diff --git a/CMakeLists.txt b/CMakeLists.txt index 8689e9add..eb0052d08 100644 --- a/CMakeLists.txt +++ b/CMakeLists.txt @@ -8,7 +8,7 @@ set(CMAKE_C_FLAGS "-O1 -Wall -std=gnu99 -fgnu89-inline") IF(${PLATFORM_TO_USE} MATCHES "PLATFORM_DESKTOP") add_definitions(-DPLATFORM_DESKTOP, -DGRAPHICS_API_OPENGL_33) - include_directories("." "src/" "external/openal_soft/include" "external/glew/include" "external/glfw3/include") + include_directories("." "src/" "external/openal_soft/include" "external/glfw3/include") ENDIF() @@ -22,7 +22,7 @@ ENDIF() IF(${PLATFORM_TO_USE} MATCHES "PLATFORM_WEB") 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() diff --git a/external/openal_soft/include/AL/alext.h b/external/openal_soft/include/AL/alext.h index 0447f2bb4..6af581aad 100644 --- a/external/openal_soft/include/AL/alext.h +++ b/external/openal_soft/include/AL/alext.h @@ -13,8 +13,8 @@ * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the - * Free Software Foundation, Inc., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. + * Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * 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 +#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 } #endif diff --git a/external/openal_soft/include/AL/efx-presets.h b/external/openal_soft/include/AL/efx-presets.h index 86dcbda2f..8539fd517 100644 --- a/external/openal_soft/include/AL/efx-presets.h +++ b/external/openal_soft/include/AL/efx-presets.h @@ -345,7 +345,7 @@ typedef struct { /* Driving Presets */ #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 \ { 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 } diff --git a/external/openal_soft/lib/win32/libopenal32.a b/external/openal_soft/lib/win32/libOpenAL32.dll.a similarity index 71% rename from external/openal_soft/lib/win32/libopenal32.a rename to external/openal_soft/lib/win32/libOpenAL32.dll.a index 5f7352aca..1c4c63c8f 100644 Binary files a/external/openal_soft/lib/win32/libopenal32.a and b/external/openal_soft/lib/win32/libOpenAL32.dll.a differ diff --git a/external/openal_soft/openal32.dll b/external/openal_soft/openal32.dll index 71ced6a21..1e3bddd56 100644 Binary files a/external/openal_soft/openal32.dll and b/external/openal_soft/openal32.dll differ diff --git a/src/audio.c b/src/audio.c index a884303c0..452459ef6 100644 --- a/src/audio.c +++ b/src/audio.c @@ -56,6 +56,9 @@ #define JAR_XM_IMPLEMENTATION #include "jar_xm.h" // XM loading functions +#define JAR_MOD_IMPLEMENTATION +#include "jar_mod.h" // For playing .mod files + //---------------------------------------------------------------------------------- // Defines and Macros //---------------------------------------------------------------------------------- @@ -95,10 +98,11 @@ typedef struct MixChannel_t { // NOTE: Anything longer than ~10 seconds should be streamed into a mix channel... typedef struct Music { 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 - int totalSamplesLeft; + unsigned int totalSamplesLeft; float totalLengthSeconds; bool loop; bool chipTune; // True if chiptune is loaded @@ -111,9 +115,9 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType; //---------------------------------------------------------------------------------- // 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 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 @@ -175,7 +179,7 @@ void CloseAudioDevice(void) { for(int index=0; index= MAX_MIX_CHANNELS) return NULL; if(!IsAudioDeviceReady()) InitAudioDevice(); - if(!mixChannelsActive_g[mixChannel]){ + if(!mixChannels_g[mixChannel]){ MixChannel_t *mixc = (MixChannel_t*)malloc(sizeof(MixChannel_t)); mixc->sampleRate = sampleRate; mixc->channels = channels; mixc->mixChannel = mixChannel; mixc->floatingPoint = floatingPoint; - mixChannelsActive_g[mixChannel] = mixc; + mixChannels_g[mixChannel] = mixc; // setup openAL format if(channels == 1) @@ -283,7 +287,7 @@ static void CloseMixChannel(MixChannel_t* mixc) //delete source and buffers alDeleteSources(1, &mixc->alSource); alDeleteBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer); - mixChannelsActive_g[mixc->mixChannel] = NULL; + mixChannels_g[mixc->mixChannel] = NULL; free(mixc); mixc = NULL; } @@ -294,7 +298,7 @@ static void CloseMixChannel(MixChannel_t* mixc) // @Returns number of samples that where processed. 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) { // 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 RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint) { int mixIndex; for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot { - if(mixChannelsActive_g[mixIndex] == NULL) break; - else if(mixIndex == MAX_MIX_CHANNELS - 1) return -1; // error + if(mixChannels_g[mixIndex] == NULL) break; + else if(mixIndex == MAX_MIX_CHANNELS - 1) return ERROR_OUT_OF_MIX_CHANNELS; // error } if(InitMixChannel(sampleRate, mixIndex, channels, floatingPoint)) return mixIndex; else - return -2; // error + return ERROR_RAW_CONTEXT_CREATION; // error } void CloseRawAudioContext(RawAudioContext ctx) { - if(mixChannelsActive_g[ctx]) - CloseMixChannel(mixChannelsActive_g[ctx]); + if(mixChannels_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; if(ctx >= 0) { - MixChannel_t* mixc = mixChannelsActive_g[ctx]; + MixChannel_t* mixc = mixChannels_g[ctx]; numBuffered = BufferMixChannel(mixc, data, numberElements); } return numBuffered; @@ -431,7 +438,10 @@ Sound LoadSound(char *fileName) if (strcmp(GetExtension(fileName),"wav") == 0) wave = LoadWAV(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) { @@ -558,6 +568,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId) if (rresFile == NULL) { TraceLog(WARNING, "[%s] rRES raylib resource file could not be opened", rresName); + sound.error = ERROR_UNABLE_TO_OPEN_RRES_FILE; //error } 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')) { TraceLog(WARNING, "[%s] This is not a valid raylib resource file", rresName); + sound.error = ERROR_INVALID_RRES_FILE; } else { @@ -662,6 +674,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId) else { TraceLog(WARNING, "[%s] Required resource do not seem to be a valid SOUND resource", rresName); + sound.error = ERROR_INVALID_RRES_RESOURCE; } } else @@ -767,105 +780,134 @@ int PlayMusicStream(int musicIndex, char *fileName) { 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 { - if(mixChannelsActive_g[mixIndex] == NULL) break; - else if(mixIndex == MAX_MIX_CHANNELS - 1) return 2; // error + if(mixChannels_g[mixIndex] == NULL) break; + else if(mixIndex == MAX_MIX_CHANNELS - 1) return ERROR_OUT_OF_MIX_CHANNELS; // error } if (strcmp(GetExtension(fileName),"ogg") == 0) { // 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); - return 3; // error + return ERROR_LOADING_OGG; // error } else { // 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 channels: %i", fileName, info.channels); 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; - currentMusic[musicIndex].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[musicIndex].stream) * info.channels; - currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream); + musicChannels_g[musicIndex].totalSamplesLeft = (unsigned int)stb_vorbis_stream_length_in_samples(musicChannels_g[musicIndex].stream) * info.channels; + musicChannels_g[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(musicChannels_g[musicIndex].stream); if (info.channels == 2){ - currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false); - currentMusic[musicIndex].mixc->playing = true; + musicChannels_g[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false); + musicChannels_g[musicIndex].mixc->playing = true; } else{ - currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false); - currentMusic[musicIndex].mixc->playing = true; + musicChannels_g[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false); + 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) { // only stereo is supported for xm - if(!jar_xm_create_context_from_file(¤tMusic[musicIndex].chipctx, 48000, fileName)) + if(!jar_xm_create_context_from_file(&musicChannels_g[musicIndex].xmctx, 48000, fileName)) { - currentMusic[musicIndex].chipTune = true; - currentMusic[musicIndex].loop = true; - jar_xm_set_max_loop_count(currentMusic[musicIndex].chipctx, 0); // infinite number of loops - currentMusic[musicIndex].totalSamplesLeft = jar_xm_get_remaining_samples(currentMusic[musicIndex].chipctx); - currentMusic[musicIndex].totalLengthSeconds = ((float)currentMusic[musicIndex].totalSamplesLeft) / 48000.f; + musicChannels_g[musicIndex].chipTune = true; + musicChannels_g[musicIndex].loop = true; + jar_xm_set_max_loop_count(musicChannels_g[musicIndex].xmctx, 0); // infinite number of loops + musicChannels_g[musicIndex].totalSamplesLeft = (unsigned int)jar_xm_get_remaining_samples(musicChannels_g[musicIndex].xmctx); + musicChannels_g[musicIndex].totalLengthSeconds = ((float)musicChannels_g[musicIndex].totalSamplesLeft) / 48000.f; musicEnabled_g = true; - TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic[musicIndex].totalSamplesLeft); - TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic[musicIndex].totalLengthSeconds); + TraceLog(INFO, "[%s] XM number of samples: %i", fileName, musicChannels_g[musicIndex].totalSamplesLeft); + TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, musicChannels_g[musicIndex].totalLengthSeconds); - currentMusic[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, false); - if(!currentMusic[musicIndex].mixc) return 5; // error - currentMusic[musicIndex].mixc->playing = true; + musicChannels_g[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, true); + if(!musicChannels_g[musicIndex].mixc) return ERROR_XM_CONTEXT_CREATION; // error + musicChannels_g[musicIndex].mixc->playing = true; } else { 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 { 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 } -// 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) { - 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 { - stb_vorbis_close(currentMusic[index].stream); + stb_vorbis_close(musicChannels_g[index].stream); } 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; - currentMusic[index].chipctx = NULL; + musicChannels_g[index].stream = NULL; + musicChannels_g[index].xmctx = NULL; } } } @@ -875,7 +917,7 @@ int getMusicStreamCount(void) { int musicCount = 0; 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; } @@ -884,11 +926,11 @@ int getMusicStreamCount(void) void PauseMusicStream(int index) { // 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"); - alSourcePause(currentMusic[index].mixc->alSource); - currentMusic[index].mixc->playing = false; + alSourcePause(musicChannels_g[index].mixc->alSource); + musicChannels_g[index].mixc->playing = false; } } @@ -897,13 +939,13 @@ void ResumeMusicStream(int index) { // Resume music playing... if music available! ALenum state; - if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ - alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state); + if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){ + alGetSourcei(musicChannels_g[index].mixc->alSource, AL_SOURCE_STATE, &state); if (state == AL_PAUSED) { TraceLog(INFO, "Resuming music stream"); - alSourcePlay(currentMusic[index].mixc->alSource); - currentMusic[index].mixc->playing = true; + alSourcePlay(musicChannels_g[index].mixc->alSource); + musicChannels_g[index].mixc->playing = true; } } } @@ -914,8 +956,8 @@ bool IsMusicPlaying(int index) bool playing = false; ALint state; - if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ - alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state); + if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){ + alGetSourcei(musicChannels_g[index].mixc->alSource, AL_SOURCE_STATE, &state); if (state == AL_PLAYING) playing = true; } @@ -925,29 +967,29 @@ bool IsMusicPlaying(int index) // Set volume for music void SetMusicVolume(int index, float volume) { - if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ - alSourcef(currentMusic[index].mixc->alSource, AL_GAIN, volume); + if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){ + alSourcef(musicChannels_g[index].mixc->alSource, AL_GAIN, volume); } } void SetMusicPitch(int index, float pitch) { - if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ - alSourcef(currentMusic[index].mixc->alSource, AL_PITCH, pitch); + if(index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc){ + 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 totalSeconds; - if (currentMusic[index].chipTune) + if (musicChannels_g[index].chipTune) { - totalSeconds = currentMusic[index].totalLengthSeconds; + totalSeconds = (float)musicChannels_g[index].totalLengthSeconds; } else { - totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[index].stream); + totalSeconds = stb_vorbis_stream_length_in_seconds(musicChannels_g[index].stream); } return totalSeconds; @@ -957,19 +999,24 @@ float GetMusicTimeLength(int index) float GetMusicTimePlayed(int index) { 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; - jar_xm_get_position(currentMusic[index].chipctx, NULL, NULL, NULL, &samples); - secondsPlayed = (float)samples / (48000 * currentMusic[index].mixc->channels); // Not sure if this is the correct value + jar_xm_get_position(musicChannels_g[index].xmctx, NULL, NULL, NULL, &samples); + 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 { - int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels; - int samplesPlayed = totalSamples - currentMusic[index].totalSamplesLeft; - secondsPlayed = (float)samplesPlayed / (currentMusic[index].mixc->sampleRate * currentMusic[index].mixc->channels); + int totalSamples = stb_vorbis_stream_length_in_samples(musicChannels_g[index].stream) * musicChannels_g[index].mixc->channels; + int samplesPlayed = totalSamples - musicChannels_g[index].totalSamplesLeft; + 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 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= MUSIC_BUFFER_SIZE_SHORT) + size = MUSIC_BUFFER_SIZE_SHORT / 2; + 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; break; @@ -1010,17 +1068,17 @@ static bool BufferMusicStream(int index, int numBuffers) } else { - if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) + if(musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) size = MUSIC_BUFFER_SIZE_SHORT; else - size = currentMusic[index].totalSamplesLeft; + size = musicChannels_g[index].totalSamplesLeft; for(int x=0; xchannels, pcm, size); - BufferMixChannel(currentMusic[index].mixc, pcm, streamedBytes * currentMusic[index].mixc->channels); - currentMusic[index].totalSamplesLeft -= streamedBytes * currentMusic[index].mixc->channels; - if(currentMusic[index].totalSamplesLeft <= 0) + int streamedBytes = stb_vorbis_get_samples_short_interleaved(musicChannels_g[index].stream, musicChannels_g[index].mixc->channels, pcm, size); + BufferMixChannel(musicChannels_g[index].mixc, pcm, streamedBytes * musicChannels_g[index].mixc->channels); + musicChannels_g[index].totalSamplesLeft -= streamedBytes * musicChannels_g[index].mixc->channels; + if(musicChannels_g[index].totalSamplesLeft <= 0) { active = false; break; @@ -1037,11 +1095,11 @@ static void EmptyMusicStream(int index) ALuint buffer = 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) { - alSourceUnqueueBuffers(currentMusic[index].mixc->alSource, 1, &buffer); + alSourceUnqueueBuffers(musicChannels_g[index].mixc->alSource, 1, &buffer); queued--; } @@ -1051,7 +1109,7 @@ static void EmptyMusicStream(int index) static int IsMusicStreamReadyForBuffering(int index) { ALint processed = 0; - alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed); + alGetSourcei(musicChannels_g[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed); return processed; } @@ -1062,20 +1120,21 @@ void UpdateMusicStream(int index) bool active = true; 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); - 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 { - stb_vorbis_seek_start(currentMusic[index].stream); - currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels; + stb_vorbis_seek_start(musicChannels_g[index].stream); + musicChannels_g[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(musicChannels_g[index].stream) * musicChannels_g[index].mixc->channels; } active = true; } @@ -1083,9 +1142,9 @@ void UpdateMusicStream(int index) 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); diff --git a/src/audio.h b/src/audio.h index 1140a60a8..a74755667 100644 --- a/src/audio.h +++ b/src/audio.h @@ -41,15 +41,35 @@ //---------------------------------------------------------------------------------- #ifndef __cplusplus // Boolean type - #ifndef true + #if !defined(_STDBOOL_H) typedef enum { false, true } bool; + #define _STDBOOL_H #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 typedef struct Sound { unsigned int source; unsigned int buffer; + AudioError error; // if there was any error during the creation or use of this Sound } Sound; // Wave type, defines audio wave data @@ -63,6 +83,7 @@ typedef struct Wave { typedef int RawAudioContext; + #ifdef __cplusplus extern "C" { // Prevents name mangling of functions #endif @@ -107,7 +128,7 @@ void SetMusicPitch(int index, float pitch); RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); 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 } diff --git a/src/jar_mod.h b/src/jar_mod.h new file mode 100644 index 000000000..2ddc61d1e --- /dev/null +++ b/src/jar_mod.h @@ -0,0 +1,1587 @@ +// jar_mod.h - v0.01 - public domain C0 - Joshua Reisenauer +// +// HISTORY: +// +// v0.01 2016-03-12 Setup +// +// +// USAGE: +// +// In ONE source file, put: +// +// #define JAR_MOD_IMPLEMENTATION +// #include "jar_mod.h" +// +// Other source files should just include jar_mod.h +// +// SAMPLE CODE: +// jar_mod_context_t modctx; +// short samplebuff[4096]; +// bool bufferFull = false; +// int intro_load(void) +// { +// jar_mod_init(&modctx); +// jar_mod_load_file(&modctx, "file.mod"); +// return 1; +// } +// int intro_unload(void) +// { +// jar_mod_unload(&modctx); +// return 1; +// } +// int intro_tick(long counter) +// { +// if(!bufferFull) +// { +// jar_mod_fillbuffer(&modctx, samplebuff, 4096, 0); +// bufferFull=true; +// } +// if(IsKeyDown(KEY_ENTER)) +// return 1; +// return 0; +// } +// +// +// LISCENSE: +// +// Written by: Jean-François DEL NERO (http://hxc2001.com/) free.fr> +// Adapted to jar_mod by: Joshua Adam Reisenauer +// This program is free software. It comes without any warranty, to the +// extent permitted by applicable law. You can redistribute it and/or +// modify it under the terms of the Do What The Fuck You Want To Public +// License, Version 2, as published by Sam Hocevar. See +// http://sam.zoy.org/wtfpl/COPYING for more details. +/////////////////////////////////////////////////////////////////////////////////// +// HxCMOD Core API: +// ------------------------------------------- +// int jar_mod_init(jar_mod_context_t * modctx) +// +// - Initialize the jar_mod_context_t buffer. Must be called before doing anything else. +// Return 1 if success. 0 in case of error. +// ------------------------------------------- +// mulong jar_mod_load_file(jar_mod_context_t * modctx, char* filename) +// +// - "Load" a MOD from file, context must already be initialized. +// Return size of file in bytes. +// ------------------------------------------- +// void jar_mod_fillbuffer( jar_mod_context_t * modctx, unsigned short * outbuffer, unsigned long nbsample, jar_mod_tracker_buffer_state * trkbuf ) +// +// - Generate and return the next samples chunk to outbuffer. +// nbsample specify the number of stereo 16bits samples you want. +// The output format is by default signed 48000Hz 16-bit Stereo PCM samples, otherwise it is changed with jar_mod_setcfg(). +// The output buffer size in bytes must be equal to ( nbsample * 2 * channels ). +// The optional trkbuf parameter can be used to get detailed status of the player. Put NULL/0 is unused. +// ------------------------------------------- +// void jar_mod_unload( jar_mod_context_t * modctx ) +// - "Unload" / clear the player status. +// ------------------------------------------- +/////////////////////////////////////////////////////////////////////////////////// + + +#ifndef INCLUDE_JAR_MOD_H +#define INCLUDE_JAR_MOD_H + +#include +#include +#include + + + +#ifdef __cplusplus +extern "C" { +#endif + + + +// Basic type +typedef unsigned char muchar; +typedef unsigned short muint; +typedef short mint; +typedef unsigned long mulong; + +#define NUMMAXCHANNELS 32 +#define MAXNOTES 12*12 +#define DEFAULT_SAMPLE_RATE 48000 +// +// MOD file structures +// + +#pragma pack(1) + +typedef struct { + muchar name[22]; + muint length; + muchar finetune; + muchar volume; + muint reppnt; + muint replen; +} sample; + +typedef struct { + muchar sampperiod; + muchar period; + muchar sampeffect; + muchar effect; +} note; + +typedef struct { + muchar title[20]; + sample samples[31]; + muchar length; // length of tablepos + muchar protracker; + muchar patterntable[128]; + muchar signature[4]; + muchar speed; +} module; + +#pragma pack() + +// +// HxCMod Internal structures +// +typedef struct { + char* sampdata; + muint sampnum; + muint length; + muint reppnt; + muint replen; + mulong samppos; + muint period; + muchar volume; + mulong ticks; + muchar effect; + muchar parameffect; + muint effect_code; + mint decalperiod; + mint portaspeed; + mint portaperiod; + mint vibraperiod; + mint Arpperiods[3]; + muchar ArpIndex; + mint oldk; + muchar volumeslide; + muchar vibraparam; + muchar vibrapointeur; + muchar finetune; + muchar cut_param; + muint patternloopcnt; + muint patternloopstartpoint; +} channel; + +typedef struct { + module song; + char* sampledata[31]; + note* patterndata[128]; + + mulong playrate; + muint tablepos; + muint patternpos; + muint patterndelay; + muint jump_loop_effect; + muchar bpm; + mulong patternticks; + mulong patterntickse; + mulong patternticksaim; + mulong sampleticksconst; + mulong samplenb; + channel channels[NUMMAXCHANNELS]; + muint number_of_channels; + muint fullperiod[MAXNOTES * 8]; + muint mod_loaded; + mint last_r_sample; + mint last_l_sample; + mint stereo; + mint stereo_separation; + mint bits; + mint filter; + + muchar *modfile; // the raw mod file + mulong modfilesize; + muint loopcount; +} jar_mod_context_t; + +// +// Player states structures +// +typedef struct track_state_ +{ + unsigned char instrument_number; + unsigned short cur_period; + unsigned char cur_volume; + unsigned short cur_effect; + unsigned short cur_parameffect; +}track_state; + +typedef struct tracker_state_ +{ + int number_of_tracks; + int bpm; + int speed; + int cur_pattern; + int cur_pattern_pos; + int cur_pattern_table_pos; + unsigned int buf_index; + track_state tracks[32]; +}tracker_state; + +typedef struct tracker_state_instrument_ +{ + char name[22]; + int active; +}tracker_state_instrument; + +typedef struct jar_mod_tracker_buffer_state_ +{ + int nb_max_of_state; + int nb_of_state; + int cur_rd_index; + int sample_step; + char name[64]; + tracker_state_instrument instruments[31]; + tracker_state * track_state_buf; +}jar_mod_tracker_buffer_state; + + + +bool jar_mod_init(jar_mod_context_t * modctx); +bool jar_mod_setcfg(jar_mod_context_t * modctx, int samplerate, int bits, int stereo, int stereo_separation, int filter); +void jar_mod_fillbuffer(jar_mod_context_t * modctx, short * outbuffer, unsigned long nbsample, jar_mod_tracker_buffer_state * trkbuf); +void jar_mod_unload(jar_mod_context_t * modctx); +mulong jar_mod_load_file(jar_mod_context_t * modctx, char* filename); +mulong jar_mod_current_samples(jar_mod_context_t * modctx); +mulong jar_mod_max_samples(jar_mod_context_t * modctx); +void jar_mod_seek_start(jar_mod_context_t * ctx); + +#ifdef __cplusplus +} +#endif +//-------------------------------------------------------------------- + + + +//------------------------------------------------------------------------------- +#ifdef JAR_MOD_IMPLEMENTATION + +// Effects list +#define EFFECT_ARPEGGIO 0x0 // Supported +#define EFFECT_PORTAMENTO_UP 0x1 // Supported +#define EFFECT_PORTAMENTO_DOWN 0x2 // Supported +#define EFFECT_TONE_PORTAMENTO 0x3 // Supported +#define EFFECT_VIBRATO 0x4 // Supported +#define EFFECT_VOLSLIDE_TONEPORTA 0x5 // Supported +#define EFFECT_VOLSLIDE_VIBRATO 0x6 // Supported +#define EFFECT_VOLSLIDE_TREMOLO 0x7 // - TO BE DONE - +#define EFFECT_SET_PANNING 0x8 // - TO BE DONE - +#define EFFECT_SET_OFFSET 0x9 // Supported +#define EFFECT_VOLUME_SLIDE 0xA // Supported +#define EFFECT_JUMP_POSITION 0xB // Supported +#define EFFECT_SET_VOLUME 0xC // Supported +#define EFFECT_PATTERN_BREAK 0xD // Supported + +#define EFFECT_EXTENDED 0xE +#define EFFECT_E_FINE_PORTA_UP 0x1 // Supported +#define EFFECT_E_FINE_PORTA_DOWN 0x2 // Supported +#define EFFECT_E_GLISSANDO_CTRL 0x3 // - TO BE DONE - +#define EFFECT_E_VIBRATO_WAVEFORM 0x4 // - TO BE DONE - +#define EFFECT_E_SET_FINETUNE 0x5 // - TO BE DONE - +#define EFFECT_E_PATTERN_LOOP 0x6 // Supported +#define EFFECT_E_TREMOLO_WAVEFORM 0x7 // - TO BE DONE - +#define EFFECT_E_SET_PANNING_2 0x8 // - TO BE DONE - +#define EFFECT_E_RETRIGGER_NOTE 0x9 // - TO BE DONE - +#define EFFECT_E_FINE_VOLSLIDE_UP 0xA // Supported +#define EFFECT_E_FINE_VOLSLIDE_DOWN 0xB // Supported +#define EFFECT_E_NOTE_CUT 0xC // Supported +#define EFFECT_E_NOTE_DELAY 0xD // - TO BE DONE - +#define EFFECT_E_PATTERN_DELAY 0xE // Supported +#define EFFECT_E_INVERT_LOOP 0xF // - TO BE DONE - +#define EFFECT_SET_SPEED 0xF0 // Supported +#define EFFECT_SET_TEMPO 0xF2 // Supported + +#define PERIOD_TABLE_LENGTH MAXNOTES +#define FULL_PERIOD_TABLE_LENGTH ( PERIOD_TABLE_LENGTH * 8 ) + +static const short periodtable[]= +{ + 27392, 25856, 24384, 23040, 21696, 20480, 19328, 18240, 17216, 16256, 15360, 14496, + 13696, 12928, 12192, 11520, 10848, 10240, 9664, 9120, 8606, 8128, 7680, 7248, + 6848, 6464, 6096, 5760, 5424, 5120, 4832, 4560, 4304, 4064, 3840, 3624, + 3424, 3232, 3048, 2880, 2712, 2560, 2416, 2280, 2152, 2032, 1920, 1812, + 1712, 1616, 1524, 1440, 1356, 1280, 1208, 1140, 1076, 1016, 960, 906, + 856, 808, 762, 720, 678, 640, 604, 570, 538, 508, 480, 453, + 428, 404, 381, 360, 339, 320, 302, 285, 269, 254, 240, 226, + 214, 202, 190, 180, 170, 160, 151, 143, 135, 127, 120, 113, + 107, 101, 95, 90, 85, 80, 75, 71, 67, 63, 60, 56, + 53, 50, 47, 45, 42, 40, 37, 35, 33, 31, 30, 28, + 27, 25, 24, 22, 21, 20, 19, 18, 17, 16, 15, 14, + 13, 13, 12, 11, 11, 10, 9, 9, 8, 8, 7, 7 +}; + +static const short sintable[]={ + 0, 24, 49, 74, 97, 120, 141,161, + 180, 197, 212, 224, 235, 244, 250,253, + 255, 253, 250, 244, 235, 224, 212,197, + 180, 161, 141, 120, 97, 74, 49, 24 +}; + +typedef struct modtype_ +{ + unsigned char signature[5]; + int numberofchannels; +}modtype; + +modtype modlist[]= +{ + { "M!K!",4}, + { "M.K.",4}, + { "FLT4",4}, + { "FLT8",8}, + { "4CHN",4}, + { "6CHN",6}, + { "8CHN",8}, + { "10CH",10}, + { "12CH",12}, + { "14CH",14}, + { "16CH",16}, + { "18CH",18}, + { "20CH",20}, + { "22CH",22}, + { "24CH",24}, + { "26CH",26}, + { "28CH",28}, + { "30CH",30}, + { "32CH",32}, + { "",0} +}; + +/////////////////////////////////////////////////////////////////////////////////// + +static void memcopy( void * dest, void *source, unsigned long size ) +{ + unsigned long i; + unsigned char * d,*s; + + d=(unsigned char*)dest; + s=(unsigned char*)source; + for(i=0;i= mod->fullperiod[i]) + { + return i; + } + } + + return MAXNOTES; +} + +static void worknote( note * nptr, channel * cptr, char t, jar_mod_context_t * mod ) +{ + muint sample, period, effect, operiod; + muint curnote, arpnote; + + sample = (nptr->sampperiod & 0xF0) | (nptr->sampeffect >> 4); + period = ((nptr->sampperiod & 0xF) << 8) | nptr->period; + effect = ((nptr->sampeffect & 0xF) << 8) | nptr->effect; + + operiod = cptr->period; + + if ( period || sample ) + { + if( sample && sample < 32 ) + { + cptr->sampnum = sample - 1; + } + + if( period || sample ) + { + cptr->sampdata = (char *) mod->sampledata[cptr->sampnum]; + cptr->length = mod->song.samples[cptr->sampnum].length; + cptr->reppnt = mod->song.samples[cptr->sampnum].reppnt; + cptr->replen = mod->song.samples[cptr->sampnum].replen; + + cptr->finetune = (mod->song.samples[cptr->sampnum].finetune)&0xF; + + if(effect>>8!=4 && effect>>8!=6) + { + cptr->vibraperiod=0; + cptr->vibrapointeur=0; + } + } + + if( (sample != 0) && ( (effect>>8) != EFFECT_VOLSLIDE_TONEPORTA ) ) + { + cptr->volume = mod->song.samples[cptr->sampnum].volume; + cptr->volumeslide = 0; + } + + if( ( (effect>>8) != EFFECT_TONE_PORTAMENTO && (effect>>8)!=EFFECT_VOLSLIDE_TONEPORTA) ) + { + if (period!=0) + cptr->samppos = 0; + } + + cptr->decalperiod = 0; + if( period ) + { + if(cptr->finetune) + { + if( cptr->finetune <= 7 ) + { + period = mod->fullperiod[getnote(mod,period,0) + cptr->finetune]; + } + else + { + period = mod->fullperiod[getnote(mod,period,0) - (16 - (cptr->finetune)) ]; + } + } + + cptr->period = period; + } + + } + + cptr->effect = 0; + cptr->parameffect = 0; + cptr->effect_code = effect; + + switch (effect >> 8) + { + case EFFECT_ARPEGGIO: + /* + [0]: Arpeggio + Where [0][x][y] means "play note, note+x semitones, note+y + semitones, then return to original note". The fluctuations are + carried out evenly spaced in one pattern division. They are usually + used to simulate chords, but this doesn't work too well. They are + also used to produce heavy vibrato. A major chord is when x=4, y=7. + A minor chord is when x=3, y=7. + */ + + if(effect&0xff) + { + cptr->effect = EFFECT_ARPEGGIO; + cptr->parameffect = effect&0xff; + + cptr->ArpIndex = 0; + + curnote = getnote(mod,cptr->period,cptr->finetune); + + cptr->Arpperiods[0] = cptr->period; + + arpnote = curnote + (((cptr->parameffect>>4)&0xF)*8); + if( arpnote >= FULL_PERIOD_TABLE_LENGTH ) + arpnote = FULL_PERIOD_TABLE_LENGTH - 1; + + cptr->Arpperiods[1] = mod->fullperiod[arpnote]; + + arpnote = curnote + (((cptr->parameffect)&0xF)*8); + if( arpnote >= FULL_PERIOD_TABLE_LENGTH ) + arpnote = FULL_PERIOD_TABLE_LENGTH - 1; + + cptr->Arpperiods[2] = mod->fullperiod[arpnote]; + } + break; + + case EFFECT_PORTAMENTO_UP: + /* + [1]: Slide up + Where [1][x][y] means "smoothly decrease the period of current + sample by x*16+y after each tick in the division". The + ticks/division are set with the 'set speed' effect (see below). If + the period of the note being played is z, then the final period + will be z - (x*16 + y)*(ticks - 1). As the slide rate depends on + the speed, changing the speed will change the slide. You cannot + slide beyond the note B3 (period 113). + */ + + cptr->effect = EFFECT_PORTAMENTO_UP; + cptr->parameffect = effect&0xff; + break; + + case EFFECT_PORTAMENTO_DOWN: + /* + [2]: Slide down + Where [2][x][y] means "smoothly increase the period of current + sample by x*16+y after each tick in the division". Similar to [1], + but lowers the pitch. You cannot slide beyond the note C1 (period + 856). + */ + + cptr->effect = EFFECT_PORTAMENTO_DOWN; + cptr->parameffect = effect&0xff; + break; + + case EFFECT_TONE_PORTAMENTO: + /* + [3]: Slide to note + Where [3][x][y] means "smoothly change the period of current sample + by x*16+y after each tick in the division, never sliding beyond + current period". The period-length in this channel's division is a + parameter to this effect, and hence is not played. Sliding to a + note is similar to effects [1] and [2], but the slide will not go + beyond the given period, and the direction is implied by that + period. If x and y are both 0, then the old slide will continue. + */ + + cptr->effect = EFFECT_TONE_PORTAMENTO; + if( (effect&0xff) != 0 ) + { + cptr->portaspeed = (short)(effect&0xff); + } + + if(period!=0) + { + cptr->portaperiod = period; + cptr->period = operiod; + } + break; + + case EFFECT_VIBRATO: + /* + [4]: Vibrato + Where [4][x][y] means "oscillate the sample pitch using a + particular waveform with amplitude y/16 semitones, such that (x * + ticks)/64 cycles occur in the division". The waveform is set using + effect [14][4]. By placing vibrato effects on consecutive + divisions, the vibrato effect can be maintained. If either x or y + are 0, then the old vibrato values will be used. + */ + + cptr->effect = EFFECT_VIBRATO; + if( ( effect & 0x0F ) != 0 ) // Depth continue or change ? + cptr->vibraparam = (cptr->vibraparam & 0xF0) | ( effect & 0x0F ); + if( ( effect & 0xF0 ) != 0 ) // Speed continue or change ? + cptr->vibraparam = (cptr->vibraparam & 0x0F) | ( effect & 0xF0 ); + + break; + + case EFFECT_VOLSLIDE_TONEPORTA: + /* + [5]: Continue 'Slide to note', but also do Volume slide + Where [5][x][y] means "either slide the volume up x*(ticks - 1) or + slide the volume down y*(ticks - 1), at the same time as continuing + the last 'Slide to note'". It is illegal for both x and y to be + non-zero. You cannot slide outside the volume range 0..64. The + period-length in this channel's division is a parameter to this + effect, and hence is not played. + */ + + if( period != 0 ) + { + cptr->portaperiod = period; + cptr->period = operiod; + } + + cptr->effect = EFFECT_VOLSLIDE_TONEPORTA; + if( ( effect & 0xFF ) != 0 ) + cptr->volumeslide = ( effect & 0xFF ); + + break; + + case EFFECT_VOLSLIDE_VIBRATO: + /* + [6]: Continue 'Vibrato', but also do Volume slide + Where [6][x][y] means "either slide the volume up x*(ticks - 1) or + slide the volume down y*(ticks - 1), at the same time as continuing + the last 'Vibrato'". It is illegal for both x and y to be non-zero. + You cannot slide outside the volume range 0..64. + */ + + cptr->effect = EFFECT_VOLSLIDE_VIBRATO; + if( (effect & 0xFF) != 0 ) + cptr->volumeslide = (effect & 0xFF); + break; + + case EFFECT_SET_OFFSET: + /* + [9]: Set sample offset + Where [9][x][y] means "play the sample from offset x*4096 + y*256". + The offset is measured in words. If no sample is given, yet one is + still playing on this channel, it should be retriggered to the new + offset using the current volume. + */ + + cptr->samppos = ((effect>>4) * 4096) + ((effect&0xF)*256); + + break; + + case EFFECT_VOLUME_SLIDE: + /* + [10]: Volume slide + Where [10][x][y] means "either slide the volume up x*(ticks - 1) or + slide the volume down y*(ticks - 1)". If both x and y are non-zero, + then the y value is ignored (assumed to be 0). You cannot slide + outside the volume range 0..64. + */ + + cptr->effect = EFFECT_VOLUME_SLIDE; + cptr->volumeslide = (effect & 0xFF); + break; + + case EFFECT_JUMP_POSITION: + /* + [11]: Position Jump + Where [11][x][y] means "stop the pattern after this division, and + continue the song at song-position x*16+y". This shifts the + 'pattern-cursor' in the pattern table (see above). Legal values for + x*16+y are from 0 to 127. + */ + + mod->tablepos = (effect & 0xFF); + if(mod->tablepos >= mod->song.length) + { + mod->tablepos = 0; + } + mod->patternpos = 0; + mod->jump_loop_effect = 1; + + break; + + case EFFECT_SET_VOLUME: + /* + [12]: Set volume + Where [12][x][y] means "set current sample's volume to x*16+y". + Legal volumes are 0..64. + */ + + cptr->volume = (effect & 0xFF); + break; + + case EFFECT_PATTERN_BREAK: + /* + [13]: Pattern Break + Where [13][x][y] means "stop the pattern after this division, and + continue the song at the next pattern at division x*10+y" (the 10 + is not a typo). Legal divisions are from 0 to 63 (note Protracker + exception above). + */ + + mod->patternpos = ( ((effect>>4)&0xF)*10 + (effect&0xF) ) * mod->number_of_channels; + mod->jump_loop_effect = 1; + mod->tablepos++; + if(mod->tablepos >= mod->song.length) + { + mod->tablepos = 0; + } + + break; + + case EFFECT_EXTENDED: + switch( (effect>>4) & 0xF ) + { + case EFFECT_E_FINE_PORTA_UP: + /* + [14][1]: Fineslide up + Where [14][1][x] means "decrement the period of the current sample + by x". The incrementing takes place at the beginning of the + division, and hence there is no actual sliding. You cannot slide + beyond the note B3 (period 113). + */ + + cptr->period -= (effect & 0xF); + if( cptr->period < 113 ) + cptr->period = 113; + break; + + case EFFECT_E_FINE_PORTA_DOWN: + /* + [14][2]: Fineslide down + Where [14][2][x] means "increment the period of the current sample + by x". Similar to [14][1] but shifts the pitch down. You cannot + slide beyond the note C1 (period 856). + */ + + cptr->period += (effect & 0xF); + if( cptr->period > 856 ) + cptr->period = 856; + break; + + case EFFECT_E_FINE_VOLSLIDE_UP: + /* + [14][10]: Fine volume slide up + Where [14][10][x] means "increment the volume of the current sample + by x". The incrementing takes place at the beginning of the + division, and hence there is no sliding. You cannot slide beyond + volume 64. + */ + + cptr->volume += (effect & 0xF); + if( cptr->volume>64 ) + cptr->volume = 64; + break; + + case EFFECT_E_FINE_VOLSLIDE_DOWN: + /* + [14][11]: Fine volume slide down + Where [14][11][x] means "decrement the volume of the current sample + by x". Similar to [14][10] but lowers volume. You cannot slide + beyond volume 0. + */ + + cptr->volume -= (effect & 0xF); + if( cptr->volume > 200 ) + cptr->volume = 0; + break; + + case EFFECT_E_PATTERN_LOOP: + /* + [14][6]: Loop pattern + Where [14][6][x] means "set the start of a loop to this division if + x is 0, otherwise after this division, jump back to the start of a + loop and play it another x times before continuing". If the start + of the loop was not set, it will default to the start of the + current pattern. Hence 'loop pattern' cannot be performed across + multiple patterns. Note that loops do not support nesting, and you + may generate an infinite loop if you try to nest 'loop pattern's. + */ + + if( effect & 0xF ) + { + if( cptr->patternloopcnt ) + { + cptr->patternloopcnt--; + if( cptr->patternloopcnt ) + { + mod->patternpos = cptr->patternloopstartpoint; + mod->jump_loop_effect = 1; + } + else + { + cptr->patternloopstartpoint = mod->patternpos ; + } + } + else + { + cptr->patternloopcnt = (effect & 0xF); + mod->patternpos = cptr->patternloopstartpoint; + mod->jump_loop_effect = 1; + } + } + else // Start point + { + cptr->patternloopstartpoint = mod->patternpos; + } + + break; + + case EFFECT_E_PATTERN_DELAY: + /* + [14][14]: Delay pattern + Where [14][14][x] means "after this division there will be a delay + equivalent to the time taken to play x divisions after which the + pattern will be resumed". The delay only relates to the + interpreting of new divisions, and all effects and previous notes + continue during delay. + */ + + mod->patterndelay = (effect & 0xF); + break; + + case EFFECT_E_NOTE_CUT: + /* + [14][12]: Cut sample + Where [14][12][x] means "after the current sample has been played + for x ticks in this division, its volume will be set to 0". This + implies that if x is 0, then you will not hear any of the sample. + If you wish to insert "silence" in a pattern, it is better to use a + "silence"-sample (see above) due to the lack of proper support for + this effect. + */ + cptr->effect = EFFECT_E_NOTE_CUT; + cptr->cut_param = (effect & 0xF); + if(!cptr->cut_param) + cptr->volume = 0; + break; + + default: + + break; + } + break; + + case 0xF: + /* + [15]: Set speed + Where [15][x][y] means "set speed to x*16+y". Though it is nowhere + near that simple. Let z = x*16+y. Depending on what values z takes, + different units of speed are set, there being two: ticks/division + and beats/minute (though this one is only a label and not strictly + true). If z=0, then what should technically happen is that the + module stops, but in practice it is treated as if z=1, because + there is already a method for stopping the module (running out of + patterns). If z<=32, then it means "set ticks/division to z" + otherwise it means "set beats/minute to z" (convention says that + this should read "If z<32.." but there are some composers out there + that defy conventions). Default values are 6 ticks/division, and + 125 beats/minute (4 divisions = 1 beat). The beats/minute tag is + only meaningful for 6 ticks/division. To get a more accurate view + of how things work, use the following formula: + 24 * beats/minute + divisions/minute = ----------------- + ticks/division + Hence divisions/minute range from 24.75 to 6120, eg. to get a value + of 2000 divisions/minute use 3 ticks/division and 250 beats/minute. + If multiple "set speed" effects are performed in a single division, + the ones on higher-numbered channels take precedence over the ones + on lower-numbered channels. This effect has a large number of + different implementations, but the one described here has the + widest usage. + */ + + if( (effect&0xFF) < 0x21 ) + { + if( effect&0xFF ) + { + mod->song.speed = effect&0xFF; + mod->patternticksaim = (long)mod->song.speed * ((mod->playrate * 5 ) / (((long)2 * (long)mod->bpm))); + } + } + + if( (effect&0xFF) >= 0x21 ) + { + /// HZ = 2 * BPM / 5 + mod->bpm = effect&0xFF; + mod->patternticksaim = (long)mod->song.speed * ((mod->playrate * 5 ) / (((long)2 * (long)mod->bpm))); + } + + break; + + default: + // Unsupported effect + break; + + } + +} + +static void workeffect( note * nptr, channel * cptr ) +{ + switch(cptr->effect) + { + case EFFECT_ARPEGGIO: + + if( cptr->parameffect ) + { + cptr->decalperiod = cptr->period - cptr->Arpperiods[cptr->ArpIndex]; + + cptr->ArpIndex++; + if( cptr->ArpIndex>2 ) + cptr->ArpIndex = 0; + } + break; + + case EFFECT_PORTAMENTO_UP: + + if(cptr->period) + { + cptr->period -= cptr->parameffect; + + if( cptr->period < 113 || cptr->period > 20000 ) + cptr->period = 113; + } + + break; + + case EFFECT_PORTAMENTO_DOWN: + + if(cptr->period) + { + cptr->period += cptr->parameffect; + + if( cptr->period > 20000 ) + cptr->period = 20000; + } + + break; + + case EFFECT_VOLSLIDE_TONEPORTA: + case EFFECT_TONE_PORTAMENTO: + + if( cptr->period && ( cptr->period != cptr->portaperiod ) && cptr->portaperiod ) + { + if( cptr->period > cptr->portaperiod ) + { + if( cptr->period - cptr->portaperiod >= cptr->portaspeed ) + { + cptr->period -= cptr->portaspeed; + } + else + { + cptr->period = cptr->portaperiod; + } + } + else + { + if( cptr->portaperiod - cptr->period >= cptr->portaspeed ) + { + cptr->period += cptr->portaspeed; + } + else + { + cptr->period = cptr->portaperiod; + } + } + + if( cptr->period == cptr->portaperiod ) + { + // If the slide is over, don't let it to be retriggered. + cptr->portaperiod = 0; + } + } + + if( cptr->effect == EFFECT_VOLSLIDE_TONEPORTA ) + { + if( cptr->volumeslide > 0x0F ) + { + cptr->volume = cptr->volume + (cptr->volumeslide>>4); + + if(cptr->volume>63) + cptr->volume = 63; + } + else + { + cptr->volume = cptr->volume - (cptr->volumeslide); + + if(cptr->volume>63) + cptr->volume=0; + } + } + break; + + case EFFECT_VOLSLIDE_VIBRATO: + case EFFECT_VIBRATO: + + cptr->vibraperiod = ( (cptr->vibraparam&0xF) * sintable[cptr->vibrapointeur&0x1F] )>>7; + + if( cptr->vibrapointeur > 31 ) + cptr->vibraperiod = -cptr->vibraperiod; + + cptr->vibrapointeur = (cptr->vibrapointeur+(((cptr->vibraparam>>4))&0xf)) & 0x3F; + + if( cptr->effect == EFFECT_VOLSLIDE_VIBRATO ) + { + if( cptr->volumeslide > 0xF ) + { + cptr->volume = cptr->volume+(cptr->volumeslide>>4); + + if( cptr->volume > 64 ) + cptr->volume = 64; + } + else + { + cptr->volume = cptr->volume - cptr->volumeslide; + + if( cptr->volume > 64 ) + cptr->volume = 0; + } + } + + break; + + case EFFECT_VOLUME_SLIDE: + + if( cptr->volumeslide > 0xF ) + { + cptr->volume += (cptr->volumeslide>>4); + + if( cptr->volume > 64 ) + cptr->volume = 64; + } + else + { + cptr->volume -= (cptr->volumeslide&0xf); + + if( cptr->volume > 64 ) + cptr->volume = 0; + } + break; + + case EFFECT_E_NOTE_CUT: + if(cptr->cut_param) + cptr->cut_param--; + + if(!cptr->cut_param) + cptr->volume = 0; + break; + + default: + break; + + } + +} + +/////////////////////////////////////////////////////////////////////////////////// +bool jar_mod_init(jar_mod_context_t * modctx) +{ + muint i,j; + + if( modctx ) + { + memclear(modctx, 0, sizeof(jar_mod_context_t)); + modctx->playrate = DEFAULT_SAMPLE_RATE; + modctx->stereo = 1; + modctx->stereo_separation = 1; + modctx->bits = 16; + modctx->filter = 1; + modctx->loopcount = 0; + + for(i=0; i < PERIOD_TABLE_LENGTH - 1; i++) + { + for(j=0; j < 8; j++) + { + modctx->fullperiod[(i*8) + j] = periodtable[i] - ((( periodtable[i] - periodtable[i+1] ) / 8) * j); + } + } + + return 1; + } + + return 0; +} + +bool jar_mod_setcfg(jar_mod_context_t * modctx, int samplerate, int bits, int stereo, int stereo_separation, int filter) +{ + if( modctx ) + { + modctx->playrate = samplerate; + + if( stereo ) + modctx->stereo = 1; + else + modctx->stereo = 0; + + if(stereo_separation < 4) + { + modctx->stereo_separation = stereo_separation; + } + + if( bits == 8 || bits == 16 ) + modctx->bits = bits; + else + modctx->bits = 16; + + if( filter ) + modctx->filter = 1; + else + modctx->filter = 0; + + return 1; + } + + return 0; +} + +// make certain that mod_data stays in memory while playing +static bool jar_mod_load( jar_mod_context_t * modctx, void * mod_data, int mod_data_size ) +{ + muint i, max; + unsigned short t; + sample *sptr; + unsigned char * modmemory,* endmodmemory; + + modmemory = (unsigned char *)mod_data; + endmodmemory = modmemory + mod_data_size; + + + + if(modmemory) + { + if( modctx ) + { + memcopy(&(modctx->song.title),modmemory,1084); + + i = 0; + modctx->number_of_channels = 0; + while(modlist[i].numberofchannels) + { + if(memcompare(modctx->song.signature,modlist[i].signature,4)) + { + modctx->number_of_channels = modlist[i].numberofchannels; + } + + i++; + } + + if( !modctx->number_of_channels ) + { + // 15 Samples modules support + // Shift the whole datas to make it look likes a standard 4 channels mod. + memcopy(&(modctx->song.signature), "M.K.", 4); + memcopy(&(modctx->song.length), &(modctx->song.samples[15]), 130); + memclear(&(modctx->song.samples[15]), 0, 480); + modmemory += 600; + modctx->number_of_channels = 4; + } + else + { + modmemory += 1084; + } + + if( modmemory >= endmodmemory ) + return 0; // End passed ? - Probably a bad file ! + + // Patterns loading + for (i = max = 0; i < 128; i++) + { + while (max <= modctx->song.patterntable[i]) + { + modctx->patterndata[max] = (note*)modmemory; + modmemory += (256*modctx->number_of_channels); + max++; + + if( modmemory >= endmodmemory ) + return 0; // End passed ? - Probably a bad file ! + } + } + + for (i = 0; i < 31; i++) + modctx->sampledata[i]=0; + + // Samples loading + for (i = 0, sptr = modctx->song.samples; i <31; i++, sptr++) + { + t= (sptr->length &0xFF00)>>8 | (sptr->length &0xFF)<<8; + sptr->length = t*2; + + t= (sptr->reppnt &0xFF00)>>8 | (sptr->reppnt &0xFF)<<8; + sptr->reppnt = t*2; + + t= (sptr->replen &0xFF00)>>8 | (sptr->replen &0xFF)<<8; + sptr->replen = t*2; + + + if (sptr->length == 0) continue; + + modctx->sampledata[i] = (char*)modmemory; + modmemory += sptr->length; + + if (sptr->replen + sptr->reppnt > sptr->length) + sptr->replen = sptr->length - sptr->reppnt; + + if( modmemory > endmodmemory ) + return 0; // End passed ? - Probably a bad file ! + } + + // States init + + modctx->tablepos = 0; + modctx->patternpos = 0; + modctx->song.speed = 6; + modctx->bpm = 125; + modctx->samplenb = 0; + + modctx->patternticks = (((long)modctx->song.speed * modctx->playrate * 5)/ (2 * modctx->bpm)) + 1; + modctx->patternticksaim = ((long)modctx->song.speed * modctx->playrate * 5) / (2 * modctx->bpm); + + modctx->sampleticksconst = 3546894UL / modctx->playrate; //8448*428/playrate; + + for(i=0; i < modctx->number_of_channels; i++) + { + modctx->channels[i].volume = 0; + modctx->channels[i].period = 0; + } + + modctx->mod_loaded = 1; + + return 1; + } + } + + return 0; +} + +void jar_mod_fillbuffer( jar_mod_context_t * modctx, short * outbuffer, unsigned long nbsample, jar_mod_tracker_buffer_state * trkbuf ) +{ + unsigned long i, j; + unsigned long k; + unsigned char c; + unsigned int state_remaining_steps; + int l,r; + int ll,lr; + int tl,tr; + short finalperiod; + note *nptr; + channel *cptr; + + if( modctx && outbuffer ) + { + if(modctx->mod_loaded) + { + state_remaining_steps = 0; + + if( trkbuf ) + { + trkbuf->cur_rd_index = 0; + + memcopy(trkbuf->name,modctx->song.title,sizeof(modctx->song.title)); + + for(i=0;i<31;i++) + { + memcopy(trkbuf->instruments[i].name,modctx->song.samples[i].name,sizeof(trkbuf->instruments[i].name)); + } + } + + ll = modctx->last_l_sample; + lr = modctx->last_r_sample; + + for (i = 0; i < nbsample; i++) + { + //--------------------------------------- + if( modctx->patternticks++ > modctx->patternticksaim ) + { + if( !modctx->patterndelay ) + { + nptr = modctx->patterndata[modctx->song.patterntable[modctx->tablepos]]; + nptr = nptr + modctx->patternpos; + cptr = modctx->channels; + + modctx->patternticks = 0; + modctx->patterntickse = 0; + + for(c=0;cnumber_of_channels;c++) + { + worknote((note*)(nptr+c), (channel*)(cptr+c),(char)(c+1),modctx); + } + + if( !modctx->jump_loop_effect ) + modctx->patternpos += modctx->number_of_channels; + else + modctx->jump_loop_effect = 0; + + if( modctx->patternpos == 64*modctx->number_of_channels ) + { + modctx->tablepos++; + modctx->patternpos = 0; + if(modctx->tablepos >= modctx->song.length) + { + modctx->tablepos = 0; + modctx->loopcount++; // count next loop + } + } + } + else + { + modctx->patterndelay--; + modctx->patternticks = 0; + modctx->patterntickse = 0; + } + + } + + if( modctx->patterntickse++ > (modctx->patternticksaim/modctx->song.speed) ) + { + nptr = modctx->patterndata[modctx->song.patterntable[modctx->tablepos]]; + nptr = nptr + modctx->patternpos; + cptr = modctx->channels; + + for(c=0;cnumber_of_channels;c++) + { + workeffect(nptr+c, cptr+c); + } + + modctx->patterntickse = 0; + } + + //--------------------------------------- + + if( trkbuf && !state_remaining_steps ) + { + if( trkbuf->nb_of_state < trkbuf->nb_max_of_state ) + { + memclear(&trkbuf->track_state_buf[trkbuf->nb_of_state], 0, sizeof(tracker_state)); + } + } + + l=0; + r=0; + + for(j =0, cptr = modctx->channels; j < modctx->number_of_channels ; j++, cptr++) + { + if( cptr->period != 0 ) + { + finalperiod = cptr->period - cptr->decalperiod - cptr->vibraperiod; + if( finalperiod ) + { + cptr->samppos += ( (modctx->sampleticksconst<<10) / finalperiod ); + } + + cptr->ticks++; + + if( cptr->replen<=2 ) + { + if( (cptr->samppos>>10) >= (cptr->length) ) + { + cptr->length = 0; + cptr->reppnt = 0; + + if( cptr->length ) + cptr->samppos = cptr->samppos % (((unsigned long)cptr->length)<<10); + else + cptr->samppos = 0; + } + } + else + { + if( (cptr->samppos>>10) >= (unsigned long)(cptr->replen+cptr->reppnt) ) + { + cptr->samppos = ((unsigned long)(cptr->reppnt)<<10) + (cptr->samppos % ((unsigned long)(cptr->replen+cptr->reppnt)<<10)); + } + } + + k = cptr->samppos >> 10; + + if( cptr->sampdata!=0 && ( ((j&3)==1) || ((j&3)==2) ) ) + { + r += ( cptr->sampdata[k] * cptr->volume ); + } + + if( cptr->sampdata!=0 && ( ((j&3)==0) || ((j&3)==3) ) ) + { + l += ( cptr->sampdata[k] * cptr->volume ); + } + + if( trkbuf && !state_remaining_steps ) + { + if( trkbuf->nb_of_state < trkbuf->nb_max_of_state ) + { + trkbuf->track_state_buf[trkbuf->nb_of_state].number_of_tracks = modctx->number_of_channels; + trkbuf->track_state_buf[trkbuf->nb_of_state].buf_index = i; + trkbuf->track_state_buf[trkbuf->nb_of_state].cur_pattern = modctx->song.patterntable[modctx->tablepos]; + trkbuf->track_state_buf[trkbuf->nb_of_state].cur_pattern_pos = modctx->patternpos / modctx->number_of_channels; + trkbuf->track_state_buf[trkbuf->nb_of_state].cur_pattern_table_pos = modctx->tablepos; + trkbuf->track_state_buf[trkbuf->nb_of_state].bpm = modctx->bpm; + trkbuf->track_state_buf[trkbuf->nb_of_state].speed = modctx->song.speed; + trkbuf->track_state_buf[trkbuf->nb_of_state].tracks[j].cur_effect = cptr->effect_code; + trkbuf->track_state_buf[trkbuf->nb_of_state].tracks[j].cur_parameffect = cptr->parameffect; + trkbuf->track_state_buf[trkbuf->nb_of_state].tracks[j].cur_period = finalperiod; + trkbuf->track_state_buf[trkbuf->nb_of_state].tracks[j].cur_volume = cptr->volume; + trkbuf->track_state_buf[trkbuf->nb_of_state].tracks[j].instrument_number = (unsigned char)cptr->sampnum; + } + } + } + } + + if( trkbuf && !state_remaining_steps ) + { + state_remaining_steps = trkbuf->sample_step; + + if(trkbuf->nb_of_state < trkbuf->nb_max_of_state) + trkbuf->nb_of_state++; + } + else + { + state_remaining_steps--; + } + + tl = (short)l; + tr = (short)r; + + if ( modctx->filter ) + { + // Filter + l = (l+ll)>>1; + r = (r+lr)>>1; + } + + if ( modctx->stereo_separation == 1 ) + { + // Left & Right Stereo panning + l = (l+(r>>1)); + r = (r+(l>>1)); + } + + // Level limitation + if( l > 32767 ) l = 32767; + if( l < -32768 ) l = -32768; + if( r > 32767 ) r = 32767; + if( r < -32768 ) r = -32768; + + // Store the final sample. + outbuffer[(i*2)] = l; + outbuffer[(i*2)+1] = r; + + ll = tl; + lr = tr; + + } + + modctx->last_l_sample = ll; + modctx->last_r_sample = lr; + + modctx->samplenb = modctx->samplenb+nbsample; + } + else + { + for (i = 0; i < nbsample; i++) + { + // Mod not loaded. Return blank buffer. + outbuffer[(i*2)] = 0; + outbuffer[(i*2)+1] = 0; + } + + if(trkbuf) + { + trkbuf->nb_of_state = 0; + trkbuf->cur_rd_index = 0; + trkbuf->name[0] = 0; + memclear(trkbuf->track_state_buf, 0, sizeof(tracker_state) * trkbuf->nb_max_of_state); + memclear(trkbuf->instruments, 0, sizeof(trkbuf->instruments)); + } + } + } +} + +//resets internals for mod context +static void jar_mod_reset( jar_mod_context_t * modctx) +{ + if(modctx) + { + memclear(&modctx->song, 0, sizeof(modctx->song)); + memclear(&modctx->sampledata, 0, sizeof(modctx->sampledata)); + memclear(&modctx->patterndata, 0, sizeof(modctx->patterndata)); + modctx->tablepos = 0; + modctx->patternpos = 0; + modctx->patterndelay = 0; + modctx->jump_loop_effect = 0; + modctx->bpm = 0; + modctx->patternticks = 0; + modctx->patterntickse = 0; + modctx->patternticksaim = 0; + modctx->sampleticksconst = 0; + modctx->loopcount = 0; + modctx->samplenb = 0; + memclear(modctx->channels, 0, sizeof(modctx->channels)); + modctx->number_of_channels = 0; + modctx->mod_loaded = 0; + modctx->last_r_sample = 0; + modctx->last_l_sample = 0; + + jar_mod_init(modctx); + } +} + +void jar_mod_unload( jar_mod_context_t * modctx) +{ + if(modctx) + { + if(modctx->modfile) + { + free(modctx->modfile); + modctx->modfile = 0; + } + jar_mod_reset(modctx); + } +} + + + +mulong jar_mod_load_file(jar_mod_context_t * modctx, char* filename) +{ + mulong fsize = 0; + if(modctx->modfile) + { + free(modctx->modfile); + modctx->modfile = 0; + } + + FILE *f = fopen(filename, "rb"); + if(f) + { + fseek(f,0,SEEK_END); + fsize = ftell(f); + fseek(f,0,SEEK_SET); + + if(fsize && fsize < 32*1024*1024) + { + modctx->modfile = malloc(fsize); + modctx->modfilesize = fsize; + memset(modctx->modfile, 0, fsize); + fread(modctx->modfile, fsize, 1, f); + fclose(f); + + if(!jar_mod_load(modctx, (void*)modctx->modfile, fsize)) fsize = 0; + } else fsize = 0; + } + return fsize; +} + +mulong jar_mod_current_samples(jar_mod_context_t * modctx) +{ + if(modctx) + return modctx->samplenb; + + return 0; +} + +// Works, however it is very slow, this data should be cached to ensure it is run only once per file +mulong jar_mod_max_samples(jar_mod_context_t * ctx) +{ + jar_mod_context_t tmpctx; + jar_mod_init(&tmpctx); + if(!jar_mod_load(&tmpctx, (void*)ctx->modfile, ctx->modfilesize)) return 0; + + muint buff[2]; + mulong lastcount = tmpctx.loopcount; + + while(1){ + jar_mod_fillbuffer( &tmpctx, buff, 1, 0 ); + if(tmpctx.loopcount > lastcount) break; + } + return tmpctx.samplenb; +} + +// move seek_val to sample index, 0 -> jar_mod_max_samples is the range +void jar_mod_seek_start(jar_mod_context_t * ctx) +{ + if(ctx) + { + jar_mod_reset(ctx); + jar_mod_load(ctx, ctx->modfile, ctx->modfilesize); + } +} + +#endif // end of JAR_MOD_IMPLEMENTATION +//------------------------------------------------------------------------------- + + +#endif //end of header file \ No newline at end of file diff --git a/src/raylib.h b/src/raylib.h index 1ef0a98ee..21892d68b 100644 --- a/src/raylib.h +++ b/src/raylib.h @@ -261,8 +261,9 @@ //---------------------------------------------------------------------------------- #ifndef __cplusplus // Boolean type - #ifndef true + #if !defined(_STDBOOL_H) typedef enum { false, true } bool; + #define _STDBOOL_H #endif #endif @@ -451,10 +452,29 @@ typedef struct Ray { Vector3 direction; } 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 typedef struct Sound { unsigned int source; unsigned int buffer; + AudioError error; // if there was any error during the creation or use of this Sound } Sound; // Wave type, defines audio wave data @@ -468,6 +488,8 @@ typedef struct Wave { typedef int RawAudioContext; + + // Texture formats // NOTE: Support depends on OpenGL version and platform typedef enum { @@ -926,7 +948,7 @@ void SetMusicPitch(int index, float pitch); RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); 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 }