diff options
Diffstat (limited to 'libs/raylib/src/raudio.c')
| -rw-r--r-- | libs/raylib/src/raudio.c | 1132 |
1 files changed, 704 insertions, 428 deletions
diff --git a/libs/raylib/src/raudio.c b/libs/raylib/src/raudio.c index 6313b16..2b6b0a1 100644 --- a/libs/raylib/src/raudio.c +++ b/libs/raylib/src/raudio.c @@ -1,6 +1,6 @@ /********************************************************************************************** * -* raudio - A simple and easy-to-use audio library based on miniaudio +* raudio v1.0 - A simple and easy-to-use audio library based on miniaudio * * FEATURES: * - Manage audio device (init/close) @@ -46,7 +46,7 @@ * * LICENSE: zlib/libpng * -* Copyright (c) 2013-2020 Ramon Santamaria (@raysan5) +* Copyright (c) 2013-2021 Ramon Santamaria (@raysan5) * * This software is provided "as-is", without any express or implied warranty. In no event * will the authors be held liable for any damages arising from the use of this software. @@ -159,7 +159,11 @@ typedef struct tagBITMAPINFOHEADER { #define MA_FREE RL_FREE #define MA_NO_JACK +#define MA_NO_WAV +#define MA_NO_FLAC +#define MA_NO_MP3 #define MINIAUDIO_IMPLEMENTATION +//#define MA_DEBUG_OUTPUT #include "external/miniaudio.h" // miniaudio library #undef PlaySound // Win32 API: windows.h > mmsystem.h defines PlaySound macro @@ -172,6 +176,20 @@ typedef struct tagBITMAPINFOHEADER { #if !defined(TRACELOG) #define TRACELOG(level, ...) (void)0 #endif + + // Allow custom memory allocators + #ifndef RL_MALLOC + #define RL_MALLOC(sz) malloc(sz) + #endif + #ifndef RL_CALLOC + #define RL_CALLOC(n,sz) calloc(n,sz) + #endif + #ifndef RL_REALLOC + #define RL_REALLOC(ptr,sz) realloc(ptr,sz) + #endif + #ifndef RL_FREE + #define RL_FREE(ptr) free(ptr) + #endif #endif #if defined(SUPPORT_FILEFORMAT_OGG) @@ -197,14 +215,13 @@ typedef struct tagBITMAPINFOHEADER { #include "external/jar_mod.h" // MOD loading functions #endif -#if defined(SUPPORT_FILEFORMAT_FLAC) - #define DRFLAC_MALLOC RL_MALLOC - #define DRFLAC_REALLOC RL_REALLOC - #define DRFLAC_FREE RL_FREE +#if defined(SUPPORT_FILEFORMAT_WAV) + #define DRWAV_MALLOC RL_MALLOC + #define DRWAV_REALLOC RL_REALLOC + #define DRWAV_FREE RL_FREE - #define DR_FLAC_IMPLEMENTATION - #define DR_FLAC_NO_WIN32_IO - #include "external/dr_flac.h" // FLAC loading functions + #define DR_WAV_IMPLEMENTATION + #include "external/dr_wav.h" // WAV loading functions #endif #if defined(SUPPORT_FILEFORMAT_MP3) @@ -216,6 +233,16 @@ typedef struct tagBITMAPINFOHEADER { #include "external/dr_mp3.h" // MP3 loading functions #endif +#if defined(SUPPORT_FILEFORMAT_FLAC) + #define DRFLAC_MALLOC RL_MALLOC + #define DRFLAC_REALLOC RL_REALLOC + #define DRFLAC_FREE RL_FREE + + #define DR_FLAC_IMPLEMENTATION + #define DR_FLAC_NO_WIN32_IO + #include "external/dr_flac.h" // FLAC loading functions +#endif + #if defined(_MSC_VER) #undef bool #endif @@ -223,11 +250,23 @@ typedef struct tagBITMAPINFOHEADER { //---------------------------------------------------------------------------------- // Defines and Macros //---------------------------------------------------------------------------------- -#define AUDIO_DEVICE_FORMAT ma_format_f32 -#define AUDIO_DEVICE_CHANNELS 2 -#define AUDIO_DEVICE_SAMPLE_RATE 44100 +#ifndef AUDIO_DEVICE_FORMAT + #define AUDIO_DEVICE_FORMAT ma_format_f32 // Device output format (float-32bit) +#endif +#ifndef AUDIO_DEVICE_CHANNELS + #define AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo +#endif + +#ifndef AUDIO_DEVICE_SAMPLE_RATE + #define AUDIO_DEVICE_SAMPLE_RATE 0 // Device output channels: stereo +#endif +#ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS + #define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Audio pool channels +#endif +#ifndef DEFAULT_AUDIO_BUFFER_SIZE + #define DEFAULT_AUDIO_BUFFER_SIZE 4096 // Default audio buffer size +#endif -#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 //---------------------------------------------------------------------------------- // Types and Structures Definition @@ -237,7 +276,8 @@ typedef struct tagBITMAPINFOHEADER { // NOTE: Depends on data structure provided by the library // in charge of reading the different file types typedef enum { - MUSIC_AUDIO_WAV = 0, + MUSIC_AUDIO_NONE = 0, + MUSIC_AUDIO_WAV, MUSIC_AUDIO_OGG, MUSIC_AUDIO_FLAC, MUSIC_AUDIO_MP3, @@ -255,7 +295,7 @@ typedef enum { LOG_ERROR, LOG_FATAL, LOG_NONE -} TraceLogType; +} TraceLogLevel; #endif // NOTE: Different logic is used when feeding data to the playback device @@ -304,8 +344,8 @@ typedef struct AudioData { int defaultSize; // Default audio buffer size for audio streams } Buffer; struct { - AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // Multichannel AudioBuffer pointers pool unsigned int poolCounter; // AudioBuffer pointers pool counter + AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // Multichannel AudioBuffer pointers pool unsigned int channels[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // AudioBuffer pool channels } MultiChannel; } AudioData; @@ -319,7 +359,7 @@ static AudioData AUDIO = { // Global AUDIO context // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a // standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough // In case of music-stalls, just increase this number - .Buffer.defaultSize = 4096 + .Buffer.defaultSize = 0 }; //---------------------------------------------------------------------------------- @@ -329,26 +369,25 @@ static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount); static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume); -static void InitAudioBufferPool(void); // Initialise the multichannel buffer pool -static void CloseAudioBufferPool(void); // Close the audio buffers pool - #if defined(SUPPORT_FILEFORMAT_WAV) -static Wave LoadWAV(const char *fileName); // Load WAV file +static Wave LoadWAV(const unsigned char *fileData, unsigned int fileSize); // Load WAV file static int SaveWAV(Wave wave, const char *fileName); // Save wave data as WAV file #endif #if defined(SUPPORT_FILEFORMAT_OGG) -static Wave LoadOGG(const char *fileName); // Load OGG file +static Wave LoadOGG(const unsigned char *fileData, unsigned int fileSize); // Load OGG file #endif #if defined(SUPPORT_FILEFORMAT_FLAC) -static Wave LoadFLAC(const char *fileName); // Load FLAC file +static Wave LoadFLAC(const unsigned char *fileData, unsigned int fileSize); // Load FLAC file #endif #if defined(SUPPORT_FILEFORMAT_MP3) -static Wave LoadMP3(const char *fileName); // Load MP3 file +static Wave LoadMP3(const unsigned char *fileData, unsigned int fileSize); // Load MP3 file #endif #if defined(RAUDIO_STANDALONE) -bool IsFileExtension(const char *fileName, const char *ext);// Check file extension -void TraceLog(int msgType, const char *text, ...); // Show trace log messages (LOG_INFO, LOG_WARNING, LOG_ERROR, LOG_DEBUG) +static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension +static unsigned char *LoadFileData(const char *fileName, unsigned int *bytesRead); // Load file data as byte array (read) +static bool SaveFileData(const char *fileName, void *data, unsigned int bytesToWrite); // Save data to file from byte array (write) +static bool SaveFileText(const char *fileName, char *text); // Save text data to file (write), string must be '\0' terminated #endif //---------------------------------------------------------------------------------- @@ -367,6 +406,7 @@ void SetAudioBufferVolume(AudioBuffer *buffer, float volume); void SetAudioBufferPitch(AudioBuffer *buffer, float pitch); void TrackAudioBuffer(AudioBuffer *buffer); void UntrackAudioBuffer(AudioBuffer *buffer); +int GetAudioStreamBufferSizeDefault(); //---------------------------------------------------------------------------------- // Module Functions Definition - Audio Device initialization and Closing @@ -383,7 +423,7 @@ void InitAudioDevice(void) ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context); if (result != MA_SUCCESS) { - TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize context"); + TRACELOG(LOG_WARNING, "AUDIO: Failed to initialize context"); return; } @@ -391,19 +431,19 @@ void InitAudioDevice(void) // NOTE: Using the default device. Format is floating point because it simplifies mixing. ma_device_config config = ma_device_config_init(ma_device_type_playback); config.playback.pDeviceID = NULL; // NULL for the default playback AUDIO.System.device. - config.playback.format = AUDIO_DEVICE_FORMAT; - config.playback.channels = AUDIO_DEVICE_CHANNELS; - config.capture.pDeviceID = NULL; // NULL for the default capture AUDIO.System.device. - config.capture.format = ma_format_s16; - config.capture.channels = 1; - config.sampleRate = AUDIO_DEVICE_SAMPLE_RATE; - config.dataCallback = OnSendAudioDataToDevice; - config.pUserData = NULL; + config.playback.format = AUDIO_DEVICE_FORMAT; + config.playback.channels = AUDIO_DEVICE_CHANNELS; + config.capture.pDeviceID = NULL; // NULL for the default capture AUDIO.System.device. + config.capture.format = ma_format_s16; + config.capture.channels = 1; + config.sampleRate = AUDIO_DEVICE_SAMPLE_RATE; + config.dataCallback = OnSendAudioDataToDevice; + config.pUserData = NULL; result = ma_device_init(&AUDIO.System.context, &config, &AUDIO.System.device); if (result != MA_SUCCESS) { - TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize playback device"); + TRACELOG(LOG_WARNING, "AUDIO: Failed to initialize playback device"); ma_context_uninit(&AUDIO.System.context); return; } @@ -413,7 +453,7 @@ void InitAudioDevice(void) result = ma_device_start(&AUDIO.System.device); if (result != MA_SUCCESS) { - TRACELOG(LOG_ERROR, "AUDIO: Failed to start playback device"); + TRACELOG(LOG_WARNING, "AUDIO: Failed to start playback device"); ma_device_uninit(&AUDIO.System.device); ma_context_uninit(&AUDIO.System.context); return; @@ -421,22 +461,28 @@ void InitAudioDevice(void) // Mixing happens on a seperate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may // want to look at something a bit smarter later on to keep everything real-time, if that's necessary. - if (ma_mutex_init(&AUDIO.System.context, &AUDIO.System.lock) != MA_SUCCESS) + if (ma_mutex_init(&AUDIO.System.lock) != MA_SUCCESS) { - TRACELOG(LOG_ERROR, "AUDIO: Failed to create mutex for mixing"); + TRACELOG(LOG_WARNING, "AUDIO: Failed to create mutex for mixing"); ma_device_uninit(&AUDIO.System.device); ma_context_uninit(&AUDIO.System.context); return; } - TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully"); - TRACELOG(LOG_INFO, " > Backend: miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend)); - TRACELOG(LOG_INFO, " > Format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat)); - TRACELOG(LOG_INFO, " > Channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels); - TRACELOG(LOG_INFO, " > Sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate); - TRACELOG(LOG_INFO, " > Periods size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods); + // Init dummy audio buffers pool for multichannel sound playing + for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) + { + // WARNING: An empty audioBuffer is created (data = 0) + // AudioBuffer data just points to loaded sound data + AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, 0, AUDIO_BUFFER_USAGE_STATIC); + } - InitAudioBufferPool(); + TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully"); + TRACELOG(LOG_INFO, " > Backend: miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend)); + TRACELOG(LOG_INFO, " > Format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat)); + TRACELOG(LOG_INFO, " > Channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels); + TRACELOG(LOG_INFO, " > Sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate); + TRACELOG(LOG_INFO, " > Periods size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods); AUDIO.System.isReady = true; } @@ -446,11 +492,25 @@ void CloseAudioDevice(void) { if (AUDIO.System.isReady) { + // Unload dummy audio buffers pool + // WARNING: They can be pointing to already unloaded data + for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) + { + //UnloadAudioBuffer(AUDIO.MultiChannel.pool[i]); + if (AUDIO.MultiChannel.pool[i] != NULL) + { + ma_data_converter_uninit(&AUDIO.MultiChannel.pool[i]->converter); + UntrackAudioBuffer(AUDIO.MultiChannel.pool[i]); + //RL_FREE(buffer->data); // Already unloaded by UnloadSound() + RL_FREE(AUDIO.MultiChannel.pool[i]); + } + } + ma_mutex_uninit(&AUDIO.System.lock); ma_device_uninit(&AUDIO.System.device); ma_context_uninit(&AUDIO.System.context); - CloseAudioBufferPool(); + AUDIO.System.isReady = false; TRACELOG(LOG_INFO, "AUDIO: Device closed successfully"); } @@ -480,21 +540,21 @@ AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam if (audioBuffer == NULL) { - TRACELOG(LOG_ERROR, "AUDIO: Failed to allocate memory for buffer"); + TRACELOG(LOG_WARNING, "AUDIO: Failed to allocate memory for buffer"); return NULL; } - audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1); + if (sizeInFrames > 0) audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1); // Audio data runs through a format converter - ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO_DEVICE_SAMPLE_RATE); + ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO.System.device.sampleRate); converterConfig.resampling.allowDynamicSampleRate = true; // Required for pitch shifting ma_result result = ma_data_converter_init(&converterConfig, &audioBuffer->converter); if (result != MA_SUCCESS) { - TRACELOG(LOG_ERROR, "AUDIO: Failed to create data conversion pipeline"); + TRACELOG(LOG_WARNING, "AUDIO: Failed to create data conversion pipeline"); RL_FREE(audioBuffer); return NULL; } @@ -593,18 +653,16 @@ void SetAudioBufferVolume(AudioBuffer *buffer, float volume) // Set pitch for an audio buffer void SetAudioBufferPitch(AudioBuffer *buffer, float pitch) { - if (buffer != NULL) + if ((buffer != NULL) && (pitch > 0.0f)) { - float pitchMul = pitch/buffer->pitch; - // Pitching is just an adjustment of the sample rate. // Note that this changes the duration of the sound: // - higher pitches will make the sound faster // - lower pitches make it slower - ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->converter.config.sampleRateOut/pitchMul); - buffer->pitch *= (float)buffer->converter.config.sampleRateOut/newOutputSampleRate; + ma_uint32 outputSampleRate = (ma_uint32)((float)buffer->converter.config.sampleRateOut/pitch); + ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, outputSampleRate); - ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, newOutputSampleRate); + buffer->pitch = pitch; } } @@ -651,20 +709,43 @@ Wave LoadWave(const char *fileName) { Wave wave = { 0 }; + // Loading file to memory + unsigned int fileSize = 0; + unsigned char *fileData = LoadFileData(fileName, &fileSize); + + if (fileData != NULL) + { + // Loading wave from memory data + wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, fileSize); + + RL_FREE(fileData); + } + + return wave; +} + +// Load wave from memory buffer, fileType refers to extension: i.e. ".wav" +Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int dataSize) +{ + Wave wave = { 0 }; + + char fileExtLower[16] = { 0 }; + strcpy(fileExtLower, TextToLower(fileType)); + if (false) { } #if defined(SUPPORT_FILEFORMAT_WAV) - else if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName); + else if (TextIsEqual(fileExtLower, ".wav")) wave = LoadWAV(fileData, dataSize); #endif #if defined(SUPPORT_FILEFORMAT_OGG) - else if (IsFileExtension(fileName, ".ogg")) wave = LoadOGG(fileName); + else if (TextIsEqual(fileExtLower, ".ogg")) wave = LoadOGG(fileData, dataSize); #endif #if defined(SUPPORT_FILEFORMAT_FLAC) - else if (IsFileExtension(fileName, ".flac")) wave = LoadFLAC(fileName); + else if (TextIsEqual(fileExtLower, ".flac")) wave = LoadFLAC(fileData, dataSize); #endif #if defined(SUPPORT_FILEFORMAT_MP3) - else if (IsFileExtension(fileName, ".mp3")) wave = LoadMP3(fileName); + else if (TextIsEqual(fileExtLower, ".mp3")) wave = LoadMP3(fileData, dataSize); #endif - else TRACELOG(LOG_WARNING, "FILEIO: [%s] File format not supported", fileName); + else TRACELOG(LOG_WARNING, "WAVE: File format not supported"); return wave; } @@ -702,17 +783,21 @@ Sound LoadSoundFromWave(Wave wave) ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32)); ma_uint32 frameCountIn = wave.sampleCount/wave.channels; - ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate); + ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate); if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion"); - AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC); - if (audioBuffer == NULL) TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer"); + AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, frameCount, AUDIO_BUFFER_USAGE_STATIC); + if (audioBuffer == NULL) + { + TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer"); + return sound; // early return to avoid dereferencing the audioBuffer null pointer + } - frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate); + frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate); if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion"); sound.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS; - sound.stream.sampleRate = AUDIO_DEVICE_SAMPLE_RATE; + sound.stream.sampleRate = AUDIO.System.device.sampleRate; sound.stream.sampleSize = 32; sound.stream.channels = AUDIO_DEVICE_CHANNELS; sound.stream.buffer = audioBuffer; @@ -750,7 +835,7 @@ void UpdateSound(Sound sound, const void *data, int samplesCount) } // Export wave data to file -void ExportWave(Wave wave, const char *fileName) +bool ExportWave(Wave wave, const char *fileName) { bool success = false; @@ -762,58 +847,68 @@ void ExportWave(Wave wave, const char *fileName) { // Export raw sample data (without header) // NOTE: It's up to the user to track wave parameters - SaveFileData(fileName, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8); - success = true; + success = SaveFileData(fileName, wave.data, wave.sampleCount*wave.sampleSize/8); } if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName); else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave data", fileName); + + return success; } // Export wave sample data to code (.h) -void ExportWaveAsCode(Wave wave, const char *fileName) +bool ExportWaveAsCode(Wave wave, const char *fileName) { - #define BYTES_TEXT_PER_LINE 20 + bool success = false; - char varFileName[256] = { 0 }; - int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; +#ifndef TEXT_BYTES_PER_LINE + #define TEXT_BYTES_PER_LINE 20 +#endif - FILE *txtFile = fopen(fileName, "wt"); + int waveDataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; - if (txtFile != NULL) - { - fprintf(txtFile, "\n//////////////////////////////////////////////////////////////////////////////////\n"); - fprintf(txtFile, "// //\n"); - fprintf(txtFile, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes //\n"); - fprintf(txtFile, "// //\n"); - fprintf(txtFile, "// more info and bugs-report: github.com/raysan5/raylib //\n"); - fprintf(txtFile, "// feedback and support: ray[at]raylib.com //\n"); - fprintf(txtFile, "// //\n"); - fprintf(txtFile, "// Copyright (c) 2018 Ramon Santamaria (@raysan5) //\n"); - fprintf(txtFile, "// //\n"); - fprintf(txtFile, "//////////////////////////////////////////////////////////////////////////////////\n\n"); + // NOTE: Text data buffer size is estimated considering wave data size in bytes + // and requiring 6 char bytes for every byte: "0x00, " + char *txtData = (char *)RL_CALLOC(6*waveDataSize + 2000, sizeof(char)); + int bytesCount = 0; + bytesCount += sprintf(txtData + bytesCount, "\n//////////////////////////////////////////////////////////////////////////////////\n"); + bytesCount += sprintf(txtData + bytesCount, "// //\n"); + bytesCount += sprintf(txtData + bytesCount, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes //\n"); + bytesCount += sprintf(txtData + bytesCount, "// //\n"); + bytesCount += sprintf(txtData + bytesCount, "// more info and bugs-report: github.com/raysan5/raylib //\n"); + bytesCount += sprintf(txtData + bytesCount, "// feedback and support: ray[at]raylib.com //\n"); + bytesCount += sprintf(txtData + bytesCount, "// //\n"); + bytesCount += sprintf(txtData + bytesCount, "// Copyright (c) 2018 Ramon Santamaria (@raysan5) //\n"); + bytesCount += sprintf(txtData + bytesCount, "// //\n"); + bytesCount += sprintf(txtData + bytesCount, "//////////////////////////////////////////////////////////////////////////////////\n\n"); + + char varFileName[256] = { 0 }; #if !defined(RAUDIO_STANDALONE) - // Get file name from path and convert variable name to uppercase - strcpy(varFileName, GetFileNameWithoutExt(fileName)); - for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; } + // Get file name from path and convert variable name to uppercase + strcpy(varFileName, GetFileNameWithoutExt(fileName)); + for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; } #else - strcpy(varFileName, fileName); + strcpy(varFileName, fileName); #endif - fprintf(txtFile, "// Wave data information\n"); - fprintf(txtFile, "#define %s_SAMPLE_COUNT %u\n", varFileName, wave.sampleCount); - fprintf(txtFile, "#define %s_SAMPLE_RATE %u\n", varFileName, wave.sampleRate); - fprintf(txtFile, "#define %s_SAMPLE_SIZE %u\n", varFileName, wave.sampleSize); - fprintf(txtFile, "#define %s_CHANNELS %u\n\n", varFileName, wave.channels); + bytesCount += sprintf(txtData + bytesCount, "// Wave data information\n"); + bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_COUNT %u\n", varFileName, wave.sampleCount); + bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_RATE %u\n", varFileName, wave.sampleRate); + bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_SIZE %u\n", varFileName, wave.sampleSize); + bytesCount += sprintf(txtData + bytesCount, "#define %s_CHANNELS %u\n\n", varFileName, wave.channels); - // Write byte data as hexadecimal text - fprintf(txtFile, "static unsigned char %s_DATA[%i] = { ", varFileName, dataSize); - for (int i = 0; i < dataSize - 1; i++) fprintf(txtFile, ((i%BYTES_TEXT_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]); - fprintf(txtFile, "0x%x };\n", ((unsigned char *)wave.data)[dataSize - 1]); + // Write byte data as hexadecimal text + bytesCount += sprintf(txtData + bytesCount, "static unsigned char %s_DATA[%i] = { ", varFileName, waveDataSize); + for (int i = 0; i < waveDataSize - 1; i++) bytesCount += sprintf(txtData + bytesCount, ((i%TEXT_BYTES_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]); + bytesCount += sprintf(txtData + bytesCount, "0x%x };\n", ((unsigned char *)wave.data)[waveDataSize - 1]); - fclose(txtFile); - } + // NOTE: Text data length exported is determined by '\0' (NULL) character + success = SaveFileText(fileName, txtData); + + RL_FREE(txtData); + + return success; } // Play a sound @@ -940,10 +1035,10 @@ void SetSoundPitch(Sound sound, float pitch) // Convert wave data to desired format void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) { - ma_format formatIn = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32)); - ma_format formatOut = (( sampleSize == 8)? ma_format_u8 : (( sampleSize == 16)? ma_format_s16 : ma_format_f32)); + ma_format formatIn = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32)); + ma_format formatOut = ((sampleSize == 8)? ma_format_u8 : ((sampleSize == 16)? ma_format_s16 : ma_format_f32)); - ma_uint32 frameCountIn = wave->sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so. + ma_uint32 frameCountIn = wave->sampleCount/wave->channels; ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate); if (frameCount == 0) @@ -961,7 +1056,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) return; } - wave->sampleCount = frameCount; + wave->sampleCount = frameCount*channels; wave->sampleSize = sampleSize; wave->sampleRate = sampleRate; wave->channels = channels; @@ -974,12 +1069,12 @@ Wave WaveCopy(Wave wave) { Wave newWave = { 0 }; - newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8*wave.channels); + newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8); if (newWave.data != NULL) { // NOTE: Size must be provided in bytes - memcpy(newWave.data, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8); + memcpy(newWave.data, wave.data, wave.sampleCount*wave.sampleSize/8); newWave.sampleCount = wave.sampleCount; newWave.sampleRate = wave.sampleRate; @@ -999,9 +1094,9 @@ void WaveCrop(Wave *wave, int initSample, int finalSample) { int sampleCount = finalSample - initSample; - void *data = RL_MALLOC(sampleCount*wave->sampleSize/8*wave->channels); + void *data = RL_MALLOC(sampleCount*wave->sampleSize/8); - memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8); + memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->sampleSize/8); RL_FREE(wave->data); wave->data = data; @@ -1009,25 +1104,31 @@ void WaveCrop(Wave *wave, int initSample, int finalSample) else TRACELOG(LOG_WARNING, "WAVE: Crop range out of bounds"); } -// Get samples data from wave as a floats array -// NOTE: Returned sample values are normalized to range [-1..1] -float *GetWaveData(Wave wave) +// Load samples data from wave as a floats array +// NOTE 1: Returned sample values are normalized to range [-1..1] +// NOTE 2: Sample data allocated should be freed with UnloadWaveSamples() +float *LoadWaveSamples(Wave wave) { - float *samples = (float *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(float)); + float *samples = (float *)RL_MALLOC(wave.sampleCount*sizeof(float)); + + // NOTE: sampleCount is the total number of interlaced samples (including channels) for (unsigned int i = 0; i < wave.sampleCount; i++) { - for (unsigned int j = 0; j < wave.channels; j++) - { - if (wave.sampleSize == 8) samples[wave.channels*i + j] = (float)(((unsigned char *)wave.data)[wave.channels*i + j] - 127)/256.0f; - else if (wave.sampleSize == 16) samples[wave.channels*i + j] = (float)((short *)wave.data)[wave.channels*i + j]/32767.0f; - else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j]; - } + if (wave.sampleSize == 8) samples[i] = (float)(((unsigned char *)wave.data)[i] - 127)/256.0f; + else if (wave.sampleSize == 16) samples[i] = (float)(((short *)wave.data)[i])/32767.0f; + else if (wave.sampleSize == 32) samples[i] = ((float *)wave.data)[i]; } return samples; } +// Unload samples data loaded with LoadWaveSamples() +void UnloadWaveSamples(float *samples) +{ + RL_FREE(samples); +} + //---------------------------------------------------------------------------------- // Module Functions Definition - Music loading and stream playing (.OGG) //---------------------------------------------------------------------------------- @@ -1039,21 +1140,44 @@ Music LoadMusicStream(const char *fileName) bool musicLoaded = false; if (false) { } +#if defined(SUPPORT_FILEFORMAT_WAV) + else if (IsFileExtension(fileName, ".wav")) + { + drwav *ctxWav = RL_CALLOC(1, sizeof(drwav)); + bool success = drwav_init_file(ctxWav, fileName, NULL); + + music.ctxType = MUSIC_AUDIO_WAV; + music.ctxData = ctxWav; + + if (success) + { + int sampleSize = ctxWav->bitsPerSample; + if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream() + + music.stream = InitAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels); + music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels; + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + } +#endif #if defined(SUPPORT_FILEFORMAT_OGG) else if (IsFileExtension(fileName, ".ogg")) { // Open ogg audio stream + music.ctxType = MUSIC_AUDIO_OGG; music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL); if (music.ctxData != NULL) { - music.ctxType = MUSIC_AUDIO_OGG; stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData); // Get Ogg file info // OGG bit rate defaults to 16 bit, it's enough for compressed format music.stream = InitAudioStream(info.sample_rate, 16, info.channels); + + // WARNING: It seems this function returns length in frames, not samples, so we multiply by channels music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels; - music.loopCount = 0; // Infinite loop by default + music.looping = true; // Looping enabled by default musicLoaded = true; } } @@ -1061,16 +1185,16 @@ Music LoadMusicStream(const char *fileName) #if defined(SUPPORT_FILEFORMAT_FLAC) else if (IsFileExtension(fileName, ".flac")) { - music.ctxData = drflac_open_file(fileName); + music.ctxType = MUSIC_AUDIO_FLAC; + music.ctxData = drflac_open_file(fileName, NULL); if (music.ctxData != NULL) { - music.ctxType = MUSIC_AUDIO_FLAC; drflac *ctxFlac = (drflac *)music.ctxData; music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels); - music.sampleCount = (unsigned int)ctxFlac->totalSampleCount; - music.loopCount = 0; // Infinite loop by default + music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount*ctxFlac->channels; + music.looping = true; // Looping enabled by default musicLoaded = true; } } @@ -1078,18 +1202,17 @@ Music LoadMusicStream(const char *fileName) #if defined(SUPPORT_FILEFORMAT_MP3) else if (IsFileExtension(fileName, ".mp3")) { - drmp3 *ctxMp3 = RL_MALLOC(sizeof(drmp3)); - music.ctxData = ctxMp3; - + drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3)); int result = drmp3_init_file(ctxMp3, fileName, NULL); + music.ctxType = MUSIC_AUDIO_MP3; + music.ctxData = ctxMp3; + if (result > 0) { - music.ctxType = MUSIC_AUDIO_MP3; - music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels); music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels; - music.loopCount = 0; // Infinite loop by default + music.looping = true; // Looping enabled by default musicLoaded = true; } } @@ -1098,60 +1221,261 @@ Music LoadMusicStream(const char *fileName) else if (IsFileExtension(fileName, ".xm")) { jar_xm_context_t *ctxXm = NULL; + int result = jar_xm_create_context_from_file(&ctxXm, AUDIO.System.device.sampleRate, fileName); - int result = jar_xm_create_context_from_file(&ctxXm, 48000, fileName); + music.ctxType = MUSIC_MODULE_XM; + music.ctxData = ctxXm; if (result == 0) // XM AUDIO.System.context created successfully { - music.ctxType = MUSIC_MODULE_XM; jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops + unsigned int bits = 32; + if (AUDIO_DEVICE_FORMAT == ma_format_s16) + bits = 16; + else if (AUDIO_DEVICE_FORMAT == ma_format_u8) + bits = 8; + // NOTE: Only stereo is supported for XM - music.stream = InitAudioStream(48000, 16, 2); - music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; - music.loopCount = 0; // Infinite loop by default + music.stream = InitAudioStream(AUDIO.System.device.sampleRate, bits, AUDIO_DEVICE_CHANNELS); + music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; // 2 channels + music.looping = true; // Looping enabled by default jar_xm_reset(ctxXm); // make sure we start at the beginning of the song musicLoaded = true; + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_MOD) + else if (IsFileExtension(fileName, ".mod")) + { + jar_mod_context_t *ctxMod = RL_CALLOC(1, sizeof(jar_mod_context_t)); + jar_mod_init(ctxMod); + int result = jar_mod_load_file(ctxMod, fileName); + + music.ctxType = MUSIC_MODULE_MOD; + music.ctxData = ctxMod; + + if (result > 0) + { + // NOTE: Only stereo is supported for MOD + music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, AUDIO_DEVICE_CHANNELS); + music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; // 2 channels + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + } +#endif + else TRACELOG(LOG_WARNING, "STREAM: [%s] Fileformat not supported", fileName); + + if (!musicLoaded) + { + if (false) { } + #if defined(SUPPORT_FILEFORMAT_WAV) + else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData); + #endif + #if defined(SUPPORT_FILEFORMAT_OGG) + else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); + #endif + #if defined(SUPPORT_FILEFORMAT_FLAC) + else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL); + #endif + #if defined(SUPPORT_FILEFORMAT_MP3) + else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); } + #endif + #if defined(SUPPORT_FILEFORMAT_XM) + else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData); + #endif + #if defined(SUPPORT_FILEFORMAT_MOD) + else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); } + #endif + + music.ctxData = NULL; + TRACELOG(LOG_WARNING, "FILEIO: [%s] Music file could not be opened", fileName); + } + else + { + // Show some music stream info + TRACELOG(LOG_INFO, "FILEIO: [%s] Music file successfully loaded:", fileName); + TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount); + TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate); + TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize); + TRACELOG(LOG_INFO, " > Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi"); + } + + return music; +} + +// extension including period ".mod" +Music LoadMusicStreamFromMemory(const char *fileType, unsigned char* data, int dataSize) +{ + Music music = { 0 }; + bool musicLoaded = false; + + char fileExtLower[16] = { 0 }; + strcpy(fileExtLower, TextToLower(fileType)); + + if (false) { } +#if defined(SUPPORT_FILEFORMAT_WAV) + else if (TextIsEqual(fileExtLower, ".wav")) + { + drwav *ctxWav = RL_CALLOC(1, sizeof(drwav)); + + bool success = drwav_init_memory(ctxWav, (const void*)data, dataSize, NULL); + + music.ctxType = MUSIC_AUDIO_WAV; + music.ctxData = ctxWav; + + if (success) + { + int sampleSize = ctxWav->bitsPerSample; + if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream() + + music.stream = InitAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels); + music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels; + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_FLAC) + else if (TextIsEqual(fileExtLower, ".flac")) + { + music.ctxType = MUSIC_AUDIO_FLAC; + music.ctxData = drflac_open_memory((const void*)data, dataSize, NULL); + + if (music.ctxData != NULL) + { + drflac *ctxFlac = (drflac *)music.ctxData; + + music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels); + music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount*ctxFlac->channels; + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_MP3) + else if (TextIsEqual(fileExtLower, ".mp3")) + { + drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3)); + int success = drmp3_init_memory(ctxMp3, (const void*)data, dataSize, NULL); + + music.ctxType = MUSIC_AUDIO_MP3; + music.ctxData = ctxMp3; + + if (success) + { + music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels); + music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels; + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_OGG) + else if (TextIsEqual(fileExtLower, ".ogg")) + { + // Open ogg audio stream + music.ctxType = MUSIC_AUDIO_OGG; + //music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL); + music.ctxData = stb_vorbis_open_memory((const unsigned char*)data, dataSize, NULL, NULL); + + if (music.ctxData != NULL) + { + stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData); // Get Ogg file info + + // OGG bit rate defaults to 16 bit, it's enough for compressed format + music.stream = InitAudioStream(info.sample_rate, 16, info.channels); + + // WARNING: It seems this function returns length in frames, not samples, so we multiply by channels + music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels; + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_XM) + else if (TextIsEqual(fileExtLower, ".xm")) + { + jar_xm_context_t *ctxXm = NULL; + int result = jar_xm_create_context_safe(&ctxXm, (const char*)data, dataSize, AUDIO.System.device.sampleRate); + if (result == 0) // XM AUDIO.System.context created successfully + { + music.ctxType = MUSIC_MODULE_XM; + jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops + + unsigned int bits = 32; + if (AUDIO_DEVICE_FORMAT == ma_format_s16) + bits = 16; + else if (AUDIO_DEVICE_FORMAT == ma_format_u8) + bits = 8; + + // NOTE: Only stereo is supported for XM + music.stream = InitAudioStream(AUDIO.System.device.sampleRate, bits, 2); + music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; // 2 channels + music.looping = true; // Looping enabled by default + jar_xm_reset(ctxXm); // make sure we start at the beginning of the song music.ctxData = ctxXm; + musicLoaded = true; } } #endif #if defined(SUPPORT_FILEFORMAT_MOD) - else if (IsFileExtension(fileName, ".mod")) + else if (TextIsEqual(fileExtLower, ".mod")) { jar_mod_context_t *ctxMod = RL_MALLOC(sizeof(jar_mod_context_t)); + int result = 0; jar_mod_init(ctxMod); - int result = jar_mod_load_file(ctxMod, fileName); + + // copy data to allocated memory for default UnloadMusicStream + unsigned char *newData = RL_MALLOC(dataSize); + int it = dataSize/sizeof(unsigned char); + for (int i = 0; i < it; i++){ + newData[i] = data[i]; + } + + // Memory loaded version for jar_mod_load_file() + if (dataSize && dataSize < 32*1024*1024) + { + ctxMod->modfilesize = dataSize; + ctxMod->modfile = newData; + if (jar_mod_load(ctxMod, (void *)ctxMod->modfile, dataSize)) result = dataSize; + } if (result > 0) { music.ctxType = MUSIC_MODULE_MOD; // NOTE: Only stereo is supported for MOD - music.stream = InitAudioStream(48000, 16, 2); - music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; - music.loopCount = 0; // Infinite loop by default + music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, 2); + music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; // 2 channels + music.looping = true; // Looping enabled by default musicLoaded = true; music.ctxData = ctxMod; + musicLoaded = true; } } #endif + else TRACELOG(LOG_WARNING, "STREAM: [%s] Fileformat not supported", fileType); if (!musicLoaded) { if (false) { } - #if defined(SUPPORT_FILEFORMAT_OGG) - else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); + #if defined(SUPPORT_FILEFORMAT_WAV) + else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData); #endif #if defined(SUPPORT_FILEFORMAT_FLAC) - else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData); + else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL); #endif #if defined(SUPPORT_FILEFORMAT_MP3) else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); } #endif + #if defined(SUPPORT_FILEFORMAT_OGG) + else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); + #endif #if defined(SUPPORT_FILEFORMAT_XM) else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData); #endif @@ -1159,12 +1483,13 @@ Music LoadMusicStream(const char *fileName) else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); } #endif - TRACELOG(LOG_WARNING, "FILEIO: [%s] Music file could not be opened", fileName); + music.ctxData = NULL; + TRACELOG(LOG_WARNING, "FILEIO: [%s] Music memory could not be opened", fileType); } else { // Show some music stream info - TRACELOG(LOG_INFO, "FILEIO: [%s] Music file successfully loaded:", fileName); + TRACELOG(LOG_INFO, "FILEIO: [%s] Music memory successfully loaded:", fileType); TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount); TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate); TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize); @@ -1179,22 +1504,28 @@ void UnloadMusicStream(Music music) { CloseAudioStream(music.stream); - if (false) { } + if (music.ctxData != NULL) + { + if (false) { } +#if defined(SUPPORT_FILEFORMAT_WAV) + else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData); +#endif #if defined(SUPPORT_FILEFORMAT_OGG) - else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); + else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); #endif #if defined(SUPPORT_FILEFORMAT_FLAC) - else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData); + else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL); #endif #if defined(SUPPORT_FILEFORMAT_MP3) else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); } #endif #if defined(SUPPORT_FILEFORMAT_XM) - else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData); + else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData); #endif #if defined(SUPPORT_FILEFORMAT_MOD) - else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); } + else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); } #endif + } } // Start music playing (open stream) @@ -1231,6 +1562,9 @@ void StopMusicStream(Music music) switch (music.ctxType) { +#if defined(SUPPORT_FILEFORMAT_WAV) + case MUSIC_AUDIO_WAV: drwav_seek_to_pcm_frame((drwav *)music.ctxData, 0); break; +#endif #if defined(SUPPORT_FILEFORMAT_OGG) case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break; #endif @@ -1253,7 +1587,11 @@ void StopMusicStream(Music music) // Update (re-fill) music buffers if data already processed void UpdateMusicStream(Music music) { - if (music.stream.buffer == NULL) return; + if (music.stream.buffer == NULL) + return; + + if (music.ctxType == MUSIC_MODULE_XM) + jar_xm_set_max_loop_count(music.ctxData, music.looping ? 0 : 1); bool streamEnding = false; @@ -1268,6 +1606,8 @@ void UpdateMusicStream(Music music) //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels; int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels); + if (music.ctxType == MUSIC_MODULE_XM && music.looping) sampleLeft = subBufferSizeInFrames*4; + while (IsAudioStreamProcessed(music.stream)) { if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels; @@ -1275,6 +1615,15 @@ void UpdateMusicStream(Music music) switch (music.ctxType) { + #if defined(SUPPORT_FILEFORMAT_WAV) + case MUSIC_AUDIO_WAV: + { + // NOTE: Returns the number of samples to process (not required) + if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, samplesCount/music.stream.channels, (short *)pcm); + else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm); + + } break; + #endif #if defined(SUPPORT_FILEFORMAT_OGG) case MUSIC_AUDIO_OGG: { @@ -1302,8 +1651,24 @@ void UpdateMusicStream(Music music) #if defined(SUPPORT_FILEFORMAT_XM) case MUSIC_MODULE_XM: { - // NOTE: Internally this function considers 2 channels generation, so samplesCount/2 - jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, samplesCount/2); + switch (AUDIO_DEVICE_FORMAT) + { + case ma_format_f32: + // NOTE: Internally this function considers 2 channels generation, so samplesCount/2 + jar_xm_generate_samples((jar_xm_context_t*)music.ctxData, (float*)pcm, samplesCount / 2); + break; + + case ma_format_s16: + // NOTE: Internally this function considers 2 channels generation, so samplesCount/2 + jar_xm_generate_samples_16bit((jar_xm_context_t*)music.ctxData, (short*)pcm, samplesCount / 2); + break; + + case ma_format_u8: + // NOTE: Internally this function considers 2 channels generation, so samplesCount/2 + jar_xm_generate_samples_8bit((jar_xm_context_t*)music.ctxData, (char*)pcm, samplesCount / 2); + break; + } + } break; #endif #if defined(SUPPORT_FILEFORMAT_MOD) @@ -1318,7 +1683,7 @@ void UpdateMusicStream(Music music) UpdateAudioStream(music.stream, pcm, samplesCount); - if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD)) + if ((music.ctxType == MUSIC_MODULE_XM) || music.ctxType == MUSIC_MODULE_MOD) { if (samplesCount > 1) sampleLeft -= samplesCount/2; else sampleLeft -= samplesCount; @@ -1338,15 +1703,8 @@ void UpdateMusicStream(Music music) // Reset audio stream for looping if (streamEnding) { - StopMusicStream(music); // Stop music (and reset) - - // Decrease loopCount to stop when required - if (music.loopCount > 1) - { - music.loopCount--; // Decrease loop count - PlayMusicStream(music); // Play again - } - else if (music.loopCount == 0) PlayMusicStream(music); + StopMusicStream(music); // Stop music (and reset) + if (music.looping) PlayMusicStream(music); // Play again } else { @@ -1371,14 +1729,7 @@ void SetMusicVolume(Music music, float volume) // Set pitch for music void SetMusicPitch(Music music, float pitch) { - SetAudioStreamPitch(music.stream, pitch); -} - -// Set music loop count (loop repeats) -// NOTE: If set to 0, means infinite loop -void SetMusicLoopCount(Music music, int count) -{ - music.loopCount = count; + SetAudioBufferPitch(music.stream.buffer, pitch); } // Get music time length (in seconds) @@ -1394,13 +1745,22 @@ float GetMusicTimeLength(Music music) // Get current music time played (in seconds) float GetMusicTimePlayed(Music music) { - float secondsPlayed = 0.0f; +#if defined(SUPPORT_FILEFORMAT_XM) + if (music.ctxType == MUSIC_MODULE_XM) + { + uint64_t samples = 0; + jar_xm_get_position(music.ctxData, NULL, NULL, NULL, &samples); + samples = samples % (music.sampleCount); + return (float)(samples)/(music.stream.sampleRate*music.stream.channels); + } +#endif + float secondsPlayed = 0.0f; if (music.stream.buffer != NULL) { //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels; unsigned int samplesPlayed = music.stream.buffer->totalFramesProcessed*music.stream.channels; - secondsPlayed = (float)samplesPlayed / (music.stream.sampleRate*music.stream.channels); + secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels); } return secondsPlayed; @@ -1419,7 +1779,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un // The size of a streaming buffer must be at least double the size of a period unsigned int periodSize = AUDIO.System.device.playback.internalPeriodSizeInFrames; - unsigned int subBufferSize = AUDIO.Buffer.defaultSize; // Default buffer size (audio stream) + unsigned int subBufferSize = GetAudioStreamBufferSizeDefault(); if (subBufferSize < periodSize) subBufferSize = periodSize; @@ -1554,6 +1914,15 @@ void SetAudioStreamBufferSizeDefault(int size) AUDIO.Buffer.defaultSize = size; } +int GetAudioStreamBufferSizeDefault() +{ + // if the buffer is not set, compute one that would give us a buffer good enough for a decent frame rate + if (AUDIO.Buffer.defaultSize == 0) + AUDIO.Buffer.defaultSize = AUDIO.System.device.sampleRate/30; + + return AUDIO.Buffer.defaultSize; +} + //---------------------------------------------------------------------------------- // Module specific Functions Definition //---------------------------------------------------------------------------------- @@ -1564,7 +1933,7 @@ static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, (void)pContext; (void)pDevice; - TRACELOG(LOG_ERROR, "miniaudio: %s", message); // All log messages from miniaudio are errors + TRACELOG(LOG_WARNING, "miniaudio: %s", message); // All log messages from miniaudio are errors } // Reads audio data from an AudioBuffer object in internal format. @@ -1661,7 +2030,7 @@ static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, f // detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output // frames. This can be achieved with ma_data_converter_get_required_input_frame_count(). ma_uint8 inputBuffer[4096]; - ma_uint32 inputBufferFrameCap = sizeof(inputBuffer) / ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn); + ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn); ma_uint32 totalOutputFramesProcessed = 0; while (totalOutputFramesProcessed < frameCount) @@ -1674,7 +2043,7 @@ static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, f inputFramesToProcessThisIteration = inputBufferFrameCap; } - float *runningFramesOut = framesOut + (totalOutputFramesProcessed * audioBuffer->converter.config.channelsOut); + float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.config.channelsOut); /* At this point we can convert the data to our mixing format. */ ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration); /* Safe cast. */ @@ -1798,322 +2167,146 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr } } -// Initialise the multichannel buffer pool -static void InitAudioBufferPool(void) -{ - // Dummy buffers - for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) - { - AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC); - } - - // TODO: Verification required for log - TRACELOG(LOG_INFO, "AUDIO: Multichannel pool size: %i", MAX_AUDIO_BUFFER_POOL_CHANNELS); -} - -// Close the audio buffers pool -static void CloseAudioBufferPool(void) -{ - for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) - { - RL_FREE(AUDIO.MultiChannel.pool[i]->data); - RL_FREE(AUDIO.MultiChannel.pool[i]); - } -} - #if defined(SUPPORT_FILEFORMAT_WAV) -// Load WAV file into Wave structure -static Wave LoadWAV(const char *fileName) -{ - // Basic WAV headers structs - typedef struct { - char chunkID[4]; - int chunkSize; - char format[4]; - } WAVRiffHeader; - - typedef struct { - char subChunkID[4]; - int subChunkSize; - short audioFormat; - short numChannels; - int sampleRate; - int byteRate; - short blockAlign; - short bitsPerSample; - } WAVFormat; - - typedef struct { - char subChunkID[4]; - int subChunkSize; - } WAVData; - - WAVRiffHeader wavRiffHeader = { 0 }; - WAVFormat wavFormat = { 0 }; - WAVData wavData = { 0 }; - +// Load WAV file data into Wave structure +// NOTE: Using dr_wav library +static Wave LoadWAV(const unsigned char *fileData, unsigned int fileSize) +{ Wave wave = { 0 }; - FILE *wavFile = NULL; + drwav wav = { 0 }; - wavFile = fopen(fileName, "rb"); + bool success = drwav_init_memory(&wav, fileData, fileSize, NULL); - if (wavFile == NULL) + if (success) { - TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open WAV file", fileName); - wave.data = NULL; + wave.sampleCount = (unsigned int)wav.totalPCMFrameCount*wav.channels; + wave.sampleRate = wav.sampleRate; + wave.sampleSize = 16; // NOTE: We are forcing conversion to 16bit + wave.channels = wav.channels; + wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short)); + drwav_read_pcm_frames_s16(&wav, wav.totalPCMFrameCount, wave.data); } - else - { - // Read in the first chunk into the struct - fread(&wavRiffHeader, sizeof(WAVRiffHeader), 1, wavFile); - - // Check for RIFF and WAVE tags - if ((wavRiffHeader.chunkID[0] != 'R') || - (wavRiffHeader.chunkID[1] != 'I') || - (wavRiffHeader.chunkID[2] != 'F') || - (wavRiffHeader.chunkID[3] != 'F') || - (wavRiffHeader.format[0] != 'W') || - (wavRiffHeader.format[1] != 'A') || - (wavRiffHeader.format[2] != 'V') || - (wavRiffHeader.format[3] != 'E')) - { - TRACELOG(LOG_WARNING, "WAVE: [%s] RIFF or WAVE header are not valid", fileName); - } - else - { - // Read in the 2nd chunk for the wave info - fread(&wavFormat, sizeof(WAVFormat), 1, wavFile); - - // Check for fmt tag - if ((wavFormat.subChunkID[0] != 'f') || (wavFormat.subChunkID[1] != 'm') || - (wavFormat.subChunkID[2] != 't') || (wavFormat.subChunkID[3] != ' ')) - { - TRACELOG(LOG_WARNING, "WAVE: [%s] Wave format header is not valid", fileName); - } - else - { - // Check for extra parameters; - if (wavFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR); - - // Read in the the last byte of data before the sound file - fread(&wavData, sizeof(WAVData), 1, wavFile); - - // Check for data tag - if ((wavData.subChunkID[0] != 'd') || (wavData.subChunkID[1] != 'a') || - (wavData.subChunkID[2] != 't') || (wavData.subChunkID[3] != 'a')) - { - TRACELOG(LOG_WARNING, "WAVE: [%s] Data header is not valid", fileName); - } - else - { - // Allocate memory for data - wave.data = RL_MALLOC(wavData.subChunkSize); - - // Read in the sound data into the soundData variable - fread(wave.data, wavData.subChunkSize, 1, wavFile); + else TRACELOG(LOG_WARNING, "WAVE: Failed to load WAV data"); - // Store wave parameters - wave.sampleRate = wavFormat.sampleRate; - wave.sampleSize = wavFormat.bitsPerSample; - wave.channels = wavFormat.numChannels; - - // NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes - if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32)) - { - TRACELOG(LOG_WARNING, "WAVE: [%s] Sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize); - WaveFormat(&wave, wave.sampleRate, 16, wave.channels); - } - - // NOTE: Only support up to 2 channels (mono, stereo) - if (wave.channels > 2) - { - WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2); - TRACELOG(LOG_WARNING, "WAVE: [%s] Channels number (%i) not supported, converted to 2 channels", fileName, wave.channels); - } - - // NOTE: subChunkSize comes in bytes, we need to translate it to number of samples - wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels; - - TRACELOG(LOG_INFO, "WAVE: [%s] File loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); - } - } - } - - fclose(wavFile); - } + drwav_uninit(&wav); return wave; } // Save wave data as WAV file +// NOTE: Using dr_wav library static int SaveWAV(Wave wave, const char *fileName) { - int success = 0; - int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; - - // Basic WAV headers structs - typedef struct { - char chunkID[4]; - int chunkSize; - char format[4]; - } RiffHeader; - - typedef struct { - char subChunkID[4]; - int subChunkSize; - short audioFormat; - short numChannels; - int sampleRate; - int byteRate; - short blockAlign; - short bitsPerSample; - } WaveFormat; - - typedef struct { - char subChunkID[4]; - int subChunkSize; - } WaveData; - - FILE *wavFile = fopen(fileName, "wb"); - - if (wavFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open audio file", fileName); - else - { - RiffHeader riffHeader; - WaveFormat waveFormat; - WaveData waveData; - - // Fill structs with data - riffHeader.chunkID[0] = 'R'; - riffHeader.chunkID[1] = 'I'; - riffHeader.chunkID[2] = 'F'; - riffHeader.chunkID[3] = 'F'; - riffHeader.chunkSize = 44 - 4 + wave.sampleCount*wave.sampleSize/8; - riffHeader.format[0] = 'W'; - riffHeader.format[1] = 'A'; - riffHeader.format[2] = 'V'; - riffHeader.format[3] = 'E'; - - waveFormat.subChunkID[0] = 'f'; - waveFormat.subChunkID[1] = 'm'; - waveFormat.subChunkID[2] = 't'; - waveFormat.subChunkID[3] = ' '; - waveFormat.subChunkSize = 16; - waveFormat.audioFormat = 1; - waveFormat.numChannels = wave.channels; - waveFormat.sampleRate = wave.sampleRate; - waveFormat.byteRate = wave.sampleRate*wave.sampleSize/8; - waveFormat.blockAlign = wave.sampleSize/8; - waveFormat.bitsPerSample = wave.sampleSize; - - waveData.subChunkID[0] = 'd'; - waveData.subChunkID[1] = 'a'; - waveData.subChunkID[2] = 't'; - waveData.subChunkID[3] = 'a'; - waveData.subChunkSize = dataSize; - - fwrite(&riffHeader, sizeof(RiffHeader), 1, wavFile); - fwrite(&waveFormat, sizeof(WaveFormat), 1, wavFile); - fwrite(&waveData, sizeof(WaveData), 1, wavFile); - - success = fwrite(wave.data, dataSize, 1, wavFile); - - fclose(wavFile); - } - - // If all data has been written correctly to file, success = 1 + int success = false; + + drwav wav = { 0 }; + drwav_data_format format = { 0 }; + format.container = drwav_container_riff; + format.format = DR_WAVE_FORMAT_PCM; + format.channels = wave.channels; + format.sampleRate = wave.sampleRate; + format.bitsPerSample = wave.sampleSize; + + void *fileData = NULL; + size_t fileDataSize = 0; + success = drwav_init_memory_write(&wav, &fileData, &fileDataSize, &format, NULL); + if (success) success = (int)drwav_write_pcm_frames(&wav, wave.sampleCount/wave.channels, wave.data); + drwav_result result = drwav_uninit(&wav); + + if (result == DRWAV_SUCCESS) success = SaveFileData(fileName, (unsigned char *)fileData, (unsigned int)fileDataSize); + + drwav_free(fileData, NULL); + return success; } #endif #if defined(SUPPORT_FILEFORMAT_OGG) -// Load OGG file into Wave structure +// Load OGG file data into Wave structure // NOTE: Using stb_vorbis library -static Wave LoadOGG(const char *fileName) +static Wave LoadOGG(const unsigned char *fileData, unsigned int fileSize) { Wave wave = { 0 }; - stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL); + stb_vorbis *oggData = stb_vorbis_open_memory((unsigned char *)fileData, fileSize, NULL, NULL); - if (oggFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open OGG file", fileName); - else + if (oggData != NULL) { - stb_vorbis_info info = stb_vorbis_get_info(oggFile); + stb_vorbis_info info = stb_vorbis_get_info(oggData); wave.sampleRate = info.sample_rate; wave.sampleSize = 16; // 16 bit per sample (short) wave.channels = info.channels; - wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggFile)*info.channels; // Independent by channel + wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData)*info.channels; // Independent by channel - float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile); - if (totalSeconds > 10) TRACELOG(LOG_WARNING, "WAVE: [%s] Ogg audio length larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds); + float totalSeconds = stb_vorbis_stream_length_in_seconds(oggData); + if (totalSeconds > 10) TRACELOG(LOG_WARNING, "WAVE: OGG audio length larger than 10 seconds (%f sec.), that's a big file in memory, consider music streaming", totalSeconds); - wave.data = (short *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(short)); + wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short)); // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!) - stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels); - TRACELOG(LOG_INFO, "WAVE: [%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); + stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.sampleCount); + TRACELOG(LOG_INFO, "WAVE: OGG data loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); - stb_vorbis_close(oggFile); + stb_vorbis_close(oggData); } + else TRACELOG(LOG_WARNING, "WAVE: Failed to load OGG data"); return wave; } #endif #if defined(SUPPORT_FILEFORMAT_FLAC) -// Load FLAC file into Wave structure +// Load FLAC file data into Wave structure // NOTE: Using dr_flac library -static Wave LoadFLAC(const char *fileName) +static Wave LoadFLAC(const unsigned char *fileData, unsigned int fileSize) { Wave wave = { 0 }; - // Decode an entire FLAC file in one go - unsigned long long int totalSampleCount = 0; - wave.data = drflac_open_file_and_read_pcm_frames_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount); + // Decode the entire FLAC file in one go + unsigned long long int totalFrameCount = 0; + wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, fileSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL); - if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load FLAC data", fileName); - else + if (wave.data != NULL) { - wave.sampleCount = (unsigned int)totalSampleCount; + wave.sampleCount = (unsigned int)totalFrameCount*wave.channels; wave.sampleSize = 16; - // NOTE: Only support up to 2 channels (mono, stereo) - if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] FLAC channels number (%i) not supported", fileName, wave.channels); - - TRACELOG(LOG_INFO, "WAVE: [%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); + TRACELOG(LOG_INFO, "WAVE: FLAC data loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); } - + else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data"); + return wave; } #endif #if defined(SUPPORT_FILEFORMAT_MP3) -// Load MP3 file into Wave structure +// Load MP3 file data into Wave structure // NOTE: Using dr_mp3 library -static Wave LoadMP3(const char *fileName) +static Wave LoadMP3(const unsigned char *fileData, unsigned int fileSize) { Wave wave = { 0 }; + drmp3_config config = { 0 }; - // Decode an entire MP3 file in one go + // Decode the entire MP3 file in one go unsigned long long int totalFrameCount = 0; - drmp3_config config = { 0 }; - wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount); + wave.data = drmp3_open_memory_and_read_pcm_frames_f32(fileData, fileSize, &config, &totalFrameCount, NULL); - if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load MP3 data", fileName); - else + if (wave.data != NULL) { - wave.channels = config.outputChannels; - wave.sampleRate = config.outputSampleRate; + wave.channels = config.channels; + wave.sampleRate = config.sampleRate; wave.sampleCount = (int)totalFrameCount*wave.channels; wave.sampleSize = 32; // NOTE: Only support up to 2 channels (mono, stereo) - if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] MP3 channels number (%i) not supported", fileName, wave.channels); + // TODO: Really? + if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: MP3 channels number (%i) not supported", wave.channels); - TRACELOG(LOG_INFO, "WAVE: [%s] MP3 file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); + TRACELOG(LOG_INFO, "WAVE: MP3 file loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); } - + else TRACELOG(LOG_WARNING, "WAVE: Failed to load MP3 data"); + return wave; } #endif @@ -2121,7 +2314,7 @@ static Wave LoadMP3(const char *fileName) // Some required functions for audio standalone module version #if defined(RAUDIO_STANDALONE) // Check file extension -bool IsFileExtension(const char *fileName, const char *ext) +static bool IsFileExtension(const char *fileName, const char *ext) { bool result = false; const char *fileExt; @@ -2133,6 +2326,89 @@ bool IsFileExtension(const char *fileName, const char *ext) return result; } + +// Load data from file into a buffer +static unsigned char *LoadFileData(const char *fileName, unsigned int *bytesRead) +{ + unsigned char *data = NULL; + *bytesRead = 0; + + if (fileName != NULL) + { + FILE *file = fopen(fileName, "rb"); + + if (file != NULL) + { + // WARNING: On binary streams SEEK_END could not be found, + // using fseek() and ftell() could not work in some (rare) cases + fseek(file, 0, SEEK_END); + int size = ftell(file); + fseek(file, 0, SEEK_SET); + + if (size > 0) + { + data = (unsigned char *)RL_MALLOC(size*sizeof(unsigned char)); + + // NOTE: fread() returns number of read elements instead of bytes, so we read [1 byte, size elements] + unsigned int count = (unsigned int)fread(data, sizeof(unsigned char), size, file); + *bytesRead = count; + + if (count != size) TRACELOG(LOG_WARNING, "FILEIO: [%s] File partially loaded", fileName); + else TRACELOG(LOG_INFO, "FILEIO: [%s] File loaded successfully", fileName); + } + else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to read file", fileName); + + fclose(file); + } + else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open file", fileName); + } + else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid"); + + return data; +} + +// Save data to file from buffer +static bool SaveFileData(const char *fileName, void *data, unsigned int bytesToWrite) +{ + if (fileName != NULL) + { + FILE *file = fopen(fileName, "wb"); + + if (file != NULL) + { + unsigned int count = (unsigned int)fwrite(data, sizeof(unsigned char), bytesToWrite, file); + + if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write file", fileName); + else if (count != bytesToWrite) TRACELOG(LOG_WARNING, "FILEIO: [%s] File partially written", fileName); + else TRACELOG(LOG_INFO, "FILEIO: [%s] File saved successfully", fileName); + + fclose(file); + } + else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open file", fileName); + } + else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid"); +} + +// Save text data to file (write), string must be '\0' terminated +static bool SaveFileText(const char *fileName, char *text) +{ + if (fileName != NULL) + { + FILE *file = fopen(fileName, "wt"); + + if (file != NULL) + { + int count = fprintf(file, "%s", text); + + if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write text file", fileName); + else TRACELOG(LOG_INFO, "FILEIO: [%s] Text file saved successfully", fileName); + + fclose(file); + } + else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open text file", fileName); + } + else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid"); +} #endif #undef AudioBuffer |