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fs2open.github.com/code/sound/ds.cpp
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/*
* Copyright (C) Volition, Inc. 1999. All rights reserved.
*
* All source code herein is the property of Volition, Inc. You may not sell
* or otherwise commercially exploit the source or things you created based on the
* source.
*
*/
#include "globalincs/pstypes.h"
#include "cfile/cfile.h"
#include "sound/openal.h"
#include "sound/ds.h"
#include "sound/ds3d.h"
#include "sound/channel.h"
#include "sound/acm.h"
#include "osapi/osapi.h"
#include "sound/dscap.h"
typedef struct sound_buffer
{
ALuint buf_id; // OpenAL buffer id
int channel_id; // Channel[] index this buffer is currently bound to
int frequency;
int bits_per_sample;
int nchannels;
int nseconds;
int nbytes;
sound_buffer():
buf_id(0), channel_id(-1), frequency(0), bits_per_sample(0),
nchannels(0), nseconds(0), nbytes(0)
{
}
} sound_buffer;
static int MAX_CHANNELS = 32; // initialized properly in ds_init_channels()
channel *Channels = NULL;
static int channel_next_sig = 1;
const int BUFFER_BUMP = 50;
SCP_vector<sound_buffer> sound_buffers;
extern int Snd_sram; // mem (in bytes) used up by storing sounds in system memory
static int Ds_use_eax = 0;
static int Ds_eax_inited = 0;
static int AL_play_position = 0;
// NOTE: these can't be static
int Ds_sound_quality = DS_SQ_MEDIUM;
int Ds_float_supported = 0;
// this is so stupid - required to get VC6 to use the following array initializer
EFXREVERBPROPERTIES::EFXREVERBPROPERTIES(const EFXREVERBPROPERTIES_list &list)
{
name = list.name;
flDensity = list.flDensity;
flDiffusion = list.flDiffusion;
flGain = list.flGain;
flGainHF = list.flGainHF;
flGainLF = list.flGainLF;
flDecayTime = list.flDecayTime;
flDecayHFRatio = list.flDecayHFRatio;
flDecayLFRatio = list.flDecayLFRatio;
flReflectionsGain = list.flReflectionsGain;
flReflectionsDelay = list.flReflectionsDelay;
flReflectionsPan[0] = list.flReflectionsPan[0];
flReflectionsPan[1] = list.flReflectionsPan[1];
flReflectionsPan[2] = list.flReflectionsPan[2];
flLateReverbGain = list.flLateReverbGain;
flLateReverbDelay = list.flLateReverbDelay;
flLateReverbPan[0] = list.flLateReverbPan[0];
flLateReverbPan[1] = list.flLateReverbPan[1];
flLateReverbPan[2] = list.flLateReverbPan[2];
flEchoTime = list.flEchoTime;
flEchoDepth = list.flEchoDepth;
flModulationTime = list.flModulationTime;
flModulationDepth = list.flModulationDepth;
flAirAbsorptionGainHF = list.flAirAbsorptionGainHF;
flHFReference = list.flHFReference;
flLFReference = list.flLFReference;
flRoomRolloffFactor = list.flRoomRolloffFactor;
iDecayHFLimit = list.iDecayHFLimit;
}
static const EFXREVERBPROPERTIES_list EFX_Reverb_Defaults[EAX_ENVIRONMENT_COUNT] =
{
{ "Generic", 1.0f, 1.0f, 0.316228f, 0.891251f, 1.0f, 1.49f, 0.83f, 1.0f, 0.050003f, 0.007f, {0.0f, 0.0f, 0.0f}, 1.258925f, 0.011f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Padded cell", 0.171500f, 1.0f, 0.316228f, 0.001f, 1.0f, 0.17f, 0.10f, 1.0f, 0.250035f, 0.001f, {0.0f, 0.0f, 0.0f}, 1.269112f, 0.002f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Room", 0.428687f, 1.0f, 0.316228f, 0.592925f, 1.0f, 0.40f, 0.83f, 1.0f, 0.150314f, 0.002f, {0.0f, 0.0f, 0.0f}, 1.062919f, 0.003f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Bathroom", 0.171500f, 1.0f, 0.316228f, 0.251189f, 1.0f, 1.49f, 0.54f, 1.0f, 0.653131f, 0.007f, {0.0f, 0.0f, 0.0f}, 3.273407f, 0.011f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Living room", 0.976563f, 1.0f, 0.316228f, 0.001f, 1.0f, 0.50f, 0.10f, 1.0f, 0.205116f, 0.003f, {0.0f, 0.0f, 0.0f}, 0.280543f, 0.004f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Stone room", 1.0f, 1.0f, 0.316228f, 0.707946f, 1.0f, 2.31f, 0.64f, 1.0f, 0.441062f, 0.012f, {0.0f, 0.0f, 0.0f}, 1.100272f, 0.017f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Auditorium", 1.0f, 1.0f, 0.316228f, 0.578096f, 1.0f, 4.32f, 0.59f, 1.0f, 0.403181f, 0.02f, {0.0f, 0.0f, 0.0f}, 0.716968f, 0.03f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Concert hall", 1.0f, 1.0f, 0.316228f, 0.562341f, 1.0f, 3.92f, 0.70f, 1.0f, 0.242661f, 0.02f, {0.0f, 0.0f, 0.0f}, 0.997700f, 0.029f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Cave", 1.0f, 1.0f, 0.316228f, 1.0f, 1.0f, 2.91f, 1.30f, 1.0f, 0.500035f, 0.015f, {0.0f, 0.0f, 0.0f}, 0.706318f, 0.022f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 0 },
{ "Arena", 1.0f, 1.0f, 0.316228f, 0.447713f, 1.0f, 7.24f, 0.33f, 1.0f, 0.261216f, 0.02f, {0.0f, 0.0f, 0.0f}, 1.018591f, 0.03f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Hangar", 1.0f, 1.0f, 0.316228f, 0.316228f, 1.0f, 10.05f, 0.23f, 1.0f, 0.500035f, 0.02f, {0.0f, 0.0f, 0.0f}, 1.256030f, 0.03f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Carpeted hallway", 0.428687f, 1.0f, 0.316228f, 0.01f, 1.0f, 0.30f, 0.10f, 1.0f, 0.121479f, 0.002f, {0.0f, 0.0f, 0.0f}, 0.153109f, 0.03f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Hallway", 0.364500f, 1.0f, 0.316228f, 0.707946f, 1.0f, 1.49f, 0.59f, 1.0f, 0.245754f, 0.007f, {0.0f, 0.0f, 0.0f}, 1.661499f, 0.011f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Stone corridor", 1.0f, 1.0f, 0.316228f, 0.761202f, 1.0f, 2.70f, 0.79f, 1.0f, 0.247172f, 0.013f, {0.0f, 0.0f, 0.0f}, 1.575796f, 0.02f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Alley", 1.0f, 0.30f, 0.316228f, 0.732825f, 1.0f, 1.49f, 0.86f, 1.0f, 0.250035f, 0.007f, {0.0f, 0.0f, 0.0f}, 0.995405f, 0.011f, {0.0f, 0.0f, 0.0f}, 0.125f, 0.95f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Forest", 1.0f, 0.30f, 0.316228f, 0.022387f, 1.0f, 1.49f, 0.54f, 1.0f, 0.052481f, 0.162f, {0.0f, 0.0f, 0.0f}, 0.768245f, 0.088f, {0.0f, 0.0f, 0.0f}, 0.125f, 1.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "City", 1.0f, 0.50f, 0.316228f, 0.398107f, 1.0f, 1.49f, 0.67f, 1.0f, 0.073030f, 0.007f, {0.0f, 0.0f, 0.0f}, 0.142725f, 0.011f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Mountains", 1.0f, 0.27f, 0.316228f, 0.056234f, 1.0f, 1.49f, 0.21f, 1.0f, 0.040738f, 0.30f, {0.0f, 0.0f, 0.0f}, 0.191867f, 0.10f, {0.0f, 0.0f, 0.0f}, 0.25f, 1.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 0 },
{ "Quarry", 1.0f, 1.0f, 0.316228f, 0.316228f, 1.0f, 1.49f, 0.83f, 1.0f, 0.0f, 0.061f, {0.0f, 0.0f, 0.0f}, 1.778279f, 0.025f, {0.0f, 0.0f, 0.0f}, 0.125f, 0.70f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Plain", 1.0f, 0.21f, 0.316228f, 0.10f, 1.0f, 1.49f, 0.50f, 1.0f, 0.058479f, 0.179f, {0.0f, 0.0f, 0.0f}, 0.108893f, 0.10f, {0.0f, 0.0f, 0.0f}, 0.25f, 1.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Parking lot", 1.0f, 1.0f, 0.316228f, 1.0f, 1.0f, 1.65f, 1.50f, 1.0f, 0.208209f, 0.008f, {0.0f, 0.0f, 0.0f}, 0.265155f, 0.012f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 0 },
{ "Sewer pipe", 0.307063f, 0.80f, 0.316228f, 0.316228f, 1.0f, 2.81f, 0.14f, 1.0f, 1.638702f, 0.014f, {0.0f, 0.0f, 0.0f}, 3.247133f, 0.021f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 0.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Underwater", 0.364500f, 1.0f, 0.316228f, 0.01f, 1.0f, 1.49f, 0.10f, 1.0f, 0.596348f, 0.007f, {0.0f, 0.0f, 0.0f}, 7.079458f, 0.011f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 1.18f, 0.348f, 0.994260f, 5000.0f, 250.0f, 0.0f, 1 },
{ "Drugged", 0.428687f, 0.50f, 0.316228f, 1.0f, 1.0f, 8.39f, 1.39f, 1.0f, 0.875992f, 0.002f, {0.0f, 0.0f, 0.0f}, 3.108136f, 0.03f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 0.25f, 1.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 0 },
{ "Dizzy", 0.364500f, 0.60f, 0.316228f, 0.630957f, 1.0f, 17.23f, 0.56f, 1.0f, 0.139155f, 0.02f, {0.0f, 0.0f, 0.0f}, 0.493742f, 0.03f, {0.0f, 0.0f, 0.0f}, 0.25f, 1.0f, 0.81f, 0.31f, 0.994260f, 5000.0f, 250.0f, 0.0f, 0 },
{ "Psychotic", 0.062500f, 0.50f, 0.316228f, 0.840427f, 1.0f, 7.56f, 0.91f, 1.0f, 0.486407f, 0.02f, {0.0f, 0.0f, 0.0f}, 2.437811f, 0.03f, {0.0f, 0.0f, 0.0f}, 0.25f, 0.0f, 4.00f, 1.0f, 0.994260f, 5000.0f, 250.0f, 0.0f, 0 }
};
SCP_vector<EFXREVERBPROPERTIES> EFX_presets;
typedef ALvoid (AL_APIENTRY * ALGENFILTERS) (ALsizei, ALuint*);
typedef ALvoid (AL_APIENTRY * ALDELETEFILTERS) (ALsizei, ALuint*);
typedef ALvoid (AL_APIENTRY * ALFILTERI) (ALuint, ALenum, ALint);
typedef ALvoid (AL_APIENTRY * ALGENEFFECTS) (ALsizei, ALuint*);
typedef ALvoid (AL_APIENTRY * ALDELETEEFFECTS) (ALsizei, ALuint*);
typedef ALvoid (AL_APIENTRY * ALEFFECTI) (ALuint, ALenum, ALint);
typedef ALvoid (AL_APIENTRY * ALEFFECTF) (ALuint, ALenum, ALfloat);
typedef ALvoid (AL_APIENTRY * ALEFFECTFV) (ALuint, ALenum, ALfloat*);
typedef ALvoid (AL_APIENTRY * ALGETEFFECTF) (ALuint, ALenum, ALfloat*);
typedef ALvoid (AL_APIENTRY * ALGENAUXILIARYEFFECTSLOTS) (ALsizei, ALuint*);
typedef ALvoid (AL_APIENTRY * ALDELETEAUXILIARYEFFECTSLOTS) (ALsizei, ALuint*);
typedef ALboolean (AL_APIENTRY * ALISAUXILIARYEFFECTSLOT) (ALuint);
typedef ALvoid (AL_APIENTRY * ALAUXILIARYEFFECTSLOTI) (ALuint, ALenum, ALint);
typedef ALvoid (AL_APIENTRY * ALAUXILIARYEFFECTSLOTIV) (ALuint, ALenum, ALint*);
typedef ALvoid (AL_APIENTRY * ALAUXILIARYEFFECTSLOTF) (ALuint, ALenum, ALfloat);
typedef ALvoid (AL_APIENTRY * ALAUXILIARYEFFECTSLOTFV) (ALuint, ALenum, ALfloat*);
ALGENFILTERS v_alGenFilters = NULL;
ALDELETEFILTERS v_alDeleteFilters = NULL;
ALFILTERI v_alFilteri = NULL;
ALGENEFFECTS v_alGenEffecs = NULL;
ALDELETEEFFECTS v_alDeleteEffects = NULL;
ALEFFECTI v_alEffecti = NULL;
ALEFFECTF v_alEffectf = NULL;
ALEFFECTFV v_alEffectfv = NULL;
ALGETEFFECTF v_alGetEffectf = NULL;
ALGENAUXILIARYEFFECTSLOTS v_alGenAuxiliaryEffectSlots = NULL;
ALDELETEAUXILIARYEFFECTSLOTS v_alDeleteAuxiliaryEffectSlots = NULL;
ALISAUXILIARYEFFECTSLOT v_alIsAuxiliaryEffectSlot = NULL;
ALAUXILIARYEFFECTSLOTI v_alAuxiliaryEffectSloti = NULL;
ALAUXILIARYEFFECTSLOTIV v_alAuxiliaryEffectSlotiv = NULL;
ALAUXILIARYEFFECTSLOTF v_alAuxiliaryEffectSlotf = NULL;
ALAUXILIARYEFFECTSLOTFV v_alAuxiliaryEffectSlotfv = NULL;
ALCdevice *ds_sound_device = NULL;
ALCcontext *ds_sound_context = NULL;
ALuint AL_EFX_aux_id = 0;
static ALuint AL_EFX_effect_id = 0;
static int Ds_active_env = -1;
static void *al_load_function(const char *func_name)
{
void *func = alGetProcAddress(func_name);
if ( !func ) {
throw func_name;
}
return func;
}
static void al_efx_load_preset(size_t presetid)
{
if ( !Ds_eax_inited ) {
return;
}
if (presetid >= EFX_presets.size()) {
return;
}
EFXREVERBPROPERTIES *prop = &EFX_presets[presetid];
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_DENSITY, prop->flDensity) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_DIFFUSION, prop->flDiffusion) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_GAIN, prop->flGain) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_GAINHF, prop->flGainHF) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_GAINLF, prop->flGainLF) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_DECAY_TIME, prop->flDecayTime) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_DECAY_HFRATIO, prop->flDecayHFRatio) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_DECAY_LFRATIO, prop->flDecayLFRatio) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_REFLECTIONS_GAIN, prop->flReflectionsGain) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_REFLECTIONS_DELAY, prop->flReflectionsDelay) );
OpenAL_ErrorPrint( v_alEffectfv(AL_EFX_effect_id, AL_EAXREVERB_REFLECTIONS_PAN, prop->flReflectionsPan) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_LATE_REVERB_GAIN, prop->flLateReverbGain) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_LATE_REVERB_DELAY, prop->flLateReverbDelay) );
OpenAL_ErrorPrint( v_alEffectfv(AL_EFX_effect_id, AL_EAXREVERB_LATE_REVERB_PAN, prop->flLateReverbPan) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_ECHO_TIME, prop->flEchoTime) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_ECHO_DEPTH, prop->flEchoDepth) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_MODULATION_TIME, prop->flModulationTime) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_MODULATION_DEPTH, prop->flModulationDepth) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_AIR_ABSORPTION_GAINHF, prop->flAirAbsorptionGainHF) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_HFREFERENCE, prop->flHFReference) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_LFREFERENCE, prop->flLFReference) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_ROOM_ROLLOFF_FACTOR, prop->flRoomRolloffFactor) );
OpenAL_ErrorPrint( v_alEffecti(AL_EFX_effect_id, AL_EAXREVERB_DECAY_HFLIMIT, prop->iDecayHFLimit) );
Ds_active_env = presetid;
}
int ds_initialized = FALSE;
/**
* @brief Parse a wave file.
*
* @param fp File of sound to parse
* @param dest Address of pointer of where to store raw sound data (output parm)
* @param dest_size Number of bytes of sound data stored (output parm)
* @param header Address of pointer to a WAVEFORMATEX struct (output parm)
* @param ogg Boolean to indicate OGG vorbis file, if false assume Wave file
* @param ovf Pointer to a OggVorbis_File struct, OGG vorbis only (output parm)
*
* @return 0 if wave file successfully parsed, -1 if an error occurred
*
* NOTE: memory is malloced for the header and dest (if not OGG) in this function. It is the responsibility
* of the caller to free this memory later.
*/
int ds_parse_sound(CFILE* fp, ubyte **dest, uint *dest_size, WAVEFORMATEX **header, bool ogg, OggVorbis_File *ovf)
{
PCMWAVEFORMAT PCM_header;
ushort cbExtra = 0;
unsigned int tag, size, next_chunk;
bool got_fmt = false, got_data = false;
// some preinit stuff, could be done from calling function but this should guarantee it's right
*dest = NULL;
*dest_size = 0;
if (fp == NULL) {
return -1;
}
// if we should have a Vorbis file then try for it
if (ogg) {
if (ovf == NULL) {
Int3();
return -1;
}
// Check for OGG Vorbis first
if ( !ov_open_callbacks(fp, ovf, NULL, 0, cfile_callbacks) ) {
// got one, now read all of the needed header info
ov_info(ovf, -1);
// we only support one logical bitstream
if ( ov_streams(ovf) != 1 ) {
nprintf(( "Sound", "SOUND ==> OGG reading error: We don't support bitstream changes!\n" ));
return -1;
}
if ( (*header = (WAVEFORMATEX *) vm_malloc ( sizeof(WAVEFORMATEX) )) != NULL ) {
(*header)->wFormatTag = OGG_FORMAT_VORBIS;
(*header)->nChannels = (ushort)ovf->vi->channels;
(*header)->nSamplesPerSec = ovf->vi->rate;
switch (Ds_sound_quality) {
case DS_SQ_HIGH:
(*header)->wBitsPerSample = Ds_float_supported ? 32 : 16;
break;
case DS_SQ_MEDIUM:
(*header)->wBitsPerSample = 16;
break;
default:
(*header)->wBitsPerSample = 8;
break;
}
(*header)->nBlockAlign = (ushort)(((*header)->wBitsPerSample / 8) * ovf->vi->channels);
(*header)->nAvgBytesPerSec = ovf->vi->rate * (*header)->nBlockAlign;
// WMC - Total samples * channels * bits/sample
ogg_int64_t pcm_total_size = ov_pcm_total(ovf, -1);
if (pcm_total_size > 0) {
*dest_size = (uint)(pcm_total_size * (*header)->nBlockAlign);
} else {
nprintf(("Sound", "SOUND ==> Size returned for this file is invalid. Please re-encode the file, as it will not work correctly.\n"));
return -1;
}
} else {
Assert( 0 );
return -1;
}
// we're all good, can leave now
return 0;
}
} else {
// Otherwise we assime Wave format
// Skip the "RIFF" tag and file size (8 bytes)
// Skip the "WAVE" tag (4 bytes)
// IMPORTANT!! Look at snd_load before even THINKING about changing this.
cfseek( fp, 12, CF_SEEK_SET );
// Now read RIFF tags until the end of file
while (1) {
if ( cfread( &tag, sizeof(uint), 1, fp ) != 1 ) {
break;
}
tag = INTEL_INT( tag );
if ( cfread( &size, sizeof(uint), 1, fp ) != 1 ) {
break;
}
size = INTEL_INT( size );
next_chunk = cftell(fp) + size;
switch (tag) {
case 0x20746d66: { // The 'fmt ' tag
PCM_header.wf.wFormatTag = cfread_ushort(fp);
PCM_header.wf.nChannels = cfread_ushort(fp);
PCM_header.wf.nSamplesPerSec = cfread_uint(fp);
PCM_header.wf.nAvgBytesPerSec = cfread_uint(fp);
PCM_header.wf.nBlockAlign = cfread_ushort(fp);
PCM_header.wBitsPerSample = cfread_ushort(fp);
// should be either: WAVE_FORMAT_PCM, WAVE_FORMAT_ADPCM, WAVE_FORMAT_IEEE_FLOAT
switch (PCM_header.wf.wFormatTag) {
case WAVE_FORMAT_PCM: {
if ( (PCM_header.wBitsPerSample != 8) && (PCM_header.wBitsPerSample != 16) ) {
nprintf(("Sound", "SOUND ==> %d-bit PCM is not supported!\n", PCM_header.wBitsPerSample));
return -1;
}
// fix block align
PCM_header.wf.nBlockAlign = (PCM_header.wBitsPerSample / 8) * PCM_header.wf.nChannels;
break;
}
case WAVE_FORMAT_IEEE_FLOAT: {
if (PCM_header.wBitsPerSample != 32) {
nprintf(("Sound", "SOUND ==> %d-bit FLOAT PCM is not supported!\n", PCM_header.wBitsPerSample));
return -1;
}
switch (Ds_sound_quality) {
case DS_SQ_HIGH:
PCM_header.wBitsPerSample = Ds_float_supported ? 32 : 16;
break;
case DS_SQ_MEDIUM:
PCM_header.wBitsPerSample = 16;
break;
default:
PCM_header.wBitsPerSample = 8;
break;
}
// fix block align
PCM_header.wf.nBlockAlign = (PCM_header.wBitsPerSample / 8) * PCM_header.wf.nChannels;
break;
}
case WAVE_FORMAT_ADPCM:
// block align doesn't get fixed here
break;
default: {
nprintf(("Sound", "SOUND ==> Format '%d' is not supported!\n", PCM_header.wf.wFormatTag));
return -1;
}
}
if (PCM_header.wf.wFormatTag == WAVE_FORMAT_ADPCM) {
cbExtra = cfread_ushort(fp);
}
// Allocate memory for WAVEFORMATEX structure + extra bytes
if ( (*header = (WAVEFORMATEX *) vm_malloc ( sizeof(WAVEFORMATEX)+cbExtra )) != NULL ) {
// Copy bytes from temporary format structure
memcpy (*header, &PCM_header, sizeof(PCM_header));
(*header)->cbSize = cbExtra;
// Read those extra bytes, append to WAVEFORMATEX structure
if (cbExtra != 0) {
cfread( ((ubyte *)(*header) + sizeof(WAVEFORMATEX)), cbExtra, 1, fp);
}
} else {
// malloc failed
return -1;
}
got_fmt = true;
break;
}
case 0x61746164: { // the 'data' tag
*dest_size = size;
(*dest) = (ubyte *)vm_malloc(size);
Assert( *dest != NULL );
cfread( *dest, size, 1, fp );
got_data = true;
break;
}
default: // unknown, skip it
break;
}
// This is here so that we can avoid reading data that we don't understand or properly handle.
// We could do this just as well by checking the RIFF size, but this is easier - taylor
if (got_fmt && got_data) {
break;
}
cfseek( fp, next_chunk, CF_SEEK_SET );
}
// we're all good, can leave now
return 0;
}
return -1;
}
/**
* Parse a sound file, any format, and store the info in s_info.
*
* @param real_filename Filename to parse
* @param s_info Storage for the sound file info
*/
int ds_parse_sound_info(char *real_filename, sound_info *s_info)
{
PCMWAVEFORMAT PCM_header;
uint tag, size, next_chunk;
bool got_fmt = false, got_data = false;
OggVorbis_File ovf;
int rc, FileSize, FileOffset;
char fullpath[MAX_PATH];
char filename[MAX_FILENAME_LEN];
const int NUM_EXT = 2;
const char *audio_ext[NUM_EXT] = { ".ogg", ".wav" };
int rval = -1;
if ( (real_filename == NULL) || (s_info == NULL) ) {
return -1;
}
// remove extension
strcpy_s( filename, real_filename );
char *p = strrchr(filename, '.');
if ( p ) *p = 0;
rc = cf_find_file_location_ext(filename, NUM_EXT, audio_ext, CF_TYPE_ANY, sizeof(fullpath) - 1, fullpath, &FileSize, &FileOffset);
if (rc < 0) {
return -1;
}
// open the file
CFILE *fp = cfopen_special(fullpath, "rb", FileSize, FileOffset);
if (fp == NULL) {
return -1;
}
// Ogg Vorbis
if (rc == 0) {
if ( !ov_open_callbacks(fp, &ovf, NULL, 0, cfile_callbacks) ) {
// got one, now read all of the needed header info
ov_info(&ovf, -1);
// we only support one logical bitstream
if ( ov_streams(&ovf) != 1 ) {
nprintf(( "Sound", "SOUND ==> OGG reading error: We don't support bitstream changes!\n" ));
ov_clear(&ovf);
goto Done;
}
s_info->format = OGG_FORMAT_VORBIS;
s_info->n_channels = (ushort)ovf.vi->channels;
s_info->sample_rate = ovf.vi->rate;
switch (Ds_sound_quality) {
case DS_SQ_HIGH:
s_info->bits = Ds_float_supported ? 32 : 16;
break;
case DS_SQ_MEDIUM:
s_info->bits = 16;
break;
default:
s_info->bits = 8;
break;
}
s_info->n_block_align = (ushort)((s_info->bits / 8) * s_info->n_channels);
s_info->avg_bytes_per_sec = s_info->sample_rate * s_info->n_block_align;
s_info->size = (uint)(ov_pcm_total(&ovf, -1) * s_info->n_block_align);
ov_clear(&ovf);
// we're all good, can leave now
rval = 0;
goto Done;
}
}
// PCM Wave
else if (rc == 1) {
// Skip the "RIFF" tag and file size (8 bytes)
// Skip the "WAVE" tag (4 bytes)
// IMPORTANT!! Look at snd_load before even THINKING about changing this.
cfseek( fp, 12, CF_SEEK_SET );
// Now read RIFF tags until the end of file
while (1) {
if ( cfread( &tag, sizeof(uint), 1, fp ) != 1 ) {
break;
}
tag = INTEL_INT( tag );
if ( cfread( &size, sizeof(uint), 1, fp ) != 1 ) {
break;
}
size = INTEL_INT( size );
next_chunk = cftell(fp) + size;
switch (tag) {
case 0x20746d66: // The 'fmt ' tag
PCM_header.wf.wFormatTag = cfread_ushort(fp);
PCM_header.wf.nChannels = cfread_ushort(fp);
PCM_header.wf.nSamplesPerSec = cfread_uint(fp);
PCM_header.wf.nAvgBytesPerSec = cfread_uint(fp);
PCM_header.wf.nBlockAlign = cfread_ushort(fp);
PCM_header.wBitsPerSample = cfread_ushort(fp);
// should be either: WAVE_FORMAT_PCM, WAVE_FORMAT_ADPCM, WAVE_FORMAT_IEEE_FLOAT
switch (PCM_header.wf.wFormatTag) {
case WAVE_FORMAT_PCM: {
if ( (PCM_header.wBitsPerSample != 8) && (PCM_header.wBitsPerSample != 16) ) {
nprintf(("Sound", "SOUND ==> %d-bit PCM is not supported!\n", PCM_header.wBitsPerSample));
goto Done;
}
// fix block align
PCM_header.wf.nBlockAlign = (PCM_header.wBitsPerSample / 8) * PCM_header.wf.nChannels;
break;
}
case WAVE_FORMAT_IEEE_FLOAT: {
if (PCM_header.wBitsPerSample != 32) {
nprintf(("Sound", "SOUND ==> %d-bit FLOAT PCM is not supported!\n", PCM_header.wBitsPerSample));
goto Done;
}
switch (Ds_sound_quality) {
case DS_SQ_HIGH:
PCM_header.wBitsPerSample = Ds_float_supported ? 32 : 16;
break;
case DS_SQ_MEDIUM:
PCM_header.wBitsPerSample = 16;
break;
default:
PCM_header.wBitsPerSample = 8;
break;
}
// fix block align
PCM_header.wf.nBlockAlign = (PCM_header.wBitsPerSample / 8) * PCM_header.wf.nChannels;
break;
}
case WAVE_FORMAT_ADPCM:
// block align doesn't get fixed here
break;
default: {
nprintf(("Sound", "SOUND ==> Format '%d' is not supported!\n", PCM_header.wf.wFormatTag));
goto Done;
}
}
s_info->format = PCM_header.wf.wFormatTag;
s_info->n_channels = PCM_header.wf.nChannels;
s_info->sample_rate = PCM_header.wf.nSamplesPerSec;
s_info->bits = PCM_header.wBitsPerSample;
s_info->avg_bytes_per_sec = PCM_header.wf.nAvgBytesPerSec;
s_info->n_block_align = PCM_header.wf.nBlockAlign;
got_fmt = true;
break;
case 0x61746164: // the 'data' tag
s_info->size = size;
got_data = true;
break;
default:
break;
}
if (got_fmt && got_data) {
rval = 0;
goto Done;
}
cfseek( fp, next_chunk, CF_SEEK_SET );
}
}
Done:
cfclose(fp);
return rval;
}
/**
*
*/
int ds_get_sid()
{
sound_buffer new_buffer;
uint i;
for (i = 0; i < sound_buffers.size(); i++) {
if (sound_buffers[i].buf_id == 0) {
return (int)i;
}
}
sound_buffers.push_back( new_buffer );
return (int)(sound_buffers.size() - 1);
}
/**
* @brief Load a secondary buffer with sound data.
* @details The sounds data for game sounds are stored in the DirectSound secondary buffers,
* and are duplicated as needed and placed in the Channels[] array to be played.
*
* @param sid Pointer to software id for sound ( output parm)
* @param final_size Pointer to storage to receive uncompressed sound size (output parm)
* @param header Pointer to a WAVEFORMATEX structure
* @param si ::sound_info structure, contains details on the sound format
* @param flags Buffer properties ( DS_HARDWARE , DS_3D )
*
* @return 1 if sound effect could not loaded into a secondary buffer, 0 if sound effect successfully loaded into a secondary buffer
*
* NOTE: this function is slow, especially when sounds are loaded into hardware. Don't call this
* function from within gameplay.
*/
int ds_load_buffer(int *sid, int *final_size, void *header, sound_info *si, int flags)
{
Assert( final_size != NULL );
Assert( header != NULL );
if (si == NULL) {
Int3();
return -1;
}
// All sounds are required to have a software buffer
*sid = ds_get_sid();
if ( *sid == -1 ) {
nprintf(("Sound","SOUND ==> No more sound buffers available\n"));
return -1;
}
ALuint pi;
OpenAL_ErrorCheck( alGenBuffers (1, &pi), return -1 );
ALenum format;
ALsizei size;
ALint bits, bps, n_channels = si->n_channels;
ALuint frequency;
ALvoid *data = NULL;
int sign, byte_order = 0, section, last_section = -1;
// the below two covnert_ variables are only used when the wav format is not
// PCM. DirectSound only takes PCM sound data, so we must convert to PCM if required
ubyte *convert_buffer = NULL; // storage for converted wav file
int convert_len; // num bytes of converted wav file
uint src_bytes_used; // number of source bytes actually converted (should always be equal to original size)
int rc;
WAVEFORMATEX *pwfx = (WAVEFORMATEX *)header;
switch (si->format) {
case WAVE_FORMAT_PCM: {
Assert( si->data != NULL );
bits = si->bits;
bps = si->avg_bytes_per_sec;
size = si->size;
#if BYTE_ORDER == BIG_ENDIAN
// swap 16-bit sound data
if (bits == 16) {
ushort *swap_tmp;
for (uint i=0; i<size; i=i+2) {
swap_tmp = (ushort*)(si->data + i);
*swap_tmp = INTEL_SHORT(*swap_tmp);
}
}
#endif
data = si->data;
break;
}
case WAVE_FORMAT_ADPCM: {
Assert( si->data != NULL );
bits = (Ds_sound_quality == DS_SQ_LOW) ? 8 : 16;
nprintf(( "Sound", "SOUND ==> Converting sound from ADPCM to PCM\n" ));
rc = ACM_convert_ADPCM_to_PCM(pwfx, si->data, si->size, &convert_buffer, 0, &convert_len, &src_bytes_used, bits);
if ( rc == -1 ) {
return -1;
}
if (src_bytes_used != si->size) {
return -1; // ACM conversion failed?
}
bps = (((si->n_channels * bits) / 8) * si->sample_rate);
size = convert_len;
data = convert_buffer;
nprintf(( "Sound", "SOUND ==> Coverted sound from ADPCM to PCM successfully\n" ));
break;
}
case WAVE_FORMAT_IEEE_FLOAT: {
Assert( si->data != NULL );
bits = si->bits;
#if BYTE_ORDER == BIG_ENDIAN
// need to byte-swap before any later conversions
float *swap_tmp;
for (uint i = 0; i < si->size; i += sizeof(float)) {
swap_tmp = (float *)(si->data + i);
*swap_tmp = INTEL_FLOAT(swap_tmp);
}
#endif
if (bits == 32) {
bps = si->avg_bytes_per_sec;
size = si->size;
data = si->data;
} else if (bits == 16) {
bps = si->avg_bytes_per_sec >> 1;
size = si->size >> 1;
convert_buffer = (ubyte*)vm_malloc_q(size);
if (convert_buffer == NULL) {
return -1;
}
float *in_p = (float*)si->data;
short *out_p = (short*)convert_buffer;
int end = si->size / sizeof(float);
for (int i = 0; i < end; i++) {
int i_val = (int)(in_p[i] * 32767.0f + 0.5f);
CLAMP(i_val, -32768, 32767);
*out_p++ = (short)i_val;
}
data = convert_buffer;
} else {
bps = si->avg_bytes_per_sec >> 2;
size = si->size >> 2;
convert_buffer = (ubyte*)vm_malloc_q(size);
if (convert_buffer == NULL) {
return -1;
}
float *in_p = (float*)si->data;
ubyte *out_p = (ubyte*)convert_buffer;
int end = si->size / sizeof(float);
for (int i = 0; i < end; i++) {
int i_val = (int)(in_p[i] * 127.0f + 0.5f) + 128;
CLAMP(i_val, 0, 255);
*out_p++ = (ubyte)i_val;
}
data = convert_buffer;
}
break;
}
case OGG_FORMAT_VORBIS: {
nprintf(( "Sound", "SOUND ==> converting sound from OGG to PCM\n" ));
sign = (si->bits == 8) ? 0 : 1;
#if BYTE_ORDER == BIG_ENDIAN
byte_order = 1;
#endif
src_bytes_used = 0;
convert_buffer = (ubyte*)vm_malloc_q(si->size);
if (convert_buffer == NULL) {
return -1;
}
while (src_bytes_used < si->size) {
float **pcm = NULL;
if (si->bits == 32) {
rc = ov_read_float(&si->ogg_info, &pcm, 1024, &section);
} else {
rc = ov_read(&si->ogg_info, (char *) convert_buffer + src_bytes_used, si->size - src_bytes_used, byte_order, si->bits / 8, sign, &section);
}
// fail if the bitstream changes, shouldn't get this far if that's the case though
if ((last_section != -1) && (last_section != section)) {
nprintf(( "Sound", "SOUND ==> OGG reading error: We don't support bitstream changes!\n" ));
ov_clear(&si->ogg_info);
vm_free(convert_buffer);
convert_buffer = NULL;
return -1;
}
if (rc == OV_EBADLINK) {
ov_clear(&si->ogg_info);
vm_free(convert_buffer);
convert_buffer = NULL;
return -1;
} else if (rc == 0) {
break;
} else if (rc > 0) {
if (si->bits == 32) {
float *out_p = (float*)(convert_buffer + src_bytes_used);
for (int i = 0; i < rc; i++) {
for (int j = 0; j < si->n_channels; j++) {
*out_p++ = pcm[j][i];
}
}
src_bytes_used += (rc * si->n_block_align);
} else {
src_bytes_used += rc;
}
last_section = section;
}
}
bits = si->bits;
bps = si->avg_bytes_per_sec;
size = (int)src_bytes_used;
data = convert_buffer;
// we're done with ogg stuff so clean it up
ov_clear(&si->ogg_info);
nprintf(( "Sound", "SOUND ==> Coverted sound from OGG successfully\n" ));
break;
}
default:
return -1;
}
// if this is supposed to play in 3D then make sure it's mono
if ( (flags & DS_3D) && (n_channels > 1) ) {
ubyte *mono_buffer = NULL;
mono_buffer = (ubyte*)vm_malloc_q(size >> 1);
if (mono_buffer == NULL) {
if (convert_buffer) {
vm_free(convert_buffer);
}
return -1;
}
if (bits == 32) {
float *in_p = (float*)data;
float *out_p = (float*)mono_buffer;
int end = size / sizeof(float);
for (int i = 0; i < end; i += 2) {
float i_val = (in_p[i] + in_p[i+1]) * 0.5f;
CLAMP(i_val, -1.0f, 1.0f);
*out_p++ = i_val;
}
} else if (bits == 16) {
short *in_p = (short*)data;
short *out_p = (short*)mono_buffer;
int end = size / sizeof(short);
for (int i = 0; i < end; i += 2) {
int i_val = (in_p[i] + in_p[i+1]) >> 1;
CLAMP(i_val, -32768, 32767);
*out_p++ = (short)i_val;
}
} else {
Assert( bits == 8 );
ubyte *in_p = (ubyte*)data;
ubyte *out_p = (ubyte*)mono_buffer;
for (int i = 0; i < size; i += 2) {
int i_val = (in_p[i] + in_p[i+1]) >> 1;
CLAMP(i_val, 0, 255);
*out_p++ = (ubyte)i_val;
}
}
n_channels = 1;
size >>= 1;
bps >>= 1;
data = mono_buffer;
if (convert_buffer) {
vm_free(convert_buffer);
convert_buffer = NULL;
}
nprintf(("Sound", "SOUND ==> Converted 3D sound from stereo to mono\n"));
}
// format is now in pcm
frequency = si->sample_rate;
format = openal_get_format(bits, n_channels);
if (format == AL_INVALID_VALUE) {
if (convert_buffer) {
vm_free(convert_buffer);
}
return -1;
}
Snd_sram += size;
if (final_size) {
*final_size = size;
}
OpenAL_ErrorCheck( alBufferData(pi, format, data, size, frequency), { if (convert_buffer) vm_free(convert_buffer); return -1; } );
sound_buffers[*sid].buf_id = pi;
sound_buffers[*sid].channel_id = -1;
sound_buffers[*sid].frequency = frequency;
sound_buffers[*sid].bits_per_sample = bits;
sound_buffers[*sid].nchannels = n_channels;
sound_buffers[*sid].nseconds = size / bps;
sound_buffers[*sid].nbytes = size;
// update sound_info struct with any changed data
si->bits = bits;
si->n_channels = n_channels;
si->size = size;
si->n_block_align = (bits / 8) * n_channels;
si->avg_bytes_per_sec = bps;
if ( convert_buffer ) {
vm_free( convert_buffer );
}
return 0;
}
/**
* Initialise the ::Channels[] array
*/
void ds_init_channels()
{
try {
Channels = new channel[MAX_CHANNELS];
} catch (std::bad_alloc) {
Error(LOCATION, "Unable to allocate %d bytes for %d audio channels.", sizeof(channel) * MAX_CHANNELS, MAX_CHANNELS);
}
}
/**
* Initialise the both the software and hardware buffers
*/
void ds_init_buffers()
{
sound_buffers.clear();
// pre-allocate for at least BUFFER_BUMP buffers
sound_buffers.reserve( BUFFER_BUMP );
}
/**
* Sound initialisation
* @return -1 if init failed, 0 if init success
*/
int ds_init()
{
ALfloat list_orien[] = { 0.0f, 0.0f, 1.0f, 0.0f, 1.0f, 0.0f };
ALCint attrList[] = { ALC_FREQUENCY, 22050, 0 };
unsigned int sample_rate = 22050;
mprintf(("Initializing OpenAL...\n"));
Ds_sound_quality = os_config_read_uint("Sound", "Quality", DS_SQ_MEDIUM);
CLAMP(Ds_sound_quality, DS_SQ_LOW, DS_SQ_HIGH);
switch (Ds_sound_quality) {
case DS_SQ_HIGH:
sample_rate = 48000;
break;
case DS_SQ_MEDIUM:
sample_rate = 44100;
break;
default:
sample_rate = 22050;
break;
}
sample_rate = os_config_read_uint("Sound", "SampleRate", sample_rate);
attrList[1] = sample_rate;
SCP_string playback_device;
SCP_string capture_device;
if ( openal_init_device(&playback_device, &capture_device) == false ) {
mprintf(("\n ERROR: Unable to find suitable playback device!\n\n"));
goto AL_InitError;
}
ds_sound_device = alcOpenDevice( (const ALCchar*) playback_device.c_str() );
if (ds_sound_device == NULL) {
mprintf((" Failed to open playback_device (%s) returning error (%s)\n", playback_device.c_str(), openal_error_string(1)));
goto AL_InitError;
}
ds_sound_context = alcCreateContext(ds_sound_device, attrList);
if (ds_sound_context == NULL) {
mprintf((" Failed to create context for playback_device (%s) with attrList = { 0x%x, %d, %d } returning error (%s)\n",
playback_device.c_str(), attrList[0], attrList[1], attrList[2], openal_error_string(1)));
goto AL_InitError;
}
alcMakeContextCurrent(ds_sound_context);
// we need to clear out all errors before moving on
alcGetError(NULL);
alGetError();
// set distance model to basically match what the D3D code does
alDistanceModel(AL_INVERSE_DISTANCE_CLAMPED);
// make sure we can actually use AL_BYTE_LOKI (Mac/Win OpenAL doesn't have it)
if ( alIsExtensionPresent( (const ALchar*)"AL_LOKI_play_position" ) == AL_TRUE ) {
mprintf((" Found extension \"AL_LOKI_play_position\".\n"));
AL_play_position = 1;
}
if ( alIsExtensionPresent( (const ALchar*)"AL_EXT_float32" ) == AL_TRUE ) {
mprintf((" Found extension \"AL_EXT_float32\".\n"));
Ds_float_supported = 1;
}
Ds_use_eax = 0;
if ( alcIsExtensionPresent(ds_sound_device, (const ALchar*)"ALC_EXT_EFX") == AL_TRUE ) {
mprintf((" Found extension \"ALC_EXT_EFX\".\n"));
Ds_use_eax = os_config_read_uint("Sound", "EnableEFX", Fred_running);
}
if (Ds_use_eax == 1) {
if (ds_eax_init() != 0) {
Ds_use_eax = 0;
}
}
// the presets always need to be available to FRED
if ( !Ds_use_eax && Fred_running ) {
EFX_presets.reserve(EAX_ENVIRONMENT_COUNT);
for (size_t i = 0; i < EAX_ENVIRONMENT_COUNT; i++) {
EFX_presets.push_back( EFX_Reverb_Defaults[i] );
}
}
// setup default listener position/orientation
// this is needed for 2D pan
OpenAL_ErrorPrint( alListener3f(AL_POSITION, 0.0, 0.0, 0.0) );
OpenAL_ErrorPrint( alListenerfv(AL_ORIENTATION, list_orien) );
// disable doppler (FIXME)
OpenAL_ErrorPrint( alDopplerFactor(0.0f) );
ds_init_channels();
ds_init_buffers();
mprintf(("\n"));
{
ALCint freq = 0;
OpenAL_ErrorPrint( alcGetIntegerv(ds_sound_device, ALC_FREQUENCY, sizeof(ALCint), &freq) );
mprintf((" Sample rate: %d (%d)\n", freq, sample_rate));
}
if (Ds_use_eax) {
ALCint major = 0, minor = 0, max_sends = 0;
alcGetIntegerv(ds_sound_device, ALC_EFX_MAJOR_VERSION, 1, &major);
alcGetIntegerv(ds_sound_device, ALC_EFX_MINOR_VERSION, 1, &minor);
alcGetIntegerv(ds_sound_device, ALC_MAX_AUXILIARY_SENDS, 1, &max_sends);
mprintf((" EFX version: %d.%d\n", (int)major, (int)minor));
mprintf((" Max auxiliary sends: %d\n", max_sends));
} else {
mprintf((" EFX enabled: NO\n"));
}
mprintf((" Playback device: %s\n", playback_device.c_str()));
mprintf((" Capture device: %s\n", (capture_device.empty()) ? "<not available>" : capture_device.c_str()));
mprintf(("... OpenAL successfully initialized!\n"));
// we need to clear out any errors before moving on
alcGetError(NULL);
alGetError();
return 0;
AL_InitError:
alcMakeContextCurrent(NULL);
if (ds_sound_context != NULL) {
alcDestroyContext(ds_sound_context);
ds_sound_context = NULL;
}
if (ds_sound_device != NULL) {
alcCloseDevice(ds_sound_device);
ds_sound_device = NULL;
}
mprintf(("... OpenAL failed to initialize!\n"));
return -1;
}
/**
* Free a single channel
*/
void ds_close_channel(int i)
{
if ( (i < 0) || (i >= MAX_CHANNELS) ) {
return;
}
if ( (Channels[i].source_id != 0) && alIsSource(Channels[i].source_id) ) {
OpenAL_ErrorPrint( alSourceStop(Channels[i].source_id) );
OpenAL_ErrorPrint( alSourcei(Channels[i].source_id, AL_BUFFER, 0) );
if (Ds_eax_inited) {
OpenAL_ErrorPrint( alSource3i(Channels[i].source_id, AL_AUXILIARY_SEND_FILTER, AL_EFFECTSLOT_NULL, 0, AL_FILTER_NULL) );
}
OpenAL_ErrorPrint( alDeleteSources(1, &Channels[i].source_id) );
if (Channels[i].sid >= 0) {
sound_buffers[Channels[i].sid].channel_id = -1;
}
Channels[i].source_id = 0;
Channels[i].sid = -1;
Channels[i].sig = -1;
Channels[i].snd_id = -1;
}
}
void ds_close_channel_fast(int i)
{
if ( (i < 0) || (i >= MAX_CHANNELS) ) {
return;
}
if ( (Channels[i].source_id != 0) && alIsSource(Channels[i].source_id) ) {
OpenAL_ErrorPrint( alSourceStop(Channels[i].source_id) );
OpenAL_ErrorPrint( alSourcei(Channels[i].source_id, AL_BUFFER, 0) );
if (Ds_eax_inited) {
OpenAL_ErrorPrint( alSource3i(Channels[i].source_id, AL_AUXILIARY_SEND_FILTER, AL_EFFECTSLOT_NULL, 0, AL_FILTER_NULL) );
}
if (Channels[i].sid >= 0) {
sound_buffers[Channels[i].sid].channel_id = -1;
}
Channels[i].sid = -1;
Channels[i].sig = -1;
Channels[i].snd_id = -1;
}
}
/**
* Free all the channel buffers
*/
void ds_close_all_channels()
{
int i;
for (i = 0; i < MAX_CHANNELS; i++) {
ds_close_channel(i);
}
}
/**
* Unload a buffer
*/
void ds_unload_buffer(int sid)
{
if ( (sid < 0) || ((size_t)sid >= sound_buffers.size()) ) {
return;
}
if (sound_buffers[sid].channel_id >= 0) {
ds_close_channel_fast(sound_buffers[sid].channel_id);
sound_buffers[sid].channel_id = -1;
}
ALuint buf_id = sound_buffers[sid].buf_id;
if ( (buf_id != 0) && alIsBuffer(buf_id) ) {
OpenAL_ErrorCheck( alDeleteBuffers(1, &buf_id), return );
}
sound_buffers[sid].buf_id = 0;
}
/**
* Unload all the channel buffers
*/
void ds_close_buffers()
{
size_t i;
for (i = 0; i < sound_buffers.size(); i++) {
ds_unload_buffer(i);
}
sound_buffers.clear();
}
/**
* Close the sound system
*/
void ds_close()
{
ds_close_all_channels();
ds_close_buffers();
ds_eax_close();
// free the Channels[] array, since it was dynamically allocated
delete [] Channels;
Channels = NULL;
alcMakeContextCurrent(NULL); // hangs on me for some reason
if (ds_sound_context != NULL) {
alcDestroyContext(ds_sound_context);
ds_sound_context = NULL;
}
if (ds_sound_device != NULL) {
alcCloseDevice(ds_sound_device);
ds_sound_device = NULL;
}
}
/**
* Find a free channel to play a sound on. If no free channels exists, free up one based on volume levels.
*
* @param new_volume Volume for sound to play at
* @param snd_id Which kind of sound to play
* @param priority ::DS_MUST_PLAY, ::DS_LIMIT_ONE, ::DS_LIMIT_TWO, ::DS_LIMIT_THREE
*
* @returns Channel number to play sound on, or -1 if no channel could be found
*
* NOTE: snd_id is needed since we limit the number of concurrent samples
*/
int ds_get_free_channel(float new_volume, int snd_id, int priority)
{
int i, first_free_channel, limit = 100;
int instance_count; // number of instances of sound already playing
int lowest_vol_index = -1, lowest_instance_vol_index = -1;
float lowest_vol = 1.0f, lowest_instance_vol = 1.0f;
channel *chp;
int status;
instance_count = 0;
first_free_channel = -1;
// determine the limit of concurrent instances of this sound
switch (priority) {
case DS_MUST_PLAY:
limit = 100;
break;
case DS_LIMIT_ONE:
limit = 1;
break;
case DS_LIMIT_TWO:
limit = 2;
break;
case DS_LIMIT_THREE:
limit = 3;
break;
default:
Int3(); // get Alan
limit = 100;
break;
}
// Look for a channel to use to play this sample
for ( i = 0; i < MAX_CHANNELS; i++ ) {
chp = &Channels[i];
// source not created yet
if (chp->source_id == 0) {
if (first_free_channel == -1) {
first_free_channel = i;
}
continue;
}
// source not bound to a buffer
if (chp->sid == -1) {
if (first_free_channel == -1) {
first_free_channel = i;
}
continue;
}
OpenAL_ErrorCheck( alGetSourcei(chp->source_id, AL_SOURCE_STATE, &status), continue );
if ( (status == AL_INITIAL) || (status == AL_STOPPED) ) {
ds_close_channel_fast(i);
if (first_free_channel == -1) {
first_free_channel = i;
}
continue;
} else {
if ( chp->snd_id == snd_id ) {
instance_count++;
if ( (chp->vol < lowest_instance_vol) && (chp->looping == FALSE) ) {
lowest_instance_vol = chp->vol;
lowest_instance_vol_index = i;
}
} else if ( chp->is_voice_msg ) {
// a playing voice message is not allowed to be preempted.
} else if ( (chp->vol < lowest_vol) && (chp->looping == FALSE) ) {
lowest_vol_index = i;
lowest_vol = chp->vol;
}
}
}
// If we've exceeded the limit, then maybe stop the duplicate if it is lower volume
if ( (instance_count >= limit) && (lowest_instance_vol_index >= 0) ) {
// If there is a lower volume duplicate, stop it.... otherwise, don't play the sound
if (lowest_instance_vol <= new_volume) {
ds_close_channel_fast(lowest_instance_vol_index);
first_free_channel = lowest_instance_vol_index;
} else {
// NOTE: yes we are preventing the sound from playing even if
// there is an available channel because we are over the limit
// requested by the rest of the engine, which means if we do
// not honour its request to limit the count then the engine
// will trip itself up by using all channels without having
// the intention of actually doing so. This means we get
// very loud sounds, missing more important sounds, etc.
// Effectivly the problem is the rest of the engine assumes
// it is still stuck in the 90s with a sound card that only has
// <=16 channels so we need to give it a sound card that
// has 16 channels (though we are actually allowing 32 channels
// just because we can).
first_free_channel = -1;
}
} else if (first_free_channel == -1) {
// there is no limit barrier to play the sound, but we have run out of channels
// stop the lowest volume instance to play our sound if priority demands it
if ( (lowest_vol_index != -1) && (priority == DS_MUST_PLAY) ) {
// Check if the lowest volume playing is less than the volume of the requested sound.
// If so, then we are going to trash the lowest volume sound.
if ( Channels[lowest_vol_index].vol <= new_volume ) {
ds_close_channel_fast(lowest_vol_index);
first_free_channel = lowest_vol_index;
}
}
}
if ( (first_free_channel >= 0) && (Channels[first_free_channel].source_id == 0) ) {
OpenAL_ErrorCheck( alGenSources(1, &Channels[first_free_channel].source_id), return -1 );
}
return first_free_channel;
}
/**
* Create a sound buffer in software, without locking any data in
*/
int ds_create_buffer(int frequency, int bits_per_sample, int nchannels, int nseconds)
{
ALuint i;
int sid;
if (!ds_initialized) {
return -1;
}
sid = ds_get_sid();
if ( sid == -1 ) {
nprintf(("Sound","SOUND ==> No more OpenAL buffers available\n"));
return -1;
}
OpenAL_ErrorCheck( alGenBuffers(1, &i), return -1 );
sound_buffers[sid].buf_id = i;
sound_buffers[sid].channel_id = -1;
sound_buffers[sid].frequency = frequency;
sound_buffers[sid].bits_per_sample = bits_per_sample;
sound_buffers[sid].nchannels = nchannels;
sound_buffers[sid].nseconds = nseconds;
sound_buffers[sid].nbytes = nseconds * (bits_per_sample / 8) * nchannels * frequency;
return sid;
}
/**
* Lock data into an existing buffer
*/
int ds_lock_data(int sid, unsigned char *data, int size)
{
if ( (sid < 0) || ((size_t)sid >= sound_buffers.size()) ) {
return -1;
}
ALenum format = openal_get_format(sound_buffers[sid].bits_per_sample, sound_buffers[sid].nchannels);
if (format == AL_INVALID_VALUE) {
return -1;
}
sound_buffers[sid].nbytes = size;
OpenAL_ErrorCheck( alBufferData(sound_buffers[sid].buf_id, format, data, size, sound_buffers[sid].frequency), return -1 );
return 0;
}
/**
* Stop a buffer from playing directly
*/
void ds_stop_easy(int sid)
{
Assert(sid >= 0);
int cid = sound_buffers[sid].channel_id;
if (cid != -1) {
ALuint source_id = Channels[cid].source_id;
OpenAL_ErrorPrint( alSourceStop(source_id) );
}
}
/**
* Play a sound without the usual baggage (used for playing back real-time voice)
*
* @param sid Software id of sound
* @param volume Volume of sound effect in linear scale
*/
int ds_play_easy(int sid, float volume)
{
if (!ds_initialized) {
return -1;
}
int ch_idx = ds_get_free_channel(volume, -1, DS_MUST_PLAY);
if (ch_idx < 0) {
return -1;
}
ALuint source_id = Channels[ch_idx].source_id;
OpenAL_ErrorPrint( alSourceStop(source_id) );
if (Channels[ch_idx].sid != sid) {
ALuint buffer_id = sound_buffers[sid].buf_id;
OpenAL_ErrorCheck( alSourcei(source_id, AL_BUFFER, buffer_id), return -1 );
}
Channels[ch_idx].sid = sid;
OpenAL_ErrorPrint( alSourcef(source_id, AL_GAIN, volume) );
OpenAL_ErrorPrint( alSourcei(source_id, AL_LOOPING, AL_FALSE) );
OpenAL_ErrorPrint( alSourcei(source_id, AL_SOURCE_RELATIVE, AL_TRUE) );
OpenAL_ErrorPrint( alSource3f(Channels[ch_idx].source_id, AL_POSITION, 0.0f, 0.0f, 0.0f) );
OpenAL_ErrorPrint( alSource3f(Channels[ch_idx].source_id, AL_VELOCITY, 0.0f, 0.0f, 0.0f) );
OpenAL_ErrorPrint( alDopplerFactor(0.0f) );
OpenAL_ErrorPrint( alSourcePlay(source_id) );
return 0;
}
/**
* Play a sound secondary buffer.
*
* @param sid Software id of sound
* @param snd_id What kind of sound this is
* @param priority ::DS_MUST_PLAY, ::DS_LIMIT_ONE, ::DS_LIMIT_TWO, ::DS_LIMIT_THREE
* @param volume Volume of sound effect in DirectSound units
* @param pan Pan of sound in sound units
* @param looping Whether the sound effect is looping or not
* @param is_voice_msg If a voice message
*
* @return 1 if sound effect could not be started, >=0 sig for sound effect successfully started
*/
int ds_play(int sid, int snd_id, int priority, float volume, float pan, int looping, bool is_voice_msg)
{
int ch_idx;
if (!ds_initialized) {
return -1;
}
ch_idx = ds_get_free_channel(volume, snd_id, priority);
if (ch_idx < 0) {
return -1;
}
if (Channels[ch_idx].source_id == 0) {
return -1;
}
OpenAL_ErrorPrint( alSource3f(Channels[ch_idx].source_id, AL_POSITION, pan, 0.0f, 0.0f) );
OpenAL_ErrorPrint( alSource3f(Channels[ch_idx].source_id, AL_VELOCITY, 0.0f, 0.0f, 0.0f) );
OpenAL_ErrorPrint( alDopplerFactor(0.0f) );
OpenAL_ErrorPrint( alSourcef(Channels[ch_idx].source_id, AL_PITCH, 1.0f) );
OpenAL_ErrorPrint( alSourcef(Channels[ch_idx].source_id, AL_GAIN, volume) );
ALint status;
OpenAL_ErrorCheck( alGetSourcei(Channels[ch_idx].source_id, AL_SOURCE_STATE, &status), return -1 );
if (status == AL_PLAYING) {
OpenAL_ErrorPrint( alSourceStop(Channels[ch_idx].source_id) );
}
OpenAL_ErrorCheck( alSourcei(Channels[ch_idx].source_id, AL_BUFFER, sound_buffers[sid].buf_id), return -1 );
OpenAL_ErrorPrint( alSourcei(Channels[ch_idx].source_id, AL_SOURCE_RELATIVE, AL_TRUE) );
OpenAL_ErrorPrint( alSourcei(Channels[ch_idx].source_id, AL_LOOPING, (looping) ? AL_TRUE : AL_FALSE) );
if (Ds_eax_inited) {
OpenAL_ErrorPrint( alSource3i(Channels[ch_idx].source_id, AL_AUXILIARY_SEND_FILTER, AL_EFX_aux_id, 0, AL_FILTER_NULL) );
}
OpenAL_ErrorPrint( alSourcePlay(Channels[ch_idx].source_id) );
sound_buffers[sid].channel_id = ch_idx;
Channels[ch_idx].sid = sid;
Channels[ch_idx].snd_id = snd_id;
Channels[ch_idx].sig = channel_next_sig++;
Channels[ch_idx].last_position = 0;
Channels[ch_idx].is_voice_msg = is_voice_msg;
Channels[ch_idx].vol = volume;
Channels[ch_idx].looping = looping;
Channels[ch_idx].priority = priority;
if (channel_next_sig < 0) {
channel_next_sig = 1;
}
return Channels[ch_idx].sig;
}
/**
* Return the channel number that is playing the sound identified by sig.
* @return Channel number, if not playing, return -1.
*/
int ds_get_channel(int sig)
{
int i;
for ( i = 0; i < MAX_CHANNELS; i++ ) {
if ( Channels[i].source_id && (Channels[i].sig == sig) ) {
if ( ds_is_channel_playing(i) == TRUE ) {
return i;
}
}
}
return -1;
}
/**
* @todo Documentation
*/
int ds_is_channel_playing(int channel)
{
if ( Channels[channel].source_id != 0 ) {
ALint status;
OpenAL_ErrorPrint( alGetSourcei(Channels[channel].source_id, AL_SOURCE_STATE, &status) );
return (status == AL_PLAYING);
}
return 0;
}
/**
* @todo Documentation
*/
void ds_stop_channel(int channel)
{
if ( Channels[channel].source_id != 0 ) {
OpenAL_ErrorPrint( alSourceStop(Channels[channel].source_id) );
}
}
/**
* @todo Documentation
*/
void ds_stop_channel_all()
{
int i;
for ( i=0; i<MAX_CHANNELS; i++ ) {
if ( Channels[i].source_id != 0 ) {
OpenAL_ErrorPrint( alSourceStop(Channels[i].source_id) );
}
}
}
/**
* @brief Set the volume for a channel. The volume is expected to be in linear scale
* @details If the sound is a 3D sound buffer, this is like re-establishing the maximum volume.
*/
void ds_set_volume( int channel, float vol )
{
if ( (channel < 0) || (channel >= MAX_CHANNELS) ) {
return;
}
ALuint source_id = Channels[channel].source_id;
if (source_id != 0) {
CAP(vol, 0.0f, 1.0f);
OpenAL_ErrorPrint( alSourcef(source_id, AL_GAIN, vol) );
}
}
/**
* Set the pan for a channel. The pan is expected to be in DirectSound units
*/
void ds_set_pan( int channel, float pan )
{
if ( (channel < 0) || (channel >= MAX_CHANNELS) ) {
return;
}
ALint state;
OpenAL_ErrorCheck( alGetSourcei(Channels[channel].source_id, AL_SOURCE_STATE, &state), return );
if (state == AL_PLAYING) {
OpenAL_ErrorPrint( alSourcei(Channels[channel].source_id, AL_SOURCE_RELATIVE, AL_TRUE) );
OpenAL_ErrorPrint( alSource3f(Channels[channel].source_id, AL_POSITION, pan, 0.0f, 0.0f) );
}
}
/**
* Get the pitch of a channel
*/
int ds_get_pitch(int channel)
{
ALint status;
ALfloat alpitch = 0;
int pitch;
if ( (channel < 0) || (channel >= MAX_CHANNELS) ) {
return -1;
}
OpenAL_ErrorCheck( alGetSourcei(Channels[channel].source_id, AL_SOURCE_STATE, &status), return -1 );
if (status == AL_PLAYING) {
OpenAL_ErrorPrint( alGetSourcef(Channels[channel].source_id, AL_PITCH, &alpitch) );
}
// convert OpenAL values to DirectSound values and return
pitch = fl2i( pow(10.0, (alpitch + 2.0)) );
return pitch;
}
/**
* Set the pitch of a channel
*/
void ds_set_pitch(int channel, int pitch)
{
ALint status;
if ( (channel < 0) || (channel >= MAX_CHANNELS) ) {
return;
}
if ( pitch < MIN_PITCH ) {
pitch = MIN_PITCH;
}
if ( pitch > MAX_PITCH ) {
pitch = MAX_PITCH;
}
OpenAL_ErrorCheck( alGetSourcei(Channels[channel].source_id, AL_SOURCE_STATE, &status), return );
if (status == AL_PLAYING) {
ALfloat alpitch = log10f((float)pitch) - 2.0f;
OpenAL_ErrorPrint( alSourcef(Channels[channel].source_id, AL_PITCH, alpitch) );
}
}
/**
* @todo Documentation
*/
void ds_chg_loop_status(int channel, int loop)
{
if ( (channel < 0) || (channel >= MAX_CHANNELS) ) {
return;
}
ALuint source_id = Channels[channel].source_id;
OpenAL_ErrorPrint( alSourcei(source_id, AL_LOOPING, loop ? AL_TRUE : AL_FALSE) );
}
/**
* Starts a ds3d sound playing
*
* @param sid Software id for sound to play
* @param snd_id Identifies what type of sound is playing
* @param pos World pos of sound
* @param vel Velocity of object emitting sound
* @param min Distance at which sound doesn't get any louder
* @param max Distance at which sound becomes inaudible
* @param looping Whether to loop the sound or not
* @param max_volume Volume (0 to 1) for 3d sound at maximum
* @param estimated_vol Manual estimated volume
* @param priority ::DS_MUST_PLAY, ::DS_LIMIT_ONE, ::DS_LIMIT_TWO, ::DS_LIMIT_THREE
*
* @return 0 if sound started successfully, -1 if sound could not be played
*/
int ds3d_play(int sid, int snd_id, vec3d *pos, vec3d *vel, float min, float max, int looping, float max_volume, float estimated_vol, int priority )
{
int channel;
if (!ds_initialized) {
return -1;
}
channel = ds_get_free_channel(estimated_vol, snd_id, priority);
if (channel < 0) {
return -1;
}
if ( Channels[channel].source_id == 0 ) {
return -1;
}
// set up 3D sound data here
ds3d_update_buffer(channel, min, max, pos, vel);
OpenAL_ErrorPrint( alSourcef(Channels[channel].source_id, AL_PITCH, 1.0f) );
OpenAL_ErrorPrint( alSourcef(Channels[channel].source_id, AL_GAIN, max_volume) );
ALint status;
OpenAL_ErrorCheck( alGetSourcei(Channels[channel].source_id, AL_SOURCE_STATE, &status), return -1 );
if (status == AL_PLAYING) {
OpenAL_ErrorPrint( alSourceStop(Channels[channel].source_id) );
}
OpenAL_ErrorCheck( alSourcei(Channels[channel].source_id, AL_BUFFER, sound_buffers[sid].buf_id), return -1 );
if (Ds_eax_inited) {
OpenAL_ErrorPrint( alSource3i(Channels[channel].source_id, AL_AUXILIARY_SEND_FILTER, AL_EFX_aux_id, 0, AL_FILTER_NULL) );
}
OpenAL_ErrorPrint( alSourcei(Channels[channel].source_id, AL_SOURCE_RELATIVE, AL_FALSE) );
OpenAL_ErrorPrint( alSourcei(Channels[channel].source_id, AL_LOOPING, (looping) ? AL_TRUE : AL_FALSE) );
OpenAL_ErrorPrint( alSourcePlay(Channels[channel].source_id) );
sound_buffers[sid].channel_id = channel;
Channels[channel].sid = sid;
Channels[channel].snd_id = snd_id;
Channels[channel].sig = channel_next_sig++;
Channels[channel].last_position = 0;
Channels[channel].is_voice_msg = false;
Channels[channel].vol = max_volume;
Channels[channel].looping = looping;
Channels[channel].priority = priority;
if (channel_next_sig < 0 ) {
channel_next_sig = 1;
}
return Channels[channel].sig;
}
/**
* @todo Documentation
*/
void ds_set_position(int channel, unsigned int offset)
{
if ( (channel < 0) || (channel >= MAX_CHANNELS) ) {
return;
}
OpenAL_ErrorPrint( alSourcei(Channels[channel].source_id, AL_BYTE_OFFSET, offset) );
}
/**
* @todo Documentation
*/
unsigned int ds_get_play_position(int channel)
{
if ( (channel < 0) || (channel >= MAX_CHANNELS) ) {
return 0;
}
ALint pos = -1;
int sid = Channels[channel].sid;
if ( (sid < 0) || ((size_t)sid >= sound_buffers.size()) ) {
return 0;
}
if (AL_play_position) {
OpenAL_ErrorPrint( alGetSourcei(Channels[channel].source_id, AL_BYTE_LOKI, &pos) );
if ( pos < 0 ) {
pos = 0;
} else if ( pos > 0 ) {
// AL_BYTE_LOKI returns position in canon format which may differ
// from our sample, so we may have to scale it
ALuint buf_id = sound_buffers[sid].buf_id;
ALint size;
OpenAL_ErrorCheck( alGetBufferi(buf_id, AL_SIZE, &size), return 0 );
pos = (ALint)(pos * ((float)sound_buffers[sid].nbytes / size));
}
} else {
OpenAL_ErrorPrint( alGetSourcei(Channels[channel].source_id, AL_BYTE_OFFSET, &pos) );
if (pos < 0) {
pos = 0;
}
}
return (unsigned int) pos;
}
/**
* @todo Documentation
*/
unsigned int ds_get_write_position(int channel)
{
return 0;
}
/**
* @todo Documentation
*/
int ds_get_channel_size(int channel)
{
if ( (channel < 0) || (channel >= MAX_CHANNELS) ) {
return 0;
}
int sid = Channels[channel].sid;
if ( (sid < 0) || ((size_t)sid >= sound_buffers.size()) ) {
return 0;
}
ALuint buf_id = sound_buffers[sid].buf_id;
ALint data_size = 0;
if ( (buf_id != 0) && alIsBuffer(buf_id)) {
OpenAL_ErrorPrint( alGetBufferi(buf_id, AL_SIZE, &data_size) );
}
return (int) data_size;
}
/**
* Returns the number of channels that are actually playing
*/
int ds_get_number_channels()
{
int i,n;
if (!ds_initialized) {
return 0;
}
n = 0;
for ( i = 0; i < MAX_CHANNELS; i++ ) {
if ( Channels[i].source_id ) {
if ( ds_is_channel_playing(i) == TRUE ) {
n++;
}
}
}
return n;
}
/**
* Retreive raw data from a sound buffer
*/
int ds_get_data(int sid, char *data)
{
return -1;
}
/**
* Return the size of the raw sound data
*/
int ds_get_size(int sid, int *size)
{
Assert(sid >= 0);
if ( (sid < 0) || ((size_t)sid >= sound_buffers.size()) ) {
return 0;
}
ALuint buf_id = sound_buffers[sid].buf_id;
ALint data_size = 0;
if ( (buf_id != 0) && alIsBuffer(buf_id)) {
OpenAL_ErrorPrint( alGetBufferi(buf_id, AL_SIZE, &data_size) );
if (size) {
*size = (int) data_size;
}
return 0;
}
return -1;
}
// --------------------
//
// EAX Functions below
//
// --------------------
/**
* Set the master volume for the reverb added to all sound sources.
*
* @param volume Volume, range from 0 to 1.0
* @returns 0 if the volume is set successfully, otherwise return -1
*/
int ds_eax_set_volume(float volume)
{
if ( !Ds_eax_inited ) {
return -1;
}
CAP(volume, 0.0f, 1.0f);
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_GAIN, volume) );
OpenAL_ErrorCheck( v_alAuxiliaryEffectSloti(AL_EFX_aux_id, AL_EFFECTSLOT_EFFECT, AL_EFX_effect_id), return -1 );
return 0;
}
/**
* Set the decay time for the EAX environment (ie all sound sources)
*
* @param seconds Decay time in seconds
* @return 0 if decay time is successfully set, otherwise return -1
*/
int ds_eax_set_decay_time(float seconds)
{
if ( !Ds_eax_inited ) {
return -1;
}
CAP(seconds, 0.1f, 20.0f);
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_DECAY_TIME, seconds) );
OpenAL_ErrorCheck( v_alAuxiliaryEffectSloti(AL_EFX_aux_id, AL_EFFECTSLOT_EFFECT, AL_EFX_effect_id), return -1 );
return 0;
}
/**
* Set the damping value for the EAX environment (ie all sound sources)
*
* @param damp Damp value from 0 to 2.0
* @return 0 if the damp value is successfully set, otherwise return -1
*/
int ds_eax_set_damping(float damp)
{
if ( !Ds_eax_inited ) {
return -1;
}
CAP(damp, 0.1f, 2.0f);
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_DECAY_HFRATIO, damp) );
OpenAL_ErrorCheck( v_alAuxiliaryEffectSloti(AL_EFX_aux_id, AL_EFFECTSLOT_EFFECT, AL_EFX_effect_id), return -1 );
return 0;
}
/**
* Set up the environment type for all sound sources.
*
* @param envid Value from the EAX_ENVIRONMENT_* enumeration in ds_eax.h
* @return Always returns 0.
* @todo Proper error reporting, otherwise make a void return type.
*/
int ds_eax_set_environment(unsigned long envid)
{
al_efx_load_preset(envid);
return 0;
}
/**
* Set up all the parameters for an environment
*
* @param id Value from the EAX_ENVIRONMENT_* enumeration
* @param vol Volume for the environment (0 to 1.0)
* @param damping Damp value for the environment (0 to 2.0)
* @param decay Decay time in seconds (0.1 to 20.0)
* @return 0 if successful, otherwise return -1
*/
int ds_eax_set_all(unsigned long id, float vol, float damping, float decay)
{
if ( !Ds_eax_inited ) {
return -1;
}
// special disabled case
if ( (id == EAX_ENVIRONMENT_GENERIC) && (vol == 0.0f) && (damping == 0.0f) && (decay == 0.0f) ) {
v_alAuxiliaryEffectSloti(AL_EFX_aux_id, AL_EFFECTSLOT_EFFECT, AL_EFFECT_NULL);
Ds_active_env = -1;
return 0;
}
al_efx_load_preset(id);
CAP(vol, 0.0f, 1.0f);
CAP(decay, 0.1f, 20.0f);
CAP(damping, 0.1f, 2.0f);
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_GAIN, vol) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_DECAY_TIME, decay) );
OpenAL_ErrorPrint( v_alEffectf(AL_EFX_effect_id, AL_EAXREVERB_DECAY_HFRATIO, damping) );
OpenAL_ErrorCheck( v_alAuxiliaryEffectSloti(AL_EFX_aux_id, AL_EFFECTSLOT_EFFECT, AL_EFX_effect_id), return -1 );
return 0;
}
int ds_eax_get_preset_id(const char *name)
{
if ( !name || !strlen(name) ) {
return -1;
}
size_t count = EFX_presets.size();
for (size_t i = 0; i < count; i++) {
if ( !stricmp(name, EFX_presets[i].name.c_str()) ) {
return i;
}
}
return -1;
}
int ds_eax_get_prop(EFXREVERBPROPERTIES **props, const char *name, const char *template_name)
{
Assert( props != NULL );
Assert( name != NULL );
Assert( strlen(name) > 0 ); //-V805
int template_id = -1;
int id = ds_eax_get_preset_id(name);
if (id >= 0) {
*props = &EFX_presets[id];
} else {
id = EFX_presets.size();
EFXREVERBPROPERTIES n_prop;
if ( (template_name != NULL) && (template_name[0] != '\0') ) {
template_id = ds_eax_get_preset_id(template_name);
}
if (template_id >= 0) {
n_prop = EFX_presets[template_id];
n_prop.name = name;
} else {
n_prop.name = name;
n_prop.flDensity = 1.0f;
n_prop.flDiffusion = 1.0f;
n_prop.flGain = 0.32f;
n_prop.flGainHF = 0.89f;
n_prop.flGainLF = 0.0f;
n_prop.flDecayTime = 1.49f;
n_prop.flDecayHFRatio = 0.83f;
n_prop.flDecayLFRatio = 1.0f;
n_prop.flReflectionsGain = 0.05f;
n_prop.flReflectionsDelay = 0.007f;
n_prop.flReflectionsPan[0] = 0.0f;
n_prop.flReflectionsPan[1] = 0.0f;
n_prop.flReflectionsPan[2] = 0.0f;
n_prop.flLateReverbGain = 1.26f;
n_prop.flLateReverbDelay = 0.011f;
n_prop.flLateReverbPan[0] = 0.0f;
n_prop.flLateReverbPan[1] = 0.0f;
n_prop.flLateReverbPan[2] = 0.0f;
n_prop.flEchoTime = 0.25f;
n_prop.flEchoDepth = 0.0f;
n_prop.flModulationTime = 0.25f;
n_prop.flModulationDepth = 0.0f;
n_prop.flAirAbsorptionGainHF = 0.994f;
n_prop.flHFReference = 5000.0f;
n_prop.flLFReference = 250.0f;
n_prop.flRoomRolloffFactor = 0.0f;
n_prop.iDecayHFLimit = AL_TRUE;
}
EFX_presets.push_back( n_prop );
*props = &EFX_presets[id];
}
if ( !stricmp(name, "default") ) {
extern unsigned int SND_ENV_DEFAULT;
SND_ENV_DEFAULT = id;
}
return 0;
}
/**
* Get up the parameters for the current environment
*
* @param er (output) Hold environment parameters
* @param id If set will get specified preset env, otherwise current env
* @return 0 if successful, otherwise return -1
*/
int ds_eax_get_all(EAX_REVERBPROPERTIES *er, int id)
{
if ( !er ) {
return -1;
}
if (id < 0) {
if (Ds_active_env < 0) {
return -1;
}
er->environment = Ds_active_env;
OpenAL_ErrorPrint( v_alGetEffectf(AL_EFX_effect_id, AL_EAXREVERB_GAIN, &er->fVolume) );
OpenAL_ErrorPrint( v_alGetEffectf(AL_EFX_effect_id, AL_EAXREVERB_DECAY_TIME, &er->fDecayTime_sec) );
OpenAL_ErrorPrint( v_alGetEffectf(AL_EFX_effect_id, AL_EAXREVERB_DECAY_HFRATIO, &er->fDamping) );
} else if (id < (int)EFX_presets.size()) {
er->environment = (unsigned int)id;
er->fVolume = EFX_presets[id].flGain;
er->fDecayTime_sec = EFX_presets[id].flDecayTime;
er->fDamping = EFX_presets[id].flDecayHFRatio;
} else {
return -1;
}
return 0;
}
/**
* Close down EAX, freeing any allocated resources
*/
void ds_eax_close()
{
if (Ds_eax_inited == 0) {
return;
}
v_alAuxiliaryEffectSloti(AL_EFX_aux_id, AL_EFFECTSLOT_EFFECT, AL_EFFECT_NULL);
v_alDeleteEffects(1, &AL_EFX_effect_id);
AL_EFX_effect_id = 0;
v_alDeleteAuxiliaryEffectSlots(1, &AL_EFX_aux_id);
AL_EFX_aux_id = 0;
Ds_eax_inited = 0;
}
/**
* Initialize EAX
* @return 0 if initialization is successful, otherwise return -1
*/
int ds_eax_init()
{
if (Ds_eax_inited) {
return 0;
}
try {
v_alGenFilters = (ALGENFILTERS) al_load_function("alGenFilters");
v_alDeleteFilters = (ALDELETEFILTERS) al_load_function("alDeleteFilters");
v_alFilteri = (ALFILTERI) al_load_function("alFilteri");
v_alGenEffecs = (ALGENEFFECTS) al_load_function("alGenEffects");
v_alDeleteEffects = (ALDELETEEFFECTS) al_load_function("alDeleteEffects");
v_alEffecti = (ALEFFECTI) al_load_function("alEffecti");
v_alEffectf = (ALEFFECTF) al_load_function("alEffectf");
v_alEffectfv = (ALEFFECTFV) al_load_function("alEffectfv");
v_alGetEffectf = (ALGETEFFECTF) al_load_function("alGetEffectf");
v_alGenAuxiliaryEffectSlots = (ALGENAUXILIARYEFFECTSLOTS) al_load_function("alGenAuxiliaryEffectSlots");
v_alDeleteAuxiliaryEffectSlots = (ALDELETEAUXILIARYEFFECTSLOTS) al_load_function("alDeleteAuxiliaryEffectSlots");
v_alIsAuxiliaryEffectSlot = (ALISAUXILIARYEFFECTSLOT) al_load_function("alIsAuxiliaryEffectSlot");
v_alAuxiliaryEffectSloti = (ALAUXILIARYEFFECTSLOTI) al_load_function("alAuxiliaryEffectSloti");
v_alAuxiliaryEffectSlotiv = (ALAUXILIARYEFFECTSLOTIV) al_load_function("alAuxiliaryEffectSlotiv");
v_alAuxiliaryEffectSlotf = (ALAUXILIARYEFFECTSLOTF) al_load_function("alAuxiliaryEffectSlotf");
v_alAuxiliaryEffectSlotfv = (ALAUXILIARYEFFECTSLOTFV) al_load_function("alAuxiliaryEffectSlotfv");
} catch (const char *err) {
mprintf(("\n EFX: Unable to load function: %s()\n", err));
Ds_eax_inited = 0;
return -1;
}
v_alGenAuxiliaryEffectSlots(1, &AL_EFX_aux_id);
if (alGetError() != AL_NO_ERROR) {
mprintf(("\n EFX: Unable to create Aux effect!\n"));
return -1;
}
v_alGenEffecs(1, &AL_EFX_effect_id);
if (alGetError() != AL_NO_ERROR) {
mprintf(("\n EFX: Unable to create effect!\n"));
return -1;
}
v_alEffecti(AL_EFX_effect_id, AL_EFFECT_TYPE, AL_EFFECT_EAXREVERB);
if (alGetError() != AL_NO_ERROR) {
mprintf(("\n EFX: EAXReverb not supported!\n"));
return -1;
}
v_alAuxiliaryEffectSloti(AL_EFX_aux_id, AL_EFFECTSLOT_EFFECT, AL_EFX_effect_id);
if (alGetError() != AL_NO_ERROR) {
mprintf(("\n EFX: Couldn't load effect!\n"));
return -1;
}
// add default presets
EFX_presets.reserve(EAX_ENVIRONMENT_COUNT);
for (int i = 0; i < EAX_ENVIRONMENT_COUNT; i++) {
EFX_presets.push_back( EFX_Reverb_Defaults[i] );
}
Ds_eax_inited = 1;
// disabled by default
ds_eax_set_all(EAX_ENVIRONMENT_GENERIC, 0.0f, 0.0f, 0.0f);
return 0;
}
/**
* @todo Documentation
*/
int ds_eax_is_inited()
{
return Ds_eax_inited;
}
/**
* Called once per game frame to make sure voice messages aren't looping
*/
void ds_do_frame()
{
if (!ds_initialized) {
return;
}
int i;
channel *cp = NULL;
for (i = 0; i < MAX_CHANNELS; i++) {
cp = &Channels[i];
Assert( cp != NULL );
if (cp->is_voice_msg) {
if( cp->source_id == 0 ) {
continue;
}
unsigned int current_position = ds_get_play_position(i);
if (current_position != 0) {
if (current_position < cp->last_position) {
ds_close_channel(i);
} else {
cp->last_position = current_position;
}
}
}
}
}
/**
* Given a valid channel return the sound id
*/
int ds_get_sound_id(int channel)
{
Assert( channel >= 0 );
return Channels[channel].snd_id;
}
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