scummvm/sword2/driver/d_sound.cpp

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/* Copyright (C) 1994-2003 Revolution Software Ltd
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*
* $Header$
*/
//=============================================================================
//
// Filename : d_sound.c
// Created : 3rd December 1996
// By : P.R.Porter
//
// Summary : This module holds the driver interface to direct sound.
//
// Version Date By Description
// ------- --------- --- -----------------------------------------------
// 1.0 03-Dec-96 PRP The sound buffer can be created, with the
// format defined by the game engine, and speech
// can be played.
//
// 1.1 05-Dec-96 PRP Sound effects now done.
//
// 1.2 19-Dec-96 PRP Added volume and pan to speech and sound
// effects. Also, added type to sound effects
// so that they can be looped. Implemented
// a CloseAllFx function which will clear out
// all sound effects.
//
// 1.3 20-Dec-96 PRP Fixed a bug in the function which clears
// spot effects when they have finished playing.
//
// 1.4 02-Jan-97 PRP Fixed a bug in ClearAllFx which was trying
// to close the speech.
//
// 1.5 08-Apr-97 PRP Added ... to the
// InitialiseSound function.
//
// 1.6 09-Apr-97 PRP Added functions to steam music from CD.
//
// 1.7 29-May-97 PSJ Added functions to save and restore the state
// of the sound drivers.
//
// 1.8 04-Jun-97 PRP Added bodge to PlayFx routine which registers
// a sound effect to remove itself from the list
// if it is the tune to leave a sequence.
//
// 1.9 06-Jun-97 PSJ Expanded volTable from 17 to 241 entries.
// Added an fx and a speech master volume level.
// Added SetFxVolume and GetFxVolume for fx master
// volume. Added SetSpeechVolume and GetSpeechVolume
// for speech master volume.
//
// 1.10 09-Jun-97 PSJ Added SetMusicVolume and GetMusicVolume.
//
// 1.11 09-Jun-97 PSJ Fixed bug in SetSpeechVolume.
//
// 1.12 10-Jun-97 PSJ Added MuteMusic, MuteSpeech, MuteFx, IsMusicMute,
// IsFxMute and IsSpeechMute.
//
// 1.13 12-Jun-97 PSJ Added PlayCompSpeech to play compressed speech
// from a speech cluster.
//
// 1.14 19-Jun-97 PSJ Added StreamCompMusic and UpdateCompSampleStreaming
// to play compressed music from a music cluster.
// Added StopMusic to fade out any music playing.
//
// 1.15 24-Jun-97 PSJ Changed PlayCompSpeech to physically check for
// playing samples rather than using the assuming the
// speechStatus flag is correct.
//
// 1.16 24-Jun-97 PSJ Fixed bug it SetSpeechVolume.
//
// 1.17 26-Jun-97 PSJ Added AmISpeaking() for lip syncing.
//
// 1.18 26-Jun-97 PSJ Tweaked the nose of the dread, killer AmISpeaking
// function.
//
// 1.19 26-Jun-97 PSJ Added PauseSpeech and UnpauseSpeech.
//
// 1.20 26-Jun-97 PSJ Fixed a bug in the muteSpeech routine.
//
// 1.21 26-Jun-97 PSJ Fixed a bug in the AmISpeaking routine.
//
// 1.22 26-Jun-97 PSJ PlayCompSpeech loads and pauses the speech
// ready to be played by UnpauseSpeech.
//
// 1.23 01-Jul-97 PSJ Fixed GetSpeechStatus to work when speech is paused
//
// 1.24 03-Jul-97 PSJ Stopped PlayCompSpeech clicking at the end of samples.
//
// 1.25 10-Jul-97 PSJ Reduced music volume by 1/4 when playing speech
//
// 1.26 10-Jul-97 PSJ GetMusicVolume return safeMusicVol if it is set.
//
// 1.27 15-Jul-97 PRP Added functions to pause and unpause the sound effects.
//
// 1.28 15-Jul-97 PRP Fixed PauseFx
//
// 1.29 16-Jul-97 PSJ Added GetCompSpeechSize and PreFetchCompSpeech
//
// 1.30 16-Jul-97 PRP Fixed setting of sound fx volume.
//
// 1.31 18-Jul-97 PRP Added speech expansion to get samples to sound the same.
//
// 1.32 18-Jul-97 PRP Hopefully fixed expansion algorithm.
//
// 1.33 18-Jul-97 JEL Fixed UnpauseFx()
//
// 1.34 18-Jul-97 JEL Fixed PlayCompSpeech()
//
// 1.35 18-Jul-97 JEL Removed speech volume enhancing (now to be done in speech compressor)
//
// 1.36 21-Jul-97 PRP Added new type of sound effect which is the music lead in.
// Also, added function to pause the sound effects
// just for sequences.
//
// 1.37 21-Jul-97 PRP Modified ClearAllFx so that it doesn't kick out
// lead in and lead out music for smacker sequences.
//
// 1.38 21-Jul-97 PRP Tried to fix the bug where the second lead in
// music will not play due to a duplicate id.
//
// 1.39 21-Jul-97 PRP Finally fixed the bug to kick out lead in music
// fx when they have finished.
//
// 1.40 25-Jul-97 JEL Fixed crashing when music paused & unpaused repeatedly
//
// 1.41 28-Jul-97 PRP Checked to see if fx are looping as well as playing!
//
// 1.42 30-Jul-97 PSJ Added Music dipping.
//
// 1.43 30-Jul-97 PSJ Added MusicTimeRemaining.
//
// 1.44 31-Jul-97 PSJ Adjusted MusicTimeRemaining to include music left in buffer.
//
// 1.45 06-Aug-97 PSJ Updated Get and Set scroll SoundStatus.
//
// 1.46 12-Aug-97 PSJ Added ReverseStereo(void)
//
// 1.47 13-Aug-97 PSJ Updated DipMusic so it fades up after speech has finished.
//
// 1.48 13-Aug-97 PRP Added IsFxOpen().
//
// 1.49 15-Aug-97 PRP Added SetFxVolumePan().
//
// 1.50 15-Aug-97 PRP Added SetFxIdVolume()
//
// 1.51 15-Aug-97 PSJ Fixed bug in PlayCompMusic();
//
// 1.52 19-Aug-97 JEL Fixed bug in MusicTimeRemaining()
//
// WE'VE SCREWED UP THE NUMBERING!
//
// 1.59 19-Aug-97 JEL Fixed bug in MusicTimeRemaining(), ;)
//
// 1.60 19-Aug-97 PSJ Updated DipMusic so it fades music a bit more.
//
// 1.61 21-Aug-97 PSJ Updated StreamCompMusic so if both streams are in use,
// the fading stream is stopped and the new tune started.
//
// 1.62 21-Aug-97 PSJ Updated StreamCompMusic so if the music is unmuted,
// the last tune is restarted if it was looping.
//
// 1.63 22-Aug-97 PSJ Update PlayFx to handle smacker leadouts.
//
// 1.64 27-Aug-97 PSJ Update PlayFx to record an fx's local volume,
// So SetFxVolume can update playing fx's with the
// correct volume.
//
// 1.65 27-Aug-97 PSJ Stopped CloseFX from closing invalid fx's.
//
// 1.66 01-Sep-97 PRP Cleared the fxPaused flag when closing fx.
//
// 1.67 01-Sep-97 PRP Fixed the fact that SetFxVolume was still
// being done even if the fx were muted.
//
// 1.68 01-Sep-97 PRP Set zero sound to -10000
//
// Functions
// ---------
//
// --------------------------------------------------------------------------
//
// int32 InitialiseSound(uint16 freq, uint16 channels, uint16 bitDepth)
//
// This function initialises DirectSound by specifying the parameters of the
// primary buffer.
//
// Freq is the sample rate - 44100, 22050 or 11025
// Channels should be 1 for mono, 2 for stereo
// BitDepth should be either 8 or 16 bits per sample.
//
// --------------------------------------------------------------------------
//
// int32 PlaySpeech(uint8 *data, uint8 vol, int8 pan)
//
// This function plays the wav file passed into it as speech. An error occurs
// if speech is already playing, or directSound comes accross problems. The
// volume is 0 for zero volume and 16 for maximum volume. The pan position
// is -16 for full left, 0 for central and 16 for full right.
//
// --------------------------------------------------------------------------
//
// int32 PlayCompSpeech(const char *filename, uint32 id, uint8 vol, int8 pan)
//
// This function loads, decompresses and plays the wav 'id' from the cluster
// 'filename'. An error occurs if speech is already playing, or directSound
// comes accross problems. 'volume' can be from 0 to 16. 'pan' can be from
// -16 (full left) to 16 (full right).
// id is the text line id used to reference the speech within the speech
// cluster.
//
// --------------------------------------------------------------------------
//
// int32 StopSpeechBS2(void)
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//
// Stops the speech from playing.
//
// --------------------------------------------------------------------------
//
// int32 GetSpeechStatus(void)
//
// Returns either RDSE_SAMPLEPLAYING or RDSE_SAMPLEFINISHED
//
// --------------------------------------------------------------------------
//
// int32 AmISpeaking(void)
//
// Returns either RDSE_QUIET or RDSE_SPEAKING
//
// --------------------------------------------------------------------------
//
// int32 PauseSpeech(void)
//
// Stops the speech dead in it's tracks.
//
// --------------------------------------------------------------------------
//
// int32 UnpauseSpeech(void)
//
// Re-starts the speech from where it was stopped.
//
// --------------------------------------------------------------------------
//
// int32 OpenFx(int32 id, uint8 *data)
//
// This function opens a sound effect ready for playing. A unique id should
// be passed in so that each effect can be referenced individually.
//
// WARNING: Zero is not a valid ID.
//
// --------------------------------------------------------------------------
//
// int32 PlayFx(int32 id, uint8 *data, uint8 vol, int8 pan, uint8 type)
//
// This function plays a sound effect. If the effect has already been opened
// then *data should be NULL, and the sound effect will simply be obtained
// from the id passed in. If the effect has not been opened, then the wav
// data should be passed in data. The sound effect will be closed when it
// has finished playing.
//
// The volume can be between 0 (minimum) and 16 (maximum). The pan defines
// the left/right balance of the sample. -16 is full left, and 16 is full
// right with 0 in the middle. The sample type can be either RDSE_FXSPOT, or
// RDSE_FXLOOP.
//
// WARNING: Zero is not a valid ID
//
// --------------------------------------------------------------------------
//
// int32 CloseFx(int32 id)
//
// This function closes a sound effect which has been previously opened for
// playing. Sound effects must be closed when they are finished with,
// otherwise you will run out of sound effect buffers.
//
// --------------------------------------------------------------------------
//
// int32 ClearAllFx(void)
//
// This function clears all of the sound effects which are currently open or
// playing, irrespective of type.
//
// --------------------------------------------------------------------------
//
// int32 StreamMusic(uint8 *filename, int32 loopFlag)
//
// Streams music from the file defined by filename. The loopFlag should
// be set to RDSE_FXLOOP if the music is to loop back to the start.
// Otherwise, it should be RDSE_FXSPOT.
// The return value must be checked for any problems.
//
// --------------------------------------------------------------------------
//
// int32 StreamCompMusic(uint8 *filename, uint32 id, int32 loopFlag)
//
// Streams music 'id' from the cluster file 'filename'. The loopFlag should
// be set to RDSE_FXLOOP if the music is to loop back to the start.
// Otherwise, it should be RDSE_FXSPOT.
// The return value must be checked for any problems.
//
// StreamCompMusic should not be used inconjunction with StreamMusic.
//
// --------------------------------------------------------------------------
//
// void StopMusic(void)
//
// Fades out and stops the music.
//
// --------------------------------------------------------------------------
//
// int32 PauseMusic(void)
//
// Stops the music dead in it's tracks.
//
// --------------------------------------------------------------------------
//
// int32 UnpauseMusic(void)
//
// Re-starts the music from where it was stopped.
//
// ---------------------------------------------------------------------------
//
// int32 MusicTimeRemaining(void)
//
// Returns the time left for the current tune.
//
// ----------------------------------------------------------------------------
//
// int32 ReverseStereo(void)
//
// This function reverse the pan table, thus reversing the stereo.
//
//=============================================================================
#define WIN32_LEAN_AND_MEAN
//#include <windows.h>
//#include <windowsx.h>
#include <stdio.h>
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#include "stdafx.h"
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#include "driver96.h"
#include "rdwin.h" // for hwnd.
#include "d_sound.h"
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// Decompression macros
#define MakeCompressedByte(shift,sign,amplitude) (((shift)<<4) + ((sign)<<3) + (amplitude))
#define GetCompressedShift(byte) ((byte)>>4)
#define GetCompressedSign(byte) (((byte)>>3) & 1)
#define GetCompressedAmplitude(byte) ((byte) & 7)
#define GetdAPower(dA,power) for (power = 15;power>0 && !((dA) & (1<<power)); power--)
/*
LPDIRECTSOUND lpDS;
LPDIRECTSOUNDBUFFER dsbPrimary;
LPDIRECTSOUNDBUFFER dsbSpeech;
LPDIRECTSOUNDBUFFER dsbFx[MAXFX];
*/
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int32 panTable[33] = {
-10000,
-1500, -1400, -1300, -1200,
-1100, -1000, -900, -800,
-700, -600, -500, -400,
-300, -200, -100, 0,
100, 200, 300, 400,
500, 600, 700, 800,
900, 1000, 1100, 1200,
1300, 1400, 1500, 10000
};
int32 volTable[241] = {
-10000, -3925, -3852, -3781, -3710, -3642, -3574, -3508, -3443, -3379, -3316, -3255, -3194, -3135, -3077, -3020, -2964, -2909, -2855, -2802, -2750, -2699, -2649, -2600, -2551, -2504, -2458, -2412, -2367, -2323, -2280, -2238, -2197, -2156, -2116, -2077, -2038, -2000, -1963, -1927, -1891, -1856, -1821, -1788, -1755, -1722, -1690, -1659, -1628, -1598, -1568, -1539, -1510, -1482, -1455, -1428, -1401, -1375, -1350, -1325,
-1300, -1290, -1279, -1269, -1259, -1249, -1239, -1229, -1219, -1209, -1199, -1190, -1180, -1171, -1161, -1152, -1142, -1133, -1124, -1115, -1106, -1097, -1088, -1080, -1071, -1062, -1054, -1045, -1037, -1028, -1020, -1012, -1004, -996, -988, -980, -972, -964, -956, -949, -941, -933, -926, -918, -911, -904, -896, -889, -882, -875, -868, -861, -854, -847, -840, -833, -827, -820, -813, -807,
-800, -791, -782, -773, -764, -755, -747, -738, -730, -721, -713, -705, -697, -689, -681, -673, -665, -658, -650, -643, -635, -628, -621, -613, -606, -599, -593, -586, -579, -572, -566, -559, -553, -546, -540, -534, -528, -522, -516, -510, -504, -498, -492, -487, -481, -476, -470, -465, -459, -454, -449, -444, -439, -434, -429, -424, -419, -414, -409, -404,
-400, -362, -328, -297, -269, -244, -221, -200, -181, -164, -148, -134, -122, -110, -100, -90, -82, -74, -67, -61, -55, -50, -45, -41, -37, -33, -30, -27, -25, -22, -20, -18, -16, -15, -13, -12, -11, -10, -9, -8, -7, -6, -6, -5, -5, -4, -4, -3, -3, -3, -2, -2, -2, -2, -1, -1, -1, -1, -1, 0
};
int32 musicVolTable[17] = {
-10000,
-5000, -3000, -2500, -2250,
-2000, -1750, -1500, -1250,
-1000, -750, -500, -350,
-200, -100, -50, 0
};
BS2Sound::BS2Sound(void) {
soundOn = 0;
speechStatus = 0;
fxPaused = 0;
speechPaused = 0;
speechVol = 14;
fxVol = 14;
speechMuted = 0;
fxMuted = 0;
compressedMusic = 0;
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volMusic[0] = 16;
volMusic[1] = 16;
musicMuted = 0;
}
/* not used seemingly - khalek
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#define SPEECH_EXPANSION
#ifdef SPEECH_EXPANSION
int16 ExpandSpeech(int16 sample)
// This code is executed to expand the speech samples to make them sound
// louder, without losing the quality of the sample
{
double x, xsquared, result;
double expansionFactor = 2.5;
x = (double) sample;
xsquared = sample * sample;
if (x < 0.0)
{
result = expansionFactor * x + (expansionFactor - 1.0) * xsquared / 32768.0;
if (result < -32767.0)
result = -32767.0;
}
else
{
result = expansionFactor * x + (1.0 - expansionFactor) * xsquared / 32768.0;
if (result > 32767.0)
result = 32767.0;
}
return (int16) result;
}
#endif
*/
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// --------------------------------------------------------------------------
// This function reverse the pan table, thus reversing the stereo.
// --------------------------------------------------------------------------
int32 BS2Sound::ReverseStereo(void)
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{
int32 i,j;
for (i = 0; i<16; i++)
{
j = panTable[i];
panTable[i] = panTable[32-i];
panTable[32-i] = j;
}
return (RD_OK);
}
// --------------------------------------------------------------------------
// This function returns the index of the sound effect with the ID passed in.
// --------------------------------------------------------------------------
int32 BS2Sound::GetFxIndex(int32 id)
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{
int32 i = 0;
while (i < MAXFX)
{
if (fxId[i] == id)
break;
i++;
}
return(i);
}
int32 BS2Sound::IsFxOpen(int32 id)
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{
int32 i = 0;
while (i < MAXFX)
{
if (fxId[i] == id)
break;
i++;
}
if (i == MAXFX)
return 1;
else
return 0;
}
// --------------------------------------------------------------------------
// This function checks the status of all current sound effects, and clears
// out the ones which are no longer required in a buffer. It is called on
// a slow timer from rdwin.c
// --------------------------------------------------------------------------
void BS2Sound::FxServer(void)
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{
warning("stub FxServer");
/*
int32 i;
int32 status;
if (!soundOn)
return;
if (musicPaused[0] + musicPaused[1] == 0)
{
if (compressedMusic == 1)
UpdateCompSampleStreaming();
else if (compressedMusic == 2)
UpdateSampleStreaming();
}
if (fxPaused)
{
for (i=0; i<MAXFX; i++)
{
if ((fxId[i] == 0xfffffffe) || (fxId[i] == 0xffffffff))
{
IDirectSoundBuffer_GetStatus(dsbFx[i], &status);
if (!(status & (DSBSTATUS_PLAYING + DSBSTATUS_LOOPING)))
{
if (fxCached[i] == RDSE_FXTOCLEAR)
{
IDirectSoundBuffer_Release(dsbFx[i]);
fxId[i] = 0;
}
}
}
}
return;
}
for (i=0; i<MAXFX; i++)
{
if (fxId[i])
{
IDirectSoundBuffer_GetStatus(dsbFx[i], &status);
if (!(status & (DSBSTATUS_PLAYING + DSBSTATUS_LOOPING)))
{
if (fxCached[i] == RDSE_FXTOCLEAR)
{
IDirectSoundBuffer_Release(dsbFx[i]);
fxId[i] = 0;
}
}
}
}
*/
}
int32 BS2Sound::InitialiseSound(uint16 freq, uint16 channels, uint16 bitDepth)
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{
warning("stub InitaliseSound( %d, %d, %d )", freq, channels, bitDepth);
/*
int32 i;
HRESULT hrz;
DSBUFFERDESC dsbd;
WAVEFORMATEX pf;
hrz = DirectSoundCreate(NULL, &lpDS, NULL);
if (hrz != DS_OK)
return(RDERR_DSOUNDCREATE);
hrz = IDirectSound_SetCooperativeLevel(lpDS, hwnd, DSSCL_EXCLUSIVE);
if (hrz != DS_OK)
{
IDirectSound_Release(lpDS);
return(RDERR_DSOUNDCOOPERATE);
}
memset(&dsbd, 0, sizeof(DSBUFFERDESC));
dsbd.dwSize = sizeof(DSBUFFERDESC);
dsbd.dwFlags = DSBCAPS_PRIMARYBUFFER;
dsbd.lpwfxFormat = NULL;
hrz = IDirectSound_CreateSoundBuffer(lpDS, &dsbd, &dsbPrimary, NULL);
if (hrz != DS_OK)
{
IDirectSound_Release(lpDS);
return(RDERR_DSOUNDPBUFFER);
}
memset(&pf, 0, sizeof(WAVEFORMATEX));
pf.wFormatTag = WAVE_FORMAT_PCM;
pf.nSamplesPerSec = freq;
pf.nChannels = channels;
pf.wBitsPerSample = bitDepth;
pf.nBlockAlign = pf.wBitsPerSample * pf.nChannels >> 3;
pf.nAvgBytesPerSec = pf.nBlockAlign * pf.nSamplesPerSec;
pf.cbSize = 0;
hrz = IDirectSoundBuffer_SetFormat(dsbPrimary, (LPWAVEFORMATEX) &pf);
if (hrz != DS_OK)
{
// We have not been able to set the primary format to the format requested!!!
// But carry on anyway, the mixer will just have to work harder :)
}
// Clear the fx id's
for (i=0; i<MAXFX; i++)
fxId[i] = 0;
soundOn = 1;
//----------------------------------
// New initialisers (James19aug97)
memset (fxId, 0, MAXFX*sizeof(int32));
memset (fxCached, 0, MAXFX*sizeof(uint8));
memset (fxiPaused, 0, MAXFX*sizeof(uint8));
memset (fxLooped, 0, MAXFX*sizeof(uint8));
memset (musStreaming, 0, MAXFX*sizeof(int16));
memset (musicPaused, 0, MAXFX*sizeof(int16));
memset (musCounter, 0, MAXFX*sizeof(int16));
memset (musFading, 0, MAXFX*sizeof(int16));
memset (musLooping, 0, MAXFX*sizeof(int16));
memset (fpMus, 0, MAXFX*sizeof(FILE*));
memset (streamCursor, 0, MAXFX*sizeof(int32));
memset (musFilePos, 0, MAXFX*sizeof(int32));
memset (musEnd, 0, MAXFX*sizeof(int32));
memset (musLastSample, 0, MAXFX*sizeof(int16));
memset (musId, 0, MAXFX*sizeof(uint32));
*/
return(RD_OK);
}
int32 BS2Sound::PlaySpeech(uint8 *data, uint8 vol, int8 pan)
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{
warning("stub PlaySpeech");
/*
uint32 dwBytes1, dwBytes2;
int32 i;
uint32 *data32;
void *lpv1, *lpv2;
_wavHeader *wav;
HRESULT hr;
DSBUFFERDESC dsbd;
PCMWAVEFORMAT wf;
wav = (_wavHeader *) data;
if (soundOn)
{
if (speechStatus)
return(RDERR_SPEECHPLAYING);
memset(&wf, 0, sizeof(PCMWAVEFORMAT));
wf.wf.wFormatTag = WAVE_FORMAT_PCM;
wf.wf.nChannels = wav->channels;
wf.wf.nSamplesPerSec = wav->samplesPerSec;
wf.wBitsPerSample = 8 * wav->blockAlign / (wav->samplesPerSec * wav->channels);
wf.wf.nBlockAlign = wf.wf.nChannels * wf.wBitsPerSample / 8;
wf.wf.nAvgBytesPerSec = wf.wf.nSamplesPerSec * wf.wf.nBlockAlign;
memset(&dsbd, 0, sizeof(DSBUFFERDESC));
dsbd.dwSize = sizeof(DSBUFFERDESC);
// dsbd.dwFlags = DSBCAPS_CTRLDEFAULT;
dsbd.lpwfxFormat = (LPWAVEFORMATEX) &wf;
// Set the sample size - search for the size of the data.
i = 0;
while (i<100)
{
if (*data == 'd')
{
data32 = (int32 *) data;
if (*data32 == 'atad')
break;
}
i += 1;
data++;
}
if (i == 100)
return(RDERR_INVALIDWAV);
dsbd.dwBufferBytes = *(data32 + 1);
// Create the speech sample buffer
hr = IDirectSound_CreateSoundBuffer(lpDS, &dsbd, &dsbSpeech, NULL);
if (hr != DS_OK)
return(RDERR_CREATESOUNDBUFFER);
// Lock the speech buffer, ready to fill it with data
hr = IDirectSoundBuffer_Lock(dsbSpeech, 0, dsbd.dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
if (hr == DSERR_BUFFERLOST)
{
IDirectSoundBuffer_Restore(dsbSpeech);
hr = IDirectSoundBuffer_Lock(dsbSpeech, 0, dsbd.dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
}
if (hr == DS_OK)
{
// Fill the speech buffer with data
memcpy((uint8 *) lpv1, (uint8 *) (data32 + 2), dwBytes1);
if (dwBytes1 != dsbd.dwBufferBytes)
{
memcpy((uint8 *) lpv1 + dwBytes1, (uint8 *) (data32 + 2) + dwBytes1, dwBytes2);
}
// Unlock the buffer now that we've filled it
IDirectSoundBuffer_Unlock(dsbSpeech, lpv1, dwBytes1, lpv2, dwBytes2);
// Modify the volume according to the master volume
if (speechMuted)
IDirectSoundBuffer_SetVolume(dsbSpeech, volTable[0]);
else
IDirectSoundBuffer_SetVolume(dsbSpeech, volTable[vol*speechVol]);
IDirectSoundBuffer_SetPan(dsbSpeech, panTable[pan+16]);
// Start the speech playing
IDirectSoundBuffer_Play(dsbSpeech, 0, 0, 0);
speechStatus = 1;
}
else
{
IDirectSoundBuffer_Release(dsbSpeech);
return(RDERR_LOCKSPEECHBUFFER);
}
}
*/
return(RD_OK);
}
int32 BS2Sound::AmISpeaking()
2003-07-28 01:47:41 +00:00
{
warning("stub AmISpeaking");
/*
int32 len;
// int32 status;
int32 readCursor, writeCursor;
int32 dwBytes1, dwBytes2;
int16 *sample;
int32 count = 0;
LPVOID lpv1, lpv2;
HRESULT hr;
#define POSITIVE_THRESHOLD 350
#define NEGATIVE_THRESHOLD -350
if ((!speechMuted) && (!speechPaused) && (dsbSpeech))
{
if (IDirectSoundBuffer_GetCurrentPosition(dsbSpeech, &readCursor, &writeCursor) != DS_OK)
{
return (RDSE_SPEAKING);
}
len = 44100 / 12;
hr = IDirectSoundBuffer_Lock(dsbSpeech, readCursor, len, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
if (hr == DS_OK)
{
for (sample = (int16*)lpv1; sample<(int16*)((int8*)lpv1+dwBytes1); sample+= 90) // 20 samples
if (*sample>POSITIVE_THRESHOLD || *sample<NEGATIVE_THRESHOLD)
count++;
IDirectSoundBuffer_Unlock(dsbSpeech,lpv1,dwBytes1,lpv2,dwBytes2);
if (count>5) // 25% of the samples
return (RDSE_SPEAKING);
}
return (RDSE_QUIET);
}
return (RDSE_SPEAKING);
*/
return RDSE_QUIET;
}
int32 BS2Sound::GetCompSpeechSize(const char *filename, uint32 speechid)
2003-07-28 01:47:41 +00:00
{
int32 i;
uint32 speechIndex[2];
FILE *fp;
// Open the speech cluster and find the data offset & size
fp = fopen(filename, "rb");
if (fp == NULL)
return(0);
if (fseek(fp, (++speechid)*8, SEEK_SET))
{
fclose(fp);
return (0);
}
if (fread(speechIndex, sizeof(uint32), 2, fp) != 2)
{
fclose(fp);
return (0);
}
if (!speechIndex[0] || !speechIndex[1])
{
fclose(fp);
return (0);
}
fclose(fp);
i = (speechIndex[1]-1)*2 + sizeof(_wavHeader) + 8;
return(i);
}
int32 BS2Sound::PreFetchCompSpeech(const char *filename, uint32 speechid, uint8 *waveMem)
2003-07-28 01:47:41 +00:00
{
uint32 i;
uint16 *data16;
uint8 *data8;
uint32 speechIndex[2];
_wavHeader *pwf = (_wavHeader *) waveMem;
FILE *fp;
// Open the speech cluster and find the data offset & size
fp = fopen(filename, "rb");
if (fp == NULL)
return(RDERR_INVALIDFILENAME);
if (fseek(fp, (++speechid)*8, SEEK_SET))
{
fclose(fp);
return (RDERR_READERROR);
}
if (fread(speechIndex, sizeof(uint32), 2, fp) != 2)
{
fclose(fp);
return (RDERR_READERROR);
}
if (!speechIndex[0] || !speechIndex[1])
{
fclose(fp);
return (RDERR_INVALIDID);
}
data16 = (uint16*)(waveMem + sizeof(_wavHeader));
memset(pwf, 0, sizeof(_wavHeader));
*((uint32*)pwf->riff) = 'FFIR';
*((uint32*)pwf->wavID) = 'EVAW';
*((uint32*)pwf->format) = ' tmf';
pwf->formatLen = 0x00000010;
pwf->formatTag = 0x0001;
pwf->channels = 0x0001;
pwf->samplesPerSec = 0x5622;
pwf->avgBytesPerSec = 0x0000;
pwf->blockAlign = 0xAC44;
pwf->unknown1 = 0x0000;
pwf->unknown2 = 0x0002;
pwf->bitsPerSample = 0x0010;
*((uint32*)data16) = 'atad';
data16 += 2;
*((uint32*)data16) = (speechIndex[1]-1)*2;
data16 += 2;
pwf->fileLength = (speechIndex[1]-1)*2 + sizeof(_wavHeader) + 8;
// Calculate position in buffer to load compressed sound into
data8 = (uint8*)data16 + (speechIndex[1]-1);
if (fseek(fp, speechIndex[0], SEEK_SET))
{
fclose(fp);
return (RDERR_INVALIDID);
}
if (fread(data8, sizeof(uint8), speechIndex[1], fp) != speechIndex[1])
{
fclose(fp);
return (RDERR_INVALIDID);
}
fclose(fp);
data16[0] = *((int16*)data8); // Starting Value
i=1;
while (i<(speechIndex[1]-1))
{
if (GetCompressedSign(data8[i+1]))
data16[i] = data16[i-1] - (GetCompressedAmplitude(data8[i+1])<<GetCompressedShift(data8[i+1]));
else
data16[i] = data16[i-1] + (GetCompressedAmplitude(data8[i+1])<<GetCompressedShift(data8[i+1]));
i++;
}
return(RD_OK);
}
int32 BS2Sound::PlayCompSpeech(const char *filename, uint32 speechid, uint8 vol, int8 pan)
2003-07-28 01:47:41 +00:00
{
warning("stub PlayCompSpeech( %s, %d, %d, %d )", filename, speechid, vol, pan);
/*
uint32 dwBytes1, dwBytes2;
uint32 i;
uint16 *data16;
uint8 *data8;
uint32 speechIndex[2];
void *lpv1, *lpv2;
HRESULT hr;
DSBUFFERDESC dsbd;
PCMWAVEFORMAT wf;
FILE *fp;
if (!speechMuted)
{
if (GetSpeechStatus() == RDERR_SPEECHPLAYING)
return RDERR_SPEECHPLAYING;
memset(&wf, 0, sizeof(PCMWAVEFORMAT));
wf.wf.wFormatTag = WAVE_FORMAT_PCM;
wf.wf.nChannels = 1;
wf.wf.nSamplesPerSec = 22050;
wf.wBitsPerSample = 16;
wf.wf.nBlockAlign = 2;
wf.wf.nAvgBytesPerSec = 44100;
memset(&dsbd, 0, sizeof(DSBUFFERDESC));
dsbd.dwSize = sizeof(DSBUFFERDESC);
// dsbd.dwFlags = DSBCAPS_CTRLDEFAULT;
dsbd.lpwfxFormat = (LPWAVEFORMATEX) &wf;
// Open the speech cluster and find the data offset & size
fp = fopen(filename, "rb");
if (fp == NULL)
return(RDERR_INVALIDFILENAME);
if (fseek(fp, (++speechid)*8, SEEK_SET))
{
fclose(fp);
return (RDERR_READERROR);
}
if (fread(speechIndex, sizeof(uint32), 2, fp) != 2)
{
fclose(fp);
return (RDERR_READERROR);
}
if (speechIndex[0]==0 || speechIndex[1]==0)
{
fclose(fp);
return (RDERR_INVALIDID);
}
dsbd.dwBufferBytes = (speechIndex[1]-1)*2;
// Create tempory buffer for compressed speech
if ((data8 = malloc(speechIndex[1])) == NULL)
{
fclose(fp);
return(RDERR_OUTOFMEMORY);
}
if (fseek(fp, speechIndex[0], SEEK_SET))
{
fclose(fp);
free(data8);
return (RDERR_INVALIDID);
}
if (fread(data8, sizeof(uint8), speechIndex[1], fp) != speechIndex[1])
{
fclose(fp);
free(data8);
return (RDERR_INVALIDID);
}
fclose(fp);
// Create the speech sample buffer
hr = IDirectSound_CreateSoundBuffer(lpDS, &dsbd, &dsbSpeech, NULL);
if (hr != DS_OK)
{
free(data8);
return(RDERR_CREATESOUNDBUFFER);
}
// Lock the speech buffer, ready to fill it with data
hr = IDirectSoundBuffer_Lock(dsbSpeech, 0, dsbd.dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
if (hr == DSERR_BUFFERLOST)
{
IDirectSoundBuffer_Restore(dsbSpeech);
hr = IDirectSoundBuffer_Lock(dsbSpeech, 0, dsbd.dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
}
if (hr == DS_OK)
{
// decompress data into speech buffer.
data16 = (uint16*)lpv1;
data16[0] = *((int16*)data8); // Starting Value
i=1;
while (i<dwBytes1/2)
{
if (GetCompressedSign(data8[i+1]))
data16[i] = data16[i-1] - (GetCompressedAmplitude(data8[i+1])<<GetCompressedShift(data8[i+1]));
else
data16[i] = data16[i-1] + (GetCompressedAmplitude(data8[i+1])<<GetCompressedShift(data8[i+1]));
i++;
}
if (dwBytes1 != dsbd.dwBufferBytes)
{
while (i<(dwBytes1+dwBytes2)/2)
{
if (GetCompressedSign(data8[i+1]))
data16[i] = data16[i-1] - (GetCompressedAmplitude(data8[i+1])<<GetCompressedShift(data8[i+1]));
else
data16[i] = data16[i-1] + (GetCompressedAmplitude(data8[i+1])<<GetCompressedShift(data8[i+1]));
i++;
}
}
free(data8);
// Unlock the buffer now that we've filled it
IDirectSoundBuffer_Unlock(dsbSpeech, lpv1, dwBytes1, lpv2, dwBytes2);
// Modify the volume according to the master volume
if (speechMuted)
IDirectSoundBuffer_SetVolume(dsbSpeech, volTable[0]);
else
IDirectSoundBuffer_SetVolume(dsbSpeech, volTable[vol*speechVol]);
IDirectSoundBuffer_SetPan(dsbSpeech, panTable[pan+16]);
// Start the speech playing
speechPaused = 1;
// IDirectSoundBuffer_Play(dsbSpeech, 0, 0, 0);
speechStatus = 1;
}
else
{
IDirectSoundBuffer_Release(dsbSpeech);
free(data8);
return(RDERR_LOCKSPEECHBUFFER);
}
}
DipMusic();
*/
return(RD_OK);
}
int32 BS2Sound::StopSpeechBS2(void)
2003-07-28 01:47:41 +00:00
{
warning("stub StopSpeechBS2");
2003-07-28 01:47:41 +00:00
/*
int32 status;
if (!soundOn)
return(RD_OK);
if (speechStatus)
{
IDirectSoundBuffer_GetStatus(dsbSpeech, &status);
if (status & DSBSTATUS_PLAYING)
{
IDirectSoundBuffer_Stop(dsbSpeech);
// SetMusicVolume(GetMusicVolume());
}
IDirectSoundBuffer_Release(dsbSpeech);
dsbSpeech = 0;
speechStatus = 0;
return(RD_OK);
}
*/
return(RDERR_SPEECHNOTPLAYING);
}
int32 BS2Sound::GetSpeechStatus(void)
2003-07-28 01:47:41 +00:00
{
warning("stub GetSpeechStatus");
/*
int32 status;
if ((!soundOn) || (!speechStatus))
return(RDSE_SAMPLEFINISHED);
if (speechPaused)
return(RDSE_SAMPLEPLAYING);
IDirectSoundBuffer_GetStatus(dsbSpeech, &status);
if (!(status & DSBSTATUS_PLAYING))
{
speechStatus = 0;
IDirectSoundBuffer_Release(dsbSpeech);
dsbSpeech = 0;
// SetMusicVolume(GetMusicVolume());
return(RDSE_SAMPLEFINISHED);
}
return(RDSE_SAMPLEPLAYING);
*/
return RDSE_SAMPLEFINISHED;
}
void BS2Sound::SetSpeechVolume(uint8 volume)
2003-07-28 01:47:41 +00:00
{
warning("stub SetSpeechVolume");
/*
speechVol = volume;
if (dsbSpeech && !speechMuted && GetSpeechStatus() == RDSE_SAMPLEPLAYING)
IDirectSoundBuffer_SetVolume(dsbSpeech, volTable[16*speechVol]);
*/
}
uint8 BS2Sound::GetSpeechVolume()
2003-07-28 01:47:41 +00:00
{
return speechVol;
}
void BS2Sound::MuteSpeech(uint8 mute)
2003-07-28 01:47:41 +00:00
{
warning("stub MuteSpeech( %d )", mute);
/*
speechMuted = mute;
if (GetSpeechStatus() == RDSE_SAMPLEPLAYING)
{
if (mute)
IDirectSoundBuffer_SetVolume(dsbSpeech, volTable[0]);
else
IDirectSoundBuffer_SetVolume(dsbSpeech, volTable[16*speechVol]);
}
*/
}
uint8 BS2Sound::IsSpeechMute(void)
2003-07-28 01:47:41 +00:00
{
return (speechMuted);
}
int32 BS2Sound::PauseSpeech(void)
2003-07-28 01:47:41 +00:00
{
warning("PauseSpeech");
/*
if (GetSpeechStatus() == RDSE_SAMPLEPLAYING)
{
speechPaused = 1;
return (IDirectSoundBuffer_Stop(dsbSpeech));
}
*/
return(RD_OK);
}
int32 BS2Sound::UnpauseSpeech(void)
2003-07-28 01:47:41 +00:00
{
warning("UnpauseSpeech");
/*
if (speechPaused)
{
speechPaused = 0;
return (IDirectSoundBuffer_Play(dsbSpeech, 0, 0, 0));
}
*/
return(RD_OK);
}
int32 BS2Sound::OpenFx(int32 id, uint8 *data)
2003-07-28 01:47:41 +00:00
{
warning("stub OpenFx( %d )", id);
/*
uint32 dwBytes1, dwBytes2;
int32 i, fxi;
uint32 *data32;
void *lpv1, *lpv2;
_wavHeader *wav;
HRESULT hr;
DSBUFFERDESC dsbd;
PCMWAVEFORMAT wf;
wav = (_wavHeader *) data;
if (soundOn)
{
// Check for a valid id.
if (id == 0)
return(RDERR_INVALIDID);
// Check that the fx is not already open
for (i=0; i<MAXFX; i++)
if (fxId[i] == id)
return(RDERR_FXALREADYOPEN);
// Now choose a free slot for the fx
fxi = 0;
while (fxi<MAXFX)
{
if (fxId[fxi] == 0)
break;
fxi++;
}
if (fxi == MAXFX)
return(RDERR_NOFREEBUFFERS);
memset(&wf, 0, sizeof(PCMWAVEFORMAT));
wf.wf.wFormatTag = WAVE_FORMAT_PCM;
wf.wf.nChannels = wav->channels;
wf.wf.nSamplesPerSec = wav->samplesPerSec;
wf.wBitsPerSample = 8 * wav->blockAlign / (wav->samplesPerSec * wav->channels);
wf.wf.nBlockAlign = wf.wf.nChannels * wf.wBitsPerSample / 8;
wf.wf.nAvgBytesPerSec = wf.wf.nSamplesPerSec * wf.wf.nBlockAlign;
memset(&dsbd, 0, sizeof(DSBUFFERDESC));
dsbd.dwSize = sizeof(DSBUFFERDESC);
// dsbd.dwFlags = DSBCAPS_CTRLDEFAULT;
dsbd.lpwfxFormat = (LPWAVEFORMATEX) &wf;
// Set the sample size - search for the size of the data.
i = 0;
while (i<100)
{
if (*data == 'd')
{
data32 = (int32 *) data;
if (*data32 == 'atad')
break;
}
i += 1;
data++;
}
if (i == 100)
return(RDERR_INVALIDWAV);
dsbd.dwBufferBytes = *(data32 + 1);
// Create the speech sample buffer
hr = IDirectSound_CreateSoundBuffer(lpDS, &dsbd, &dsbFx[fxi], NULL);
if (hr != DS_OK)
return(RDERR_CREATESOUNDBUFFER);
// Lock the speech buffer, ready to fill it with data
hr = IDirectSoundBuffer_Lock(dsbFx[fxi], 0, dsbd.dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
if (hr == DSERR_BUFFERLOST)
{
IDirectSoundBuffer_Restore(dsbFx[fxi]);
hr = IDirectSoundBuffer_Lock(dsbFx[fxi], 0, dsbd.dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
}
if (hr == DS_OK)
{
// Fill the speech buffer with data
memcpy((uint8 *) lpv1, (uint8 *) (data32 + 2), dwBytes1);
if (dwBytes1 != dsbd.dwBufferBytes)
{
memcpy((uint8 *) lpv1 + dwBytes1, (uint8 *) (data32 + 2) + dwBytes1, dwBytes2);
}
// Unlock the buffer now that we've filled it
IDirectSoundBuffer_Unlock(dsbFx[fxi], lpv1, dwBytes1, lpv2, dwBytes2);
}
else
{
IDirectSoundBuffer_Release(dsbFx[fxi]);
return(RDERR_LOCKSPEECHBUFFER);
}
fxId[fxi] = id;
fxCached[fxi] = RDSE_FXCACHED;
}
*/
return(RD_OK);
}
int32 BS2Sound::PlayFx(int32 id, uint8 *data, uint8 vol, int8 pan, uint8 type)
2003-07-28 01:47:41 +00:00
{
warning("stub PlayFx( %d, %d, %d, %d )", id, vol, pan, type);
/*
int32 i, loop;
HRESULT hr;
if (type == RDSE_FXLOOP)
loop = DSBPLAY_LOOPING;
else
loop = 0;
if (soundOn)
{
if (data == NULL)
{
if (type == RDSE_FXLEADOUT)
{
id = 0xffffffff;
i = GetFxIndex(id);
if (i == MAXFX)
return(RDERR_FXNOTOPEN);
fxLooped[i] = 0;
// Start the sound effect playing
if (musicMuted)
IDirectSoundBuffer_SetVolume(dsbFx[i], volTable[0]);
else
IDirectSoundBuffer_SetVolume(dsbFx[i], musicVolTable[volMusic[0]]);
IDirectSoundBuffer_SetPan(dsbFx[i], 0);
IDirectSoundBuffer_Play(dsbFx[i], 0, 0, 0);
fxCached[i] = RDSE_FXTOCLEAR;
}
else
{
i = GetFxIndex(id);
if (i == MAXFX)
return(RDERR_FXNOTOPEN);
fxLooped[i] = loop;
fxVolume[i] = vol;
// Start the sound effect playing
if (fxMuted)
IDirectSoundBuffer_SetVolume(dsbFx[i], volTable[0]);
else
IDirectSoundBuffer_SetVolume(dsbFx[i], volTable[vol*fxVol]);
IDirectSoundBuffer_SetPan(dsbFx[i], panTable[pan+16]);
IDirectSoundBuffer_Play(dsbFx[i], 0, 0, loop);
if (id == 0xffffffff)
fxCached[i] = RDSE_FXTOCLEAR;
}
}
else
{
if (type == RDSE_FXLEADIN)
{
id = 0xfffffffe;
hr = OpenFx(id, data);
if (hr != RD_OK)
return hr;
i = GetFxIndex(id);
if (i == MAXFX)
return RDERR_FXFUCKED;
fxCached[i] = RDSE_FXTOCLEAR;
if (musicMuted)
IDirectSoundBuffer_SetVolume(dsbFx[i], volTable[0]);
else
IDirectSoundBuffer_SetVolume(dsbFx[i], musicVolTable[volMusic[0]]);
IDirectSoundBuffer_SetPan(dsbFx[i], 0);
IDirectSoundBuffer_Play(dsbFx[i], 0, 0, 0);
}
else
{
hr = OpenFx(id, data);
if (hr != RD_OK)
return(hr);
i = GetFxIndex(id);
if (i == MAXFX)
return(RDERR_FXFUCKED);
fxCached[i] = RDSE_FXTOCLEAR;
fxLooped[i] = loop;
fxVolume[i] = vol;
// Start the sound effect playing
if (fxMuted)
IDirectSoundBuffer_SetVolume(dsbFx[i], volTable[0]);
else
IDirectSoundBuffer_SetVolume(dsbFx[i], volTable[vol*fxVol]);
IDirectSoundBuffer_SetPan(dsbFx[i], panTable[pan+16]);
IDirectSoundBuffer_Play(dsbFx[i], 0, 0, loop);
}
}
}
*/
return(RD_OK);
}
int32 BS2Sound::SetFxVolumePan(int32 id, uint8 vol, int8 pan)
2003-07-28 01:47:41 +00:00
{
warning("stub SetFxVolumePan( %d, %d, %d )", id, vol, pan);
/*
int32 i = GetFxIndex(id);
if (i == MAXFX)
return RDERR_FXNOTOPEN;
fxVolume[i] = vol;
if (!fxMuted)
IDirectSoundBuffer_SetVolume(dsbFx[i], volTable[vol*fxVol]);
IDirectSoundBuffer_SetPan(dsbFx[i], panTable[pan+16]);
*/
return RD_OK;
}
int32 BS2Sound::SetFxIdVolume(int32 id, uint8 vol)
2003-07-28 01:47:41 +00:00
{
warning("stub SetFxIdVolume( %d, %d )", id, vol);
/*
int32 i = GetFxIndex(id);
if (i == MAXFX)
return RDERR_FXNOTOPEN;
fxVolume[i] = vol;
if (!fxMuted)
IDirectSoundBuffer_SetVolume(dsbFx[i], volTable[vol*fxVol]);
*/
return RD_OK;
}
int32 BS2Sound::ClearAllFx(void)
2003-07-28 01:47:41 +00:00
{
warning("stub ClearAllFx");
/*
int32 status;
int32 i;
if (!soundOn)
return(RD_OK);
i = 0;
while (i < MAXFX)
{
if ((fxId[i]) && (fxId[i] != 0xfffffffe) && (fxId[i] != 0xffffffff))
{
IDirectSoundBuffer_GetStatus(dsbFx[i], &status);
if (status & (DSBSTATUS_PLAYING + DSBSTATUS_LOOPING))
{
IDirectSoundBuffer_Stop(dsbFx[i]);
}
IDirectSoundBuffer_Release(dsbFx[i]);
fxId[i] = 0;
fxiPaused[i] = 0;
}
i++;
}
*/
return(RD_OK);
}
int32 BS2Sound::CloseFx(int32 id)
2003-07-28 01:47:41 +00:00
{
warning("stub CloseFx( %d )", id);
/*
int32 i;
int32 status;
if (!soundOn)
return(RD_OK);
i = GetFxIndex(id);
if (i<MAXFX)
{
IDirectSoundBuffer_GetStatus(dsbFx[i], &status);
if (status & (DSBSTATUS_PLAYING + DSBSTATUS_LOOPING))
{
IDirectSoundBuffer_Stop(dsbFx[i]);
}
IDirectSoundBuffer_Release(dsbFx[i]);
fxId[i] = 0;
fxiPaused[i] = 0;
}
*/
return(RD_OK);
}
int32 BS2Sound::PauseFx(void)
2003-07-28 01:47:41 +00:00
{
warning("stub PauseFx");
/*
int32 i;
int32 status;
if (!fxPaused)
{
for (i=0; i<MAXFX; i++)
{
if (fxId[i])
{
IDirectSoundBuffer_GetStatus(dsbFx[i], &status);
if (status & (DSBSTATUS_PLAYING + DSBSTATUS_LOOPING))
{
fxiPaused[i] = 1;
if (IDirectSoundBuffer_Stop(dsbFx[i]) != RD_OK)
return(RDERR_FXFUCKED);
}
}
else
{
fxiPaused[i] = 0;
}
}
fxPaused = 1;
}
*/
return (RD_OK);
}
int32 BS2Sound::PauseFxForSequence(void)
2003-07-28 01:47:41 +00:00
{
warning("stub PauseFxForSequence");
/*
int32 i;
int32 status;
if (!fxPaused)
{
for (i=0; i<MAXFX; i++)
{
if ((fxId[i]) && (fxId[i] != 0xfffffffe))
{
IDirectSoundBuffer_GetStatus(dsbFx[i], &status);
if (status & (DSBSTATUS_PLAYING + DSBSTATUS_LOOPING))
{
fxiPaused[i] = 1;
IDirectSoundBuffer_Stop(dsbFx[i]);
}
}
else
{
fxiPaused[i] = 0;
}
}
fxPaused = 1;
}
*/
return (RD_OK);
}
int32 BS2Sound::UnpauseFx(void)
2003-07-28 01:47:41 +00:00
{
warning("stub UnpauseFx");
/*
int32 i;
if (fxPaused)
{
for (i=0; i<MAXFX; i++)
{
if (fxiPaused[i] && fxId[i])
{
if (IDirectSoundBuffer_Play(dsbFx[i], 0, 0, fxLooped[i]) != RD_OK)
return(RDERR_FXFUCKED);
}
}
fxPaused = 0;
}
*/
return (RD_OK);
}
uint8 BS2Sound::GetFxVolume()
2003-07-28 01:47:41 +00:00
{
return fxVol;
}
void BS2Sound::SetFxVolume(uint8 volume)
2003-07-28 01:47:41 +00:00
{
warning("stub SetFxVolume( %d )", volume);
/*
int32 fxi;
fxVol = volume;
// Now update the volume of any fxs playing
for (fxi = 0; fxi<MAXFX; fxi++)
{
if (fxId[fxi] && !fxMuted)
IDirectSoundBuffer_SetVolume(dsbFx[fxi], volTable[fxVolume[fxi]*fxVol]);
}
*/
}
void BS2Sound::MuteFx(uint8 mute)
2003-07-28 01:47:41 +00:00
{
warning("stub MuteFx( %d )");
/*
int32 fxi;
fxMuted = mute;
// Now update the volume of any fxs playing
for (fxi = 0; fxi<MAXFX; fxi++)
{
if (fxId[fxi])
{
if (mute)
IDirectSoundBuffer_SetVolume(dsbFx[fxi], volTable[0]);
else
IDirectSoundBuffer_SetVolume(dsbFx[fxi], volTable[fxVolume[fxi]*fxVol]);
}
}
*/
}
uint8 BS2Sound::IsFxMute(void)
2003-07-28 01:47:41 +00:00
{
return (fxMuted);
}
void BS2Sound::StartMusicFadeDown(int i)
2003-07-28 01:47:41 +00:00
{
// IDirectSoundBuffer_Stop(lpDsbMus[i]);
// IDirectSoundBuffer_Release(lpDsbMus[i]);
musFading[i] = -16;
// musStreaming[i] = 0;
fclose(fpMus[i]);
}
int32 BS2Sound::StreamMusic(uint8 *filename, int32 looping)
2003-07-28 01:47:41 +00:00
{
warning("stub StreamMusic( %s, %d )", filename, looping);
/*
HRESULT hr;
LPVOID lpv1, lpv2;
DWORD dwBytes1, dwBytes2;
int32 i;
int32 v0, v1;
int32 bytes;
_wavHeader head;
// Do not allow compressed and uncompressed music to be streamed at the same time.
if (compressedMusic == 1)
return (RDERR_FXFUCKED);
compressedMusic = 2;
if (musStreaming[0] + musStreaming[1] == 0)
{
i = 0;
fpMus[i] = fopen(filename, "rb");
if (fpMus[i] == NULL)
return(RDERR_INVALIDFILENAME);
fread(&head, sizeof(_wavHeader), 1, fpMus[i]);
streamCursor[i] = 0;
musLooping[i] = looping;
memset(&wfMus[i], 0, sizeof(PCMWAVEFORMAT));
wfMus[i].wf.wFormatTag = WAVE_FORMAT_PCM;
wfMus[i].wf.nChannels = head.channels;
wfMus[i].wf.nSamplesPerSec = head.samplesPerSec;
wfMus[i].wBitsPerSample = 8 * head.blockAlign / (head.samplesPerSec * head.channels);
wfMus[i].wf.nBlockAlign = wfMus[i].wf.nChannels * wfMus[i].wBitsPerSample / 8;
wfMus[i].wf.nAvgBytesPerSec = wfMus[i].wf.nSamplesPerSec * wfMus[i].wf.nBlockAlign;
// Reset the sample format and size
memset(&dsbdMus[i], 0, sizeof(DSBUFFERDESC));
dsbdMus[i].dwSize = sizeof(DSBUFFERDESC);
// dsbdMus[i].dwFlags = DSBCAPS_CTRLDEFAULT;
dsbdMus[i].dwBufferBytes = 3 * wfMus[i].wf.nAvgBytesPerSec; // 3 seconds
dsbdMus[i].lpwfxFormat = (LPWAVEFORMATEX) &wfMus[i];
// Create the sound effect sample buffer
hr = IDirectSound_CreateSoundBuffer(lpDS, &dsbdMus[i], &lpDsbMus[i], NULL);
if (hr == DS_OK)
{
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], 0, dsbdMus[i].dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
if (hr == DSERR_BUFFERLOST)
{
IDirectSoundBuffer_Restore(lpDsbMus[i]);
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], 0, dsbdMus[i].dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
}
if (hr == DS_OK)
{
// Fill the speech buffer with data
bytes = fread(lpv1, 1, dwBytes1, fpMus[i]);
// memcpy((uint8 *) lpv1, (uint8 *) wavData + sizeof(wavHeader), dwBytes1);
// Unlock the buffer now that we've filled it
IDirectSoundBuffer_Unlock(lpDsbMus[i], lpv1, dwBytes1, lpv2, dwBytes2);
// Modify the volume according to the master volume and music mute state
if (musicMuted)
v0 = v1 = 0;
else
{
v0 = volMusic[0];
v1 = volMusic[1];
}
if (v0 > v1)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v0]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], musicVolTable[v1*16/v0]);
}
else
{
if (v1 > v0)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], -musicVolTable[v0*16/v1]);
}
else
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], 0);
}
}
// Start the sound effect playing
IDirectSoundBuffer_Play(lpDsbMus[i], 0, 0, DSBPLAY_LOOPING);
musStreaming[i] = 1;
musCounter[i] = 250;
strcpy(musFilename[i], filename);
// and exit the function.
}
else
{
// Pdebug("Failed to lock sound buffer upon creation - (%d)", hr & 0x0000ffff);
// DirectSoundDebug("Error - ", hr);
fclose(fpMus[i]);
return(RDERR_LOCKFAILED);
}
}
else
{
// Pdebug("Failed to create sound buffer - (%d)", hr & 0x0000ffff);
// Pdebug("Error - ", hr);
fclose(fpMus[i]);
return(RDERR_CREATESOUNDBUFFER);
}
}
else if (musStreaming[0] + musStreaming[1] == 1)
{
i = musStreaming[0];
musLooping[i] = looping;
if (!musFading[1-i])
StartMusicFadeDown(1 - i);
fpMus[i] = fopen(filename, "rb");
if (fpMus[i] == NULL)
return(RDERR_INVALIDFILENAME);
fread(&head, sizeof(_wavHeader), 1, fpMus[i]);
streamCursor[i] = 0;
memset(&wfMus[i], 0, sizeof(PCMWAVEFORMAT));
wfMus[i].wf.wFormatTag = WAVE_FORMAT_PCM;
wfMus[i].wf.nChannels = head.channels;
wfMus[i].wf.nSamplesPerSec = head.samplesPerSec;
wfMus[i].wBitsPerSample = 8 * head.blockAlign / (head.samplesPerSec * head.channels);
wfMus[i].wf.nBlockAlign = wfMus[i].wf.nChannels * wfMus[i].wBitsPerSample / 8;
wfMus[i].wf.nAvgBytesPerSec = wfMus[i].wf.nSamplesPerSec * wfMus[i].wf.nBlockAlign;
// Reset the sample format and size
memset(&dsbdMus[i], 0, sizeof(DSBUFFERDESC));
dsbdMus[i].dwSize = sizeof(DSBUFFERDESC);
// dsbdMus[i].dwFlags = DSBCAPS_CTRLDEFAULT;
dsbdMus[i].dwBufferBytes = 6 * wfMus[i].wf.nAvgBytesPerSec; // 3 seconds
dsbdMus[i].lpwfxFormat = (LPWAVEFORMATEX) &wfMus[i];
// Create the sound effect sample buffer
hr = IDirectSound_CreateSoundBuffer(lpDS, &dsbdMus[i], &lpDsbMus[i], NULL);
if (hr == DS_OK)
{
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], 0, dsbdMus[i].dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
if (hr == DSERR_BUFFERLOST)
{
IDirectSoundBuffer_Restore(lpDsbMus[i]);
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], 0, dsbdMus[i].dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
}
if (hr == DS_OK)
{
// Fill the speech buffer with data
bytes = fread(lpv1, 1, dwBytes1, fpMus[i]);
// Pdebug("Read %d bytes\n", bytes);
// memcpy((uint8 *) lpv1, (uint8 *) wavData + sizeof(_wavHeader), dwBytes1);
// Unlock the buffer now that we've filled it
IDirectSoundBuffer_Unlock(lpDsbMus[i], lpv1, dwBytes1, lpv2, dwBytes2);
// Modify the volume according to the master volume and music mute state
if (musicMuted)
v0 = v1 = 0;
else
{
v0 = volMusic[0];
v1 = volMusic[1];
}
if (v0 > v1)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v0]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], musicVolTable[v1*16/v0]);
}
else
{
if (v1 > v0)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], -musicVolTable[v0*16/v1]);
}
else
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], 0);
}
}
// Start the sound effect playing
IDirectSoundBuffer_Play(lpDsbMus[i], 0, 0, DSBPLAY_LOOPING);
musStreaming[i] = 1;
musCounter[i] = 250;
strcpy(musFilename[i], filename);
}
else
{
// Pdebug("Failed to lock sound buffer upon creation - (%d)", hr & 0x0000ffff);
// DirectSoundDebug("Error - ", hr);
fclose(fpMus[i]);
return(RDERR_LOCKFAILED);
}
}
else
{
// Pdebug("Failed to create sound buffer - (%d)", hr & 0x0000ffff);
// Pdebug("Error - ", hr);
fclose(fpMus[i]);
return(RDERR_CREATESOUNDBUFFER);
}
}
*/
return(RD_OK);
}
void BS2Sound::UpdateSampleStreaming(void)
2003-07-28 01:47:41 +00:00
{
warning("stub UpdateSampleStreaming");
/*
int32 i;
int32 v0, v1;
int32 readLen;
int32 len;
int32 readCursor, writeCursor;
int32 dwBytes1, dwBytes2;
LPVOID lpv1, lpv2;
HRESULT hr;
for (i=0; i<MAXMUS; i++)
{
if (musStreaming[i])
{
if (musFading[i])
{
if (musFading[i] < 0)
{
if (++musFading[i] == 0)
{
IDirectSoundBuffer_Stop(lpDsbMus[i]);
IDirectSoundBuffer_Release(lpDsbMus[i]);
musStreaming[i] = 0;
}
else
{
// Modify the volume according to the master volume and music mute state
if (musicMuted)
v0 = v1 = 0;
else
{
v0 = (volMusic[0] * (0 - musFading[i]) / 16);
v1 = (volMusic[1] * (0 - musFading[i]) / 16);
}
if (v0 > v1)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v0]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], musicVolTable[v1*16/v0]);
}
else
{
if (v1 > v0)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], -musicVolTable[v0*16/v1]);
}
else
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], 0);
}
}
}
}
}
else
{
if (IDirectSoundBuffer_GetCurrentPosition(lpDsbMus[i], &readCursor, &writeCursor) != DS_OK)
{
// Pdebug ("Stopping sample %d cos cant get position", i);
IDirectSoundBuffer_Stop(lpDsbMus[i]);
}
len = readCursor - streamCursor[i];
if (len < 0)
{
len += dsbdMus[i].dwBufferBytes;
}
if (len > 0)
{
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], streamCursor[i], len, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
if (hr == DSERR_BUFFERLOST)
{
IDirectSoundBuffer_Restore(lpDsbMus[i]);
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], streamCursor[i], len, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
}
if (hr == DS_OK)
{
streamCursor[i] += len;
if (streamCursor[i] >= (int32) dsbdMus[i].dwBufferBytes)
streamCursor[i] -= dsbdMus[i].dwBufferBytes;
if (len > dwBytes1)
{
readLen = fread(lpv1, 1, dwBytes1, fpMus[i]);
if (readLen == dwBytes1)
{
readLen = fread(lpv2, 1, dwBytes2, fpMus[i]);
if (readLen != dwBytes2)
{
IDirectSoundBuffer_Unlock(lpDsbMus[i], lpv1, dwBytes1, lpv2, dwBytes2);
StartMusicFadeDown(i);
if (musLooping[i])
{
StreamMusic(musFilename[i], musLooping[i]);
}
}
else
{
IDirectSoundBuffer_Unlock(lpDsbMus[i], lpv1, dwBytes1, lpv2, dwBytes2);
}
}
else
{
IDirectSoundBuffer_Unlock(lpDsbMus[i], lpv1, dwBytes1, lpv2, dwBytes2);
StartMusicFadeDown(i);
if (musLooping[i])
{
StreamMusic(musFilename[i], musLooping[i]);
}
}
}
else
{
readLen = fread(lpv1, 1, len, fpMus[i]);
if (readLen != len)
{
IDirectSoundBuffer_Unlock(lpDsbMus[i], lpv1, dwBytes1, lpv2, dwBytes2);
StartMusicFadeDown(i);
if (musLooping[i])
{
StreamMusic(musFilename[i], musLooping[i]);
}
}
else
{
IDirectSoundBuffer_Unlock(lpDsbMus[i], lpv1, dwBytes1, lpv2, dwBytes2);
}
}
}
// else
// {
// DirectSoundDebug("Failed to lock sound buffer to write bytes", hr);
// Pdebug("Stream cursor %d", streamCursor[i]);
// Pdebug("len %d", len);
// }
}
//}
}
}
}
*/
}
int32 BS2Sound::StreamCompMusic(const char *filename, uint32 musicId, int32 looping)
2003-07-28 01:47:41 +00:00
{
warning("stub StreamCompMusic( %s, %d, %d )", filename, musicId, looping);
/*
HRESULT hr;
LPVOID lpv1, lpv2;
DWORD dwBytes1, dwBytes2;
uint32 i,j;
int32 v0, v1;
uint16 *data16;
uint8 *data8;
// Do not allow compressed and uncompressed music to be streamed at the same time.
if (compressedMusic == 2)
return (RDERR_FXFUCKED);
compressedMusic = 1;
if (musStreaming[0] + musStreaming[1] == 0) // No music streaming at present.
{
i = 0;
musLooping[i] = looping; // Save looping info
strcpy(musFilename[i], filename); // And tune id's
musId[i] = musicId;
if (IsMusicMute()) // Don't start streaming if the volume is off.
return (RD_OK);
if (!fpMus[0])
fpMus[0] = fopen(filename, "rb"); // Always use fpMus[0] (all music in one cluster) musFilePos[i] for different pieces of music.
if (fpMus[0] == NULL)
return(RDERR_INVALIDFILENAME);
if (fseek(fpMus[0], (musicId+1)*8, SEEK_SET)) // Seek to music index
{
fclose(fpMus[0]);
fpMus[0] = 0;
return (RDERR_READERROR);
}
if (fread(&musFilePos[i], sizeof(uint32), 1, fpMus[0]) != 1) // Read music index
{
fclose(fpMus[0]);
fpMus[0] = 0;
return (RDERR_READERROR);
}
if (fread(&musEnd[i], sizeof(uint32), 1, fpMus[0]) != 1) // Read music length
{
fclose(fpMus[0]);
fpMus[0] = 0;
return (RDERR_READERROR);
}
if (!musEnd[i] || !musFilePos[i]) // Check that music is valid (has length & offset)
{
fclose(fpMus[0]);
fpMus[0] = 0;
return (RDERR_INVALIDID);
}
musEnd[i] += musFilePos[i]; // Calculate the file position of the end of the music
streamCursor[i] = 0; // Reset streaming cursor and store looping flag
memset(&wfMus[i], 0, sizeof(PCMWAVEFORMAT)); // Set up wave format (no headers in cluster)
wfMus[i].wf.wFormatTag = WAVE_FORMAT_PCM;
wfMus[i].wf.nChannels = 1;
wfMus[i].wf.nSamplesPerSec = 22050;
wfMus[i].wBitsPerSample = 16;
wfMus[i].wf.nBlockAlign = 2;
wfMus[i].wf.nAvgBytesPerSec = 44100;
// Reset the sample format and size
memset(&dsbdMus[i], 0, sizeof(DSBUFFERDESC));
dsbdMus[i].dwSize = sizeof(DSBUFFERDESC);
// dsbdMus[i].dwFlags = DSBCAPS_CTRLDEFAULT;
dsbdMus[i].dwBufferBytes = 3 * wfMus[i].wf.nAvgBytesPerSec; // 3 seconds
dsbdMus[i].lpwfxFormat = (LPWAVEFORMATEX) &wfMus[i];
// Create a temporary buffer
if ((data8 = malloc(dsbdMus[i].dwBufferBytes/2)) == NULL) // Allocate a compressed data buffer
{
fclose(fpMus[0]);
fpMus[0] = 0;
return(RDERR_OUTOFMEMORY);
}
// Seek to start of the compressed music
if (fseek(fpMus[0], musFilePos[i], SEEK_SET))
{
fclose(fpMus[0]);
fpMus[0] = 0;
free(data8);
return (RDERR_INVALIDID);
}
// Read the compressed data in to the buffer
if (fread(data8, sizeof(uint8), dsbdMus[i].dwBufferBytes/2, fpMus[0]) != dsbdMus[i].dwBufferBytes/2)
{
fclose(fpMus[0]);
fpMus[0] = 0;
free(data8);
return (RDERR_INVALIDID);
}
// Store the current position in the file for future streaming
musFilePos[i] = ftell(fpMus[0]);
// Create the music buffer
hr = IDirectSound_CreateSoundBuffer(lpDS, &dsbdMus[i], &lpDsbMus[i], NULL);
if (hr == DS_OK)
{
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], 0, dsbdMus[i].dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
if (hr == DSERR_BUFFERLOST)
{
IDirectSoundBuffer_Restore(lpDsbMus[i]);
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], 0, dsbdMus[i].dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
}
if (hr == DS_OK)
{
// decompress the music into the music buffer.
data16 = (uint16*)lpv1;
data16[0] = *((int16*)data8); // First sample value
j=1;
while (j<(dwBytes1/2)-1)
{
if (GetCompressedSign(data8[j+1]))
data16[j] = data16[j-1] - (GetCompressedAmplitude(data8[j+1])<<GetCompressedShift(data8[j+1]));
else
data16[j] = data16[j-1] + (GetCompressedAmplitude(data8[j+1])<<GetCompressedShift(data8[j+1]));
j++;
}
// Never need to fill lpv2 because we started at the begining of the sound buffer
// Store the value of the last sample ready for next batch of decompression
musLastSample[i] = data16[j-1];
// Free the decompression buffer and unlock the buffer now that we've filled it
free(data8);
IDirectSoundBuffer_Unlock(lpDsbMus[i], lpv1, dwBytes1, lpv2, dwBytes2);
// Modify the volume according to the master volume and music mute state
if (musicMuted)
v0 = v1 = 0;
else
{
v0 = volMusic[0];
v1 = volMusic[1];
}
if (v0 > v1)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v0]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], musicVolTable[v1*16/v0]);
}
else
{
if (v1 > v0)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], -musicVolTable[v0*16/v1]);
}
else
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], 0);
}
}
// Start the sound effect playing
IDirectSoundBuffer_Play(lpDsbMus[i], 0, 0, DSBPLAY_LOOPING);
// Recorder some last variables
musStreaming[i] = 1;
musCounter[i] = 250;
// and exit the function.
}
else
{
// Opps Failed to lock the sound buffer
fclose(fpMus[0]);
fpMus[0] = 0;
return(RDERR_LOCKFAILED);
}
}
else
{
// Opps Failed to create the sound buffer
fclose(fpMus[0]);
fpMus[0] = 0;
return(RDERR_CREATESOUNDBUFFER);
}
}
else
{
if (musStreaming[0] + musStreaming[1] == 2) // Both streams in use, try to find a fading stream
{
if (musFading[0])
i = 0;
else
i = 1;
musFading[i] = 0;
IDirectSoundBuffer_Stop(lpDsbMus[i]);
IDirectSoundBuffer_Release(lpDsbMus[i]);
musStreaming[i] = 0;
}
if (musStreaming[0] + musStreaming[1] == 1) // Some music is already streaming
{
i = musStreaming[0]; // Set i to the free channel
musLooping[i] = looping; // Save looping info
strcpy(musFilename[i], filename); // And tune id's
musId[i] = musicId;
if (IsMusicMute()) // Don't start streaming if the volume is off.
return (RD_OK);
if (!fpMus[0])
fpMus[0] = fopen(filename, "rb"); // Always use fpMus[0] (all music in one cluster) musFilePos[i] for different pieces of music.
if (fpMus[0] == NULL)
return(RDERR_INVALIDFILENAME);
if (!musFading[1-i]) // Start other music stream fading out
musFading[1 - i] = -16;
streamCursor[i] = 0; // Reset the streaming cursor for this sample
if (fseek(fpMus[0], (musicId+1)*8, SEEK_SET)) // Seek to music index
{
fclose(fpMus[0]);
fpMus[0] = 0;
return (RDERR_READERROR);
}
if (fread(&musFilePos[i], sizeof(uint32), 1, fpMus[0]) != 1) // Read music index
{
fclose(fpMus[0]);
fpMus[0] = 0;
return (RDERR_READERROR);
}
if (fread(&musEnd[i], sizeof(uint32), 1, fpMus[0]) != 1) // Read music length
{
fclose(fpMus[0]);
fpMus[0] = 0;
return (RDERR_READERROR);
}
if (!musEnd[i] || !musFilePos[i]) // Check that music is valid (has length & offset)
{
fclose(fpMus[0]);
fpMus[0] = 0;
return (RDERR_INVALIDID);
}
musEnd[i] += musFilePos[i]; // Calculate the file position of the end of the music
memset(&wfMus[i], 0, sizeof(PCMWAVEFORMAT)); // Set up the music format info
wfMus[i].wf.wFormatTag = WAVE_FORMAT_PCM;
wfMus[i].wf.nChannels = 1;
wfMus[i].wf.nSamplesPerSec = 22050;
wfMus[i].wBitsPerSample = 16;
wfMus[i].wf.nBlockAlign = 2;
wfMus[i].wf.nAvgBytesPerSec = 44100;
// Reset the sample format and size
memset(&dsbdMus[i], 0, sizeof(DSBUFFERDESC));
dsbdMus[i].dwSize = sizeof(DSBUFFERDESC);
// dsbdMus[i].dwFlags = DSBCAPS_CTRLDEFAULT;
dsbdMus[i].dwBufferBytes = 3 * wfMus[i].wf.nAvgBytesPerSec; // 3 seconds
dsbdMus[i].lpwfxFormat = (LPWAVEFORMATEX) &wfMus[i];
// Allocate a compressed data buffer
if ((data8 = malloc(dsbdMus[i].dwBufferBytes/2)) == NULL)
{
fclose(fpMus[0]);
fpMus[0] = 0;
return(RDERR_OUTOFMEMORY);
}
// Seek to start of the compressed music
if (fseek(fpMus[0], musFilePos[i], SEEK_SET))
{
fclose(fpMus[0]);
fpMus[0] = 0;
free(data8);
return (RDERR_INVALIDID);
}
// Read the compressed data in to the buffer
if (fread(data8, sizeof(uint8), dsbdMus[i].dwBufferBytes/2, fpMus[0]) != dsbdMus[i].dwBufferBytes/2)
{
fclose(fpMus[0]);
fpMus[0] = 0;
free(data8);
return (RDERR_INVALIDID);
}
// Store the current position in the file for future streaming
musFilePos[i] = ftell(fpMus[0]);
// Create the sound effect sample buffer
hr = IDirectSound_CreateSoundBuffer(lpDS, &dsbdMus[i], &lpDsbMus[i], NULL);
if (hr == DS_OK)
{
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], 0, dsbdMus[i].dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
if (hr == DSERR_BUFFERLOST)
{
IDirectSoundBuffer_Restore(lpDsbMus[i]);
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], 0, dsbdMus[i].dwBufferBytes, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
}
if (hr == DS_OK)
{
// decompress the music into the music buffer.
data16 = (uint16*)lpv1;
data16[0] = *((int16*)data8); // First sample value
j=1;
while (j<(dwBytes1/2)-1)
{
if (GetCompressedSign(data8[j+1]))
data16[j] = data16[j-1] - (GetCompressedAmplitude(data8[j+1])<<GetCompressedShift(data8[j+1]));
else
data16[j] = data16[j-1] + (GetCompressedAmplitude(data8[j+1])<<GetCompressedShift(data8[j+1]));
j++;
}
// Never need to fill lpv2 because we started at the begining of the sound buffer
// Store the value of the last sample ready for next batch of decompression
musLastSample[i] = data16[j-1];
// Free the compressiong buffer and unlock the buffer now that we've filled it
free(data8);
IDirectSoundBuffer_Unlock(lpDsbMus[i], lpv1, dwBytes1, lpv2, dwBytes2);
// Modify the volume according to the master volume and music mute state
if (musicMuted)
v0 = v1 = 0;
else
{
v0 = volMusic[0];
v1 = volMusic[1];
}
if (v0 > v1)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v0]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], musicVolTable[v1*16/v0]);
}
else
{
if (v1 > v0)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], -musicVolTable[v0*16/v1]);
}
else
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], 0);
}
}
// Start the sound effect playing
IDirectSoundBuffer_Play(lpDsbMus[i], 0, 0, DSBPLAY_LOOPING);
// Record the last variables for streaming and looping
musStreaming[i] = 1;
musCounter[i] = 250;
}
else
{
// Opps failed to lock the sound buffer
fclose(fpMus[0]);
fpMus[0] = 0;
return(RDERR_LOCKFAILED);
}
}
else
{
// Opps failed to create the sound buffer
fclose(fpMus[0]);
fpMus[0] = 0;
return(RDERR_CREATESOUNDBUFFER);
}
}
}
*/
return(RD_OK);
}
void BS2Sound::UpdateCompSampleStreaming(void)
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{
warning("stub UpdateCompSampleStreaming");
/*
uint32 i,j,k;
int32 v0, v1;
int32 len;
int32 readCursor, writeCursor;
int32 dwBytes1, dwBytes2;
LPVOID lpv1, lpv2;
HRESULT hr;
uint16 *data16;
uint8 *data8;
int fade;
for (i=0; i<MAXMUS; i++)
{
if (musStreaming[i])
{
if (musFading[i])
{
if (musFading[i] < 0)
{
if (++musFading[i] == 0)
{
IDirectSoundBuffer_Stop(lpDsbMus[i]);
IDirectSoundBuffer_Release(lpDsbMus[i]);
musStreaming[i] = 0;
musLooping[i] = 0;
}
else
{
// Modify the volume according to the master volume and music mute state
if (musicMuted)
v0 = v1 = 0;
else
{
v0 = (volMusic[0] * (0 - musFading[i]) / 16);
v1 = (volMusic[1] * (0 - musFading[i]) / 16);
}
if (v0 > v1)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v0]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], musicVolTable[v1*16/v0]);
}
else
{
if (v1 > v0)
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], -musicVolTable[v0*16/v1]);
}
else
{
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[v1]);
IDirectSoundBuffer_SetPan(lpDsbMus[i], 0);
}
}
}
}
}
else
{
if (IDirectSoundBuffer_GetCurrentPosition(lpDsbMus[i], &readCursor, &writeCursor) != DS_OK)
{
// Failed to get read and write positions
IDirectSoundBuffer_Stop(lpDsbMus[i]);
}
// Caluculate the amount of data to load into the sound buffer
len = readCursor - streamCursor[i];
if (len < 0)
{
len += dsbdMus[i].dwBufferBytes; // Wrap around !
}
// Reduce length if it requires reading past the end of the music
if (musFilePos[i]+len >= musEnd[i])
{
len = musEnd[i] - musFilePos[i];
fade = 1; // End of music reaced so we'll need to fade and repeat
}
else
fade = 0;
if (len > 0)
{
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], streamCursor[i], len, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
if (hr == DSERR_BUFFERLOST)
{
IDirectSoundBuffer_Restore(lpDsbMus[i]);
hr = IDirectSoundBuffer_Lock(lpDsbMus[i], streamCursor[i], len, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0);
}
if (hr == DS_OK)
{
streamCursor[i] += len;
if (streamCursor[i] >= (int32) dsbdMus[i].dwBufferBytes)
streamCursor[i] -= dsbdMus[i].dwBufferBytes;
// Allocate a compressed data buffer
if ((data8 = malloc(len/2)) == NULL)
{
fclose(fpMus[0]);
fpMus[0] = 0;
musFading[i] = -16;
}
// Seek to update position of compressed music when neccassary (probably never occurs)
if (ftell(fpMus[0]) != musFilePos[i])
fseek(fpMus[0], musFilePos[i], SEEK_SET);
// Read the compressed data in to the buffer
if (fread(data8, sizeof(uint8), len/2, fpMus[0]) != (size_t)len/2)
{
fclose(fpMus[0]);
fpMus[0] = 0;
free(data8);
musFading[i] = -16;
return;
}
// Update the current position in the file for future streaming
musFilePos[i] = ftell(fpMus[0]);
// decompress the music into the music buffer.
data16 = (uint16*)lpv1;
// Decompress the first byte using the last decompressed sample
if (GetCompressedSign(data8[0]))
data16[0] = musLastSample[i] - (GetCompressedAmplitude(data8[0])<<GetCompressedShift(data8[0]));
else
data16[0] = musLastSample[i] + (GetCompressedAmplitude(data8[0])<<GetCompressedShift(data8[0]));
j = 1;
// Decompress the rest of lpv1
while (j<(uint32)dwBytes1/2)
{
if (GetCompressedSign(data8[j]))
data16[j] = data16[j-1] - (GetCompressedAmplitude(data8[j])<<GetCompressedShift(data8[j]));
else
data16[j] = data16[j-1] + (GetCompressedAmplitude(data8[j])<<GetCompressedShift(data8[j]));
j++;
}
// Store the value of the last sample ready for next batch of decompression
musLastSample[i] = data16[j-1];
if (dwBytes1 < len) // The buffer has wrapped so we need to decompress to lpv2 as well
{
data16 = (uint16*)lpv2;
// Decompress first sample int lpv2 from lastsample in lpv1
if (GetCompressedSign(data8[j]))
data16[0] = musLastSample[i] - (GetCompressedAmplitude(data8[j])<<GetCompressedShift(data8[j]));
else
data16[0] = musLastSample[i] + (GetCompressedAmplitude(data8[j])<<GetCompressedShift(data8[j]));
j++;
k = 1;
// Decompress the rest of lpv2
while (k<(uint32)dwBytes2/2)
{
if (GetCompressedSign(data8[j]))
data16[k] = data16[k-1] - (GetCompressedAmplitude(data8[j])<<GetCompressedShift(data8[j]));
else
data16[k] = data16[k-1] + (GetCompressedAmplitude(data8[j])<<GetCompressedShift(data8[j]));
j++;
k++;
}
// Store the value of the last sample ready for next batch of decompression
musLastSample[i] = data16[k-1];
}
// Free the compressed data buffer and unlock the sound buffer.
free(data8);
IDirectSoundBuffer_Unlock(lpDsbMus[i], lpv1, dwBytes1, lpv2, dwBytes2);
// End of the music so we need to start fading and start the music again
if (fade)
{
musFading[i] = -16; // Fade the old music
// Close the music cluster if it's open
if (fpMus[0])
{
fclose(fpMus[0]);
fpMus[0] = 0;
}
// Loop if neccassary
if (musLooping[i])
StreamCompMusic(musFilename[i], musId[i], musLooping[i]);
}
}
}
}
}
}
DipMusic();
*/
}
int32 BS2Sound::DipMusic()
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{
warning("stub DipMusic");
/*
int32 len;
int32 readCursor, writeCursor;
int32 dwBytes1, dwBytes2;
int16 *sample;
int32 total = 0;
int32 i;
int32 status;
LPVOID lpv1, lpv2;
HRESULT hr = DS_OK;
LPDIRECTSOUNDBUFFER dsbMusic = NULL;
int32 currentMusicVol = musicVolTable[volMusic[0]];
int32 minMusicVol;
// Find which music buffer is currently playing
for (i = 0; i<MAXMUS && !dsbMusic; i++)
{
if (musStreaming[i] && musFading[i] == 0)
dsbMusic = lpDsbMus[i];
}
if ((!musicMuted) && dsbMusic && (!speechMuted) && (volMusic[0]>2))
{
minMusicVol = musicVolTable[volMusic[0] - 3];
if (speechStatus)
{
IDirectSoundBuffer_GetStatus(dsbSpeech, &status);
if ((hr = IDirectSoundBuffer_GetCurrentPosition(dsbMusic, &readCursor, &writeCursor)) != DS_OK)
return hr;
len = 44100 / 12 ;// 12th of a second
if ((hr = IDirectSoundBuffer_Lock(dsbMusic, readCursor, len, &lpv1, &dwBytes1, &lpv2, &dwBytes2, 0)) != DS_OK)
return hr;
for (i = 0, sample = (int16*)lpv1; sample<(int16*)((int8*)lpv1+dwBytes1); sample+= 30, i++) // 60 samples
{
if (*sample>0)
total += *sample;
else
total -= *sample;
}
total /= i;
total = minMusicVol + ( ( (currentMusicVol - minMusicVol) * total ) / 8000);
if (total > currentMusicVol)
total = currentMusicVol;
IDirectSoundBuffer_SetVolume(dsbMusic, total);
IDirectSoundBuffer_Unlock(dsbMusic,lpv1,dwBytes1,lpv2,dwBytes2);
}
else
{
IDirectSoundBuffer_GetVolume(dsbMusic, &total);
total += 50;
if (total > currentMusicVol)
total = currentMusicVol;
IDirectSoundBuffer_SetVolume(dsbMusic, total);
}
}
return (hr);
*/
return RD_OK;
}
int32 BS2Sound::MusicTimeRemaining()
2003-07-28 01:47:41 +00:00
{
warning("stub MusicTimeRemaaining");
/*
int32 writeCursor;
int32 i;
int32 readCursor;
for (i=0; i<MAXMUS && !musStreaming[i]; i++)
{
// this is meant to be empty! (James19aug97)
}
if (i == MAXMUS)
return 0;
if ((IDirectSoundBuffer_GetCurrentPosition(lpDsbMus[i], &readCursor, &writeCursor)) != DS_OK)
return 0;
return (((132300-readCursor)/2 + (musEnd[i] - musFilePos[i])) / 22050);
*/
return 0;
}
void BS2Sound::StopMusic(void)
2003-07-28 01:47:41 +00:00
{
int32 i;
switch (compressedMusic)
{
case 1: // compressed music streaming
for (i = 0; i<MAXMUS; i++)
{
if (musStreaming[i])
musFading[i] = -16;
else
// If the music is muted, make sure the tune doesn't restart.
musLooping[i] = 0;
}
if (fpMus[0])
{
fclose(fpMus[0]);
fpMus[0] = 0;
}
break;
case 2:
for (i = 0; i<MAXMUS; i++)
{
if (musStreaming[i])
StartMusicFadeDown(i);
}
break;
default:
break;
}
}
int32 BS2Sound::PauseMusic(void)
2003-07-28 01:47:41 +00:00
{
warning("stub PauseMusic");
/*
int32 i;
if (soundOn)
{
for (i=0; i<2; i++)
{
if (musStreaming[i])
{
musicPaused[i] = TRUE;
if (IDirectSoundBuffer_Stop(lpDsbMus[i]) != RD_OK)
return(RDERR_FXFUCKED);
}
else
{
musicPaused[i] = FALSE;
}
}
}
*/
return(RD_OK);
}
int32 BS2Sound::UnpauseMusic(void)
2003-07-28 01:47:41 +00:00
{
warning("stub UnpauseMusic");
/*
int32 i;
if (soundOn)
{
for (i=0; i<2; i++)
{
if (musicPaused[i])
{
if (IDirectSoundBuffer_Play(lpDsbMus[i], 0, 0, DSBPLAY_LOOPING) != RD_OK)
return(RDERR_FXFUCKED);
musicPaused[i] = FALSE;
}
}
}
*/
return(RD_OK);
}
void BS2Sound::SetMusicVolume(uint8 volume)
2003-07-28 01:47:41 +00:00
{
warning("stub SetMusicVolume( %d )", volume);
/*
int32 i;
for (i = 0; i<MAXMUS; i++)
{
volMusic[i] = volume;
if (musStreaming[i] && !musFading[i] && !musicMuted)
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[volume]);
}
*/
}
uint8 BS2Sound::GetMusicVolume()
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{
2003-07-28 11:21:25 +00:00
return (uint8) volMusic[0];
2003-07-28 01:47:41 +00:00
}
void BS2Sound::MuteMusic(uint8 mute)
2003-07-28 01:47:41 +00:00
{
warning("stub MuteMusic( %d )", mute);
/*
int32 i;
musicMuted = mute;
for (i = 0; i<MAXMUS; i++)
{
if (!mute)
{
if (!musStreaming[i] && musLooping[i])
StreamCompMusic(musFilename[i], musId[i], musLooping[i]);
}
if (musStreaming[i] && !musFading[i])
{
if (mute)
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[0]);
else
IDirectSoundBuffer_SetVolume(lpDsbMus[i], musicVolTable[volMusic[i]]);
}
}
*/
}
uint8 BS2Sound::IsMusicMute(void)
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{
return (musicMuted);
}
void BS2Sound::GetSoundStatus(_drvSoundStatus *s)
2003-07-28 01:47:41 +00:00
{
int i;
// s->hwnd = hwnd;
// s->lpDS = lpDS;
// s->dsbPrimary = dsbPrimary;
// s->dsbSpeech = dsbSpeech;
s->soundOn = soundOn;
s->speechStatus = speechStatus;
s->fxPaused = fxPaused;
s->speechPaused = speechPaused;
s->speechVol = speechVol;
s->fxVol = fxVol;
s->speechMuted = speechMuted;
s->fxMuted = fxMuted;
s->compressedMusic = compressedMusic;
s->musicMuted = musicMuted;
memcpy(s->fxId, fxId, sizeof(int32) * MAXFX);
memcpy(s->fxCached, fxCached, sizeof(uint8) * MAXFX);
// memcpy(s->dsbFx, dsbFx, sizeof(LPDIRECTSOUNDBUFFER) * MAXFX);
memcpy(s->fxiPaused, fxiPaused, sizeof(uint8) * MAXFX);
memcpy(s->fxLooped, fxLooped, sizeof(uint8) * MAXFX);
memcpy(s->musStreaming, musStreaming, sizeof(int16) * MAXMUS);
memcpy(s->musicPaused, musicPaused, sizeof(int16) * MAXMUS);
memcpy(s->musCounter, musCounter, sizeof(int16) * MAXMUS);
memcpy(s->musFading, musFading, sizeof(int16) * MAXMUS);
memcpy(s->musLooping, musLooping, sizeof(int16) * MAXMUS);
memcpy(s->musLastSample,musLastSample, sizeof(int16) * MAXMUS);
memcpy(s->streamCursor, streamCursor, sizeof(int32) * MAXMUS);
memcpy(s->musFilePos, musFilePos, sizeof(int32) * MAXMUS);
memcpy(s->musEnd, musEnd, sizeof(int32) * MAXMUS);
memcpy(s->musId, musId, sizeof(uint32) * MAXMUS);
memcpy(s->volMusic, volMusic, sizeof(uint32) * 2);
// memcpy(s->dsbdMus, dsbdMus, sizeof(DSBUFFERDESC) * MAXMUS);
// memcpy(s->lpDsbMus, lpDsbMus, sizeof(LPDIRECTSOUNDBUFFER) * MAXMUS);
memcpy(s->fpMus, fpMus, sizeof(FILE*) * MAXMUS);
// memcpy(s->wfMus, wfMus, sizeof(PCMWAVEFORMAT) * MAXMUS);
for (i = 0; i<MAXMUS; i++)
memcpy(s->musFilename[i], musFilename[i], sizeof(char) * 256);
}
void BS2Sound::SetSoundStatus(_drvSoundStatus *s)
2003-07-28 01:47:41 +00:00
{
int i;
// hwnd = s->hwnd;
// lpDS = s->lpDS;
// dsbPrimary = s->dsbPrimary;
// dsbSpeech = s->dsbSpeech;
soundOn = s->soundOn;
speechStatus = s->speechStatus;
fxPaused = s->fxPaused;
speechPaused = s->speechPaused;
speechVol = s->speechVol;
fxVol = s->fxVol;
speechMuted = s->speechMuted;
fxMuted = s->fxMuted;
compressedMusic = s->compressedMusic;
musicMuted = s->musicMuted;
memcpy(fxId, s->fxId, sizeof(int32) * MAXFX);
memcpy(fxCached, s->fxCached, sizeof(uint8) * MAXFX);
// memcpy(dsbFx, s->dsbFx, sizeof(LPDIRECTSOUNDBUFFER) * MAXFX);
memcpy(fxiPaused, s->fxiPaused, sizeof(uint8) * MAXFX);
memcpy(fxLooped, s->fxLooped, sizeof(uint8) * MAXFX);
memcpy(musStreaming, s->musStreaming, sizeof(int16) * MAXMUS);
memcpy(musicPaused, s->musicPaused, sizeof(int16) * MAXMUS);
memcpy(musCounter, s->musCounter, sizeof(int16) * MAXMUS);
memcpy(musFading, s->musFading, sizeof(int16) * MAXMUS);
memcpy(musLooping, s->musLooping, sizeof(int16) * MAXMUS);
memcpy(musLastSample,s->musLastSample, sizeof(int16) * MAXMUS);
memcpy(streamCursor, s->streamCursor, sizeof(int32) * MAXMUS);
memcpy(musFilePos, s->musFilePos, sizeof(int32) * MAXMUS);
memcpy(musEnd, s->musEnd, sizeof(int32) * MAXMUS);
memcpy(musId, s->musId, sizeof(uint32) * MAXMUS);
memcpy(volMusic, s->volMusic, sizeof(uint32) * 2);
// memcpy(dsbdMus, s->dsbdMus, sizeof(DSBUFFERDESC) * MAXMUS);
// memcpy(lpDsbMus, s->lpDsbMus, sizeof(LPDIRECTSOUNDBUFFER) * MAXMUS);
// memcpy(fpMus, s->fpMus, sizeof(FILE*) * MAXMUS);
// memcpy(wfMus, s->wfMus, sizeof(PCMWAVEFORMAT) * MAXMUS);
for (i = 0; i<MAXMUS; i++)
memcpy(musFilename[i], s->musFilename[i], sizeof(char) * 256);
}