diff options
Diffstat (limited to 'src/audio.c')
| -rw-r--r-- | src/audio.c | 526 |
1 files changed, 308 insertions, 218 deletions
diff --git a/src/audio.c b/src/audio.c index 0c61c0fa..ad1d7eba 100644 --- a/src/audio.c +++ b/src/audio.c @@ -35,26 +35,29 @@ #include "raylib.h" #endif -#include "AL/al.h" // OpenAL basic header -#include "AL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work) -#include "AL/alext.h" // extensions for other format types +#include "AL/al.h" // OpenAL basic header +#include "AL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work) +#include "AL/alext.h" // OpenAL extensions for other format types -#include <stdlib.h> // Declares malloc() and free() for memory management -#include <string.h> // Required for strcmp() -#include <stdio.h> // Used for .WAV loading +#include <stdlib.h> // Required for: malloc(), free() +#include <string.h> // Required for: strcmp(), strncmp() +#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread() #if defined(AUDIO_STANDALONE) - #include <stdarg.h> // Used for functions with variable number of parameters (TraceLog()) + #include <stdarg.h> // Required for: va_list, va_start(), vfprintf(), va_end() #else - #include "utils.h" // rRES data decompression utility function - // NOTE: Includes Android fopen function map + #include "utils.h" // Required for: DecompressData() + // NOTE: Includes Android fopen() function map #endif //#define STB_VORBIS_HEADER_ONLY -#include "stb_vorbis.h" // OGG loading functions +#include "external/stb_vorbis.h" // OGG loading functions #define JAR_XM_IMPLEMENTATION -#include "jar_xm.h" // For playing .xm files +#include "external/jar_xm.h" // XM loading functions + +#define JAR_MOD_IMPLEMENTATION +#include "external/jar_mod.h" // MOD loading functions //---------------------------------------------------------------------------------- // Defines and Macros @@ -86,24 +89,45 @@ typedef struct MixChannel_t { unsigned char mixChannel; // 0-3 or mixA-mixD, each mix channel can receive up to one dedicated audio stream bool floatingPoint; // if false then the short datatype is used instead bool playing; // false if paused - ALenum alFormat; // openAL format specifier - ALuint alSource; // openAL source - ALuint alBuffer[MAX_STREAM_BUFFERS]; // openAL sample buffer + + ALenum alFormat; // OpenAL format specifier + ALuint alSource; // OpenAL source + ALuint alBuffer[MAX_STREAM_BUFFERS]; // OpenAL sample buffer } MixChannel_t; // Music type (file streaming from memory) // 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; // XM chiptune context + jar_mod_context_t modctx; // MOD chiptune context MixChannel_t *mixc; // mix channel - int totalSamplesLeft; + unsigned int totalSamplesLeft; float totalLengthSeconds; bool loop; - bool chipTune; // True if chiptune is loaded + bool chipTune; // chiptune is loaded? } Music; +// Audio errors registered +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; + #if defined(AUDIO_STANDALONE) typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType; #endif @@ -111,9 +135,11 @@ 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 Music musicChannels_g[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same time +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 int lastAudioError = 0; // Registers last audio error //---------------------------------------------------------------------------------- // Module specific Functions Declaration @@ -125,10 +151,9 @@ static void UnloadWave(Wave wave); // Unload wave data static bool BufferMusicStream(int index, int numBuffers); // Fill music buffers with data static void EmptyMusicStream(int index); // Empty music buffers - -static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); // For streaming into mix channels. -static void CloseMixChannel(MixChannel_t* mixc); // Frees mix channel -static int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses +static MixChannel_t *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); // For streaming into mix channels. +static void CloseMixChannel(MixChannel_t *mixc); // Frees mix channel +static int BufferMixChannel(MixChannel_t *mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses static int FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer); // Fill buffer with zeros, returns number processed static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // Pass two arrays of the same legnth in static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // Pass two arrays of same length in @@ -149,13 +174,13 @@ void InitAudioDevice(void) // Open and initialize a device with default settings ALCdevice *device = alcOpenDevice(NULL); - if(!device) TraceLog(ERROR, "Audio device could not be opened"); + if (!device) TraceLog(ERROR, "Audio device could not be opened"); ALCcontext *context = alcCreateContext(device, NULL); - if(context == NULL || alcMakeContextCurrent(context) == ALC_FALSE) + if ((context == NULL) || (alcMakeContextCurrent(context) == ALC_FALSE)) { - if(context != NULL) alcDestroyContext(context); + if (context != NULL) alcDestroyContext(context); alcCloseDevice(device); @@ -173,11 +198,10 @@ void InitAudioDevice(void) // Close the audio device for all contexts void CloseAudioDevice(void) { - for(int index=0; index<MAX_MUSIC_STREAMS; index++) + for (int index=0; index<MAX_MUSIC_STREAMS; index++) { - if(currentMusic[index].mixc) StopMusicStream(index); // Stop music streaming and close current stream + if (musicChannels_g[index].mixc) StopMusicStream(index); // Stop music streaming and close current stream } - ALCdevice *device; ALCcontext *context = alcGetCurrentContext(); @@ -195,9 +219,12 @@ void CloseAudioDevice(void) bool IsAudioDeviceReady(void) { ALCcontext *context = alcGetCurrentContext(); + if (context == NULL) return false; - else{ + else + { ALCdevice *device = alcGetContextsDevice(context); + if (device == NULL) return false; else return true; } @@ -210,33 +237,30 @@ bool IsAudioDeviceReady(void) // For streaming into mix channels. // The mixChannel is what audio muxing channel you want to operate on, 0-3 are the ones available. Each mix channel can only be used one at a time. // exmple usage is InitMixChannel(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point -static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint) +static MixChannel_t *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint) { - if(mixChannel >= MAX_MIX_CHANNELS) return NULL; - if(!IsAudioDeviceReady()) InitAudioDevice(); + if (mixChannel >= MAX_MIX_CHANNELS) return NULL; + if (!IsAudioDeviceReady()) InitAudioDevice(); - if(!mixChannelsActive_g[mixChannel]){ - MixChannel_t *mixc = (MixChannel_t*)malloc(sizeof(MixChannel_t)); + 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) + // Setup OpenAL format + if (channels == 1) { - if(floatingPoint) - mixc->alFormat = AL_FORMAT_MONO_FLOAT32; - else - mixc->alFormat = AL_FORMAT_MONO16; + if (floatingPoint) mixc->alFormat = AL_FORMAT_MONO_FLOAT32; + else mixc->alFormat = AL_FORMAT_MONO16; } - else if(channels == 2) + else if (channels == 2) { - if(floatingPoint) - mixc->alFormat = AL_FORMAT_STEREO_FLOAT32; - else - mixc->alFormat = AL_FORMAT_STEREO16; + if (floatingPoint) mixc->alFormat = AL_FORMAT_STEREO_FLOAT32; + else mixc->alFormat = AL_FORMAT_STEREO16; } // Create an audio source @@ -249,10 +273,8 @@ static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mix // Create Buffer alGenBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer); - //fill buffers - int x; - for(x=0;x<MAX_STREAM_BUFFERS;x++) - FillAlBufferWithSilence(mixc, mixc->alBuffer[x]); + // Fill buffers + for (int i = 0; i < MAX_STREAM_BUFFERS; i++) FillAlBufferWithSilence(mixc, mixc->alBuffer[i]); alSourceQueueBuffers(mixc->alSource, MAX_STREAM_BUFFERS, mixc->alBuffer); mixc->playing = true; @@ -260,30 +282,33 @@ static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mix return mixc; } + return NULL; } // Frees buffer in mix channel -static void CloseMixChannel(MixChannel_t* mixc) +static void CloseMixChannel(MixChannel_t *mixc) { - if(mixc){ + if (mixc) + { alSourceStop(mixc->alSource); mixc->playing = false; - //flush out all queued buffers + // Flush out all queued buffers ALuint buffer = 0; int queued = 0; alGetSourcei(mixc->alSource, AL_BUFFERS_QUEUED, &queued); + while (queued > 0) { alSourceUnqueueBuffers(mixc->alSource, 1, &buffer); queued--; } - //delete source and buffers + // 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; } @@ -292,39 +317,46 @@ static void CloseMixChannel(MixChannel_t* mixc) // Pushes more audio data into mixc mix channel, only one buffer per call // Call "BufferMixChannel(mixc, NULL, 0)" if you want to pause the audio. // @Returns number of samples that where processed. -static int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements) +static int BufferMixChannel(MixChannel_t *mixc, void *data, int numberElements) { - if(!mixc || mixChannelsActive_g[mixc->mixChannel] != mixc) return 0; // when there is two channels there must be an even number of samples + if (!mixc || (mixChannels_g[mixc->mixChannel] != mixc)) return 0; // When there is two channels there must be an even number of samples - if (!data || !numberElements) - { // pauses audio until data is given - if(mixc->playing){ + if (!data || !numberElements) + { + // Pauses audio until data is given + if (mixc->playing) + { alSourcePause(mixc->alSource); mixc->playing = false; } + return 0; } - else if(!mixc->playing) - { // restart audio otherwise + else if (!mixc->playing) + { + // Restart audio otherwise alSourcePlay(mixc->alSource); mixc->playing = true; } - - + ALuint buffer = 0; alSourceUnqueueBuffers(mixc->alSource, 1, &buffer); - if(!buffer) return 0; - if(mixc->floatingPoint) // process float buffers + if (!buffer) return 0; + + if (mixc->floatingPoint) { - float *ptr = (float*)data; + // Process float buffers + float *ptr = (float *)data; alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(float), mixc->sampleRate); } - else // process short buffers + else { - short *ptr = (short*)data; + // Process short buffers + short *ptr = (short *)data; alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(short), mixc->sampleRate); } + alSourceQueueBuffers(mixc->alSource, 1, &buffer); return numberElements; @@ -333,15 +365,18 @@ static int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements) // fill buffer with zeros, returns number processed static int FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer) { - if(mixc->floatingPoint){ - float pcm[MUSIC_BUFFER_SIZE_FLOAT] = {0.f}; + if (mixc->floatingPoint) + { + float pcm[MUSIC_BUFFER_SIZE_FLOAT] = { 0.0f }; alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), mixc->sampleRate); + return MUSIC_BUFFER_SIZE_FLOAT; } else { - short pcm[MUSIC_BUFFER_SIZE_SHORT] = {0}; + short pcm[MUSIC_BUFFER_SIZE_SHORT] = { 0 }; alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), mixc->sampleRate); + return MUSIC_BUFFER_SIZE_SHORT; } } @@ -351,13 +386,10 @@ static int FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer) // ResampleShortToFloat(sh,fl,3); static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len) { - int x; - for(x=0;x<len;x++) + for (int i = 0; i < len; i++) { - if(shorts[x] < 0) - floats[x] = (float)shorts[x] / 32766.f; - else - floats[x] = (float)shorts[x] / 32767.f; + if (shorts[i] < 0) floats[i] = (float)shorts[i]/32766.0f; + else floats[i] = (float)shorts[i]/32767.0f; } } @@ -366,54 +398,49 @@ static void ResampleShortToFloat(short *shorts, float *floats, unsigned short le // ResampleByteToFloat(ch,fl,3); static void ResampleByteToFloat(char *chars, float *floats, unsigned short len) { - int x; - for(x=0;x<len;x++) + for (int i = 0; i < len; i++) { - if(chars[x] < 0) - floats[x] = (float)chars[x] / 127.f; - else - floats[x] = (float)chars[x] / 128.f; + if (chars[i] < 0) floats[i] = (float)chars[i]/127.0f; + else floats[i] = (float)chars[i]/128.0f; } } -// 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 + 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 + if (InitMixChannel(sampleRate, mixIndex, channels, floatingPoint)) return mixIndex; + else 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) + + if (ctx >= 0) { - MixChannel_t* mixc = mixChannelsActive_g[ctx]; + MixChannel_t* mixc = mixChannels_g[ctx]; numBuffered = BufferMixChannel(mixc, data, numberElements); } + return numBuffered; } - - - - //---------------------------------------------------------------------------------- // Module Functions Definition - Sounds loading and playing (.WAV) //---------------------------------------------------------------------------------- @@ -431,7 +458,13 @@ 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); + + // TODO: Find a better way to register errors (similar to glGetError()) + lastAudioError = ERROR_EXTENSION_NOT_RECOGNIZED; + } if (wave.data != NULL) { @@ -558,6 +591,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId) if (rresFile == NULL) { TraceLog(WARNING, "[%s] rRES raylib resource file could not be opened", rresName); + lastAudioError = ERROR_UNABLE_TO_OPEN_RRES_FILE; } else { @@ -572,6 +606,7 @@ 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); + lastAudioError = ERROR_INVALID_RRES_FILE; } else { @@ -662,6 +697,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId) else { TraceLog(WARNING, "[%s] Required resource do not seem to be a valid SOUND resource", rresName); + lastAudioError = ERROR_INVALID_RRES_RESOURCE; } } else @@ -763,119 +799,156 @@ void SetSoundPitch(Sound sound, float pitch) // Start music playing (open stream) // returns 0 on success -int PlayMusicStream(int musicIndex, char *fileName) +int PlayMusicStream(int index, char *fileName) { int mixIndex; - if(currentMusic[musicIndex].stream || currentMusic[musicIndex].chipctx) return 1; // error + if (musicChannels_g[index].stream || musicChannels_g[index].xmctx) return ERROR_UNINITIALIZED_CHANNELS; // error - for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot + for (mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot { - if(mixChannelsActive_g[mixIndex] == NULL) break; - 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[index].stream = stb_vorbis_open_filename(fileName, NULL, NULL); - if (currentMusic[musicIndex].stream == NULL) + if (musicChannels_g[index].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[index].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[index].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[index].totalSamplesLeft = (unsigned int)stb_vorbis_stream_length_in_samples(musicChannels_g[index].stream) * info.channels; + musicChannels_g[index].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(musicChannels_g[index].stream); - if (info.channels == 2){ - currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false); - currentMusic[musicIndex].mixc->playing = true; + if (info.channels == 2) + { + musicChannels_g[index].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false); + musicChannels_g[index].mixc->playing = true; } - else{ - currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false); - currentMusic[musicIndex].mixc->playing = true; + else + { + musicChannels_g[index].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false); + musicChannels_g[index].mixc->playing = true; } - if(!currentMusic[musicIndex].mixc) return 4; // error + + if (!musicChannels_g[index].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[index].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[index].chipTune = true; + musicChannels_g[index].loop = true; + jar_xm_set_max_loop_count(musicChannels_g[index].xmctx, 0); // infinite number of loops + musicChannels_g[index].totalSamplesLeft = (unsigned int)jar_xm_get_remaining_samples(musicChannels_g[index].xmctx); + musicChannels_g[index].totalLengthSeconds = ((float)musicChannels_g[index].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[index].totalSamplesLeft); + TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, musicChannels_g[index].totalLengthSeconds); + + musicChannels_g[index].mixc = InitMixChannel(48000, mixIndex, 2, true); + + if (!musicChannels_g[index].mixc) return ERROR_XM_CONTEXT_CREATION; // error - currentMusic[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, false); - if(!currentMusic[musicIndex].mixc) return 5; // error - currentMusic[musicIndex].mixc->playing = true; + musicChannels_g[index].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[index].modctx); + + if (jar_mod_load_file(&musicChannels_g[index].modctx, fileName)) + { + musicChannels_g[index].chipTune = true; + musicChannels_g[index].loop = true; + musicChannels_g[index].totalSamplesLeft = (unsigned int)jar_mod_max_samples(&musicChannels_g[index].modctx); + musicChannels_g[index].totalLengthSeconds = ((float)musicChannels_g[index].totalSamplesLeft) / 48000.f; + musicEnabled_g = true; + + TraceLog(INFO, "[%s] MOD number of samples: %i", fileName, musicChannels_g[index].totalSamplesLeft); + TraceLog(INFO, "[%s] MOD track length: %11.6f sec", fileName, musicChannels_g[index].totalLengthSeconds); + + musicChannels_g[index].mixc = InitMixChannel(48000, mixIndex, 2, false); + + if (!musicChannels_g[index].mixc) return ERROR_MOD_CONTEXT_CREATION; // error + + musicChannels_g[index].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) - { - jar_xm_free_context(currentMusic[index].chipctx); - } - else + if (musicChannels_g[index].chipTune && musicChannels_g[index].xmctx) { - stb_vorbis_close(currentMusic[index].stream); + 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(musicChannels_g[index].stream); + + if (!GetMusicStreamCount()) musicEnabled_g = false; - if(!getMusicStreamCount()) musicEnabled_g = false; - if(currentMusic[index].stream || currentMusic[index].chipctx) + if (musicChannels_g[index].stream || musicChannels_g[index].xmctx) { - currentMusic[index].stream = NULL; - currentMusic[index].chipctx = NULL; + musicChannels_g[index].stream = NULL; + musicChannels_g[index].xmctx = NULL; } } } //get number of music channels active at this time, this does not mean they are playing -int getMusicStreamCount(void) +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++; + + // Find empty music slot + for (int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++) + { + if(musicChannels_g[musicIndex].stream != NULL || musicChannels_g[musicIndex].chipTune) musicCount++; + } return musicCount; } @@ -884,11 +957,11 @@ 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 +970,16 @@ 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 +990,10 @@ 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; } @@ -925,30 +1003,28 @@ 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); } } +// Set pitch for music 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) - { - totalSeconds = currentMusic[index].totalLengthSeconds; - } - else - { - totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[index].stream); - } + + if (musicChannels_g[index].chipTune) totalSeconds = (float)musicChannels_g[index].totalLengthSeconds; + else totalSeconds = stb_vorbis_stream_length_in_seconds(musicChannels_g[index].stream); return totalSeconds; } @@ -957,19 +1033,25 @@ 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); } } @@ -986,22 +1068,33 @@ static bool BufferMusicStream(int index, int numBuffers) short pcm[MUSIC_BUFFER_SIZE_SHORT]; float pcmf[MUSIC_BUFFER_SIZE_FLOAT]; - 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) + 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++) + for (int i = 0; i < numBuffers; i++) { - 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 +1103,16 @@ static bool BufferMusicStream(int index, int numBuffers) } else { - if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) - size = MUSIC_BUFFER_SIZE_SHORT; - else - size = currentMusic[index].totalSamplesLeft; + if (musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) size = MUSIC_BUFFER_SIZE_SHORT; + else size = musicChannels_g[index].totalSamplesLeft; - for(int x=0; x<numBuffers; x++) + for (int i = 0; i < numBuffers; i++) { - 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; @@ -1037,21 +1129,21 @@ 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--; } } -//determine if a music stream is ready to be written to +// Determine if a music stream is ready to be written 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; } @@ -1062,37 +1154,35 @@ 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; } - 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); } - else - return; - } // Load WAV file into Wave structure |