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Diffstat (limited to 'src/audio.c')
| -rw-r--r-- | src/audio.c | 1958 |
1 files changed, 0 insertions, 1958 deletions
diff --git a/src/audio.c b/src/audio.c deleted file mode 100644 index a76481b4..00000000 --- a/src/audio.c +++ /dev/null @@ -1,1958 +0,0 @@ -/********************************************************************************************** -* -* 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 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. -* -**********************************************************************************************/ - -#if defined(AUDIO_STANDALONE) - #include "audio.h" - #include <stdarg.h> // Required for: va_list, va_start(), vfprintf(), va_end() -#else - #include "raylib.h" // Declares module functions -// Check if config flags have been externally provided on compilation line -#if !defined(EXTERNAL_CONFIG_FLAGS) - #include "config.h" // Defines module configuration flags -#endif - #include "utils.h" // Required for: fopen() Android mapping -#endif - -#include "external/mini_al.h" // mini_al audio library - // NOTE: Cannot be implement here because it conflicts with - // Win32 APIs: Rectangle, CloseWindow(), ShowCursor(), PlaySoundA() - -#include <stdlib.h> // Required for: malloc(), free() -#include <string.h> // Required for: strcmp(), strncmp() -#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread() - -#if defined(SUPPORT_FILEFORMAT_OGG) - #define STB_VORBIS_IMPLEMENTATION - #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 - -#if defined(_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) - -//---------------------------------------------------------------------------------- -// 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 -#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 -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 -#endif -#if defined(SUPPORT_FILEFORMAT_FLAC) -static Wave LoadFLAC(const char *fileName); // Load FLAC file -#endif -#if defined(SUPPORT_FILEFORMAT_MP3) -static Wave LoadMP3(const char *fileName); // Load MP3 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 -//---------------------------------------------------------------------------------- -#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_bytes_per_sample(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; - } - - mal_uint32 framesJustRead = (mal_uint32)mal_dsp_read(&audioBuffer->dsp, framesToReadRightNow, tempBuffer, audioBuffer->dsp.pUserData); - 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_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn)*audioBuffer->dsp.formatConverterIn.config.channels; - - // 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; - } - } -} - -//---------------------------------------------------------------------------------- -// Module Functions Definition - Audio Device initialization and Closing -//---------------------------------------------------------------------------------- -// Initialize audio device -void InitAudioDevice(void) -{ - // 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; -} - -// Close the audio device for all contexts -void CloseAudioDevice(void) -{ - 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); - - TraceLog(LOG_INFO, "Audio device closed successfully"); -} - -// Check if device has been initialized successfully -bool IsAudioDeviceReady(void) -{ - return isAudioInitialized; -} - -// Set master volume (listener) -void SetMasterVolume(float volume) -{ - if (volume < 0.0f) volume = 0.0f; - else if (volume > 1.0f) volume = 1.0f; - - masterVolume = volume; -} - -//---------------------------------------------------------------------------------- -// Module Functions Definition - Audio Buffer management -//---------------------------------------------------------------------------------- - -// 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_bytes_per_sample(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; - dspConfig.onRead = OnAudioBufferDSPRead; - dspConfig.pUserData = audioBuffer; - dspConfig.allowDynamicSampleRate = MAL_TRUE; // <-- Required for pitch shifting. - mal_result resultMAL = mal_dsp_init(&dspConfig, &audioBuffer->dsp); - if (resultMAL != MAL_SUCCESS) - { - TraceLog(LOG_ERROR, "CreateAudioBuffer() : 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, "DeleteAudioBuffer() : 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, "IsAudioBufferPlaying() : 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, "StopAudioBuffer() : 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, "PauseAudioBuffer() : No audio buffer"); - return; - } - - audioBuffer->paused = true; -} - -// Resume an audio buffer -void ResumeAudioBuffer(AudioBuffer *audioBuffer) -{ - if (audioBuffer == NULL) - { - TraceLog(LOG_ERROR, "ResumeAudioBuffer() : 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, "SetAudioBufferVolume() : 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, "SetAudioBufferPitch() : 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.src.config.sampleRateOut / (float)audioBuffer->dsp.src.config.sampleRateIn) / pitch) * audioBuffer->dsp.src.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); -} - -//---------------------------------------------------------------------------------- -// 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 -#if defined(SUPPORT_FILEFORMAT_MP3) - else if (IsFileExtension(fileName, ".mp3")) wave = LoadMP3(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) - { - // 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/wave.channels; - - mal_uint32 frameCount = (mal_uint32)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_uint32)mal_convert_frames(audioBuffer->buffer, audioBuffer->dsp.formatConverterIn.config.formatIn, audioBuffer->dsp.formatConverterIn.config.channels, audioBuffer->dsp.src.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn); - if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed"); - - sound.audioBuffer = audioBuffer; - } - - 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) -{ - DeleteAudioBuffer((AudioBuffer *)sound.audioBuffer); - - 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) -{ - 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.formatConverterIn.config.channels*mal_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn)); -} - -// Export wave data to file -void ExportWave(Wave wave, const char *fileName) -{ - bool success = false; - - if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName); - else if (IsFileExtension(fileName, ".raw")) - { - // Export raw sample data (without header) - // NOTE: It's up to the user to track wave parameters - FILE *rawFile = fopen(fileName, "wb"); - success = fwrite(wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8, 1, rawFile); - fclose(rawFile); - } - - if (success) TraceLog(LOG_INFO, "Wave exported successfully: %s", fileName); - else TraceLog(LOG_WARNING, "Wave could not be exported."); -} - -// Export wave sample data to code (.h) -void ExportWaveAsCode(Wave wave, const char *fileName) -{ - #define BYTES_TEXT_PER_LINE 20 - - char varFileName[256] = { 0 }; - int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; - - FILE *txtFile = fopen(fileName, "wt"); - - 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"); - - // 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; } - - fprintf(txtFile, "// Wave data information\n"); - fprintf(txtFile, "#define %s_SAMPLE_COUNT %i\n", varFileName, wave.sampleCount); - fprintf(txtFile, "#define %s_SAMPLE_RATE %i\n", varFileName, wave.sampleRate); - fprintf(txtFile, "#define %s_SAMPLE_SIZE %i\n", varFileName, wave.sampleSize); - fprintf(txtFile, "#define %s_CHANNELS %i\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]); - - fclose(txtFile); -} - -// Play a sound -void PlaySound(Sound sound) -{ - PlayAudioBuffer((AudioBuffer *)sound.audioBuffer); -} - -// Pause a sound -void PauseSound(Sound sound) -{ - PauseAudioBuffer((AudioBuffer *)sound.audioBuffer); -} - -// Resume a paused sound -void ResumeSound(Sound sound) -{ - ResumeAudioBuffer((AudioBuffer *)sound.audioBuffer); -} - -// Stop reproducing a sound -void StopSound(Sound sound) -{ - StopAudioBuffer((AudioBuffer *)sound.audioBuffer); -} - -// Check if a sound is playing -bool IsSoundPlaying(Sound sound) -{ - return IsAudioBufferPlaying((AudioBuffer *)sound.audioBuffer); -} - -// Set volume for a sound -void SetSoundVolume(Sound sound, float volume) -{ - SetAudioBufferVolume((AudioBuffer *)sound.audioBuffer, volume); -} - -// Set pitch for a sound -void SetSoundPitch(Sound sound, float pitch) -{ - SetAudioBufferPitch((AudioBuffer *)sound.audioBuffer, pitch); -} - -// Convert wave data to desired format -void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) -{ - 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_uint32)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_uint32)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; -} - -// 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) && ((unsigned int)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 (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]; - } - } - - 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)); - bool musicLoaded = true; - - if (IsFileExtension(fileName, ".ogg")) - { - // Open ogg audio stream - music->ctxOgg = stb_vorbis_open_filename(fileName, NULL, NULL); - - if (music->ctxOgg == NULL) musicLoaded = false; - 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)*info.channels; - music->samplesLeft = music->totalSamples; - music->ctxType = MUSIC_AUDIO_OGG; - music->loopCount = -1; // Infinite loop by default - - TraceLog(LOG_DEBUG, "[%s] OGG 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) musicLoaded = false; - else - { - music->stream = InitAudioStream(music->ctxFlac->sampleRate, music->ctxFlac->bitsPerSample, music->ctxFlac->channels); - music->totalSamples = (unsigned int)music->ctxFlac->totalSampleCount; - 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")) - { - int result = drmp3_init_file(&music->ctxMp3, fileName, NULL); - - if (!result) musicLoaded = false; - else - { - TraceLog(LOG_INFO, "[%s] MP3 sample rate: %i", fileName, music->ctxMp3.sampleRate); - TraceLog(LOG_INFO, "[%s] MP3 bits per sample: %i", fileName, 32); - TraceLog(LOG_INFO, "[%s] MP3 channels: %i", fileName, music->ctxMp3.channels); - - music->stream = InitAudioStream(music->ctxMp3.sampleRate, 32, music->ctxMp3.channels); - - // TODO: There is not an easy way to compute the total number of samples available - // in an MP3, frames size could be variable... we tried with a 60 seconds music... but crashes... - music->totalSamples = drmp3_get_pcm_frame_count(&music->ctxMp3)*music->ctxMp3.channels; - music->samplesLeft = music->totalSamples; - music->ctxType = MUSIC_AUDIO_MP3; - music->loopCount = -1; // Infinite loop by default - - TraceLog(LOG_INFO, "[%s] MP3 total samples: %i", fileName, music->totalSamples); - } - } -#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_INFO, "[%s] XM number of samples: %i", fileName, music->totalSamples); - TraceLog(LOG_INFO, "[%s] XM track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f); - } - else musicLoaded = false; - } -#endif -#if defined(SUPPORT_FILEFORMAT_MOD) - else if (IsFileExtension(fileName, ".mod")) - { - jar_mod_init(&music->ctxMod); - - if (jar_mod_load_file(&music->ctxMod, fileName)) - { - // NOTE: Only stereo is supported for MOD - 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_INFO, "[%s] MOD number of samples: %i", fileName, music->samplesLeft); - TraceLog(LOG_INFO, "[%s] MOD track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f); - } - else musicLoaded = false; - } -#endif - else musicLoaded = false; - - if (!musicLoaded) - { - 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); - music = NULL; - - TraceLog(LOG_WARNING, "[%s] Music file could not be opened", fileName); - } - - return music; -} - -// Unload music stream -void UnloadMusicStream(Music music) -{ - if (music == NULL) return; - - 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 (music != NULL) - { - 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; - } -} - -// Pause music playing -void PauseMusicStream(Music music) -{ - if (music != NULL) PauseAudioStream(music->stream); -} - -// Resume music playing -void ResumeMusicStream(Music music) -{ - if (music != NULL) ResumeAudioStream(music->stream); -} - -// Stop music playing (close stream) -// TODO: To clear a buffer, make sure they have been already processed! -void StopMusicStream(Music music) -{ - if (music == NULL) return; - - StopAudioStream(music->stream); - - // Restart music context - switch (music->ctxType) - { - case MUSIC_AUDIO_OGG: stb_vorbis_seek_start(music->ctxOgg); break; -#if defined(SUPPORT_FILEFORMAT_FLAC) - case MUSIC_AUDIO_FLAC: /* TODO: Restart FLAC context */ break; -#endif -#if defined(SUPPORT_FILEFORMAT_MP3) - case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame(&music->ctxMp3, 0); 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 (music == NULL) return; - - 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.channels*music->stream.sampleSize/8, 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/music->stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music->stream.channels; - 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); - - } 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, (short *)pcm); - - } break; - #endif - #if defined(SUPPORT_FILEFORMAT_MP3) - case MUSIC_AUDIO_MP3: - { - // NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed - drmp3_read_pcm_frames_f32(&music->ctxMp3, samplesCount/music->stream.channels, (float *)pcm); - - } break; - #endif - #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(music->ctxXm, (short *)pcm, samplesCount/2); - } break; - #endif - #if defined(SUPPORT_FILEFORMAT_MOD) - case MUSIC_MODULE_MOD: - { - // NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2 - jar_mod_fillbuffer(&music->ctxMod, (short *)pcm, samplesCount/2, 0); - } break; - #endif - default: break; - } - - - UpdateAudioStream(music->stream, pcm, samplesCount); - if ((music->ctxType == MUSIC_MODULE_XM) || (music->ctxType == MUSIC_MODULE_MOD)) - { - if (samplesCount > 1) music->samplesLeft -= samplesCount/2; - else music->samplesLeft -= samplesCount; - } - else 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); - } -} - -// Check if any music is playing -bool IsMusicPlaying(Music music) -{ - if (music == NULL) return false; - else return IsAudioStreamPlaying(music->stream); -} - -// Set volume for music -void SetMusicVolume(Music music, float volume) -{ - if (music != NULL) SetAudioStreamVolume(music->stream, volume); -} - -// Set pitch for music -void SetMusicPitch(Music music, float pitch) -{ - if (music != NULL) SetAudioStreamPitch(music->stream, pitch); -} - -// Set music loop count (loop repeats) -// NOTE: If set to -1, means infinite loop -void SetMusicLoopCount(Music music, int count) -{ - if (music != NULL) music->loopCount = count; -} - -// Get music time length (in seconds) -float GetMusicTimeLength(Music music) -{ - float totalSeconds = 0.0f; - - if (music != NULL) totalSeconds = (float)music->totalSamples/(music->stream.sampleRate*music->stream.channels); - - return totalSeconds; -} - -// Get current music time played (in seconds) -float GetMusicTimePlayed(Music music) -{ - float secondsPlayed = 0.0f; - - if (music != NULL) - { - unsigned int samplesPlayed = music->totalSamples - music->samplesLeft; - secondsPlayed = (float)samplesPlayed/(music->stream.sampleRate*music->stream.channels); - } - - 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 - } - - 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; - - 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) -{ - DeleteAudioBuffer((AudioBuffer *)stream.audioBuffer); - - 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) -{ - 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/stream.channels) - { - mal_uint32 framesToWrite = subBufferSizeInFrames; - - if (framesToWrite > ((mal_uint32)samplesCount/stream.channels)) framesToWrite = (mal_uint32)samplesCount/stream.channels; - - 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; - } -} - -// Check if any audio stream buffers requires refill -bool IsAudioBufferProcessed(AudioStream stream) -{ - AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer; - if (audioBuffer == NULL) - { - TraceLog(LOG_ERROR, "IsAudioBufferProcessed() : No audio buffer"); - return false; - } - - return audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1]; -} - -// Play audio stream -void PlayAudioStream(AudioStream stream) -{ - PlayAudioBuffer((AudioBuffer *)stream.audioBuffer); -} - -// Play audio stream -void PauseAudioStream(AudioStream stream) -{ - PauseAudioBuffer((AudioBuffer *)stream.audioBuffer); -} - -// Resume audio stream playing -void ResumeAudioStream(AudioStream stream) -{ - ResumeAudioBuffer((AudioBuffer *)stream.audioBuffer); -} - -// Check if audio stream is playing. -bool IsAudioStreamPlaying(AudioStream stream) -{ - return IsAudioBufferPlaying((AudioBuffer *)stream.audioBuffer); -} - -// Stop audio stream -void StopAudioStream(AudioStream stream) -{ - StopAudioBuffer((AudioBuffer *)stream.audioBuffer); -} - -void SetAudioStreamVolume(AudioStream stream, float volume) -{ - SetAudioBufferVolume((AudioBuffer *)stream.audioBuffer, volume); -} - -void SetAudioStreamPitch(AudioStream stream, float pitch) -{ - SetAudioBufferPitch((AudioBuffer *)stream.audioBuffer, pitch); -} - -//---------------------------------------------------------------------------------- -// 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; -} - -// Save wave data as WAV file -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, "[%s] WAV audio file could not be created", 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; - - success = fwrite(&riffHeader, sizeof(RiffHeader), 1, wavFile); - success = fwrite(&waveFormat, sizeof(WaveFormat), 1, wavFile); - success = 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 - return success; -} -#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 = (unsigned int)stb_vorbis_stream_length_in_samples(oggFile)*info.channels; // 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 = (unsigned int)totalSampleCount; - 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 - -#if defined(SUPPORT_FILEFORMAT_MP3) -// Load MP3 file into Wave structure -// NOTE: Using dr_mp3 library -static Wave LoadMP3(const char *fileName) -{ - Wave wave = { 0 }; - - // Decode an entire MP3 file in one go - uint64_t totalFrameCount = 0; - drmp3_config config = { 0 }; - wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount); - - wave.channels = config.outputChannels; - wave.sampleRate = config.outputSampleRate; - 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, "[%s] MP3 channels number (%i) not supported", fileName, wave.channels); - - if (wave.data == NULL) TraceLog(LOG_WARNING, "[%s] MP3 data could not be loaded", fileName); - else TraceLog(LOG_INFO, "[%s] MP3 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 |