/********************************************************************************************** * * raylib.audio - Basic funtionality to work with audio * * FEATURES: * - Manage audio device (init/close) * - Load and unload audio files * - Format wave data (sample rate, size, channels) * - Play/Stop/Pause/Resume loaded audio * - Manage mixing channels * - Manage raw audio context * * CONFIGURATION: * * #define AUDIO_STANDALONE * Define to use the module as standalone library (independently of raylib). * Required types and functions are defined in the same module. * * #define USE_OPENAL_BACKEND * Use OpenAL Soft audio backend usage * * #define SUPPORT_FILEFORMAT_WAV * #define SUPPORT_FILEFORMAT_OGG * #define SUPPORT_FILEFORMAT_XM * #define SUPPORT_FILEFORMAT_MOD * #define SUPPORT_FILEFORMAT_FLAC * #define SUPPORT_FILEFORMAT_MP3 * Selected desired fileformats to be supported for loading. Some of those formats are * supported by default, to remove support, just comment unrequired #define in this module * * LIMITATIONS (only OpenAL Soft): * Only up to two channels supported: MONO and STEREO (for additional channels, use AL_EXT_MCFORMATS) * Only the following sample sizes supported: 8bit PCM, 16bit PCM, 32-bit float PCM (using AL_EXT_FLOAT32) * * DEPENDENCIES: * mini_al - Audio device/context management (https://github.com/dr-soft/mini_al) * stb_vorbis - OGG audio files loading (http://www.nothings.org/stb_vorbis/) * jar_xm - XM module file loading * jar_mod - MOD audio file loading * dr_flac - FLAC audio file loading * * *OpenAL Soft - Audio device management, still used on HTML5 and OSX platforms * * CONTRIBUTORS: * David Reid (github: @mackron) (Nov. 2017): * - Complete port to mini_al library * * Joshua Reisenauer (github: @kd7tck) (2015) * - XM audio module support (jar_xm) * - MOD audio module support (jar_mod) * - Mixing channels support * - Raw audio context support * * * LICENSE: zlib/libpng * * Copyright (c) 2014-2018 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. * * Permission is granted to anyone to use this software for any purpose, including commercial * applications, and to alter it and redistribute it freely, subject to the following restrictions: * * 1. The origin of this software must not be misrepresented; you must not claim that you * wrote the original software. If you use this software in a product, an acknowledgment * in the product documentation would be appreciated but is not required. * * 2. Altered source versions must be plainly marked as such, and must not be misrepresented * as being the original software. * * 3. This notice may not be removed or altered from any source distribution. * **********************************************************************************************/ #include "config.h" #if !defined(USE_OPENAL_BACKEND) #define USE_MINI_AL 1 // Set to 1 to use mini_al; 0 to use OpenAL. #endif #if defined(AUDIO_STANDALONE) #include "audio.h" #include // Required for: va_list, va_start(), vfprintf(), va_end() #else #include "config.h" // Defines module configuration flags #include "raylib.h" // Declares module functions #include "utils.h" // Required for: fopen() Android mapping #endif #include "external/mini_al.h" // Implemented in mini_al.c. Cannot implement this here because it conflicts with Win32 APIs such as CloseWindow(), etc. #if !defined(USE_MINI_AL) || (USE_MINI_AL == 0) #if defined(__APPLE__) #include "OpenAL/al.h" // OpenAL basic header #include "OpenAL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work) #else #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 header, required for AL_EXT_FLOAT32 and AL_EXT_MCFORMATS #endif // OpenAL extension: AL_EXT_FLOAT32 - Support for 32bit float samples // OpenAL extension: AL_EXT_MCFORMATS - Support for multi-channel formats (Quad, 5.1, 6.1, 7.1) #endif #include // Required for: malloc(), free() #include // Required for: strcmp(), strncmp() #include // Required for: FILE, fopen(), fclose(), fread() #if defined(SUPPORT_FILEFORMAT_OGG) //#define STB_VORBIS_HEADER_ONLY #include "external/stb_vorbis.h" // OGG loading functions #endif #if defined(SUPPORT_FILEFORMAT_XM) #define JAR_XM_IMPLEMENTATION #include "external/jar_xm.h" // XM loading functions #endif #if defined(SUPPORT_FILEFORMAT_MOD) #define JAR_MOD_IMPLEMENTATION #include "external/jar_mod.h" // MOD loading functions #endif #if defined(SUPPORT_FILEFORMAT_FLAC) #define DR_FLAC_IMPLEMENTATION #define DR_FLAC_NO_WIN32_IO #include "external/dr_flac.h" // FLAC loading functions #endif #if defined(SUPPORT_FILEFORMAT_MP3) #define DR_MP3_IMPLEMENTATION #include "external/dr_mp3.h" // MP3 loading functions #endif #ifdef _MSC_VER #undef bool #endif //---------------------------------------------------------------------------------- // Defines and Macros //---------------------------------------------------------------------------------- #define MAX_STREAM_BUFFERS 2 // Number of buffers for each audio stream // NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds // and double-buffering system, I concluded that a 4096 samples buffer should be enough // In case of music-stalls, just increase this number #define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. 16bit, Mono: 8Kb) // Support uncompressed PCM data in 32-bit float IEEE format // NOTE: This definition is included in "AL/alext.h", but some OpenAL implementations // could not provide the extensions header (Android), so its defined here #if !defined(AL_EXT_float32) #define AL_EXT_float32 1 #define AL_FORMAT_MONO_FLOAT32 0x10010 #define AL_FORMAT_STEREO_FLOAT32 0x10011 #endif //---------------------------------------------------------------------------------- // Types and Structures Definition //---------------------------------------------------------------------------------- typedef enum { MUSIC_AUDIO_OGG = 0, MUSIC_AUDIO_FLAC, MUSIC_AUDIO_MP3, MUSIC_MODULE_XM, MUSIC_MODULE_MOD } MusicContextType; // Music type (file streaming from memory) typedef struct MusicData { MusicContextType ctxType; // Type of music context (OGG, XM, MOD) #if defined(SUPPORT_FILEFORMAT_OGG) stb_vorbis *ctxOgg; // OGG audio context #endif #if defined(SUPPORT_FILEFORMAT_FLAC) drflac *ctxFlac; // FLAC audio context #endif #if defined(SUPPORT_FILEFORMAT_MP3) drmp3 ctxMp3; // MP3 audio context #endif #if defined(SUPPORT_FILEFORMAT_XM) jar_xm_context_t *ctxXm; // XM chiptune context #endif #if defined(SUPPORT_FILEFORMAT_MOD) jar_mod_context_t ctxMod; // MOD chiptune context #endif AudioStream stream; // Audio stream (double buffering) int loopCount; // Loops count (times music repeats), -1 means infinite loop unsigned int totalSamples; // Total number of samples unsigned int samplesLeft; // Number of samples left to end } MusicData; #if defined(AUDIO_STANDALONE) typedef enum { LOG_INFO = 0, LOG_ERROR, LOG_WARNING, LOG_DEBUG, LOG_OTHER } TraceLogType; #endif //---------------------------------------------------------------------------------- // Global Variables Definition //---------------------------------------------------------------------------------- // ... //---------------------------------------------------------------------------------- // Module specific Functions Declaration //---------------------------------------------------------------------------------- #if defined(SUPPORT_FILEFORMAT_WAV) static Wave LoadWAV(const char *fileName); // Load WAV file #endif #if defined(SUPPORT_FILEFORMAT_OGG) static Wave LoadOGG(const char *fileName); // Load OGG file #endif #if defined(SUPPORT_FILEFORMAT_FLAC) static Wave LoadFLAC(const char *fileName); // Load FLAC file #endif #if defined(AUDIO_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) #endif //---------------------------------------------------------------------------------- // mini_al AudioBuffer Functionality //---------------------------------------------------------------------------------- #if USE_MINI_AL #define DEVICE_FORMAT mal_format_f32 #define DEVICE_CHANNELS 2 #define DEVICE_SAMPLE_RATE 44100 typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioBufferUsage; // Audio buffer structure // NOTE: Slightly different logic is used when feeding data to the playback device depending on whether or not data is streamed typedef struct AudioBuffer AudioBuffer; struct AudioBuffer { mal_dsp dsp; // Required for format conversion float volume; float pitch; bool playing; bool paused; bool looping; // Always true for AudioStreams int usage; // AudioBufferUsage type bool isSubBufferProcessed[2]; unsigned int frameCursorPos; unsigned int bufferSizeInFrames; AudioBuffer *next; AudioBuffer *prev; unsigned char buffer[1]; }; // mini_al global variables static mal_context context; static mal_device device; static mal_mutex audioLock; static bool isAudioInitialized = MAL_FALSE; static float masterVolume = 1.0f; // Audio buffers are tracked in a linked list static AudioBuffer *firstAudioBuffer = NULL; static AudioBuffer *lastAudioBuffer = NULL; // mini_al functions declaration static void OnLog(mal_context *pContext, mal_device *pDevice, const char *message); static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameCount, void *pFramesOut); static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, void *pFramesOut, void *pUserData); static void MixAudioFrames(float *framesOut, const float *framesIn, mal_uint32 frameCount, float localVolume); // AudioBuffer management functions declaration // NOTE: Those functions are not exposed by raylib... for the moment AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint32 sampleRate, mal_uint32 bufferSizeInFrames, AudioBufferUsage usage); void DeleteAudioBuffer(AudioBuffer *audioBuffer); bool IsAudioBufferPlaying(AudioBuffer *audioBuffer); void PlayAudioBuffer(AudioBuffer *audioBuffer); void StopAudioBuffer(AudioBuffer *audioBuffer); void PauseAudioBuffer(AudioBuffer *audioBuffer); void ResumeAudioBuffer(AudioBuffer *audioBuffer); void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume); void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch); void TrackAudioBuffer(AudioBuffer *audioBuffer); void UntrackAudioBuffer(AudioBuffer *audioBuffer); // Log callback function static void OnLog(mal_context *pContext, mal_device *pDevice, const char *message) { (void)pContext; (void)pDevice; TraceLog(LOG_ERROR, message); // All log messages from mini_al are errors } // Sending audio data to device callback function static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameCount, void *pFramesOut) { // This is where all of the mixing takes place. (void)pDevice; // Mixing is basically just an accumulation. We need to initialize the output buffer to 0. memset(pFramesOut, 0, frameCount*pDevice->channels*mal_get_sample_size_in_bytes(pDevice->format)); // Using a mutex here for thread-safety which makes things not real-time. This is unlikely to be necessary for this project, but may // want to consider how you might want to avoid this. mal_mutex_lock(&audioLock); { for (AudioBuffer *audioBuffer = firstAudioBuffer; audioBuffer != NULL; audioBuffer = audioBuffer->next) { // Ignore stopped or paused sounds. if (!audioBuffer->playing || audioBuffer->paused) continue; mal_uint32 framesRead = 0; for (;;) { if (framesRead > frameCount) { TraceLog(LOG_DEBUG, "Mixed too many frames from audio buffer"); break; } if (framesRead == frameCount) break; // Just read as much data as we can from the stream. mal_uint32 framesToRead = (frameCount - framesRead); while (framesToRead > 0) { float tempBuffer[1024]; // 512 frames for stereo. mal_uint32 framesToReadRightNow = framesToRead; if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS) { framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS; } // If we're not looping, we need to make sure we flush the internal buffers of the DSP pipeline to ensure we get the // last few samples. bool flushDSP = !audioBuffer->looping; mal_uint32 framesJustRead = mal_dsp_read_frames_ex(&audioBuffer->dsp, framesToReadRightNow, tempBuffer, flushDSP); if (framesJustRead > 0) { float *framesOut = (float *)pFramesOut + (framesRead*device.channels); float *framesIn = tempBuffer; MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume); framesToRead -= framesJustRead; framesRead += framesJustRead; } // If we weren't able to read all the frames we requested, break. if (framesJustRead < framesToReadRightNow) { if (!audioBuffer->looping) { StopAudioBuffer(audioBuffer); break; } else { // Should never get here, but just for safety, // move the cursor position back to the start and continue the loop. audioBuffer->frameCursorPos = 0; continue; } } } // If for some reason we weren't able to read every frame we'll need to break from the loop. // Not doing this could theoretically put us into an infinite loop. if (framesToRead > 0) break; } } } mal_mutex_unlock(&audioLock); return frameCount; // We always output the same number of frames that were originally requested. } // DSP read from audio buffer callback function static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, void *pFramesOut, void *pUserData) { AudioBuffer *audioBuffer = (AudioBuffer *)pUserData; mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2; mal_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames; if (currentSubBufferIndex > 1) { TraceLog(LOG_DEBUG, "Frame cursor position moved too far forward in audio stream"); return 0; } // Another thread can update the processed state of buffers so we just take a copy here to try and avoid potential synchronization problems. bool isSubBufferProcessed[2]; isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0]; isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1]; mal_uint32 frameSizeInBytes = mal_get_sample_size_in_bytes(audioBuffer->dsp.config.formatIn)*audioBuffer->dsp.config.channelsIn; // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0. mal_uint32 framesRead = 0; for (;;) { // We break from this loop differently depending on the buffer's usage. For static buffers, we simply fill as much data as we can. For // streaming buffers we only fill the halves of the buffer that are processed. Unprocessed halves must keep their audio data in-tact. if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) { if (framesRead >= frameCount) break; } else { if (isSubBufferProcessed[currentSubBufferIndex]) break; } mal_uint32 totalFramesRemaining = (frameCount - framesRead); if (totalFramesRemaining == 0) break; mal_uint32 framesRemainingInOutputBuffer; if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) { framesRemainingInOutputBuffer = audioBuffer->bufferSizeInFrames - audioBuffer->frameCursorPos; } else { mal_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames * currentSubBufferIndex; framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer); } mal_uint32 framesToRead = totalFramesRemaining; if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer; memcpy((unsigned char *)pFramesOut + (framesRead*frameSizeInBytes), audioBuffer->buffer + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes); audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead) % audioBuffer->bufferSizeInFrames; framesRead += framesToRead; // If we've read to the end of the buffer, mark it as processed. if (framesToRead == framesRemainingInOutputBuffer) { audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true; isSubBufferProcessed[currentSubBufferIndex] = true; currentSubBufferIndex = (currentSubBufferIndex + 1)%2; // We need to break from this loop if we're not looping. if (!audioBuffer->looping) { StopAudioBuffer(audioBuffer); break; } } } // Zero-fill excess. mal_uint32 totalFramesRemaining = (frameCount - framesRead); if (totalFramesRemaining > 0) { memset((unsigned char*)pFramesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes); // For static buffers we can fill the remaining frames with silence for safety, but we don't want // to report those frames as "read". The reason for this is that the caller uses the return value // to know whether or not a non-looping sound has finished playback. if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining; } return framesRead; } // This is the main mixing function. Mixing is pretty simple in this project - it's just an accumulation. // NOTE: framesOut is both an input and an output. It will be initially filled with zeros outside of this function. static void MixAudioFrames(float *framesOut, const float *framesIn, mal_uint32 frameCount, float localVolume) { for (mal_uint32 iFrame = 0; iFrame < frameCount; ++iFrame) { for (mal_uint32 iChannel = 0; iChannel < device.channels; ++iChannel) { float *frameOut = framesOut + (iFrame*device.channels); const float *frameIn = framesIn + (iFrame*device.channels); frameOut[iChannel] += frameIn[iChannel]*masterVolume*localVolume; } } } #endif //---------------------------------------------------------------------------------- // Module Functions Definition - Audio Device initialization and Closing //---------------------------------------------------------------------------------- // Initialize audio device void InitAudioDevice(void) { #if USE_MINI_AL // Context. mal_context_config contextConfig = mal_context_config_init(OnLog); mal_result result = mal_context_init(NULL, 0, &contextConfig, &context); if (result != MAL_SUCCESS) { TraceLog(LOG_ERROR, "Failed to initialize audio context"); return; } // Device. Using the default device. Format is floating point because it simplifies mixing. mal_device_config deviceConfig = mal_device_config_init(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, OnSendAudioDataToDevice); result = mal_device_init(&context, mal_device_type_playback, NULL, &deviceConfig, NULL, &device); if (result != MAL_SUCCESS) { TraceLog(LOG_ERROR, "Failed to initialize audio playback device"); mal_context_uninit(&context); return; } // Keep the device running the whole time. May want to consider doing something a bit smarter and only have the device running // while there's at least one sound being played. result = mal_device_start(&device); if (result != MAL_SUCCESS) { TraceLog(LOG_ERROR, "Failed to start audio playback device"); mal_device_uninit(&device); mal_context_uninit(&context); return; } // 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 (mal_mutex_init(&context, &audioLock) != MAL_SUCCESS) { TraceLog(LOG_ERROR, "Failed to create mutex for audio mixing"); mal_device_uninit(&device); mal_context_uninit(&context); return; } TraceLog(LOG_INFO, "Audio device initialized successfully: %s", device.name); TraceLog(LOG_INFO, "Audio backend: mini_al / %s", mal_get_backend_name(context.backend)); TraceLog(LOG_INFO, "Audio format: %s -> %s", mal_get_format_name(device.format), mal_get_format_name(device.internalFormat)); TraceLog(LOG_INFO, "Audio channels: %d -> %d", device.channels, device.internalChannels); TraceLog(LOG_INFO, "Audio sample rate: %d -> %d", device.sampleRate, device.internalSampleRate); TraceLog(LOG_INFO, "Audio buffer size: %d", device.bufferSizeInFrames); isAudioInitialized = MAL_TRUE; #else // Open and initialize a device with default settings ALCdevice *device = alcOpenDevice(NULL); if (!device) TraceLog(LOG_ERROR, "Audio device could not be opened"); else { ALCcontext *context = alcCreateContext(device, NULL); if ((context == NULL) || (alcMakeContextCurrent(context) == ALC_FALSE)) { if (context != NULL) alcDestroyContext(context); alcCloseDevice(device); TraceLog(LOG_ERROR, "Could not initialize audio context"); } else { TraceLog(LOG_INFO, "Audio device and context initialized successfully: %s", alcGetString(device, ALC_DEVICE_SPECIFIER)); // Listener definition (just for 2D) alListener3f(AL_POSITION, 0.0f, 0.0f, 0.0f); alListener3f(AL_VELOCITY, 0.0f, 0.0f, 0.0f); alListener3f(AL_ORIENTATION, 0.0f, 0.0f, -1.0f); alListenerf(AL_GAIN, 1.0f); if (alIsExtensionPresent("AL_EXT_float32")) TraceLog(LOG_INFO, "[EXTENSION] AL_EXT_float32 supported"); else TraceLog(LOG_INFO, "[EXTENSION] AL_EXT_float32 not supported"); } } #endif } // Close the audio device for all contexts void CloseAudioDevice(void) { #if USE_MINI_AL if (!isAudioInitialized) { TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized"); return; } mal_mutex_uninit(&audioLock); mal_device_uninit(&device); mal_context_uninit(&context); #else ALCdevice *device; ALCcontext *context = alcGetCurrentContext(); if (context == NULL) TraceLog(LOG_WARNING, "Could not get current audio context for closing"); device = alcGetContextsDevice(context); alcMakeContextCurrent(NULL); alcDestroyContext(context); alcCloseDevice(device); #endif TraceLog(LOG_INFO, "Audio device closed successfully"); } // Check if device has been initialized successfully bool IsAudioDeviceReady(void) { #if USE_MINI_AL return isAudioInitialized; #else ALCcontext *context = alcGetCurrentContext(); if (context == NULL) return false; else { ALCdevice *device = alcGetContextsDevice(context); if (device == NULL) return false; else return true; } #endif } // Set master volume (listener) void SetMasterVolume(float volume) { if (volume < 0.0f) volume = 0.0f; else if (volume > 1.0f) volume = 1.0f; #if USE_MINI_AL masterVolume = volume; #else alListenerf(AL_GAIN, volume); #endif } //---------------------------------------------------------------------------------- // Module Functions Definition - Audio Buffer management //---------------------------------------------------------------------------------- #if USE_MINI_AL // Create a new audio buffer. Initially filled with silence AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint32 sampleRate, mal_uint32 bufferSizeInFrames, AudioBufferUsage usage) { AudioBuffer *audioBuffer = (AudioBuffer *)calloc(sizeof(*audioBuffer) + (bufferSizeInFrames*channels*mal_get_sample_size_in_bytes(format)), 1); if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to allocate memory for audio buffer"); return NULL; } // We run audio data through a format converter. mal_dsp_config dspConfig; memset(&dspConfig, 0, sizeof(dspConfig)); dspConfig.formatIn = format; dspConfig.formatOut = DEVICE_FORMAT; dspConfig.channelsIn = channels; dspConfig.channelsOut = DEVICE_CHANNELS; dspConfig.sampleRateIn = sampleRate; dspConfig.sampleRateOut = DEVICE_SAMPLE_RATE; mal_result resultMAL = mal_dsp_init(&dspConfig, OnAudioBufferDSPRead, audioBuffer, &audioBuffer->dsp); if (resultMAL != MAL_SUCCESS) { TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to create data conversion pipeline"); free(audioBuffer); return NULL; } audioBuffer->volume = 1; audioBuffer->pitch = 1; audioBuffer->playing = 0; audioBuffer->paused = 0; audioBuffer->looping = 0; audioBuffer->usage = usage; audioBuffer->bufferSizeInFrames = bufferSizeInFrames; audioBuffer->frameCursorPos = 0; // Buffers should be marked as processed by default so that a call to UpdateAudioStream() immediately after initialization works correctly. audioBuffer->isSubBufferProcessed[0] = true; audioBuffer->isSubBufferProcessed[1] = true; TrackAudioBuffer(audioBuffer); return audioBuffer; } // Delete an audio buffer void DeleteAudioBuffer(AudioBuffer *audioBuffer) { if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer"); return; } UntrackAudioBuffer(audioBuffer); free(audioBuffer); } // Check if an audio buffer is playing bool IsAudioBufferPlaying(AudioBuffer *audioBuffer) { if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer"); return false; } return audioBuffer->playing && !audioBuffer->paused; } // Play an audio buffer // NOTE: Buffer is restarted to the start. // Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained. void PlayAudioBuffer(AudioBuffer *audioBuffer) { if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer"); return; } audioBuffer->playing = true; audioBuffer->paused = false; audioBuffer->frameCursorPos = 0; } // Stop an audio buffer void StopAudioBuffer(AudioBuffer *audioBuffer) { if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer"); return; } // Don't do anything if the audio buffer is already stopped. if (!IsAudioBufferPlaying(audioBuffer)) return; audioBuffer->playing = false; audioBuffer->paused = false; audioBuffer->frameCursorPos = 0; audioBuffer->isSubBufferProcessed[0] = true; audioBuffer->isSubBufferProcessed[1] = true; } // Pause an audio buffer void PauseAudioBuffer(AudioBuffer *audioBuffer) { if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer"); return; } audioBuffer->paused = true; } // Resume an audio buffer void ResumeAudioBuffer(AudioBuffer *audioBuffer) { if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer"); return; } audioBuffer->paused = false; } // Set volume for an audio buffer void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume) { if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer"); return; } audioBuffer->volume = volume; } // Set pitch for an audio buffer void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch) { if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer"); return; } audioBuffer->pitch = 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. mal_uint32 newOutputSampleRate = (mal_uint32)((((float)audioBuffer->dsp.config.sampleRateOut / (float)audioBuffer->dsp.config.sampleRateIn) / pitch) * audioBuffer->dsp.config.sampleRateIn); mal_dsp_set_output_sample_rate(&audioBuffer->dsp, newOutputSampleRate); } // Track audio buffer to linked list next position void TrackAudioBuffer(AudioBuffer *audioBuffer) { mal_mutex_lock(&audioLock); { if (firstAudioBuffer == NULL) firstAudioBuffer = audioBuffer; else { lastAudioBuffer->next = audioBuffer; audioBuffer->prev = lastAudioBuffer; } lastAudioBuffer = audioBuffer; } mal_mutex_unlock(&audioLock); } // Untrack audio buffer from linked list void UntrackAudioBuffer(AudioBuffer *audioBuffer) { mal_mutex_lock(&audioLock); { if (audioBuffer->prev == NULL) firstAudioBuffer = audioBuffer->next; else audioBuffer->prev->next = audioBuffer->next; if (audioBuffer->next == NULL) lastAudioBuffer = audioBuffer->prev; else audioBuffer->next->prev = audioBuffer->prev; audioBuffer->prev = NULL; audioBuffer->next = NULL; } mal_mutex_unlock(&audioLock); } #endif //---------------------------------------------------------------------------------- // Module Functions Definition - Sounds loading and playing (.WAV) //---------------------------------------------------------------------------------- // Load wave data from file Wave LoadWave(const char *fileName) { Wave wave = { 0 }; if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName); #if defined(SUPPORT_FILEFORMAT_OGG) else if (IsFileExtension(fileName, ".ogg")) wave = LoadOGG(fileName); #endif #if defined(SUPPORT_FILEFORMAT_FLAC) else if (IsFileExtension(fileName, ".flac")) wave = LoadFLAC(fileName); #endif else TraceLog(LOG_WARNING, "[%s] Audio fileformat not supported, it can't be loaded", fileName); return wave; } // Load wave data from raw array data Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int channels) { Wave wave; wave.data = data; wave.sampleCount = sampleCount; wave.sampleRate = sampleRate; wave.sampleSize = sampleSize; wave.channels = channels; // NOTE: Copy wave data to work with, user is responsible of input data to free Wave cwave = WaveCopy(wave); WaveFormat(&cwave, sampleRate, sampleSize, channels); return cwave; } // Load sound from file // NOTE: The entire file is loaded to memory to be played (no-streaming) Sound LoadSound(const char *fileName) { Wave wave = LoadWave(fileName); Sound sound = LoadSoundFromWave(wave); UnloadWave(wave); // Sound is loaded, we can unload wave return sound; } // Load sound from wave data // NOTE: Wave data must be unallocated manually Sound LoadSoundFromWave(Wave wave) { Sound sound = { 0 }; if (wave.data != NULL) { #if USE_MINI_AL // When using mini_al we need to do our own mixing. To simplify this we need convert the format of each sound to be consistent with // the format used to open the playback device. We can do this two ways: // // 1) Convert the whole sound in one go at load time (here). // 2) Convert the audio data in chunks at mixing time. // // I have decided on the first option because it offloads work required for the format conversion to the to the loading stage. The // downside to this is that it uses more memory if the original sound is u8 or s16. mal_format formatIn = ((wave.sampleSize == 8) ? mal_format_u8 : ((wave.sampleSize == 16) ? mal_format_s16 : mal_format_f32)); mal_uint32 frameCountIn = wave.sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so. mal_uint32 frameCount = mal_convert_frames(NULL, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, formatIn, wave.channels, wave.sampleRate, frameCountIn); if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Failed to get frame count for format conversion"); AudioBuffer* audioBuffer = CreateAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC); if (audioBuffer == NULL) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Failed to create audio buffer"); frameCount = mal_convert_frames(audioBuffer->buffer, audioBuffer->dsp.config.formatIn, audioBuffer->dsp.config.channelsIn, audioBuffer->dsp.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn); if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed"); sound.audioBuffer = audioBuffer; #else ALenum format = 0; // The OpenAL format is worked out by looking at the number of channels and the sample size (bits per sample) if (wave.channels == 1) { switch (wave.sampleSize) { case 8: format = AL_FORMAT_MONO8; break; case 16: format = AL_FORMAT_MONO16; break; case 32: format = AL_FORMAT_MONO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 default: TraceLog(LOG_WARNING, "Wave sample size not supported: %i", wave.sampleSize); break; } } else if (wave.channels == 2) { switch (wave.sampleSize) { case 8: format = AL_FORMAT_STEREO8; break; case 16: format = AL_FORMAT_STEREO16; break; case 32: format = AL_FORMAT_STEREO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 default: TraceLog(LOG_WARNING, "Wave sample size not supported: %i", wave.sampleSize); break; } } else TraceLog(LOG_WARNING, "Wave number of channels not supported: %i", wave.channels); // Create an audio source ALuint source; alGenSources(1, &source); // Generate pointer to audio source alSourcef(source, AL_PITCH, 1.0f); alSourcef(source, AL_GAIN, 1.0f); alSource3f(source, AL_POSITION, 0.0f, 0.0f, 0.0f); alSource3f(source, AL_VELOCITY, 0.0f, 0.0f, 0.0f); alSourcei(source, AL_LOOPING, AL_FALSE); // Convert loaded data to OpenAL buffer //---------------------------------------- ALuint buffer; alGenBuffers(1, &buffer); // Generate pointer to buffer unsigned int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; // Size in bytes // Upload sound data to buffer alBufferData(buffer, format, wave.data, dataSize, wave.sampleRate); // Attach sound buffer to source alSourcei(source, AL_BUFFER, buffer); TraceLog(LOG_INFO, "[SND ID %i][BUFR ID %i] Sound data loaded successfully (%i Hz, %i bit, %s)", source, buffer, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); sound.source = source; sound.buffer = buffer; sound.format = format; #endif } return sound; } // Unload wave data void UnloadWave(Wave wave) { if (wave.data != NULL) free(wave.data); TraceLog(LOG_INFO, "Unloaded wave data from RAM"); } // Unload sound void UnloadSound(Sound sound) { #if USE_MINI_AL DeleteAudioBuffer((AudioBuffer *)sound.audioBuffer); #else alSourceStop(sound.source); alDeleteSources(1, &sound.source); alDeleteBuffers(1, &sound.buffer); #endif TraceLog(LOG_INFO, "[SND ID %i][BUFR ID %i] Unloaded sound data from RAM", sound.source, sound.buffer); } // Update sound buffer with new data // NOTE: data must match sound.format void UpdateSound(Sound sound, const void *data, int samplesCount) { #if USE_MINI_AL AudioBuffer *audioBuffer = (AudioBuffer *)sound.audioBuffer; if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer"); return; } StopAudioBuffer(audioBuffer); // TODO: May want to lock/unlock this since this data buffer is read at mixing time. memcpy(audioBuffer->buffer, data, samplesCount*audioBuffer->dsp.config.channelsIn*mal_get_sample_size_in_bytes(audioBuffer->dsp.config.formatIn)); #else ALint sampleRate, sampleSize, channels; alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate); alGetBufferi(sound.buffer, AL_BITS, &sampleSize); // It could also be retrieved from sound.format alGetBufferi(sound.buffer, AL_CHANNELS, &channels); // It could also be retrieved from sound.format TraceLog(LOG_DEBUG, "UpdateSound() : AL_FREQUENCY: %i", sampleRate); TraceLog(LOG_DEBUG, "UpdateSound() : AL_BITS: %i", sampleSize); TraceLog(LOG_DEBUG, "UpdateSound() : AL_CHANNELS: %i", channels); unsigned int dataSize = samplesCount*channels*sampleSize/8; // Size of data in bytes alSourceStop(sound.source); // Stop sound alSourcei(sound.source, AL_BUFFER, 0); // Unbind buffer from sound to update //alDeleteBuffers(1, &sound.buffer); // Delete current buffer data //alGenBuffers(1, &sound.buffer); // Generate new buffer // Upload new data to sound buffer alBufferData(sound.buffer, sound.format, data, dataSize, sampleRate); // Attach sound buffer to source again alSourcei(sound.source, AL_BUFFER, sound.buffer); #endif } // Play a sound void PlaySound(Sound sound) { #if USE_MINI_AL PlayAudioBuffer((AudioBuffer *)sound.audioBuffer); #else alSourcePlay(sound.source); // Play the sound #endif //TraceLog(LOG_INFO, "Playing sound"); // Find the current position of the sound being played // NOTE: Only work when the entire file is in a single buffer //int byteOffset; //alGetSourcei(sound.source, AL_BYTE_OFFSET, &byteOffset); // //int sampleRate; //alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate); // AL_CHANNELS, AL_BITS (bps) //float seconds = (float)byteOffset/sampleRate; // Number of seconds since the beginning of the sound //or //float result; //alGetSourcef(sound.source, AL_SEC_OFFSET, &result); // AL_SAMPLE_OFFSET } // Pause a sound void PauseSound(Sound sound) { #if USE_MINI_AL PauseAudioBuffer((AudioBuffer *)sound.audioBuffer); #else alSourcePause(sound.source); #endif } // Resume a paused sound void ResumeSound(Sound sound) { #if USE_MINI_AL ResumeAudioBuffer((AudioBuffer *)sound.audioBuffer); #else ALenum state; alGetSourcei(sound.source, AL_SOURCE_STATE, &state); if (state == AL_PAUSED) alSourcePlay(sound.source); #endif } // Stop reproducing a sound void StopSound(Sound sound) { #if USE_MINI_AL StopAudioBuffer((AudioBuffer *)sound.audioBuffer); #else alSourceStop(sound.source); #endif } // Check if a sound is playing bool IsSoundPlaying(Sound sound) { #if USE_MINI_AL return IsAudioBufferPlaying((AudioBuffer *)sound.audioBuffer); #else bool playing = false; ALint state; alGetSourcei(sound.source, AL_SOURCE_STATE, &state); if (state == AL_PLAYING) playing = true; return playing; #endif } // Set volume for a sound void SetSoundVolume(Sound sound, float volume) { #if USE_MINI_AL SetAudioBufferVolume((AudioBuffer *)sound.audioBuffer, volume); #else alSourcef(sound.source, AL_GAIN, volume); #endif } // Set pitch for a sound void SetSoundPitch(Sound sound, float pitch) { #if USE_MINI_AL SetAudioBufferPitch((AudioBuffer *)sound.audioBuffer, pitch); #else alSourcef(sound.source, AL_PITCH, pitch); #endif } // Convert wave data to desired format void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) { #if USE_MINI_AL mal_format formatIn = ((wave->sampleSize == 8) ? mal_format_u8 : ((wave->sampleSize == 16) ? mal_format_s16 : mal_format_f32)); mal_format formatOut = (( sampleSize == 8) ? mal_format_u8 : (( sampleSize == 16) ? mal_format_s16 : mal_format_f32)); mal_uint32 frameCountIn = wave->sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so. mal_uint32 frameCount = mal_convert_frames(NULL, formatOut, channels, sampleRate, NULL, formatIn, wave->channels, wave->sampleRate, frameCountIn); if (frameCount == 0) { TraceLog(LOG_ERROR, "WaveFormat() : Failed to get frame count for format conversion."); return; } void *data = malloc(frameCount*channels*(sampleSize/8)); frameCount = mal_convert_frames(data, formatOut, channels, sampleRate, wave->data, formatIn, wave->channels, wave->sampleRate, frameCountIn); if (frameCount == 0) { TraceLog(LOG_ERROR, "WaveFormat() : Format conversion failed."); return; } wave->sampleCount = frameCount; wave->sampleSize = sampleSize; wave->sampleRate = sampleRate; wave->channels = channels; free(wave->data); wave->data = data; #else // Format sample rate // NOTE: Only supported 22050 <--> 44100 if (wave->sampleRate != sampleRate) { // TODO: Resample wave data (upsampling or downsampling) // NOTE 1: To downsample, you have to drop samples or average them. // NOTE 2: To upsample, you have to interpolate new samples. wave->sampleRate = sampleRate; } // Format sample size // NOTE: Only supported 8 bit <--> 16 bit <--> 32 bit if (wave->sampleSize != sampleSize) { void *data = malloc(wave->sampleCount*wave->channels*sampleSize/8); for (int i = 0; i < wave->sampleCount; i++) { for (int j = 0; j < wave->channels; j++) { if (sampleSize == 8) { if (wave->sampleSize == 16) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float)(((short *)wave->data)[wave->channels*i + j])/32767.0f)*256); else if (wave->sampleSize == 32) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float *)wave->data)[wave->channels*i + j]*127.0f + 127); } else if (sampleSize == 16) { if (wave->sampleSize == 8) ((short *)data)[wave->channels*i + j] = (short)(((float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f)*32767); else if (wave->sampleSize == 32) ((short *)data)[wave->channels*i + j] = (short)((((float *)wave->data)[wave->channels*i + j])*32767); } else if (sampleSize == 32) { if (wave->sampleSize == 8) ((float *)data)[wave->channels*i + j] = (float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f; else if (wave->sampleSize == 16) ((float *)data)[wave->channels*i + j] = (float)(((short *)wave->data)[wave->channels*i + j])/32767.0f; } } } wave->sampleSize = sampleSize; free(wave->data); wave->data = data; } // Format channels (interlaced mode) // NOTE: Only supported mono <--> stereo if (wave->channels != channels) { void *data = malloc(wave->sampleCount*wave->sampleSize/8*channels); if ((wave->channels == 1) && (channels == 2)) // mono ---> stereo (duplicate mono information) { for (int i = 0; i < wave->sampleCount; i++) { for (int j = 0; j < channels; j++) { if (wave->sampleSize == 8) ((unsigned char *)data)[channels*i + j] = ((unsigned char *)wave->data)[i]; else if (wave->sampleSize == 16) ((short *)data)[channels*i + j] = ((short *)wave->data)[i]; else if (wave->sampleSize == 32) ((float *)data)[channels*i + j] = ((float *)wave->data)[i]; } } } else if ((wave->channels == 2) && (channels == 1)) // stereo ---> mono (mix stereo channels) { for (int i = 0, j = 0; i < wave->sampleCount; i++, j += 2) { if (wave->sampleSize == 8) ((unsigned char *)data)[i] = (((unsigned char *)wave->data)[j] + ((unsigned char *)wave->data)[j + 1])/2; else if (wave->sampleSize == 16) ((short *)data)[i] = (((short *)wave->data)[j] + ((short *)wave->data)[j + 1])/2; else if (wave->sampleSize == 32) ((float *)data)[i] = (((float *)wave->data)[j] + ((float *)wave->data)[j + 1])/2.0f; } } // TODO: Add/remove additional interlaced channels wave->channels = channels; free(wave->data); wave->data = data; } #endif } // Copy a wave to a new wave Wave WaveCopy(Wave wave) { Wave newWave = { 0 }; newWave.data = malloc(wave.sampleCount*wave.sampleSize/8*wave.channels); if (newWave.data != NULL) { // NOTE: Size must be provided in bytes memcpy(newWave.data, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8); newWave.sampleCount = wave.sampleCount; newWave.sampleRate = wave.sampleRate; newWave.sampleSize = wave.sampleSize; newWave.channels = wave.channels; } return newWave; } // Crop a wave to defined samples range // NOTE: Security check in case of out-of-range void WaveCrop(Wave *wave, int initSample, int finalSample) { if ((initSample >= 0) && (initSample < finalSample) && (finalSample > 0) && (finalSample < wave->sampleCount)) { int sampleCount = finalSample - initSample; void *data = malloc(sampleCount*wave->sampleSize/8*wave->channels); memcpy(data, (unsigned char*)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8); free(wave->data); wave->data = data; } 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) { float *samples = (float *)malloc(wave.sampleCount*wave.channels*sizeof(float)); for (int i = 0; i < wave.sampleCount; i++) { for (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]; } } return samples; } //---------------------------------------------------------------------------------- // Module Functions Definition - Music loading and stream playing (.OGG) //---------------------------------------------------------------------------------- // Load music stream from file Music LoadMusicStream(const char *fileName) { Music music = (MusicData *)malloc(sizeof(MusicData)); if (IsFileExtension(fileName, ".ogg")) { // Open ogg audio stream music->ctxOgg = stb_vorbis_open_filename(fileName, NULL, NULL); if (music->ctxOgg == NULL) TraceLog(LOG_WARNING, "[%s] OGG audio file could not be opened", fileName); else { stb_vorbis_info info = stb_vorbis_get_info(music->ctxOgg); // 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); music->totalSamples = (unsigned int)stb_vorbis_stream_length_in_samples(music->ctxOgg); // Independent by channel music->samplesLeft = music->totalSamples; music->ctxType = MUSIC_AUDIO_OGG; music->loopCount = -1; // Infinite loop by default TraceLog(LOG_DEBUG, "[%s] FLAC total samples: %i", fileName, music->totalSamples); TraceLog(LOG_DEBUG, "[%s] OGG sample rate: %i", fileName, info.sample_rate); TraceLog(LOG_DEBUG, "[%s] OGG channels: %i", fileName, info.channels); TraceLog(LOG_DEBUG, "[%s] OGG memory required: %i", fileName, info.temp_memory_required); } } #if defined(SUPPORT_FILEFORMAT_FLAC) else if (IsFileExtension(fileName, ".flac")) { music->ctxFlac = drflac_open_file(fileName); if (music->ctxFlac == NULL) TraceLog(LOG_WARNING, "[%s] FLAC audio file could not be opened", fileName); else { music->stream = InitAudioStream(music->ctxFlac->sampleRate, music->ctxFlac->bitsPerSample, music->ctxFlac->channels); music->totalSamples = (unsigned int)music->ctxFlac->totalSampleCount/music->ctxFlac->channels; music->samplesLeft = music->totalSamples; music->ctxType = MUSIC_AUDIO_FLAC; music->loopCount = -1; // Infinite loop by default TraceLog(LOG_DEBUG, "[%s] FLAC total samples: %i", fileName, music->totalSamples); TraceLog(LOG_DEBUG, "[%s] FLAC sample rate: %i", fileName, music->ctxFlac->sampleRate); TraceLog(LOG_DEBUG, "[%s] FLAC bits per sample: %i", fileName, music->ctxFlac->bitsPerSample); TraceLog(LOG_DEBUG, "[%s] FLAC channels: %i", fileName, music->ctxFlac->channels); } } #endif #if defined(SUPPORT_FILEFORMAT_MP3) else if (IsFileExtension(fileName, ".mp3")) { drmp3_init_file(&music->ctxMp3, fileName, NULL) if (music->ctxMp3 == NULL) TraceLog(LOG_WARNING, "[%s] MP3 audio file could not be opened", fileName); else { music->stream = InitAudioStream(music->ctxMp3.sampleRate, 16, music->ctxMp3.channels); //music->totalSamples = (unsigned int)music->ctxMp3.totalSampleCount/music->ctxMp3.channels; //TODO! music->samplesLeft = music->totalSamples; music->ctxType = MUSIC_AUDIO_MP3; music->loopCount = -1; // Infinite loop by default TraceLog(LOG_DEBUG, "[%s] MP3 total samples: %i", fileName, music->totalSamples); TraceLog(LOG_DEBUG, "[%s] MP3 sample rate: %i", fileName, music->ctxMp3.sampleRate); //TraceLog(LOG_DEBUG, "[%s] MP3 bits per sample: %i", fileName, music->ctxMp3.bitsPerSample); TraceLog(LOG_DEBUG, "[%s] MP3 channels: %i", fileName, music->ctxMp3.channels); } } #endif #if defined(SUPPORT_FILEFORMAT_XM) else if (IsFileExtension(fileName, ".xm")) { int result = jar_xm_create_context_from_file(&music->ctxXm, 48000, fileName); if (!result) // XM context created successfully { jar_xm_set_max_loop_count(music->ctxXm, 0); // Set infinite number of loops // NOTE: Only stereo is supported for XM music->stream = InitAudioStream(48000, 16, 2); music->totalSamples = (unsigned int)jar_xm_get_remaining_samples(music->ctxXm); music->samplesLeft = music->totalSamples; music->ctxType = MUSIC_MODULE_XM; music->loopCount = -1; // Infinite loop by default TraceLog(LOG_DEBUG, "[%s] XM number of samples: %i", fileName, music->totalSamples); TraceLog(LOG_DEBUG, "[%s] XM track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f); } else TraceLog(LOG_WARNING, "[%s] XM file could not be opened", fileName); } #endif #if defined(SUPPORT_FILEFORMAT_MOD) else if (IsFileExtension(fileName, ".mod")) { jar_mod_init(&music->ctxMod); if (jar_mod_load_file(&music->ctxMod, fileName)) { music->stream = InitAudioStream(48000, 16, 2); music->totalSamples = (unsigned int)jar_mod_max_samples(&music->ctxMod); music->samplesLeft = music->totalSamples; music->ctxType = MUSIC_MODULE_MOD; music->loopCount = -1; // Infinite loop by default TraceLog(LOG_DEBUG, "[%s] MOD number of samples: %i", fileName, music->samplesLeft); TraceLog(LOG_DEBUG, "[%s] MOD track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f); } else TraceLog(LOG_WARNING, "[%s] MOD file could not be opened", fileName); } #endif else TraceLog(LOG_WARNING, "[%s] Audio fileformat not supported, it can't be loaded", fileName); return music; } // Unload music stream void UnloadMusicStream(Music music) { CloseAudioStream(music->stream); if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close(music->ctxOgg); #if defined(SUPPORT_FILEFORMAT_FLAC) else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_free(music->ctxFlac); #endif #if defined(SUPPORT_FILEFORMAT_MP3) else if (music->ctxType == MUSIC_AUDIO_MP3) drmp3_uninit(&music->ctxMp3); #endif #if defined(SUPPORT_FILEFORMAT_XM) else if (music->ctxType == MUSIC_MODULE_XM) jar_xm_free_context(music->ctxXm); #endif #if defined(SUPPORT_FILEFORMAT_MOD) else if (music->ctxType == MUSIC_MODULE_MOD) jar_mod_unload(&music->ctxMod); #endif free(music); } // Start music playing (open stream) void PlayMusicStream(Music music) { #if USE_MINI_AL AudioBuffer *audioBuffer = (AudioBuffer *)music->stream.audioBuffer; if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer"); return; } // For music streams, we need to make sure we maintain the frame cursor position. This is hack for this section of code in UpdateMusicStream() // // NOTE: In case window is minimized, music stream is stopped, // // just make sure to play again on window restore // if (IsMusicPlaying(music)) PlayMusicStream(music); mal_uint32 frameCursorPos = audioBuffer->frameCursorPos; PlayAudioStream(music->stream); // <-- This resets the cursor position. audioBuffer->frameCursorPos = frameCursorPos; #else alSourcePlay(music->stream.source); #endif } // Pause music playing void PauseMusicStream(Music music) { #if USE_MINI_AL PauseAudioStream(music->stream); #else alSourcePause(music->stream.source); #endif } // Resume music playing void ResumeMusicStream(Music music) { #if USE_MINI_AL ResumeAudioStream(music->stream); #else ALenum state; alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state); if (state == AL_PAUSED) { TraceLog(LOG_INFO, "[AUD ID %i] Resume music stream playing", music->stream.source); alSourcePlay(music->stream.source); } #endif } // Stop music playing (close stream) // TODO: To clear a buffer, make sure they have been already processed! void StopMusicStream(Music music) { #if USE_MINI_AL StopAudioStream(music->stream); #else alSourceStop(music->stream.source); /* // Clear stream buffers // WARNING: Queued buffers must have been processed before unqueueing and reloaded with data!!! void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, 1); for (int i = 0; i < MAX_STREAM_BUFFERS; i++) { //UpdateAudioStream(music->stream, pcm, AUDIO_BUFFER_SIZE); // Update one buffer at a time alBufferData(music->stream.buffers[i], music->stream.format, pcm, AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, music->stream.sampleRate); } free(pcm); */ #endif // Restart music context switch (music->ctxType) { case MUSIC_AUDIO_OGG: stb_vorbis_seek_start(music->ctxOgg); break; #if defined(SUPPORT_FILEFORMAT_FLAC) case MUSIC_MODULE_FLAC: /* TODO: Restart FLAC context */ break; #endif #if defined(SUPPORT_FILEFORMAT_XM) case MUSIC_MODULE_XM: /* TODO: Restart XM context */ break; #endif #if defined(SUPPORT_FILEFORMAT_MOD) case MUSIC_MODULE_MOD: jar_mod_seek_start(&music->ctxMod); break; #endif default: break; } music->samplesLeft = music->totalSamples; } // Update (re-fill) music buffers if data already processed // TODO: Make sure buffers are ready for update... check music state void UpdateMusicStream(Music music) { #if USE_MINI_AL bool streamEnding = false; unsigned int subBufferSizeInFrames = ((AudioBuffer *)music->stream.audioBuffer)->bufferSizeInFrames/2; // NOTE: Using dynamic allocation because it could require more than 16KB void *pcm = calloc(subBufferSizeInFrames*music->stream.sampleSize/8*music->stream.channels, 1); int samplesCount = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts while (IsAudioBufferProcessed(music->stream)) { if (music->samplesLeft >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames; else samplesCount = music->samplesLeft; // TODO: Really don't like ctxType thingy... switch (music->ctxType) { case MUSIC_AUDIO_OGG: { // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!) stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount*music->stream.channels); } break; #if defined(SUPPORT_FILEFORMAT_FLAC) case MUSIC_AUDIO_FLAC: { // NOTE: Returns the number of samples to process unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, samplesCount*music->stream.channels, (short *)pcm); } break; #endif #if defined(SUPPORT_FILEFORMAT_MP3) case MUSIC_AUDIO_MP3: { // NOTE: Returns the number of samples to process unsigned int numSamplesMp3 = (unsigned int)drmp3_read_f32(music->ctxMp3, samplesCount*music->stream.channels, (short *)pcm); } break; #endif #if defined(SUPPORT_FILEFORMAT_XM) case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, samplesCount); break; #endif #if defined(SUPPORT_FILEFORMAT_MOD) case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, samplesCount, 0); break; #endif default: break; } UpdateAudioStream(music->stream, pcm, samplesCount); music->samplesLeft -= samplesCount; if (music->samplesLeft <= 0) { streamEnding = true; break; } } // Free allocated pcm data free(pcm); // Reset audio stream for looping if (streamEnding) { StopMusicStream(music); // Stop music (and reset) // Decrease loopCount to stop when required if (music->loopCount > 0) { music->loopCount--; // Decrease loop count PlayMusicStream(music); // Play again } else { if (music->loopCount == -1) PlayMusicStream(music); } } else { // NOTE: In case window is minimized, music stream is stopped, // just make sure to play again on window restore if (IsMusicPlaying(music)) PlayMusicStream(music); } #else ALenum state; ALint processed = 0; alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state); // Get music stream state alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed); // Get processed buffers if (processed > 0) { bool streamEnding = false; // NOTE: Using dynamic allocation because it could require more than 16KB void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, 1); int numBuffersToProcess = processed; int samplesCount = 0; // Total size of data steamed in L+R samples for xm floats, // individual L or R for ogg shorts for (int i = 0; i < numBuffersToProcess; i++) { if (music->samplesLeft >= AUDIO_BUFFER_SIZE) samplesCount = AUDIO_BUFFER_SIZE; else samplesCount = music->samplesLeft; // TODO: Really don't like ctxType thingy... switch (music->ctxType) { case MUSIC_AUDIO_OGG: { // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!) int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount*music->stream.channels); } break; #if defined(SUPPORT_FILEFORMAT_FLAC) case MUSIC_AUDIO_FLAC: { // NOTE: Returns the number of samples to process unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, samplesCount*music->stream.channels, (short *)pcm); } break; #endif #if defined(SUPPORT_FILEFORMAT_XM) case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, samplesCount); break; #endif #if defined(SUPPORT_FILEFORMAT_MOD) case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, samplesCount, 0); break; #endif default: break; } UpdateAudioStream(music->stream, pcm, samplesCount); music->samplesLeft -= samplesCount; if (music->samplesLeft <= 0) { streamEnding = true; break; } } // Free allocated pcm data free(pcm); // Reset audio stream for looping if (streamEnding) { StopMusicStream(music); // Stop music (and reset) // Decrease loopCount to stop when required if (music->loopCount > 0) { music->loopCount--; // Decrease loop count PlayMusicStream(music); // Play again } else { if (music->loopCount == -1) { PlayMusicStream(music); } } } else { // NOTE: In case window is minimized, music stream is stopped, // just make sure to play again on window restore if (state != AL_PLAYING) PlayMusicStream(music); } } #endif } // Check if any music is playing bool IsMusicPlaying(Music music) { #if USE_MINI_AL return IsAudioStreamPlaying(music->stream); #else bool playing = false; ALint state; alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state); if (state == AL_PLAYING) playing = true; return playing; #endif } // Set volume for music void SetMusicVolume(Music music, float volume) { #if USE_MINI_AL SetAudioStreamVolume(music->stream, volume); #else alSourcef(music->stream.source, AL_GAIN, volume); #endif } // Set pitch for music void SetMusicPitch(Music music, float pitch) { #if USE_MINI_AL SetAudioStreamPitch(music->stream, pitch); #else alSourcef(music->stream.source, AL_PITCH, pitch); #endif } // Set music loop count (loop repeats) // NOTE: If set to -1, means infinite loop void SetMusicLoopCount(Music music, int count) { music->loopCount = count; } // Get music time length (in seconds) float GetMusicTimeLength(Music music) { float totalSeconds = (float)music->totalSamples/music->stream.sampleRate; return totalSeconds; } // Get current music time played (in seconds) float GetMusicTimePlayed(Music music) { float secondsPlayed = 0.0f; unsigned int samplesPlayed = music->totalSamples - music->samplesLeft; secondsPlayed = (float)samplesPlayed/music->stream.sampleRate; return secondsPlayed; } // Init audio stream (to stream audio pcm data) AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels) { AudioStream stream = { 0 }; stream.sampleRate = sampleRate; stream.sampleSize = sampleSize; // Only mono and stereo channels are supported, more channels require AL_EXT_MCFORMATS extension if ((channels > 0) && (channels < 3)) stream.channels = channels; else { TraceLog(LOG_WARNING, "Init audio stream: Number of channels not supported: %i", channels); stream.channels = 1; // Fallback to mono channel } #if USE_MINI_AL mal_format formatIn = ((stream.sampleSize == 8) ? mal_format_u8 : ((stream.sampleSize == 16) ? mal_format_s16 : mal_format_f32)); // The size of a streaming buffer must be at least double the size of a period. unsigned int periodSize = device.bufferSizeInFrames / device.periods; unsigned int subBufferSize = AUDIO_BUFFER_SIZE; if (subBufferSize < periodSize) subBufferSize = periodSize; AudioBuffer *audioBuffer = CreateAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM); if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer"); return stream; } audioBuffer->looping = true; // Always loop for streaming buffers. stream.audioBuffer = audioBuffer; #else // Setup OpenAL format if (stream.channels == 1) { switch (sampleSize) { case 8: stream.format = AL_FORMAT_MONO8; break; case 16: stream.format = AL_FORMAT_MONO16; break; case 32: stream.format = AL_FORMAT_MONO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 default: TraceLog(LOG_WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break; } } else if (stream.channels == 2) { switch (sampleSize) { case 8: stream.format = AL_FORMAT_STEREO8; break; case 16: stream.format = AL_FORMAT_STEREO16; break; case 32: stream.format = AL_FORMAT_STEREO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 default: TraceLog(LOG_WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break; } } // Create an audio source alGenSources(1, &stream.source); alSourcef(stream.source, AL_PITCH, 1.0f); alSourcef(stream.source, AL_GAIN, 1.0f); alSource3f(stream.source, AL_POSITION, 0.0f, 0.0f, 0.0f); alSource3f(stream.source, AL_VELOCITY, 0.0f, 0.0f, 0.0f); // Create Buffers (double buffering) alGenBuffers(MAX_STREAM_BUFFERS, stream.buffers); // Initialize buffer with zeros by default // NOTE: Using dynamic allocation because it requires more than 16KB void *pcm = calloc(AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, 1); for (int i = 0; i < MAX_STREAM_BUFFERS; i++) { alBufferData(stream.buffers[i], stream.format, pcm, AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, stream.sampleRate); } free(pcm); alSourceQueueBuffers(stream.source, MAX_STREAM_BUFFERS, stream.buffers); #endif TraceLog(LOG_INFO, "[AUD ID %i] Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.source, stream.sampleRate, stream.sampleSize, (stream.channels == 1) ? "Mono" : "Stereo"); return stream; } // Close audio stream and free memory void CloseAudioStream(AudioStream stream) { #if USE_MINI_AL DeleteAudioBuffer((AudioBuffer *)stream.audioBuffer); #else // Stop playing channel alSourceStop(stream.source); // Flush out all queued buffers int queued = 0; alGetSourcei(stream.source, AL_BUFFERS_QUEUED, &queued); ALuint buffer = 0; while (queued > 0) { alSourceUnqueueBuffers(stream.source, 1, &buffer); queued--; } // Delete source and buffers alDeleteSources(1, &stream.source); alDeleteBuffers(MAX_STREAM_BUFFERS, stream.buffers); #endif TraceLog(LOG_INFO, "[AUD ID %i] Unloaded audio stream data", stream.source); } // Update audio stream buffers with data // NOTE 1: Only updates one buffer of the stream source: unqueue -> update -> queue // NOTE 2: To unqueue a buffer it needs to be processed: IsAudioBufferProcessed() void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) { #if USE_MINI_AL AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer; if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "UpdateAudioStream() : No audio buffer"); return; } if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1]) { mal_uint32 subBufferToUpdate; if (audioBuffer->isSubBufferProcessed[0] && audioBuffer->isSubBufferProcessed[1]) { // Both buffers are available for updating. Update the first one and make sure the cursor is moved back to the front. subBufferToUpdate = 0; audioBuffer->frameCursorPos = 0; } else { // Just update whichever sub-buffer is processed. subBufferToUpdate = (audioBuffer->isSubBufferProcessed[0]) ? 0 : 1; } mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2; unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate); // Does this API expect a whole buffer to be updated in one go? Assuming so, but if not will need to change this logic. if (subBufferSizeInFrames >= (mal_uint32)samplesCount) { mal_uint32 framesToWrite = subBufferSizeInFrames; if (framesToWrite > (mal_uint32)samplesCount) framesToWrite = (mal_uint32)samplesCount; mal_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8); memcpy(subBuffer, data, bytesToWrite); // Any leftover frames should be filled with zeros. mal_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite; if (leftoverFrameCount > 0) { memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8)); } audioBuffer->isSubBufferProcessed[subBufferToUpdate] = false; } else { TraceLog(LOG_ERROR, "UpdateAudioStream() : Attempting to write too many frames to buffer"); return; } } else { TraceLog(LOG_ERROR, "Audio buffer not available for updating"); return; } #else ALuint buffer = 0; alSourceUnqueueBuffers(stream.source, 1, &buffer); // Check if any buffer was available for unqueue if (alGetError() != AL_INVALID_VALUE) { alBufferData(buffer, stream.format, data, samplesCount*stream.sampleSize/8*stream.channels, stream.sampleRate); alSourceQueueBuffers(stream.source, 1, &buffer); } else TraceLog(LOG_WARNING, "[AUD ID %i] Audio buffer not available for unqueuing", stream.source); #endif } // Check if any audio stream buffers requires refill bool IsAudioBufferProcessed(AudioStream stream) { #if USE_MINI_AL AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer; if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "IsAudioBufferProcessed() : No audio buffer"); return false; } return audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1]; #else ALint processed = 0; // Determine if music stream is ready to be written alGetSourcei(stream.source, AL_BUFFERS_PROCESSED, &processed); return (processed > 0); #endif } // Play audio stream void PlayAudioStream(AudioStream stream) { #if USE_MINI_AL PlayAudioBuffer((AudioBuffer *)stream.audioBuffer); #else alSourcePlay(stream.source); #endif } // Play audio stream void PauseAudioStream(AudioStream stream) { #if USE_MINI_AL PauseAudioBuffer((AudioBuffer *)stream.audioBuffer); #else alSourcePause(stream.source); #endif } // Resume audio stream playing void ResumeAudioStream(AudioStream stream) { #if USE_MINI_AL ResumeAudioBuffer((AudioBuffer *)stream.audioBuffer); #else ALenum state; alGetSourcei(stream.source, AL_SOURCE_STATE, &state); if (state == AL_PAUSED) alSourcePlay(stream.source); #endif } // Check if audio stream is playing. bool IsAudioStreamPlaying(AudioStream stream) { #if USE_MINI_AL return IsAudioBufferPlaying((AudioBuffer *)stream.audioBuffer); #else bool playing = false; ALint state; alGetSourcei(stream.source, AL_SOURCE_STATE, &state); if (state == AL_PLAYING) playing = true; return playing; #endif } // Stop audio stream void StopAudioStream(AudioStream stream) { #if USE_MINI_AL StopAudioBuffer((AudioBuffer *)stream.audioBuffer); #else alSourceStop(stream.source); #endif } void SetAudioStreamVolume(AudioStream stream, float volume) { #if USE_MINI_AL SetAudioBufferVolume((AudioBuffer *)stream.audioBuffer, volume); #else alSourcef(stream.source, AL_GAIN, volume); #endif } void SetAudioStreamPitch(AudioStream stream, float pitch) { #if USE_MINI_AL SetAudioBufferPitch((AudioBuffer *)stream.audioBuffer, pitch); #else alSourcef(stream.source, AL_PITCH, pitch); #endif } //---------------------------------------------------------------------------------- // Module specific Functions Definition //---------------------------------------------------------------------------------- #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; WAVFormat wavFormat; WAVData wavData; Wave wave = { 0 }; FILE *wavFile; wavFile = fopen(fileName, "rb"); if (wavFile == NULL) { TraceLog(LOG_WARNING, "[%s] WAV file could not be opened", fileName); wave.data = NULL; } else { // Read in the first chunk into the struct fread(&wavRiffHeader, sizeof(WAVRiffHeader), 1, wavFile); // Check for RIFF and WAVE tags if (strncmp(wavRiffHeader.chunkID, "RIFF", 4) || strncmp(wavRiffHeader.format, "WAVE", 4)) { TraceLog(LOG_WARNING, "[%s] Invalid RIFF or WAVE Header", 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, "[%s] Invalid Wave format", 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, "[%s] Invalid data header", fileName); } else { // Allocate memory for data wave.data = malloc(wavData.subChunkSize); // Read in the sound data into the soundData variable fread(wave.data, wavData.subChunkSize, 1, wavFile); // 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, "[%s] WAV 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, "[%s] WAV 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, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); } } } fclose(wavFile); } return wave; } #endif #if defined(SUPPORT_FILEFORMAT_OGG) // Load OGG file into Wave structure // NOTE: Using stb_vorbis library static Wave LoadOGG(const char *fileName) { Wave wave = { 0 }; stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL); if (oggFile == NULL) TraceLog(LOG_WARNING, "[%s] OGG file could not be opened", fileName); else { stb_vorbis_info info = stb_vorbis_get_info(oggFile); wave.sampleRate = info.sample_rate; wave.sampleSize = 16; // 16 bit per sample (short) wave.channels = info.channels; wave.sampleCount = (int)stb_vorbis_stream_length_in_samples(oggFile); // Independent by channel float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile); if (totalSeconds > 10) TraceLog(LOG_WARNING, "[%s] Ogg audio length is larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds); wave.data = (short *)malloc(wave.sampleCount*wave.channels*sizeof(short)); // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!) int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels); TraceLog(LOG_DEBUG, "[%s] Samples obtained: %i", fileName, numSamplesOgg); TraceLog(LOG_INFO, "[%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); stb_vorbis_close(oggFile); } return wave; } #endif #if defined(SUPPORT_FILEFORMAT_FLAC) // Load FLAC file into Wave structure // NOTE: Using dr_flac library static Wave LoadFLAC(const char *fileName) { Wave wave; // Decode an entire FLAC file in one go uint64_t totalSampleCount; wave.data = drflac_open_and_decode_file_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount); wave.sampleCount = (int)totalSampleCount/wave.channels; wave.sampleSize = 16; // NOTE: Only support up to 2 channels (mono, stereo) if (wave.channels > 2) TraceLog(LOG_WARNING, "[%s] FLAC channels number (%i) not supported", fileName, wave.channels); if (wave.data == NULL) TraceLog(LOG_WARNING, "[%s] FLAC data could not be loaded", fileName); else TraceLog(LOG_INFO, "[%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); return wave; } #endif // Some required functions for audio standalone module version #if defined(AUDIO_STANDALONE) // Check file extension bool IsFileExtension(const char *fileName, const char *ext) { bool result = false; const char *fileExt; if ((fileExt = strrchr(fileName, '.')) != NULL) { if (strcmp(fileExt, ext) == 0) result = true; } return result; } // Show trace log messages (LOG_INFO, LOG_WARNING, LOG_ERROR, LOG_DEBUG) void TraceLog(int msgType, const char *text, ...) { va_list args; va_start(args, text); switch (msgType) { case LOG_INFO: fprintf(stdout, "INFO: "); break; case LOG_ERROR: fprintf(stdout, "ERROR: "); break; case LOG_WARNING: fprintf(stdout, "WARNING: "); break; case LOG_DEBUG: fprintf(stdout, "DEBUG: "); break; default: break; } vfprintf(stdout, text, args); fprintf(stdout, "\n"); va_end(args); if (msgType == LOG_ERROR) exit(1); } #endif