SAMPLEINFO has no root-key field, and RP's authored content predates NoteAudioControlID entirely (its AudioControlID enum stops at AttackTimeAudioControlID), so every source runs at DEFAULT_NOTE=60 and the factor is identically 1.0. It is applied anyway for engine parity, clearly marked as inert for current content.

F3 was the highest-leverage single change: restoring the authored curve also repairs the distance-blind transient cull, the voice-steal weighting, and the mix-ducking chain, all of which were treating far sources as full-presence.

F22 is the one where RP is the lead repo rather than the follower. BT411 classes it low-priority because it "matters mostly for pod-hardware target" — which is precisely what this project is.

Verified on this machine: ALC_EXT_EFX is present and all nine EFX entry points resolve, so F9/F11 are live rather than silently inert. The driver grants 256 mono sources — OpenAL Soft's default budget, which BT411 raised explicitly via context attributes. RP412 still accepts the default; worth revisiting if voice starvation shows up in a busy match.

Asset-side — measured against RP's own banks

RP's banks turned out to differ from BT's in ways that matter, so BT's prevalences do not transfer. Measured directly (tools/rp_sf2extract.py --stats):

	RP total
Presets	154 (67 in bank 1, 87 in bank 2)
Instrument zones	395
Multi-zone presets	130 / 154 (84%)
Key-splits	zero
Max zones in any preset	4
Looping zones	129
Zones with an authored low-pass	154
Zones with layer attenuation	97
Zones with releaseVolEnv	349

Two RP-specific findings that change the work:

1. RP's banks contain no key-splits at all. Every multi-zone preset is a pure layer stack whose zones share one key range, all covering note 60. Combined with RP's content never authoring a note (§9 above), key ranges are unreachable here — so zone selection is a non-issue and every zone is simply a simultaneous voice. BT's F1, which is largely about key-splits, mostly does not apply; what applies is the plain zone count.
2. Max 4 zones per preset, which fits PRESETINFO.samples[5] as it stands. No structural change was needed — and the engine's long-standing Warn(GetVoiceCount() > 4) ("AWE appears to only play 1st 4 voices", L4AUDLVL.cpp:29) matches the bank data exactly.

#	Gap	RP status
F1	Zones dropped	fixed — 93 presets were short; 176 zones recovered, 219 → 395
F2	Root key / tuning dropped	fixed — every shipped WAV was flat 44100 Hz; 202 of 219 checked zones were off, worst ~9 semitones. Tuning is now baked into each WAV's declared rate
F14	Per-zone generators	partly fixed — initialAttenuation (SBK inverted scale) and the authored initialFilterFc/Q resonant low-pass are baked into the PCM. Volume envelopes still dropped
F13	Loop regions and release envelopes	open — needs SAMPLEINFO fields plus engine work (see below)

F13 remains the outstanding asset-side item. It needs loopStart/loopEnd and releaseSec on SAMPLEINFO, AL_SOFT_loop_points at buffer setup, and a gain ramp on the stop path. 349 of 395 zones carry an authored release envelope that is currently an instant cut.

The ordering hazard BT flagged — loop regions must land with or before multi-zone extraction, or whole-buffer looping over a short loop region replays an entire sample every cycle — is worth re-checking for RP now that zone counts have gone up. RP's 129 looping zones should be measured for loop-region coverage before F13 lands.

The churn this work introduced — now fixed (2026-08-05). Recovering the zones took voice demand per sound from ~1.1 to ~2.6, roughly doubling the allocation churn: RequestAudioChannels called alGenSources per sound event and ReleaseSourceSet called alDeleteSources on release. BT411 hit exactly this, and its measured conclusion was that raising the source budget was not the fix — pooling was, and a net CPU win besides.

Sources are now generated once and recycled through a free list (RPAudioPoolAcquire / RPAudioPoolRelease, L4AUDRND.cpp), capped at 240 against the driver's 256-mono grant. Steady-state play costs no allocation: measured 3 sources generated across 12,000 acquisitions.

Two real bugs were sitting underneath it, both verified against this driver rather than assumed:

• The bulk delete was atomic and leaked whole sets. alDeleteSources(3, {valid, valid, 0}) returns an error and deletes nothing — both live sources survive. The old ReleaseSourceSet passed the whole fixed-size array and then parked slots at -1 (0xFFFFFFFF), so any partial set, or any double release, leaked its entire allocation. Release is now per-source, and slots park at 0, which is never a valid AL name.
• SourceSet.sources[] was never initialized. The constructor set only count, and RequestAudioChannels decided whether a slot was already filled by asking alIsSource about uninitialized stack garbage. A value that happened to match a live name meant two sources silently sharing one — a latent hazard that pooling would have made more likely, since recycling keeps small integer names in circulation.

Recycled sources are scrubbed before being parked: stopped, buffer detached, looping/gain/pitch/relative/position/velocity reset, and the EFX direct filter and reverb send cleared — without that last part a dry cockpit sound could inherit the wet send of the 3D source that held the name before it. Verified: a source deliberately dirtied then released comes back with looping=0, gain=1.0, pitch=1.0, relative=0.

9a. Tuning knobs — standing in for hardware the pod had

The cabinets ran the game at unity gain and did all their volume and tone shaping outside it, in an external amplifier and a 3-way crossover. That is why there is no master volume anywhere in the original code, and why AUDIO.INI has no level control: the operator turned a knob on an amp.

A desktop player has neither, so the port has to provide them. Two env vars, both documented in environ.ini, both defaulting to leaving the mix exactly as the pod played it:

Knob	Stands in for	Range	Default
RP412AUDIOVOLUME / PgUp, PgDn	the amplifier's volume	0.0 – 2.0	1.0 (unity, as the pod ran)
RP412AUDIOBASS / Home, End	the crossover's low band	0.0 – 1.0	1.0 (as authored)

Both step live by 0.05 and persist beside the exe (volume.cfg, bass.cfg), which then win over environ.ini next launch.

RP412AUDIOVOLUME is a straight alListenerf(AL_GAIN, …) at renderer init. There was no listener gain call at all before, so the default is a genuine no-op. PgUp/PgDn step it live by 0.05 and persist to volume.cfg beside the exe, which then wins over environ.ini on the next launch — the env var decides where an untouched machine starts, the keys are the knob, and a knob stays where it was left.

The keys are polled (GetAsyncKeyState in LBE4ControlsManager::Execute), not taken off the key-message path, and that is not a style choice. RP's keyboard pump consumes only WM_KEYUP/WM_SYSKEYUP/WM_CHAR from the front of the queue while the front-end runs message loops of its own, so key messages are raced for and routinely lost — measured at roughly two of every six presses arriving when the volume was first wired through that path. Survivable for a one-shot like the abort chord; not for a control you tap repeatedly to find a level. This is a pre-existing property of the input path, worth knowing before binding anything else to it.

RP412AUDIOBASS is not an EFX filter, and the reason is worth recording: the OpenAL this game ships implements only AL_FILTER_LOWPASS. It is Creative's (installed by oalinst.exe; the renderer reports "Generic Software"), not OpenAL Soft, and it rejects both AL_FILTER_HIGHPASS and AL_FILTER_BANDPASS — verified on the build machine. A bandpass would have been the neat answer, carrying the authored brightness model on GAINHF and the trim on GAINLF across the one direct filter a source gets. It is not available.

So the trim is a per-zone gain applied in the mix. That works because of how RP's low end is built: the weight sits in discrete deep layer zones whose per-zone tuning bakes out to a very low playback rate — 13 zones below 8 kHz, 3.4 to 5.2 octaves below their recorded pitch, against 81% of the set at 22 kHz and above. A zone's baked rate is a reliable proxy for which band it occupies, so attenuating the low-rate zones is a real low-band trim rather than a blunt overall cut. Each buffer's depth is fixed at load from its rate; the trim itself is read at mix time, which is what lets Home/End move it while sounds are playing. The ramp is untouched at/above 22050 Hz, full trim at/below 5512 Hz, log-interpolated between. At 0.7 that is −3.1 dB on the deepest layers, −1.4 dB at 11 kHz, nothing from 22 kHz up.

(An earlier revision scaled the PCM at load instead. That could never be a live knob — the samples are already in OpenAL buffers by the time a key is pressed — and it risked quantisation on top. The gain form is both live and cleaner.)

A caution for anyone extending the EFX work: EFX_Initialize reads alGetError() after configuring the scratch filter, so asking for a filter type this driver does not support leaves an error pending and takes the whole bridge down with it — reverb included. That is not hypothetical; it is exactly what the bandpass attempt did before the filter-type probe caught it.

AUDIO.INI is also a live mixing desk now, for the first time since 1995 — those constants used to be computed and discarded. global_reverb_scale is the wet amount, amplitude_rolloff/_knee/_distance_scale set how loud distant things are, high_frequency_rolloff* how dull, compression_* the ducking, clipping_radius the cull. Read once at init. Editing it diverges from the authored 1995 values, which is a real cost — it is byte-identical to the shipping original today.

10. A recovery path, in order

1. Engine-side fidelity first. Done (2026-08-05). F3, F4, F9, F10, F11, F12 and P1 all landed; L4AUDEFX.cpp/.h ported and added to Munga_L4.vcxproj. Builds clean on VS2022 Release|Win32; smoke-tested against vRIO on COM1 with RP412STEAM=0 — reaches gameplay and holds a steady frame loop.

2. Wire RP's banks in. Done (2026-08-05). Both banks are now in assets/RP411/AUDIO/, hash-identical to the 1996 originals. tools/rp_sf2extract.py extracts all 395 zones with tuning, layer attenuation and the authored low-pass baked in, and regenerates RP_L4/WTPresets.cpp. Verified: 176 zones recovered with none lost (the 46 preset slots that disappeared were all empty placeholders); every extreme baked rate (1228 Hz – 88200 Hz) accepted by libsndfile → alBufferData → alSourcePlay on the real runtime path. Still open here: F13.

   Confirmed by ear (2026-08-05): markedly more bass. That is the expected signature of the tuning fix — the deepest layers were the worst offenders, a collision sub-thud playing at 44100 Hz where the bank says 1228 — compounded by the 176 recovered zones, which are disproportionately the low rumble layers sitting under collisions and explosions.

3. Recover the .SCP sources from sda4/RPLIVE/AUDIO/ into the asset pipeline, so authored audio becomes editable again rather than frozen in RPL4.RES.

4. Then quad. With the above in place:

   • Ask ALC for a multichannel format instead of accepting the stereo default (MUNGA_L4/L4AUDRND.cpp:380).
   • Place four AL_SOURCE_RELATIVE sources at fixed corner positions and drive their AL_GAIN from the existing GetFrontLeftScale() family, bypassing OpenAL's panner.
   • Feed the ITD offsets to AL_PITCH — or implement a real fractional delay, which a software mixer can do and the EMU8000 could not. The detune path is already written and is the authentic behaviour.
   • Re-derive azimuth_max from the actual cabinet speaker angles instead of the hardcoded 45°.

Steps 1 and 2 are where nearly all the audible improvement is. Step 4 is what made the pod feel like the sound was in the room with you.

11. Verifying any of this

File	What's in it
MUNGA_L4/L4AUDHDW.h	AWE/MIDI constants, AudioCard/AudioHardware (commented), SourceSet
MUNGA_L4/L4AUDHDW.cpp	MPU-401 port I/O, BLASTER parsing (:269), card init (:935)
MUNGA_L4/L4AUDIO.cpp	CalculateSpatialization (:60), ITD pitch (:414), dead quad CC7 path (:1964)
MUNGA_L4/L4AUDRND.cpp	renderer, OpenAL init (:380), dead quad channel allocator (:1309)
MUNGA_L4/L4AUDRES.cpp	resource manager, WAV → AL buffers, SCP compile path (:769)
MUNGA_L4/L4APP.cpp:505	the dead two-card gate on renderer creation
MUNGA_L4/sos/	HMI SOS driver headers (bc4 + wc), SOSMAWE.C bank uploader
dist/AUDIO/AUDIO.INI	every tuning constant — byte-identical to the 1995 original
../TeslaRel410/CODE/RP/MUNGA_L4/	the original DOS source, for anything the comments don't answer
../BT411/docs/AUDIO_FIDELITY.md	the full 23-finding audit this section maps from

The commented-out regions are a faithful copy of the original — verified against TeslaRel410/CODE/RP/MUNGA_L4/L4AUDIO.CPP, which matches character-for-character in the spatialization and ITD paths. They were preserved deliberately and they describe exactly how the pod's audio hardware was driven.