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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
View file

@ -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()

87
external/openal_soft/include/AL/alext.h vendored
View file

@ -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

2
external/openal_soft/include/AL/efx-presets.h vendored
View file

@ -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 }

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
#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_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(!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(&currentMusic[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<numBuffers; x++)
{
jar_xm_generate_samples_16bit(currentMusic[index].chipctx, pcm, size); // reads 2*readlen shorts and moves them to buffer+size memory location
BufferMixChannel(currentMusic[index].mixc, pcm, size * 2);
currentMusic[index].totalSamplesLeft -= size * 2;
if(currentMusic[index].totalSamplesLeft <= 0)
if(musicChannels_g[index].modctx.mod_loaded){
if(musicChannels_g[index].totalSamplesLeft >= 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; x<numBuffers; x++)
{
int streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].mixc->channels, 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);

25
src/audio.h
View file

@ -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
}

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
// 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
}