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Audio: ENABLE sound -- real OpenAL + in-tree WAV loader; both backend DLLs were no-op STUBS (task #50)

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Root cause of "no sound, ever": the two audio backend DLLs shipped in the repo
are fakes. libsndfile-1.dll exports 15 funcs BY ORDINAL ONLY (no names) and
sf_open() always returns NULL; OpenAL32.dll (72KB) imports only KERNEL32 -- no
dsound/wasapi/winmm -- so it is a pure no-op that returns fake handles
(ctx=0x00000001, alGenSources->0) and never touches the hardware. The whole
render->device->buffer->source->play chain ran clean and silent.

Fixes:
- OpenAL32.dll: replace the stub with the real OpenAL Soft 1.25.2 Win32 build
  (imports AVRT/ole32/WINMM, real WASAPI backend). The exe imports the 25 AL
  funcs by NAME so it is a drop-in; alGenSources now yields a live source and
  alSourcePlay reaches AL_PLAYING.
- libsndfile: DROPPED entirely. It is replaced by LoadWavPCM() in L4AUDRES --
  a tiny RIFF/WAVE fmt+data reader that loads our soundbank WAVs (16-bit PCM)
  straight into the AL buffer. Removed the .lib/.dll from the link + copy and
  git-rm'd the stub. (This also kills the "ordinal 50 could not be located in
  libsndfile-1.dll" load-failure popup: adding an sf_strerror import bound to
  an ordinal the 2..16-only stub could not satisfy.)
- Soundbank: 241 samples cracked from AUDIO1/2.RES (SF2 v1.0) by
  tools/sf2extract.py into content/AUDIO/*.wav + the allPresets[2][128] table
  (audiopresets.cpp), replacing the zero-init btstubs stub. All 241 now load
  (alErr=0). BT_AUDIO_TEST plays buffer0 as proof-of-life; BT_AUDIO_LOG traces
  the chain.

Remaining: in-game triggering (AudioEntities on fire/step/engine/explosion)
so PlayNote fires during play -- next audio wave.

Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
This commit is contained in:
arcattack
2026-07-15 18:55:33 -05:00
co-authored by Claude Opus 4.8
parent 52440e13b0
commit 5b46655b82
250 changed files with 1193 additions and 73 deletions
Download Patch File Download Diff File Expand all files Collapse all files
engine/MUNGA_L4/L4AUDLVL.h
+1 -1
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@@ -35,7 +35,7 @@ struct PRESETINFO
bool is3d;
};
extern PRESETINFO allPresets[2][100];
extern PRESETINFO allPresets[2][128];
bool PRESET_isImplemented(int bank, int preset);
int PRESET_getNumSamples(int bank, int preset);
engine/MUNGA_L4/L4AUDRES.cpp
+104 -66
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@@ -1,3 +1,6 @@
#include <stdio.h>
#include <direct.h>
#include <string.h>
#include <cstdlib>
#include "mungal4.h"
#pragma hdrstop
@@ -14,11 +17,65 @@
#include "..\munga\app.h"
#include "..\munga\notation.h"
#include "openal\al.h"
#include "sndfile.h"
#include "openal\alc.h"
ALuint *g_buffers;
int g_numBuffers;
//
// Minimal canonical-PCM WAV reader. The repo's libsndfile-1.dll is a STUB
// (exports 15 funcs by ordinal only, no names -- sf_open always returns NULL
// and sf_strerror(NULL) faults), so the original sf_open path loaded nothing.
// Our soundbank samples (AUDIO1/2.RES, cracked by scratchpad/sf2extract.py)
// are written as plain 16-bit mono PCM WAVs, so a tiny RIFF/WAVE fmt+data
// parser loads the real sample data directly into the AL buffer -- no
// external dependency, no stub. Handles 8/16-bit mono/stereo PCM.
//
static bool LoadWavPCM(const char *path, char **outData, int *outBytes,
int *outChannels, int *outBits, int *outRate)
{
FILE *fp = fopen(path, "rb");
if (!fp) return false;
fseek(fp, 0, SEEK_END); long flen = ftell(fp); fseek(fp, 0, SEEK_SET);
if (flen < 44) { fclose(fp); return false; }
unsigned char *buf = new unsigned char[flen];
size_t got = fread(buf, 1, flen, fp);
fclose(fp);
if ((long)got != flen) { delete [] buf; return false; }
if (memcmp(buf, "RIFF", 4) != 0 || memcmp(buf + 8, "WAVE", 4) != 0) { delete [] buf; return false; }
int channels = 0, bits = 0, rate = 0;
char *pcm = 0; int pcmBytes = 0;
bool haveFmt = false;
long p = 12;
while (p + 8 <= flen)
{
const unsigned char *id = buf + p;
unsigned int csz = buf[p+4] | (buf[p+5]<<8) | (buf[p+6]<<16) | ((unsigned)buf[p+7]<<24);
long body = p + 8;
if (memcmp(id, "fmt ", 4) == 0 && csz >= 16)
{
channels = buf[body+2] | (buf[body+3]<<8);
rate = buf[body+4] | (buf[body+5]<<8) | (buf[body+6]<<16) | ((unsigned)buf[body+7]<<24);
bits = buf[body+14] | (buf[body+15]<<8);
haveFmt = true;
}
else if (memcmp(id, "data", 4) == 0)
{
long avail = flen - body;
pcmBytes = ((long)csz <= avail) ? (int)csz : (int)avail;
if (pcmBytes < 0) pcmBytes = 0;
pcm = new char[pcmBytes > 0 ? pcmBytes : 1];
memcpy(pcm, buf + body, pcmBytes);
}
p = body + csz + (csz & 1); // RIFF chunks are word-aligned
}
delete [] buf;
if (!haveFmt || pcm == 0) { delete [] pcm; return false; }
*outData = pcm; *outBytes = pcmBytes;
*outChannels = channels; *outBits = bits; *outRate = rate;
return true;
}
//#############################################################################
//####################### AudioObjectStream #############################
//#############################################################################
@@ -543,7 +600,7 @@ void
for(int i=1; i <= 2; i++)
{
for (int j=0; j < 100; j++)
for (int j=0; j < 128; j++)
{
if (PRESET_isImplemented(i,j))
{
@@ -574,7 +631,7 @@ void
//Load buffers
for(int i=1; i<=2 && bufferInd < g_numBuffers; i++)
{
for(int j=0; j<100 && bufferInd < g_numBuffers; j++)
for(int j=0; j<128 && bufferInd < g_numBuffers; j++)
{
if (PRESET_isImplemented(i,j))
{
@@ -582,84 +639,65 @@ void
{
SAMPLEINFO info = PRESET_getSampleInfo(i,j,k);
//Load WAV to buffer
char *data;
int size;
//Open WAV
char fullname[50];
strcpy_s(fullname,50,"AUDIO\\");
strcpy_s(fullname,50,"AUDIO/");
strcat_s(fullname,50,info.file);
SF_INFO *sfInfo = new SF_INFO[2];
//SF_INFO is working different than expected!
sfInfo->format = 0;
SNDFILE *file = sf_open(fullname,SFM_READ,sfInfo);
if (file == NULL)
char *data = 0; int size = 0;
int wavCh = 0, wavBits = 0, wavRate = 0;
if (!LoadWavPCM(fullname, &data, &size, &wavCh, &wavBits, &wavRate))
{
DEBUG_STREAM << "Failed to open." << std::endl;
if (getenv("BT_AUDIO_LOG") && bufferInd == 0) {
char cwd[512]; cwd[0]=0; _getcwd(cwd, sizeof(cwd));
DEBUG_STREAM << "[audio] DIAG cwd=[" << cwd << "] path=[" << fullname << "] WAV load FAILED\n" << std::flush;
}
DEBUG_STREAM << "[audio] Failed to open AUDIO/" << info.file << " -- skipping\n" << std::flush;
PRESET_setBufferIndex(i,j,k,bufferInd);
bufferInd++;
continue;
}
unsigned long formatBits = 0;
unsigned long sampleRate = sfInfo->samplerate;
size = sfInfo->frames;
bool isMono = (sfInfo->channels == 1);
if (sfInfo->format & SF_FORMAT_PCM_S8)
{
formatBits = 8;
} else if (sfInfo->format & SF_FORMAT_PCM_16)
{
formatBits = 16;
} else
{
DEBUG_STREAM << "BAD FORMAT" << std::endl;
}
ALenum format;
if (formatBits == 8)
{
if (isMono)
{
format = AL_FORMAT_MONO8;
} else
{
size *= 2;
format = AL_FORMAT_STEREO8;
}
} else if (formatBits == 16)
{
size *= 2;
if (isMono)
{
format = AL_FORMAT_MONO16;
} else
{
size *= 2;
format = AL_FORMAT_STEREO16;
}
}
if (wavBits == 8)
format = (wavCh == 1) ? AL_FORMAT_MONO8 : AL_FORMAT_STEREO8;
else
format = (wavCh == 1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
ALsizei alSampleRate = sampleRate;
//Load size & data
delete [] sfInfo;
data = new char[size];
sf_read_raw(file,data,size);
sf_close(file);
//Feed the buffer
alBufferData(g_buffers[bufferInd],format,data,size,alSampleRate);
//Feed the buffer with the real PCM sample data
alBufferData(g_buffers[bufferInd], format, data, size, wavRate);
if (getenv("BT_AUDIO_LOG") && bufferInd < 3)
DEBUG_STREAM << "[audio] loaded buf" << bufferInd << " " << info.file
<< " ch=" << wavCh << " bits=" << wavBits << " rate=" << wavRate
<< " bytes=" << size << " alErr=" << alGetError() << "\n" << std::flush;
PRESET_setBufferIndex(i,j,k,bufferInd);
bufferInd++;
delete [] data;
delete [] data;
}
}
}
}
if (getenv("BT_AUDIO_TEST") && g_numBuffers > 0)
{
// PROOF OF LIFE: play the first loaded sample so we can confirm the
// SF2 -> WAV -> OpenAL buffer -> speakers path works end to end.
ALCcontext *ctx = alcGetCurrentContext();
alGetError();
ALuint testsrc = 0; alGenSources(1, &testsrc);
ALenum genErr = alGetError();
alSourcei(testsrc, AL_BUFFER, g_buffers[0]);
ALenum bufErr = alGetError();
alSourcei(testsrc, AL_LOOPING, AL_TRUE);
alSourcef(testsrc, AL_GAIN, 1.0f);
alSourcePlay(testsrc);
ALenum playErr = alGetError();
ALint st = 0; alGetSourcei(testsrc, AL_SOURCE_STATE, &st);
DEBUG_STREAM << "[audio] TEST-PLAY buffer0 src=" << testsrc
<< " ctx=" << (void*)ctx
<< " genErr=" << genErr << " bufErr=" << bufErr << " playErr=" << playErr
<< " state=" << st << " playing=" << (int)(st==AL_PLAYING) << "\n" << std::flush;
}
//
//----------------------------------------------------------------------
// Load the sound banks
engine/lib/OpenAL32.dll
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engine/lib/libsndfile-1.dll
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engine/lib/libsndfile-1.lib
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