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Merge pull request #123 from kd7tck/develop

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mod player
This commit is contained in:
Ray 2016-06-02 16:18:53 +02:00
commit 7447b3e1da
9 changed files with 1888 additions and 116 deletions
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4
CMakeLists.txt
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@ -8,7 +8,7 @@ set(CMAKE_C_FLAGS "-O1 -Wall -std=gnu99 -fgnu89-inline")
IF(${PLATFORM_TO_USE} MATCHES "PLATFORM_DESKTOP") IF(${PLATFORM_TO_USE} MATCHES "PLATFORM_DESKTOP")
add_definitions(-DPLATFORM_DESKTOP, -DGRAPHICS_API_OPENGL_33) add_definitions(-DPLATFORM_DESKTOP, -DGRAPHICS_API_OPENGL_33)
include_directories("." "src/" "external/openal_soft/include" "external/glew/include" "external/glfw3/include") include_directories("." "src/" "external/openal_soft/include" "external/glfw3/include")
ENDIF() ENDIF()
@ -22,7 +22,7 @@ ENDIF()
IF(${PLATFORM_TO_USE} MATCHES "PLATFORM_WEB") IF(${PLATFORM_TO_USE} MATCHES "PLATFORM_WEB")
add_definitions(-DPLATFORM_WEB, -GRAPHICS_API_OPENGL_ES2) add_definitions(-DPLATFORM_WEB, -GRAPHICS_API_OPENGL_ES2)
include_directories("." "src/" "external/openal_soft/include" "external/glew/include" "external/glfw3/include") include_directories("." "src/" "external/openal_soft/include" "external/glfw3/include")
ENDIF() ENDIF()

87
external/openal_soft/include/AL/alext.h vendored
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@ -13,8 +13,8 @@
* *
* You should have received a copy of the GNU Library General Public * You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the * License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Free Software Foundation, Inc.,
* Boston, MA 02111-1307, USA. * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html * Or go to http://www.gnu.org/copyleft/lgpl.html
*/ */
@ -348,6 +348,89 @@ AL_API void AL_APIENTRY alGetSourcei64vSOFT(ALuint source, ALenum param, ALint64
#endif #endif
#endif #endif
#ifndef ALC_EXT_DEFAULT_FILTER_ORDER
#define ALC_EXT_DEFAULT_FILTER_ORDER 1
#define ALC_DEFAULT_FILTER_ORDER 0x1100
#endif
#ifndef AL_SOFT_deferred_updates
#define AL_SOFT_deferred_updates 1
#define AL_DEFERRED_UPDATES_SOFT 0xC002
typedef ALvoid (AL_APIENTRY*LPALDEFERUPDATESSOFT)(void);
typedef ALvoid (AL_APIENTRY*LPALPROCESSUPDATESSOFT)(void);
#ifdef AL_ALEXT_PROTOTYPES
AL_API ALvoid AL_APIENTRY alDeferUpdatesSOFT(void);
AL_API ALvoid AL_APIENTRY alProcessUpdatesSOFT(void);
#endif
#endif
#ifndef AL_SOFT_block_alignment
#define AL_SOFT_block_alignment 1
#define AL_UNPACK_BLOCK_ALIGNMENT_SOFT 0x200C
#define AL_PACK_BLOCK_ALIGNMENT_SOFT 0x200D
#endif
#ifndef AL_SOFT_MSADPCM
#define AL_SOFT_MSADPCM 1
#define AL_FORMAT_MONO_MSADPCM_SOFT 0x1302
#define AL_FORMAT_STEREO_MSADPCM_SOFT 0x1303
#endif
#ifndef AL_SOFT_source_length
#define AL_SOFT_source_length 1
/*#define AL_BYTE_LENGTH_SOFT 0x2009*/
/*#define AL_SAMPLE_LENGTH_SOFT 0x200A*/
/*#define AL_SEC_LENGTH_SOFT 0x200B*/
#endif
#ifndef ALC_SOFT_pause_device
#define ALC_SOFT_pause_device 1
typedef void (ALC_APIENTRY*LPALCDEVICEPAUSESOFT)(ALCdevice *device);
typedef void (ALC_APIENTRY*LPALCDEVICERESUMESOFT)(ALCdevice *device);
#ifdef AL_ALEXT_PROTOTYPES
ALC_API void ALC_APIENTRY alcDevicePauseSOFT(ALCdevice *device);
ALC_API void ALC_APIENTRY alcDeviceResumeSOFT(ALCdevice *device);
#endif
#endif
#ifndef AL_EXT_BFORMAT
#define AL_EXT_BFORMAT 1
#define AL_FORMAT_BFORMAT2D_8 0x20021
#define AL_FORMAT_BFORMAT2D_16 0x20022
#define AL_FORMAT_BFORMAT2D_FLOAT32 0x20023
#define AL_FORMAT_BFORMAT3D_8 0x20031
#define AL_FORMAT_BFORMAT3D_16 0x20032
#define AL_FORMAT_BFORMAT3D_FLOAT32 0x20033
#endif
#ifndef AL_EXT_MULAW_BFORMAT
#define AL_EXT_MULAW_BFORMAT 1
#define AL_FORMAT_BFORMAT2D_MULAW 0x10031
#define AL_FORMAT_BFORMAT3D_MULAW 0x10032
#endif
#ifndef ALC_SOFT_HRTF
#define ALC_SOFT_HRTF 1
#define ALC_HRTF_SOFT 0x1992
#define ALC_DONT_CARE_SOFT 0x0002
#define ALC_HRTF_STATUS_SOFT 0x1993
#define ALC_HRTF_DISABLED_SOFT 0x0000
#define ALC_HRTF_ENABLED_SOFT 0x0001
#define ALC_HRTF_DENIED_SOFT 0x0002
#define ALC_HRTF_REQUIRED_SOFT 0x0003
#define ALC_HRTF_HEADPHONES_DETECTED_SOFT 0x0004
#define ALC_HRTF_UNSUPPORTED_FORMAT_SOFT 0x0005
#define ALC_NUM_HRTF_SPECIFIERS_SOFT 0x1994
#define ALC_HRTF_SPECIFIER_SOFT 0x1995
#define ALC_HRTF_ID_SOFT 0x1996
typedef const ALCchar* (ALC_APIENTRY*LPALCGETSTRINGISOFT)(ALCdevice *device, ALCenum paramName, ALCsizei index);
typedef ALCboolean (ALC_APIENTRY*LPALCRESETDEVICESOFT)(ALCdevice *device, const ALCint *attribs);
#ifdef AL_ALEXT_PROTOTYPES
ALC_API const ALCchar* ALC_APIENTRY alcGetStringiSOFT(ALCdevice *device, ALCenum paramName, ALCsizei index);
ALC_API ALCboolean ALC_APIENTRY alcResetDeviceSOFT(ALCdevice *device, const ALCint *attribs);
#endif
#endif
#ifdef __cplusplus #ifdef __cplusplus
} }
#endif #endif

2
external/openal_soft/include/AL/efx-presets.h vendored
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@ -345,7 +345,7 @@ typedef struct {
/* Driving Presets */ /* Driving Presets */
#define EFX_REVERB_PRESET_DRIVING_COMMENTATOR \ #define EFX_REVERB_PRESET_DRIVING_COMMENTATOR \
{ 1.0000f, 0.0000f, 3.1623f, 0.5623f, 0.5012f, 2.4200f, 0.8800f, 0.6800f, 0.1995f, 0.0930f, { 0.0000f, 0.0000f, 0.0000f }, 0.2512f, 0.0170f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9886f, 5000.0000f, 250.0000f, 0.0000f, 0x1 } { 1.0000f, 0.0000f, 0.3162f, 0.5623f, 0.5012f, 2.4200f, 0.8800f, 0.6800f, 0.1995f, 0.0930f, { 0.0000f, 0.0000f, 0.0000f }, 0.2512f, 0.0170f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9886f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_DRIVING_PITGARAGE \ #define EFX_REVERB_PRESET_DRIVING_PITGARAGE \
{ 0.4287f, 0.5900f, 0.3162f, 0.7079f, 0.5623f, 1.7200f, 0.9300f, 0.8700f, 0.5623f, 0.0000f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0160f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.1100f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 } { 0.4287f, 0.5900f, 0.3162f, 0.7079f, 0.5623f, 1.7200f, 0.9300f, 0.8700f, 0.5623f, 0.0000f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0160f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.1100f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }

BIN
external/openal_soft/lib/win32/libopenal32.a → external/openal_soft/lib/win32/libOpenAL32.dll.a vendored
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external/openal_soft/openal32.dll vendored
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273
src/audio.c
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@ -56,6 +56,9 @@
#define JAR_XM_IMPLEMENTATION #define JAR_XM_IMPLEMENTATION
#include "jar_xm.h" // XM loading functions #include "jar_xm.h" // XM loading functions
#define JAR_MOD_IMPLEMENTATION
#include "jar_mod.h" // For playing .mod files
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Defines and Macros // Defines and Macros
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
@ -95,10 +98,11 @@ typedef struct MixChannel_t {
// NOTE: Anything longer than ~10 seconds should be streamed into a mix channel... // NOTE: Anything longer than ~10 seconds should be streamed into a mix channel...
typedef struct Music { typedef struct Music {
stb_vorbis *stream; stb_vorbis *stream;
jar_xm_context_t *chipctx; // Stores jar_xm mixc jar_xm_context_t *xmctx; // Stores jar_xm mixc, XM chiptune context
jar_mod_context_t modctx; // Stores mod chiptune context
MixChannel_t *mixc; // mix channel MixChannel_t *mixc; // mix channel
int totalSamplesLeft; unsigned int totalSamplesLeft;
float totalLengthSeconds; float totalLengthSeconds;
bool loop; bool loop;
bool chipTune; // True if chiptune is loaded bool chipTune; // True if chiptune is loaded
@ -111,9 +115,9 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Global Variables Definition // Global Variables Definition
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
static MixChannel_t* mixChannelsActive_g[MAX_MIX_CHANNELS]; // What mix channels are currently active static MixChannel_t* mixChannels_g[MAX_MIX_CHANNELS]; // What mix channels are currently active
static bool musicEnabled_g = false; static bool musicEnabled_g = false;
static Music currentMusic[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same time static Music musicChannels_g[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same time
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Module specific Functions Declaration // Module specific Functions Declaration
@ -175,7 +179,7 @@ void CloseAudioDevice(void)
{ {
for(int index=0; index<MAX_MUSIC_STREAMS; index++) for(int index=0; index<MAX_MUSIC_STREAMS; index++)
{ {
if(currentMusic[index].mixc) StopMusicStream(index); // Stop music streaming and close current stream if(musicChannels_g[index].mixc) StopMusicStream(index); // Stop music streaming and close current stream
} }
@ -215,13 +219,13 @@ static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mix
if(mixChannel >= MAX_MIX_CHANNELS) return NULL; if(mixChannel >= MAX_MIX_CHANNELS) return NULL;
if(!IsAudioDeviceReady()) InitAudioDevice(); if(!IsAudioDeviceReady()) InitAudioDevice();
if(!mixChannelsActive_g[mixChannel]){ if(!mixChannels_g[mixChannel]){
MixChannel_t *mixc = (MixChannel_t*)malloc(sizeof(MixChannel_t)); MixChannel_t *mixc = (MixChannel_t*)malloc(sizeof(MixChannel_t));
mixc->sampleRate = sampleRate; mixc->sampleRate = sampleRate;
mixc->channels = channels; mixc->channels = channels;
mixc->mixChannel = mixChannel; mixc->mixChannel = mixChannel;
mixc->floatingPoint = floatingPoint; mixc->floatingPoint = floatingPoint;
mixChannelsActive_g[mixChannel] = mixc; mixChannels_g[mixChannel] = mixc;
// setup openAL format // setup openAL format
if(channels == 1) if(channels == 1)
@ -283,7 +287,7 @@ static void CloseMixChannel(MixChannel_t* mixc)
//delete source and buffers //delete source and buffers
alDeleteSources(1, &mixc->alSource); alDeleteSources(1, &mixc->alSource);
alDeleteBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer); alDeleteBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
mixChannelsActive_g[mixc->mixChannel] = NULL; mixChannels_g[mixc->mixChannel] = NULL;
free(mixc); free(mixc);
mixc = NULL; mixc = NULL;
} }
@ -294,7 +298,7 @@ static void CloseMixChannel(MixChannel_t* mixc)
// @Returns number of samples that where processed. // @Returns number of samples that where processed.
static int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements) static int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements)
{ {
if(!mixc || mixChannelsActive_g[mixc->mixChannel] != mixc) return 0; // when there is two channels there must be an even number of samples if(!mixc || mixChannels_g[mixc->mixChannel] != mixc) return 0; // when there is two channels there must be an even number of samples
if (!data || !numberElements) if (!data || !numberElements)
{ // pauses audio until data is given { // pauses audio until data is given
@ -376,35 +380,38 @@ static void ResampleByteToFloat(char *chars, float *floats, unsigned short len)
} }
} }
// used to output raw audio streams, returns negative numbers on error // used to output raw audio streams, returns negative numbers on error, + number represents the mix channel index
// if floating point is false the data size is 16bit short, otherwise it is float 32bit // if floating point is false the data size is 16bit short, otherwise it is float 32bit
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint) RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint)
{ {
int mixIndex; int mixIndex;
for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
{ {
if(mixChannelsActive_g[mixIndex] == NULL) break; if(mixChannels_g[mixIndex] == NULL) break;
else if(mixIndex == MAX_MIX_CHANNELS - 1) return -1; // error else if(mixIndex == MAX_MIX_CHANNELS - 1) return ERROR_OUT_OF_MIX_CHANNELS; // error
} }
if(InitMixChannel(sampleRate, mixIndex, channels, floatingPoint)) if(InitMixChannel(sampleRate, mixIndex, channels, floatingPoint))
return mixIndex; return mixIndex;
else else
return -2; // error return ERROR_RAW_CONTEXT_CREATION; // error
} }
void CloseRawAudioContext(RawAudioContext ctx) void CloseRawAudioContext(RawAudioContext ctx)
{ {
if(mixChannelsActive_g[ctx]) if(mixChannels_g[ctx])
CloseMixChannel(mixChannelsActive_g[ctx]); CloseMixChannel(mixChannels_g[ctx]);
} }
int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements) // if 0 is returned, the buffers are still full and you need to keep trying with the same data until a + number is returned.
// any + number returned is the number of samples that was processed and passed into buffer.
// data either needs to be array of floats or shorts.
int BufferRawAudioContext(RawAudioContext ctx, void *data, unsigned short numberElements)
{ {
int numBuffered = 0; int numBuffered = 0;
if(ctx >= 0) if(ctx >= 0)
{ {
MixChannel_t* mixc = mixChannelsActive_g[ctx]; MixChannel_t* mixc = mixChannels_g[ctx];
numBuffered = BufferMixChannel(mixc, data, numberElements); numBuffered = BufferMixChannel(mixc, data, numberElements);
} }
return numBuffered; return numBuffered;
@ -431,7 +438,10 @@ Sound LoadSound(char *fileName)
if (strcmp(GetExtension(fileName),"wav") == 0) wave = LoadWAV(fileName); if (strcmp(GetExtension(fileName),"wav") == 0) wave = LoadWAV(fileName);
else if (strcmp(GetExtension(fileName),"ogg") == 0) wave = LoadOGG(fileName); else if (strcmp(GetExtension(fileName),"ogg") == 0) wave = LoadOGG(fileName);
else TraceLog(WARNING, "[%s] Sound extension not recognized, it can't be loaded", fileName); else{
TraceLog(WARNING, "[%s] Sound extension not recognized, it can't be loaded", fileName);
sound.error = ERROR_EXTENSION_NOT_RECOGNIZED; //error
}
if (wave.data != NULL) if (wave.data != NULL)
{ {
@ -558,6 +568,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
if (rresFile == NULL) if (rresFile == NULL)
{ {
TraceLog(WARNING, "[%s] rRES raylib resource file could not be opened", rresName); TraceLog(WARNING, "[%s] rRES raylib resource file could not be opened", rresName);
sound.error = ERROR_UNABLE_TO_OPEN_RRES_FILE; //error
} }
else else
{ {
@ -572,6 +583,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
if ((id[0] != 'r') && (id[1] != 'R') && (id[2] != 'E') &&(id[3] != 'S')) if ((id[0] != 'r') && (id[1] != 'R') && (id[2] != 'E') &&(id[3] != 'S'))
{ {
TraceLog(WARNING, "[%s] This is not a valid raylib resource file", rresName); TraceLog(WARNING, "[%s] This is not a valid raylib resource file", rresName);
sound.error = ERROR_INVALID_RRES_FILE;
} }
else else
{ {
@ -662,6 +674,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
else else
{ {
TraceLog(WARNING, "[%s] Required resource do not seem to be a valid SOUND resource", rresName); TraceLog(WARNING, "[%s] Required resource do not seem to be a valid SOUND resource", rresName);
sound.error = ERROR_INVALID_RRES_RESOURCE;
} }
} }
else else
@ -767,105 +780,134 @@ int PlayMusicStream(int musicIndex, char *fileName)
{ {
int mixIndex; int mixIndex;
if(currentMusic[musicIndex].stream || currentMusic[musicIndex].chipctx) return 1; // error if(musicChannels_g[musicIndex].stream || musicChannels_g[musicIndex].xmctx) return ERROR_UNINITIALIZED_CHANNELS; // error
for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
{ {
if(mixChannelsActive_g[mixIndex] == NULL) break; if(mixChannels_g[mixIndex] == NULL) break;
else if(mixIndex == MAX_MIX_CHANNELS - 1) return 2; // error else if(mixIndex == MAX_MIX_CHANNELS - 1) return ERROR_OUT_OF_MIX_CHANNELS; // error
} }
if (strcmp(GetExtension(fileName),"ogg") == 0) if (strcmp(GetExtension(fileName),"ogg") == 0)
{ {
// Open audio stream // Open audio stream
currentMusic[musicIndex].stream = stb_vorbis_open_filename(fileName, NULL, NULL); musicChannels_g[musicIndex].stream = stb_vorbis_open_filename(fileName, NULL, NULL);
if (currentMusic[musicIndex].stream == NULL) if (musicChannels_g[musicIndex].stream == NULL)
{ {
TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName); TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName);
return 3; // error return ERROR_LOADING_OGG; // error
} }
else else
{ {
// Get file info // Get file info
stb_vorbis_info info = stb_vorbis_get_info(currentMusic[musicIndex].stream); stb_vorbis_info info = stb_vorbis_get_info(musicChannels_g[musicIndex].stream);
TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate); TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels); TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels);
TraceLog(DEBUG, "[%s] Temp memory required: %i", fileName, info.temp_memory_required); TraceLog(DEBUG, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
currentMusic[musicIndex].loop = true; // We loop by default musicChannels_g[musicIndex].loop = true; // We loop by default
musicEnabled_g = true; musicEnabled_g = true;
currentMusic[musicIndex].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[musicIndex].stream) * info.channels; musicChannels_g[musicIndex].totalSamplesLeft = (unsigned int)stb_vorbis_stream_length_in_samples(musicChannels_g[musicIndex].stream) * info.channels;
currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream); musicChannels_g[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(musicChannels_g[musicIndex].stream);
if (info.channels == 2){ if (info.channels == 2){
currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false); musicChannels_g[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false);
currentMusic[musicIndex].mixc->playing = true; musicChannels_g[musicIndex].mixc->playing = true;
} }
else{ else{
currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false); musicChannels_g[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false);
currentMusic[musicIndex].mixc->playing = true; musicChannels_g[musicIndex].mixc->playing = true;
} }
if(!currentMusic[musicIndex].mixc) return 4; // error if(!musicChannels_g[musicIndex].mixc) return ERROR_LOADING_OGG; // error
} }
} }
else if (strcmp(GetExtension(fileName),"xm") == 0) else if (strcmp(GetExtension(fileName),"xm") == 0)
{ {
// only stereo is supported for xm // only stereo is supported for xm
if(!jar_xm_create_context_from_file(&currentMusic[musicIndex].chipctx, 48000, fileName)) if(!jar_xm_create_context_from_file(&musicChannels_g[musicIndex].xmctx, 48000, fileName))
{ {
currentMusic[musicIndex].chipTune = true; musicChannels_g[musicIndex].chipTune = true;
currentMusic[musicIndex].loop = true; musicChannels_g[musicIndex].loop = true;
jar_xm_set_max_loop_count(currentMusic[musicIndex].chipctx, 0); // infinite number of loops jar_xm_set_max_loop_count(musicChannels_g[musicIndex].xmctx, 0); // infinite number of loops
currentMusic[musicIndex].totalSamplesLeft = jar_xm_get_remaining_samples(currentMusic[musicIndex].chipctx); musicChannels_g[musicIndex].totalSamplesLeft = (unsigned int)jar_xm_get_remaining_samples(musicChannels_g[musicIndex].xmctx);
currentMusic[musicIndex].totalLengthSeconds = ((float)currentMusic[musicIndex].totalSamplesLeft) / 48000.f; musicChannels_g[musicIndex].totalLengthSeconds = ((float)musicChannels_g[musicIndex].totalSamplesLeft) / 48000.f;
musicEnabled_g = true; musicEnabled_g = true;
TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic[musicIndex].totalSamplesLeft); TraceLog(INFO, "[%s] XM number of samples: %i", fileName, musicChannels_g[musicIndex].totalSamplesLeft);
TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic[musicIndex].totalLengthSeconds); TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, musicChannels_g[musicIndex].totalLengthSeconds);
currentMusic[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, false); musicChannels_g[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, true);
if(!currentMusic[musicIndex].mixc) return 5; // error if(!musicChannels_g[musicIndex].mixc) return ERROR_XM_CONTEXT_CREATION; // error
currentMusic[musicIndex].mixc->playing = true; musicChannels_g[musicIndex].mixc->playing = true;
} }
else else
{ {
TraceLog(WARNING, "[%s] XM file could not be opened", fileName); TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
return 6; // error return ERROR_LOADING_XM; // error
}
}
else if (strcmp(GetExtension(fileName),"mod") == 0)
{
jar_mod_init(&musicChannels_g[musicIndex].modctx);
if(jar_mod_load_file(&musicChannels_g[musicIndex].modctx, fileName))
{
musicChannels_g[musicIndex].chipTune = true;
musicChannels_g[musicIndex].loop = true;
musicChannels_g[musicIndex].totalSamplesLeft = (unsigned int)jar_mod_max_samples(&musicChannels_g[musicIndex].modctx);
musicChannels_g[musicIndex].totalLengthSeconds = ((float)musicChannels_g[musicIndex].totalSamplesLeft) / 48000.f;
musicEnabled_g = true;
TraceLog(INFO, "[%s] MOD number of samples: %i", fileName, musicChannels_g[musicIndex].totalSamplesLeft);
TraceLog(INFO, "[%s] MOD track length: %11.6f sec", fileName, musicChannels_g[musicIndex].totalLengthSeconds);
musicChannels_g[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, false);
if(!musicChannels_g[musicIndex].mixc) return ERROR_MOD_CONTEXT_CREATION; // error
musicChannels_g[musicIndex].mixc->playing = true;
}
else
{
TraceLog(WARNING, "[%s] MOD file could not be opened", fileName);
return ERROR_LOADING_MOD; // error
} }
} }
else else
{ {
TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName); TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
return 7; // error return ERROR_EXTENSION_NOT_RECOGNIZED; // error
} }
return 0; // normal return return 0; // normal return
} }
// Stop music playing for individual music index of currentMusic array (close stream) // Stop music playing for individual music index of musicChannels_g array (close stream)
void StopMusicStream(int index) void StopMusicStream(int index)
{ {
if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc) if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
{ {
CloseMixChannel(currentMusic[index].mixc); CloseMixChannel(musicChannels_g[index].mixc);
if (currentMusic[index].chipTune) if (musicChannels_g[index].chipTune && musicChannels_g[index].xmctx)
{ {
jar_xm_free_context(currentMusic[index].chipctx); jar_xm_free_context(musicChannels_g[index].xmctx);
musicChannels_g[index].xmctx = 0;
}
else if(musicChannels_g[index].chipTune && musicChannels_g[index].modctx.mod_loaded)
{
jar_mod_unload(&musicChannels_g[index].modctx);
} }
else else
{ {
stb_vorbis_close(currentMusic[index].stream); stb_vorbis_close(musicChannels_g[index].stream);
} }
if(!getMusicStreamCount()) musicEnabled_g = false; if(!getMusicStreamCount()) musicEnabled_g = false;
if(currentMusic[index].stream || currentMusic[index].chipctx) if(musicChannels_g[index].stream || musicChannels_g[index].xmctx)
{ {
currentMusic[index].stream = NULL; musicChannels_g[index].stream = NULL;
currentMusic[index].chipctx = NULL; musicChannels_g[index].xmctx = NULL;
} }
} }
} }
@ -875,7 +917,7 @@ int getMusicStreamCount(void)
{ {
int musicCount = 0; int musicCount = 0;
for(int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++) // find empty music slot for(int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++) // find empty music slot
if(currentMusic[musicIndex].stream != NULL || currentMusic[musicIndex].chipTune) musicCount++; if(musicChannels_g[musicIndex].stream != NULL || musicChannels_g[musicIndex].chipTune) musicCount++;
return musicCount; return musicCount;
} }
@ -884,11 +926,11 @@ int getMusicStreamCount(void)
void PauseMusicStream(int index) void PauseMusicStream(int index)
{ {
// Pause music stream if music available! // Pause music stream if music available!
if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc && musicEnabled_g) if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc && musicEnabled_g)
{ {
TraceLog(INFO, "Pausing music stream"); TraceLog(INFO, "Pausing music stream");
alSourcePause(currentMusic[index].mixc->alSource); alSourcePause(musicChannels_g[index].mixc->alSource);
currentMusic[index].mixc->playing = false; musicChannels_g[index].mixc->playing = false;
} }
} }
@ -897,13 +939,13 @@ void ResumeMusicStream(int index)
{ {
// Resume music playing... if music available! // Resume music playing... if music available!
ALenum state; ALenum state;
if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){
alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state); alGetSourcei(musicChannels_g[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED) if (state == AL_PAUSED)
{ {
TraceLog(INFO, "Resuming music stream"); TraceLog(INFO, "Resuming music stream");
alSourcePlay(currentMusic[index].mixc->alSource); alSourcePlay(musicChannels_g[index].mixc->alSource);
currentMusic[index].mixc->playing = true; musicChannels_g[index].mixc->playing = true;
} }
} }
} }
@ -914,8 +956,8 @@ bool IsMusicPlaying(int index)
bool playing = false; bool playing = false;
ALint state; ALint state;
if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){
alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state); alGetSourcei(musicChannels_g[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true; if (state == AL_PLAYING) playing = true;
} }
@ -925,29 +967,29 @@ bool IsMusicPlaying(int index)
// Set volume for music // Set volume for music
void SetMusicVolume(int index, float volume) void SetMusicVolume(int index, float volume)
{ {
if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){
alSourcef(currentMusic[index].mixc->alSource, AL_GAIN, volume); alSourcef(musicChannels_g[index].mixc->alSource, AL_GAIN, volume);
} }
} }
void SetMusicPitch(int index, float pitch) void SetMusicPitch(int index, float pitch)
{ {
if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){
alSourcef(currentMusic[index].mixc->alSource, AL_PITCH, pitch); alSourcef(musicChannels_g[index].mixc->alSource, AL_PITCH, pitch);
} }
} }
// Get current music time length (in seconds) // Get music time length (in seconds)
float GetMusicTimeLength(int index) float GetMusicTimeLength(int index)
{ {
float totalSeconds; float totalSeconds;
if (currentMusic[index].chipTune) if (musicChannels_g[index].chipTune)
{ {
totalSeconds = currentMusic[index].totalLengthSeconds; totalSeconds = (float)musicChannels_g[index].totalLengthSeconds;
} }
else else
{ {
totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[index].stream); totalSeconds = stb_vorbis_stream_length_in_seconds(musicChannels_g[index].stream);
} }
return totalSeconds; return totalSeconds;
@ -957,19 +999,24 @@ float GetMusicTimeLength(int index)
float GetMusicTimePlayed(int index) float GetMusicTimePlayed(int index)
{ {
float secondsPlayed = 0.0f; float secondsPlayed = 0.0f;
if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc) if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
{ {
if (currentMusic[index].chipTune) if (musicChannels_g[index].chipTune && musicChannels_g[index].xmctx)
{ {
uint64_t samples; uint64_t samples;
jar_xm_get_position(currentMusic[index].chipctx, NULL, NULL, NULL, &samples); jar_xm_get_position(musicChannels_g[index].xmctx, NULL, NULL, NULL, &samples);
secondsPlayed = (float)samples / (48000 * currentMusic[index].mixc->channels); // Not sure if this is the correct value secondsPlayed = (float)samples / (48000.f * musicChannels_g[index].mixc->channels); // Not sure if this is the correct value
}
else if(musicChannels_g[index].chipTune && musicChannels_g[index].modctx.mod_loaded)
{
long numsamp = jar_mod_current_samples(&musicChannels_g[index].modctx);
secondsPlayed = (float)numsamp / (48000.f);
} }
else else
{ {
int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels; int totalSamples = stb_vorbis_stream_length_in_samples(musicChannels_g[index].stream) * musicChannels_g[index].mixc->channels;
int samplesPlayed = totalSamples - currentMusic[index].totalSamplesLeft; int samplesPlayed = totalSamples - musicChannels_g[index].totalSamplesLeft;
secondsPlayed = (float)samplesPlayed / (currentMusic[index].mixc->sampleRate * currentMusic[index].mixc->channels); secondsPlayed = (float)samplesPlayed / (musicChannels_g[index].mixc->sampleRate * musicChannels_g[index].mixc->channels);
} }
} }
@ -989,19 +1036,30 @@ static bool BufferMusicStream(int index, int numBuffers)
int size = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts int size = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts
bool active = true; // We can get more data from stream (not finished) bool active = true; // We can get more data from stream (not finished)
if (currentMusic[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes. if (musicChannels_g[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
{ {
if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT)
size = MUSIC_BUFFER_SIZE_SHORT / 2;
else
size = currentMusic[index].totalSamplesLeft / 2;
for(int x=0; x<numBuffers; x++) for(int x=0; x<numBuffers; x++)
{ {
jar_xm_generate_samples_16bit(currentMusic[index].chipctx, pcm, size); // reads 2*readlen shorts and moves them to buffer+size memory location if(musicChannels_g[index].modctx.mod_loaded){
BufferMixChannel(currentMusic[index].mixc, pcm, size * 2); if(musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT)
currentMusic[index].totalSamplesLeft -= size * 2; size = MUSIC_BUFFER_SIZE_SHORT / 2;
if(currentMusic[index].totalSamplesLeft <= 0) else
size = musicChannels_g[index].totalSamplesLeft / 2;
jar_mod_fillbuffer(&musicChannels_g[index].modctx, pcm, size, 0 );
BufferMixChannel(musicChannels_g[index].mixc, pcm, size * 2);
}
else if(musicChannels_g[index].xmctx){
if(musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_FLOAT)
size = MUSIC_BUFFER_SIZE_FLOAT / 2;
else
size = musicChannels_g[index].totalSamplesLeft / 2;
jar_xm_generate_samples(musicChannels_g[index].xmctx, pcmf, size); // reads 2*readlen shorts and moves them to buffer+size memory location
BufferMixChannel(musicChannels_g[index].mixc, pcmf, size * 2);
}
musicChannels_g[index].totalSamplesLeft -= size;
if(musicChannels_g[index].totalSamplesLeft <= 0)
{ {
active = false; active = false;
break; break;
@ -1010,17 +1068,17 @@ static bool BufferMusicStream(int index, int numBuffers)
} }
else else
{ {
if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) if(musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT)
size = MUSIC_BUFFER_SIZE_SHORT; size = MUSIC_BUFFER_SIZE_SHORT;
else else
size = currentMusic[index].totalSamplesLeft; size = musicChannels_g[index].totalSamplesLeft;
for(int x=0; x<numBuffers; x++) for(int x=0; x<numBuffers; x++)
{ {
int streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].mixc->channels, pcm, size); int streamedBytes = stb_vorbis_get_samples_short_interleaved(musicChannels_g[index].stream, musicChannels_g[index].mixc->channels, pcm, size);
BufferMixChannel(currentMusic[index].mixc, pcm, streamedBytes * currentMusic[index].mixc->channels); BufferMixChannel(musicChannels_g[index].mixc, pcm, streamedBytes * musicChannels_g[index].mixc->channels);
currentMusic[index].totalSamplesLeft -= streamedBytes * currentMusic[index].mixc->channels; musicChannels_g[index].totalSamplesLeft -= streamedBytes * musicChannels_g[index].mixc->channels;
if(currentMusic[index].totalSamplesLeft <= 0) if(musicChannels_g[index].totalSamplesLeft <= 0)
{ {
active = false; active = false;
break; break;
@ -1037,11 +1095,11 @@ static void EmptyMusicStream(int index)
ALuint buffer = 0; ALuint buffer = 0;
int queued = 0; int queued = 0;
alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_QUEUED, &queued); alGetSourcei(musicChannels_g[index].mixc->alSource, AL_BUFFERS_QUEUED, &queued);
while (queued > 0) while (queued > 0)
{ {
alSourceUnqueueBuffers(currentMusic[index].mixc->alSource, 1, &buffer); alSourceUnqueueBuffers(musicChannels_g[index].mixc->alSource, 1, &buffer);
queued--; queued--;
} }
@ -1051,7 +1109,7 @@ static void EmptyMusicStream(int index)
static int IsMusicStreamReadyForBuffering(int index) static int IsMusicStreamReadyForBuffering(int index)
{ {
ALint processed = 0; ALint processed = 0;
alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed); alGetSourcei(musicChannels_g[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed);
return processed; return processed;
} }
@ -1062,20 +1120,21 @@ void UpdateMusicStream(int index)
bool active = true; bool active = true;
int numBuffers = IsMusicStreamReadyForBuffering(index); int numBuffers = IsMusicStreamReadyForBuffering(index);
if (currentMusic[index].mixc->playing && index < MAX_MUSIC_STREAMS && musicEnabled_g && currentMusic[index].mixc && numBuffers) if (musicChannels_g[index].mixc->playing && index < MAX_MUSIC_STREAMS && musicEnabled_g && musicChannels_g[index].mixc && numBuffers)
{ {
active = BufferMusicStream(index, numBuffers); active = BufferMusicStream(index, numBuffers);
if (!active && currentMusic[index].loop) if (!active && musicChannels_g[index].loop)
{ {
if (currentMusic[index].chipTune) if (musicChannels_g[index].chipTune)
{ {
currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * 48000; if(musicChannels_g[index].modctx.mod_loaded) jar_mod_seek_start(&musicChannels_g[index].modctx);
musicChannels_g[index].totalSamplesLeft = musicChannels_g[index].totalLengthSeconds * 48000;
} }
else else
{ {
stb_vorbis_seek_start(currentMusic[index].stream); stb_vorbis_seek_start(musicChannels_g[index].stream);
currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels; musicChannels_g[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(musicChannels_g[index].stream) * musicChannels_g[index].mixc->channels;
} }
active = true; active = true;
} }
@ -1083,9 +1142,9 @@ void UpdateMusicStream(int index)
if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data..."); if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state); alGetSourcei(musicChannels_g[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING && active) alSourcePlay(currentMusic[index].mixc->alSource); if (state != AL_PLAYING && active) alSourcePlay(musicChannels_g[index].mixc->alSource);
if (!active) StopMusicStream(index); if (!active) StopMusicStream(index);

25
src/audio.h
View file

@ -41,15 +41,35 @@
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
#ifndef __cplusplus #ifndef __cplusplus
// Boolean type // Boolean type
#ifndef true #if !defined(_STDBOOL_H)
typedef enum { false, true } bool; typedef enum { false, true } bool;
#define _STDBOOL_H
#endif #endif
#endif #endif
typedef enum {
ERROR_RAW_CONTEXT_CREATION = 1,
ERROR_XM_CONTEXT_CREATION = 2,
ERROR_MOD_CONTEXT_CREATION = 4,
ERROR_MIX_CHANNEL_CREATION = 8,
ERROR_MUSIC_CHANNEL_CREATION = 16,
ERROR_LOADING_XM = 32,
ERROR_LOADING_MOD = 64,
ERROR_LOADING_WAV = 128,
ERROR_LOADING_OGG = 256,
ERROR_OUT_OF_MIX_CHANNELS = 512,
ERROR_EXTENSION_NOT_RECOGNIZED = 1024,
ERROR_UNABLE_TO_OPEN_RRES_FILE = 2048,
ERROR_INVALID_RRES_FILE = 4096,
ERROR_INVALID_RRES_RESOURCE = 8192,
ERROR_UNINITIALIZED_CHANNELS = 16384
} AudioError;
// Sound source type // Sound source type
typedef struct Sound { typedef struct Sound {
unsigned int source; unsigned int source;
unsigned int buffer; unsigned int buffer;
AudioError error; // if there was any error during the creation or use of this Sound
} Sound; } Sound;
// Wave type, defines audio wave data // Wave type, defines audio wave data
@ -63,6 +83,7 @@ typedef struct Wave {
typedef int RawAudioContext; typedef int RawAudioContext;
#ifdef __cplusplus #ifdef __cplusplus
extern "C" { // Prevents name mangling of functions extern "C" { // Prevents name mangling of functions
#endif #endif
@ -107,7 +128,7 @@ void SetMusicPitch(int index, float pitch);
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint);
void CloseRawAudioContext(RawAudioContext ctx); void CloseRawAudioContext(RawAudioContext ctx);
int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements); // returns number of elements buffered int BufferRawAudioContext(RawAudioContext ctx, void *data, unsigned short numberElements); // returns number of elements buffered
#ifdef __cplusplus #ifdef __cplusplus
} }

1587
src/jar_mod.h Normal file
View file

File diff suppressed because it is too large Load diff

26
src/raylib.h
View file

@ -261,8 +261,9 @@
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
#ifndef __cplusplus #ifndef __cplusplus
// Boolean type // Boolean type
#ifndef true #if !defined(_STDBOOL_H)
typedef enum { false, true } bool; typedef enum { false, true } bool;
#define _STDBOOL_H
#endif #endif
#endif #endif
@ -451,10 +452,29 @@ typedef struct Ray {
Vector3 direction; Vector3 direction;
} Ray; } Ray;
typedef enum { // allows errors to be & together
ERROR_RAW_CONTEXT_CREATION = 1,
ERROR_XM_CONTEXT_CREATION = 2,
ERROR_MOD_CONTEXT_CREATION = 4,
ERROR_MIX_CHANNEL_CREATION = 8,
ERROR_MUSIC_CHANNEL_CREATION = 16,
ERROR_LOADING_XM = 32,
ERROR_LOADING_MOD = 64,
ERROR_LOADING_WAV = 128,
ERROR_LOADING_OGG = 256,
ERROR_OUT_OF_MIX_CHANNELS = 512,
ERROR_EXTENSION_NOT_RECOGNIZED = 1024,
ERROR_UNABLE_TO_OPEN_RRES_FILE = 2048,
ERROR_INVALID_RRES_FILE = 4096,
ERROR_INVALID_RRES_RESOURCE = 8192,
ERROR_UNINITIALIZED_CHANNELS = 16384
} AudioError;
// Sound source type // Sound source type
typedef struct Sound { typedef struct Sound {
unsigned int source; unsigned int source;
unsigned int buffer; unsigned int buffer;
AudioError error; // if there was any error during the creation or use of this Sound
} Sound; } Sound;
// Wave type, defines audio wave data // Wave type, defines audio wave data
@ -468,6 +488,8 @@ typedef struct Wave {
typedef int RawAudioContext; typedef int RawAudioContext;
// Texture formats // Texture formats
// NOTE: Support depends on OpenGL version and platform // NOTE: Support depends on OpenGL version and platform
typedef enum { typedef enum {
@ -926,7 +948,7 @@ void SetMusicPitch(int index, float pitch);
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint);
void CloseRawAudioContext(RawAudioContext ctx); void CloseRawAudioContext(RawAudioContext ctx);
int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements); // returns number of elements buffered int BufferRawAudioContext(RawAudioContext ctx, void *data, unsigned short numberElements); // returns number of elements buffered
#ifdef __cplusplus #ifdef __cplusplus
} }