Files
RP412/docs/SOUND.md
T
CydandClaude Opus 5 523f713a30 Volume and bass knobs, for players without an amplifier
The cabinets ran the game at unity and shaped volume and tone 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 none in AUDIO.INI - an
operator turned a knob on an amp. A desktop player has no amp and no
crossover, and the recovered soundbanks are a good deal livelier than what
4.12 shipped with, so the game has to offer the two controls the pod got
from hardware.

RP412AUDIOVOLUME, 0.0 to 4.0, is the amplifier: a listener gain, which the
port had never set at all. RP412AUDIOBASS, 0.0 to 1.0, is the crossover's
low band. Both default to leaving the mix exactly as the pod played it, so
neither changes anything for anyone who does not go looking.

The bass trim is not a filter, and the reason is worth writing down: the
OpenAL we ship is Creative's, not OpenAL Soft, and it implements only
AL_FILTER_LOWPASS. It rejects highpass and bandpass outright. A bandpass
would have been the tidy answer, carrying the authored brightness model on
GAINHF and the trim on GAINLF across the single direct filter a source
gets. It is not on offer.

So the trim scales sample data as it loads, which suits how this low end is
actually built: the weight lives in discrete deep layer zones whose per-zone
tuning bakes out to a very low playback rate - thirteen zones below 8kHz,
three to five octaves under their recorded pitch, against four fifths of the
set at 22kHz and up. Baked rate is a dependable proxy for band, so pulling
down the low-rate zones is a real low-band trim and not a blunt cut. It eases
in below 22kHz and reaches full depth at 5.5kHz.

Caught while building this, and the reason for the probe: EFX_Initialize
checks alGetError after configuring the scratch filter, so asking for a
filter type the driver refuses leaves an error pending and takes the entire
bridge down - reverb included. The bandpass attempt did precisely that and
would have silently killed the reverb and brightness work. Initialize now
survives losing the filter and says so.

Builds clean, runs with both knobs set and with neither.

Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
2026-08-05 23:45:29 -05:00

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# 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`):
```cpp
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`):
```cpp
#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`):
```cpp
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`:
```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`):
```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:
```cpp
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:
```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`):
```cpp
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 (1008000 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`):
```cpp
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 | §45 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` | the amplifier's volume | 0.0 4.0 | 1.0 (unity, as the pod ran) |
| `RP412AUDIOBASS` | the crossover's low band | 0.0 1.0 | 1.0 (as authored) |
`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.
`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 scales sample data as buffers load. 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. 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.
**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.