1 | /********************************************************************************************** |
2 | * |
3 | * raudio - A simple and easy-to-use audio library based on miniaudio |
4 | * |
5 | * FEATURES: |
6 | * - Manage audio device (init/close) |
7 | * - Manage raw audio context |
8 | * - Manage mixing channels |
9 | * - Load and unload audio files |
10 | * - Format wave data (sample rate, size, channels) |
11 | * - Play/Stop/Pause/Resume loaded audio |
12 | * |
13 | * CONFIGURATION: |
14 | * |
15 | * #define RAUDIO_STANDALONE |
16 | * Define to use the module as standalone library (independently of raylib). |
17 | * Required types and functions are defined in the same module. |
18 | * |
19 | * #define SUPPORT_FILEFORMAT_WAV |
20 | * #define SUPPORT_FILEFORMAT_OGG |
21 | * #define SUPPORT_FILEFORMAT_XM |
22 | * #define SUPPORT_FILEFORMAT_MOD |
23 | * #define SUPPORT_FILEFORMAT_FLAC |
24 | * #define SUPPORT_FILEFORMAT_MP3 |
25 | * Selected desired fileformats to be supported for loading. Some of those formats are |
26 | * supported by default, to remove support, just comment unrequired #define in this module |
27 | * |
28 | * DEPENDENCIES: |
29 | * miniaudio.h - Audio device management lib (https://github.com/dr-soft/miniaudio) |
30 | * stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/) |
31 | * dr_mp3.h - MP3 audio file loading (https://github.com/mackron/dr_libs) |
32 | * dr_flac.h - FLAC audio file loading (https://github.com/mackron/dr_libs) |
33 | * jar_xm.h - XM module file loading |
34 | * jar_mod.h - MOD audio file loading |
35 | * |
36 | * CONTRIBUTORS: |
37 | * David Reid (github: @mackron) (Nov. 2017): |
38 | * - Complete port to miniaudio library |
39 | * |
40 | * Joshua Reisenauer (github: @kd7tck) (2015) |
41 | * - XM audio module support (jar_xm) |
42 | * - MOD audio module support (jar_mod) |
43 | * - Mixing channels support |
44 | * - Raw audio context support |
45 | * |
46 | * |
47 | * LICENSE: zlib/libpng |
48 | * |
49 | * Copyright (c) 2013-2020 Ramon Santamaria (@raysan5) |
50 | * |
51 | * This software is provided "as-is", without any express or implied warranty. In no event |
52 | * will the authors be held liable for any damages arising from the use of this software. |
53 | * |
54 | * Permission is granted to anyone to use this software for any purpose, including commercial |
55 | * applications, and to alter it and redistribute it freely, subject to the following restrictions: |
56 | * |
57 | * 1. The origin of this software must not be misrepresented; you must not claim that you |
58 | * wrote the original software. If you use this software in a product, an acknowledgment |
59 | * in the product documentation would be appreciated but is not required. |
60 | * |
61 | * 2. Altered source versions must be plainly marked as such, and must not be misrepresented |
62 | * as being the original software. |
63 | * |
64 | * 3. This notice may not be removed or altered from any source distribution. |
65 | * |
66 | **********************************************************************************************/ |
67 | |
68 | #if defined(RAUDIO_STANDALONE) |
69 | #include "raudio.h" |
70 | #include <stdarg.h> // Required for: va_list, va_start(), vfprintf(), va_end() |
71 | #else |
72 | #include "raylib.h" // Declares module functions |
73 | |
74 | // Check if config flags have been externally provided on compilation line |
75 | #if !defined(EXTERNAL_CONFIG_FLAGS) |
76 | #include "config.h" // Defines module configuration flags |
77 | #endif |
78 | #include "utils.h" // Required for: fopen() Android mapping |
79 | #endif |
80 | |
81 | #if defined(_WIN32) |
82 | // To avoid conflicting windows.h symbols with raylib, some flags are defined |
83 | // WARNING: Those flags avoid inclusion of some Win32 headers that could be required |
84 | // by user at some point and won't be included... |
85 | //------------------------------------------------------------------------------------- |
86 | |
87 | // If defined, the following flags inhibit definition of the indicated items. |
88 | #define NOGDICAPMASKS // CC_*, LC_*, PC_*, CP_*, TC_*, RC_ |
89 | #define NOVIRTUALKEYCODES // VK_* |
90 | #define NOWINMESSAGES // WM_*, EM_*, LB_*, CB_* |
91 | #define NOWINSTYLES // WS_*, CS_*, ES_*, LBS_*, SBS_*, CBS_* |
92 | #define NOSYSMETRICS // SM_* |
93 | #define NOMENUS // MF_* |
94 | #define NOICONS // IDI_* |
95 | #define NOKEYSTATES // MK_* |
96 | #define NOSYSCOMMANDS // SC_* |
97 | #define NORASTEROPS // Binary and Tertiary raster ops |
98 | #define NOSHOWWINDOW // SW_* |
99 | #define OEMRESOURCE // OEM Resource values |
100 | #define NOATOM // Atom Manager routines |
101 | #define NOCLIPBOARD // Clipboard routines |
102 | #define NOCOLOR // Screen colors |
103 | #define NOCTLMGR // Control and Dialog routines |
104 | #define NODRAWTEXT // DrawText() and DT_* |
105 | #define NOGDI // All GDI defines and routines |
106 | #define NOKERNEL // All KERNEL defines and routines |
107 | #define NOUSER // All USER defines and routines |
108 | //#define NONLS // All NLS defines and routines |
109 | #define NOMB // MB_* and MessageBox() |
110 | #define NOMEMMGR // GMEM_*, LMEM_*, GHND, LHND, associated routines |
111 | #define NOMETAFILE // typedef METAFILEPICT |
112 | #define NOMINMAX // Macros min(a,b) and max(a,b) |
113 | #define NOMSG // typedef MSG and associated routines |
114 | #define NOOPENFILE // OpenFile(), OemToAnsi, AnsiToOem, and OF_* |
115 | #define NOSCROLL // SB_* and scrolling routines |
116 | #define NOSERVICE // All Service Controller routines, SERVICE_ equates, etc. |
117 | #define NOSOUND // Sound driver routines |
118 | #define NOTEXTMETRIC // typedef TEXTMETRIC and associated routines |
119 | #define NOWH // SetWindowsHook and WH_* |
120 | #define NOWINOFFSETS // GWL_*, GCL_*, associated routines |
121 | #define NOCOMM // COMM driver routines |
122 | #define NOKANJI // Kanji support stuff. |
123 | #define NOHELP // Help engine interface. |
124 | #define NOPROFILER // Profiler interface. |
125 | #define NODEFERWINDOWPOS // DeferWindowPos routines |
126 | #define NOMCX // Modem Configuration Extensions |
127 | |
128 | // Type required before windows.h inclusion |
129 | typedef struct tagMSG *LPMSG; |
130 | |
131 | #include <windows.h> |
132 | |
133 | // Type required by some unused function... |
134 | typedef struct tagBITMAPINFOHEADER { |
135 | DWORD biSize; |
136 | LONG biWidth; |
137 | LONG biHeight; |
138 | WORD biPlanes; |
139 | WORD biBitCount; |
140 | DWORD biCompression; |
141 | DWORD biSizeImage; |
142 | LONG biXPelsPerMeter; |
143 | LONG biYPelsPerMeter; |
144 | DWORD biClrUsed; |
145 | DWORD biClrImportant; |
146 | } BITMAPINFOHEADER, *PBITMAPINFOHEADER; |
147 | |
148 | #include <objbase.h> |
149 | #include <mmreg.h> |
150 | #include <mmsystem.h> |
151 | |
152 | // Some required types defined for MSVC/TinyC compiler |
153 | #if defined(_MSC_VER) || defined(__TINYC__) |
154 | #include "propidl.h" |
155 | #endif |
156 | #endif |
157 | |
158 | #define MA_MALLOC RL_MALLOC |
159 | #define MA_FREE RL_FREE |
160 | |
161 | #define MA_NO_JACK |
162 | #define MINIAUDIO_IMPLEMENTATION |
163 | #include "external/miniaudio.h" // miniaudio library |
164 | #undef PlaySound // Win32 API: windows.h > mmsystem.h defines PlaySound macro |
165 | |
166 | #include <stdlib.h> // Required for: malloc(), free() |
167 | #include <stdio.h> // Required for: FILE, fopen(), fclose(), fread() |
168 | |
169 | #if defined(RAUDIO_STANDALONE) |
170 | #include <string.h> // Required for: strcmp() [Used in IsFileExtension()] |
171 | |
172 | #if !defined(TRACELOG) |
173 | #define TRACELOG(level, ...) (void)0 |
174 | #endif |
175 | #endif |
176 | |
177 | #if defined(SUPPORT_FILEFORMAT_OGG) |
178 | // TODO: Remap malloc()/free() calls to RL_MALLOC/RL_FREE |
179 | |
180 | #define STB_VORBIS_IMPLEMENTATION |
181 | #include "external/stb_vorbis.h" // OGG loading functions |
182 | #endif |
183 | |
184 | #if defined(SUPPORT_FILEFORMAT_XM) |
185 | #define JARXM_MALLOC RL_MALLOC |
186 | #define JARXM_FREE RL_FREE |
187 | |
188 | #define JAR_XM_IMPLEMENTATION |
189 | #include "external/jar_xm.h" // XM loading functions |
190 | #endif |
191 | |
192 | #if defined(SUPPORT_FILEFORMAT_MOD) |
193 | #define JARMOD_MALLOC RL_MALLOC |
194 | #define JARMOD_FREE RL_FREE |
195 | |
196 | #define JAR_MOD_IMPLEMENTATION |
197 | #include "external/jar_mod.h" // MOD loading functions |
198 | #endif |
199 | |
200 | #if defined(SUPPORT_FILEFORMAT_FLAC) |
201 | #define DRFLAC_MALLOC RL_MALLOC |
202 | #define DRFLAC_REALLOC RL_REALLOC |
203 | #define DRFLAC_FREE RL_FREE |
204 | |
205 | #define DR_FLAC_IMPLEMENTATION |
206 | #define DR_FLAC_NO_WIN32_IO |
207 | #include "external/dr_flac.h" // FLAC loading functions |
208 | #endif |
209 | |
210 | #if defined(SUPPORT_FILEFORMAT_MP3) |
211 | #define DRMP3_MALLOC RL_MALLOC |
212 | #define DRMP3_REALLOC RL_REALLOC |
213 | #define DRMP3_FREE RL_FREE |
214 | |
215 | #define DR_MP3_IMPLEMENTATION |
216 | #include "external/dr_mp3.h" // MP3 loading functions |
217 | #endif |
218 | |
219 | #if defined(_MSC_VER) |
220 | #undef bool |
221 | #endif |
222 | |
223 | //---------------------------------------------------------------------------------- |
224 | // Defines and Macros |
225 | //---------------------------------------------------------------------------------- |
226 | #define AUDIO_DEVICE_FORMAT ma_format_f32 |
227 | #define AUDIO_DEVICE_CHANNELS 2 |
228 | #define AUDIO_DEVICE_SAMPLE_RATE 44100 |
229 | |
230 | #define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 |
231 | |
232 | //---------------------------------------------------------------------------------- |
233 | // Types and Structures Definition |
234 | //---------------------------------------------------------------------------------- |
235 | |
236 | // Music context type |
237 | // NOTE: Depends on data structure provided by the library |
238 | // in charge of reading the different file types |
239 | typedef enum { |
240 | MUSIC_AUDIO_WAV = 0, |
241 | MUSIC_AUDIO_OGG, |
242 | MUSIC_AUDIO_FLAC, |
243 | MUSIC_AUDIO_MP3, |
244 | MUSIC_MODULE_XM, |
245 | MUSIC_MODULE_MOD |
246 | } MusicContextType; |
247 | |
248 | #if defined(RAUDIO_STANDALONE) |
249 | typedef enum { |
250 | LOG_ALL, |
251 | LOG_TRACE, |
252 | LOG_DEBUG, |
253 | LOG_INFO, |
254 | LOG_WARNING, |
255 | LOG_ERROR, |
256 | LOG_FATAL, |
257 | LOG_NONE |
258 | } TraceLogType; |
259 | #endif |
260 | |
261 | // NOTE: Different logic is used when feeding data to the playback device |
262 | // depending on whether or not data is streamed (Music vs Sound) |
263 | typedef enum { |
264 | AUDIO_BUFFER_USAGE_STATIC = 0, |
265 | AUDIO_BUFFER_USAGE_STREAM |
266 | } AudioBufferUsage; |
267 | |
268 | // Audio buffer structure |
269 | struct rAudioBuffer { |
270 | ma_data_converter converter; // Audio data converter |
271 | |
272 | float volume; // Audio buffer volume |
273 | float pitch; // Audio buffer pitch |
274 | |
275 | bool playing; // Audio buffer state: AUDIO_PLAYING |
276 | bool paused; // Audio buffer state: AUDIO_PAUSED |
277 | bool looping; // Audio buffer looping, always true for AudioStreams |
278 | int usage; // Audio buffer usage mode: STATIC or STREAM |
279 | |
280 | bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer) |
281 | unsigned int sizeInFrames; // Total buffer size in frames |
282 | unsigned int frameCursorPos; // Frame cursor position |
283 | unsigned int totalFramesProcessed; // Total frames processed in this buffer (required for play timing) |
284 | |
285 | unsigned char *data; // Data buffer, on music stream keeps filling |
286 | |
287 | rAudioBuffer *next; // Next audio buffer on the list |
288 | rAudioBuffer *prev; // Previous audio buffer on the list |
289 | }; |
290 | |
291 | #define AudioBuffer rAudioBuffer // HACK: To avoid CoreAudio (macOS) symbol collision |
292 | |
293 | // Audio data context |
294 | typedef struct AudioData { |
295 | struct { |
296 | ma_context context; // miniaudio context data |
297 | ma_device device; // miniaudio device |
298 | ma_mutex lock; // miniaudio mutex lock |
299 | bool isReady; // Check if audio device is ready |
300 | } System; |
301 | struct { |
302 | AudioBuffer *first; // Pointer to first AudioBuffer in the list |
303 | AudioBuffer *last; // Pointer to last AudioBuffer in the list |
304 | int defaultSize; // Default audio buffer size for audio streams |
305 | } Buffer; |
306 | struct { |
307 | AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // Multichannel AudioBuffer pointers pool |
308 | unsigned int poolCounter; // AudioBuffer pointers pool counter |
309 | unsigned int channels[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // AudioBuffer pool channels |
310 | } MultiChannel; |
311 | } AudioData; |
312 | |
313 | //---------------------------------------------------------------------------------- |
314 | // Global Variables Definition |
315 | //---------------------------------------------------------------------------------- |
316 | static AudioData AUDIO = { // Global AUDIO context |
317 | |
318 | // NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number |
319 | // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a |
320 | // standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough |
321 | // In case of music-stalls, just increase this number |
322 | .Buffer.defaultSize = 4096 |
323 | }; |
324 | |
325 | //---------------------------------------------------------------------------------- |
326 | // Module specific Functions Declaration |
327 | //---------------------------------------------------------------------------------- |
328 | static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message); |
329 | static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount); |
330 | static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume); |
331 | |
332 | static void InitAudioBufferPool(void); // Initialise the multichannel buffer pool |
333 | static void CloseAudioBufferPool(void); // Close the audio buffers pool |
334 | |
335 | #if defined(SUPPORT_FILEFORMAT_WAV) |
336 | static Wave LoadWAV(const char *fileName); // Load WAV file |
337 | static int SaveWAV(Wave wave, const char *fileName); // Save wave data as WAV file |
338 | #endif |
339 | #if defined(SUPPORT_FILEFORMAT_OGG) |
340 | static Wave LoadOGG(const char *fileName); // Load OGG file |
341 | #endif |
342 | #if defined(SUPPORT_FILEFORMAT_FLAC) |
343 | static Wave LoadFLAC(const char *fileName); // Load FLAC file |
344 | #endif |
345 | #if defined(SUPPORT_FILEFORMAT_MP3) |
346 | static Wave LoadMP3(const char *fileName); // Load MP3 file |
347 | #endif |
348 | |
349 | #if defined(RAUDIO_STANDALONE) |
350 | bool IsFileExtension(const char *fileName, const char *ext);// Check file extension |
351 | void TraceLog(int msgType, const char *text, ...); // Show trace log messages (LOG_INFO, LOG_WARNING, LOG_ERROR, LOG_DEBUG) |
352 | #endif |
353 | |
354 | //---------------------------------------------------------------------------------- |
355 | // AudioBuffer management functions declaration |
356 | // NOTE: Those functions are not exposed by raylib... for the moment |
357 | //---------------------------------------------------------------------------------- |
358 | AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage); |
359 | void UnloadAudioBuffer(AudioBuffer *buffer); |
360 | |
361 | bool IsAudioBufferPlaying(AudioBuffer *buffer); |
362 | void PlayAudioBuffer(AudioBuffer *buffer); |
363 | void StopAudioBuffer(AudioBuffer *buffer); |
364 | void PauseAudioBuffer(AudioBuffer *buffer); |
365 | void ResumeAudioBuffer(AudioBuffer *buffer); |
366 | void SetAudioBufferVolume(AudioBuffer *buffer, float volume); |
367 | void SetAudioBufferPitch(AudioBuffer *buffer, float pitch); |
368 | void TrackAudioBuffer(AudioBuffer *buffer); |
369 | void UntrackAudioBuffer(AudioBuffer *buffer); |
370 | |
371 | //---------------------------------------------------------------------------------- |
372 | // Module Functions Definition - Audio Device initialization and Closing |
373 | //---------------------------------------------------------------------------------- |
374 | // Initialize audio device |
375 | void InitAudioDevice(void) |
376 | { |
377 | // TODO: Load AUDIO context memory dynamically? |
378 | |
379 | // Init audio context |
380 | ma_context_config ctxConfig = ma_context_config_init(); |
381 | ctxConfig.logCallback = OnLog; |
382 | |
383 | ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context); |
384 | if (result != MA_SUCCESS) |
385 | { |
386 | TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize context" ); |
387 | return; |
388 | } |
389 | |
390 | // Init audio device |
391 | // NOTE: Using the default device. Format is floating point because it simplifies mixing. |
392 | ma_device_config config = ma_device_config_init(ma_device_type_playback); |
393 | config.playback.pDeviceID = NULL; // NULL for the default playback AUDIO.System.device. |
394 | config.playback.format = AUDIO_DEVICE_FORMAT; |
395 | config.playback.channels = AUDIO_DEVICE_CHANNELS; |
396 | config.capture.pDeviceID = NULL; // NULL for the default capture AUDIO.System.device. |
397 | config.capture.format = ma_format_s16; |
398 | config.capture.channels = 1; |
399 | config.sampleRate = AUDIO_DEVICE_SAMPLE_RATE; |
400 | config.dataCallback = OnSendAudioDataToDevice; |
401 | config.pUserData = NULL; |
402 | |
403 | result = ma_device_init(&AUDIO.System.context, &config, &AUDIO.System.device); |
404 | if (result != MA_SUCCESS) |
405 | { |
406 | TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize playback device" ); |
407 | ma_context_uninit(&AUDIO.System.context); |
408 | return; |
409 | } |
410 | |
411 | // Keep the device running the whole time. May want to consider doing something a bit smarter and only have the device running |
412 | // while there's at least one sound being played. |
413 | result = ma_device_start(&AUDIO.System.device); |
414 | if (result != MA_SUCCESS) |
415 | { |
416 | TRACELOG(LOG_ERROR, "AUDIO: Failed to start playback device" ); |
417 | ma_device_uninit(&AUDIO.System.device); |
418 | ma_context_uninit(&AUDIO.System.context); |
419 | return; |
420 | } |
421 | |
422 | // Mixing happens on a seperate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may |
423 | // want to look at something a bit smarter later on to keep everything real-time, if that's necessary. |
424 | if (ma_mutex_init(&AUDIO.System.context, &AUDIO.System.lock) != MA_SUCCESS) |
425 | { |
426 | TRACELOG(LOG_ERROR, "AUDIO: Failed to create mutex for mixing" ); |
427 | ma_device_uninit(&AUDIO.System.device); |
428 | ma_context_uninit(&AUDIO.System.context); |
429 | return; |
430 | } |
431 | |
432 | TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully" ); |
433 | TRACELOG(LOG_INFO, " > Backend: miniaudio / %s" , ma_get_backend_name(AUDIO.System.context.backend)); |
434 | TRACELOG(LOG_INFO, " > Format: %s -> %s" , ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat)); |
435 | TRACELOG(LOG_INFO, " > Channels: %d -> %d" , AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels); |
436 | TRACELOG(LOG_INFO, " > Sample rate: %d -> %d" , AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate); |
437 | TRACELOG(LOG_INFO, " > Periods size: %d" , AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods); |
438 | |
439 | InitAudioBufferPool(); |
440 | |
441 | AUDIO.System.isReady = true; |
442 | } |
443 | |
444 | // Close the audio device for all contexts |
445 | void CloseAudioDevice(void) |
446 | { |
447 | if (AUDIO.System.isReady) |
448 | { |
449 | ma_mutex_uninit(&AUDIO.System.lock); |
450 | ma_device_uninit(&AUDIO.System.device); |
451 | ma_context_uninit(&AUDIO.System.context); |
452 | |
453 | CloseAudioBufferPool(); |
454 | |
455 | TRACELOG(LOG_INFO, "AUDIO: Device closed successfully" ); |
456 | } |
457 | else TRACELOG(LOG_WARNING, "AUDIO: Device could not be closed, not currently initialized" ); |
458 | } |
459 | |
460 | // Check if device has been initialized successfully |
461 | bool IsAudioDeviceReady(void) |
462 | { |
463 | return AUDIO.System.isReady; |
464 | } |
465 | |
466 | // Set master volume (listener) |
467 | void SetMasterVolume(float volume) |
468 | { |
469 | ma_device_set_master_volume(&AUDIO.System.device, volume); |
470 | } |
471 | |
472 | //---------------------------------------------------------------------------------- |
473 | // Module Functions Definition - Audio Buffer management |
474 | //---------------------------------------------------------------------------------- |
475 | |
476 | // Initialize a new audio buffer (filled with silence) |
477 | AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage) |
478 | { |
479 | AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer)); |
480 | |
481 | if (audioBuffer == NULL) |
482 | { |
483 | TRACELOG(LOG_ERROR, "AUDIO: Failed to allocate memory for buffer" ); |
484 | return NULL; |
485 | } |
486 | |
487 | audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1); |
488 | |
489 | // Audio data runs through a format converter |
490 | ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO_DEVICE_SAMPLE_RATE); |
491 | converterConfig.resampling.allowDynamicSampleRate = true; // Required for pitch shifting |
492 | |
493 | ma_result result = ma_data_converter_init(&converterConfig, &audioBuffer->converter); |
494 | |
495 | if (result != MA_SUCCESS) |
496 | { |
497 | TRACELOG(LOG_ERROR, "AUDIO: Failed to create data conversion pipeline" ); |
498 | RL_FREE(audioBuffer); |
499 | return NULL; |
500 | } |
501 | |
502 | // Init audio buffer values |
503 | audioBuffer->volume = 1.0f; |
504 | audioBuffer->pitch = 1.0f; |
505 | audioBuffer->playing = false; |
506 | audioBuffer->paused = false; |
507 | audioBuffer->looping = false; |
508 | audioBuffer->usage = usage; |
509 | audioBuffer->frameCursorPos = 0; |
510 | audioBuffer->sizeInFrames = sizeInFrames; |
511 | |
512 | // Buffers should be marked as processed by default so that a call to |
513 | // UpdateAudioStream() immediately after initialization works correctly |
514 | audioBuffer->isSubBufferProcessed[0] = true; |
515 | audioBuffer->isSubBufferProcessed[1] = true; |
516 | |
517 | // Track audio buffer to linked list next position |
518 | TrackAudioBuffer(audioBuffer); |
519 | |
520 | return audioBuffer; |
521 | } |
522 | |
523 | // Delete an audio buffer |
524 | void UnloadAudioBuffer(AudioBuffer *buffer) |
525 | { |
526 | if (buffer != NULL) |
527 | { |
528 | ma_data_converter_uninit(&buffer->converter); |
529 | UntrackAudioBuffer(buffer); |
530 | RL_FREE(buffer->data); |
531 | RL_FREE(buffer); |
532 | } |
533 | } |
534 | |
535 | // Check if an audio buffer is playing |
536 | bool IsAudioBufferPlaying(AudioBuffer *buffer) |
537 | { |
538 | bool result = false; |
539 | |
540 | if (buffer != NULL) result = (buffer->playing && !buffer->paused); |
541 | |
542 | return result; |
543 | } |
544 | |
545 | // Play an audio buffer |
546 | // NOTE: Buffer is restarted to the start. |
547 | // Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained. |
548 | void PlayAudioBuffer(AudioBuffer *buffer) |
549 | { |
550 | if (buffer != NULL) |
551 | { |
552 | buffer->playing = true; |
553 | buffer->paused = false; |
554 | buffer->frameCursorPos = 0; |
555 | } |
556 | } |
557 | |
558 | // Stop an audio buffer |
559 | void StopAudioBuffer(AudioBuffer *buffer) |
560 | { |
561 | if (buffer != NULL) |
562 | { |
563 | if (IsAudioBufferPlaying(buffer)) |
564 | { |
565 | buffer->playing = false; |
566 | buffer->paused = false; |
567 | buffer->frameCursorPos = 0; |
568 | buffer->totalFramesProcessed = 0; |
569 | buffer->isSubBufferProcessed[0] = true; |
570 | buffer->isSubBufferProcessed[1] = true; |
571 | } |
572 | } |
573 | } |
574 | |
575 | // Pause an audio buffer |
576 | void PauseAudioBuffer(AudioBuffer *buffer) |
577 | { |
578 | if (buffer != NULL) buffer->paused = true; |
579 | } |
580 | |
581 | // Resume an audio buffer |
582 | void ResumeAudioBuffer(AudioBuffer *buffer) |
583 | { |
584 | if (buffer != NULL) buffer->paused = false; |
585 | } |
586 | |
587 | // Set volume for an audio buffer |
588 | void SetAudioBufferVolume(AudioBuffer *buffer, float volume) |
589 | { |
590 | if (buffer != NULL) buffer->volume = volume; |
591 | } |
592 | |
593 | // Set pitch for an audio buffer |
594 | void SetAudioBufferPitch(AudioBuffer *buffer, float pitch) |
595 | { |
596 | if (buffer != NULL) |
597 | { |
598 | float pitchMul = pitch/buffer->pitch; |
599 | |
600 | // Pitching is just an adjustment of the sample rate. |
601 | // Note that this changes the duration of the sound: |
602 | // - higher pitches will make the sound faster |
603 | // - lower pitches make it slower |
604 | ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->converter.config.sampleRateOut/pitchMul); |
605 | buffer->pitch *= (float)buffer->converter.config.sampleRateOut/newOutputSampleRate; |
606 | |
607 | ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, newOutputSampleRate); |
608 | } |
609 | } |
610 | |
611 | // Track audio buffer to linked list next position |
612 | void TrackAudioBuffer(AudioBuffer *buffer) |
613 | { |
614 | ma_mutex_lock(&AUDIO.System.lock); |
615 | { |
616 | if (AUDIO.Buffer.first == NULL) AUDIO.Buffer.first = buffer; |
617 | else |
618 | { |
619 | AUDIO.Buffer.last->next = buffer; |
620 | buffer->prev = AUDIO.Buffer.last; |
621 | } |
622 | |
623 | AUDIO.Buffer.last = buffer; |
624 | } |
625 | ma_mutex_unlock(&AUDIO.System.lock); |
626 | } |
627 | |
628 | // Untrack audio buffer from linked list |
629 | void UntrackAudioBuffer(AudioBuffer *buffer) |
630 | { |
631 | ma_mutex_lock(&AUDIO.System.lock); |
632 | { |
633 | if (buffer->prev == NULL) AUDIO.Buffer.first = buffer->next; |
634 | else buffer->prev->next = buffer->next; |
635 | |
636 | if (buffer->next == NULL) AUDIO.Buffer.last = buffer->prev; |
637 | else buffer->next->prev = buffer->prev; |
638 | |
639 | buffer->prev = NULL; |
640 | buffer->next = NULL; |
641 | } |
642 | ma_mutex_unlock(&AUDIO.System.lock); |
643 | } |
644 | |
645 | //---------------------------------------------------------------------------------- |
646 | // Module Functions Definition - Sounds loading and playing (.WAV) |
647 | //---------------------------------------------------------------------------------- |
648 | |
649 | // Load wave data from file |
650 | Wave LoadWave(const char *fileName) |
651 | { |
652 | Wave wave = { 0 }; |
653 | |
654 | if (false) { } |
655 | #if defined(SUPPORT_FILEFORMAT_WAV) |
656 | else if (IsFileExtension(fileName, ".wav" )) wave = LoadWAV(fileName); |
657 | #endif |
658 | #if defined(SUPPORT_FILEFORMAT_OGG) |
659 | else if (IsFileExtension(fileName, ".ogg" )) wave = LoadOGG(fileName); |
660 | #endif |
661 | #if defined(SUPPORT_FILEFORMAT_FLAC) |
662 | else if (IsFileExtension(fileName, ".flac" )) wave = LoadFLAC(fileName); |
663 | #endif |
664 | #if defined(SUPPORT_FILEFORMAT_MP3) |
665 | else if (IsFileExtension(fileName, ".mp3" )) wave = LoadMP3(fileName); |
666 | #endif |
667 | else TRACELOG(LOG_WARNING, "FILEIO: [%s] File format not supported" , fileName); |
668 | |
669 | return wave; |
670 | } |
671 | |
672 | // Load sound from file |
673 | // NOTE: The entire file is loaded to memory to be played (no-streaming) |
674 | Sound LoadSound(const char *fileName) |
675 | { |
676 | Wave wave = LoadWave(fileName); |
677 | |
678 | Sound sound = LoadSoundFromWave(wave); |
679 | |
680 | UnloadWave(wave); // Sound is loaded, we can unload wave |
681 | |
682 | return sound; |
683 | } |
684 | |
685 | // Load sound from wave data |
686 | // NOTE: Wave data must be unallocated manually |
687 | Sound LoadSoundFromWave(Wave wave) |
688 | { |
689 | Sound sound = { 0 }; |
690 | |
691 | if (wave.data != NULL) |
692 | { |
693 | // When using miniaudio we need to do our own mixing. |
694 | // To simplify this we need convert the format of each sound to be consistent with |
695 | // the format used to open the playback AUDIO.System.device. We can do this two ways: |
696 | // |
697 | // 1) Convert the whole sound in one go at load time (here). |
698 | // 2) Convert the audio data in chunks at mixing time. |
699 | // |
700 | // First option has been selected, format conversion is done on the loading stage. |
701 | // The downside is that it uses more memory if the original sound is u8 or s16. |
702 | ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32)); |
703 | ma_uint32 frameCountIn = wave.sampleCount/wave.channels; |
704 | |
705 | ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate); |
706 | if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion" ); |
707 | |
708 | AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC); |
709 | if (audioBuffer == NULL) TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer" ); |
710 | |
711 | frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate); |
712 | if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion" ); |
713 | |
714 | sound.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS; |
715 | sound.stream.sampleRate = AUDIO_DEVICE_SAMPLE_RATE; |
716 | sound.stream.sampleSize = 32; |
717 | sound.stream.channels = AUDIO_DEVICE_CHANNELS; |
718 | sound.stream.buffer = audioBuffer; |
719 | } |
720 | |
721 | return sound; |
722 | } |
723 | |
724 | // Unload wave data |
725 | void UnloadWave(Wave wave) |
726 | { |
727 | if (wave.data != NULL) RL_FREE(wave.data); |
728 | |
729 | TRACELOG(LOG_INFO, "WAVE: Unloaded wave data from RAM" ); |
730 | } |
731 | |
732 | // Unload sound |
733 | void UnloadSound(Sound sound) |
734 | { |
735 | UnloadAudioBuffer(sound.stream.buffer); |
736 | |
737 | TRACELOG(LOG_INFO, "WAVE: Unloaded sound data from RAM" ); |
738 | } |
739 | |
740 | // Update sound buffer with new data |
741 | void UpdateSound(Sound sound, const void *data, int samplesCount) |
742 | { |
743 | if (sound.stream.buffer != NULL) |
744 | { |
745 | StopAudioBuffer(sound.stream.buffer); |
746 | |
747 | // TODO: May want to lock/unlock this since this data buffer is read at mixing time |
748 | memcpy(sound.stream.buffer->data, data, samplesCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.config.formatIn, sound.stream.buffer->converter.config.channelsIn)); |
749 | } |
750 | } |
751 | |
752 | // Export wave data to file |
753 | void ExportWave(Wave wave, const char *fileName) |
754 | { |
755 | bool success = false; |
756 | |
757 | if (false) { } |
758 | #if defined(SUPPORT_FILEFORMAT_WAV) |
759 | else if (IsFileExtension(fileName, ".wav" )) success = SaveWAV(wave, fileName); |
760 | #endif |
761 | else if (IsFileExtension(fileName, ".raw" )) |
762 | { |
763 | // Export raw sample data (without header) |
764 | // NOTE: It's up to the user to track wave parameters |
765 | SaveFileData(fileName, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8); |
766 | success = true; |
767 | } |
768 | |
769 | if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully" , fileName); |
770 | else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave data" , fileName); |
771 | } |
772 | |
773 | // Export wave sample data to code (.h) |
774 | void ExportWaveAsCode(Wave wave, const char *fileName) |
775 | { |
776 | #define BYTES_TEXT_PER_LINE 20 |
777 | |
778 | char varFileName[256] = { 0 }; |
779 | int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; |
780 | |
781 | FILE *txtFile = fopen(fileName, "wt" ); |
782 | |
783 | if (txtFile != NULL) |
784 | { |
785 | fprintf(txtFile, "\n//////////////////////////////////////////////////////////////////////////////////\n" ); |
786 | fprintf(txtFile, "// //\n" ); |
787 | fprintf(txtFile, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes //\n" ); |
788 | fprintf(txtFile, "// //\n" ); |
789 | fprintf(txtFile, "// more info and bugs-report: github.com/raysan5/raylib //\n" ); |
790 | fprintf(txtFile, "// feedback and support: ray[at]raylib.com //\n" ); |
791 | fprintf(txtFile, "// //\n" ); |
792 | fprintf(txtFile, "// Copyright (c) 2018 Ramon Santamaria (@raysan5) //\n" ); |
793 | fprintf(txtFile, "// //\n" ); |
794 | fprintf(txtFile, "//////////////////////////////////////////////////////////////////////////////////\n\n" ); |
795 | |
796 | #if !defined(RAUDIO_STANDALONE) |
797 | // Get file name from path and convert variable name to uppercase |
798 | strcpy(varFileName, GetFileNameWithoutExt(fileName)); |
799 | for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; } |
800 | #else |
801 | strcpy(varFileName, fileName); |
802 | #endif |
803 | |
804 | fprintf(txtFile, "// Wave data information\n" ); |
805 | fprintf(txtFile, "#define %s_SAMPLE_COUNT %u\n" , varFileName, wave.sampleCount); |
806 | fprintf(txtFile, "#define %s_SAMPLE_RATE %u\n" , varFileName, wave.sampleRate); |
807 | fprintf(txtFile, "#define %s_SAMPLE_SIZE %u\n" , varFileName, wave.sampleSize); |
808 | fprintf(txtFile, "#define %s_CHANNELS %u\n\n" , varFileName, wave.channels); |
809 | |
810 | // Write byte data as hexadecimal text |
811 | fprintf(txtFile, "static unsigned char %s_DATA[%i] = { " , varFileName, dataSize); |
812 | 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]); |
813 | fprintf(txtFile, "0x%x };\n" , ((unsigned char *)wave.data)[dataSize - 1]); |
814 | |
815 | fclose(txtFile); |
816 | } |
817 | } |
818 | |
819 | // Play a sound |
820 | void PlaySound(Sound sound) |
821 | { |
822 | PlayAudioBuffer(sound.stream.buffer); |
823 | } |
824 | |
825 | // Play a sound in the multichannel buffer pool |
826 | void PlaySoundMulti(Sound sound) |
827 | { |
828 | int index = -1; |
829 | unsigned int oldAge = 0; |
830 | int oldIndex = -1; |
831 | |
832 | // find the first non playing pool entry |
833 | for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) |
834 | { |
835 | if (AUDIO.MultiChannel.channels[i] > oldAge) |
836 | { |
837 | oldAge = AUDIO.MultiChannel.channels[i]; |
838 | oldIndex = i; |
839 | } |
840 | |
841 | if (!IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i])) |
842 | { |
843 | index = i; |
844 | break; |
845 | } |
846 | } |
847 | |
848 | // If no none playing pool members can be index choose the oldest |
849 | if (index == -1) |
850 | { |
851 | TRACELOG(LOG_WARNING, "SOUND: Buffer pool is already full, count: %i" , AUDIO.MultiChannel.poolCounter); |
852 | |
853 | if (oldIndex == -1) |
854 | { |
855 | // Shouldn't be able to get here... but just in case something odd happens! |
856 | TRACELOG(LOG_WARNING, "SOUND: Buffer pool could not determine oldest buffer not playing sound" ); |
857 | return; |
858 | } |
859 | |
860 | index = oldIndex; |
861 | |
862 | // Just in case... |
863 | StopAudioBuffer(AUDIO.MultiChannel.pool[index]); |
864 | } |
865 | |
866 | // Experimentally mutex lock doesn't seem to be needed this makes sense |
867 | // as pool[index] isn't playing and the only stuff we're copying |
868 | // shouldn't be changing... |
869 | |
870 | AUDIO.MultiChannel.channels[index] = AUDIO.MultiChannel.poolCounter; |
871 | AUDIO.MultiChannel.poolCounter++; |
872 | |
873 | AUDIO.MultiChannel.pool[index]->volume = sound.stream.buffer->volume; |
874 | AUDIO.MultiChannel.pool[index]->pitch = sound.stream.buffer->pitch; |
875 | AUDIO.MultiChannel.pool[index]->looping = sound.stream.buffer->looping; |
876 | AUDIO.MultiChannel.pool[index]->usage = sound.stream.buffer->usage; |
877 | AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[0] = false; |
878 | AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[1] = false; |
879 | AUDIO.MultiChannel.pool[index]->sizeInFrames = sound.stream.buffer->sizeInFrames; |
880 | AUDIO.MultiChannel.pool[index]->data = sound.stream.buffer->data; |
881 | |
882 | PlayAudioBuffer(AUDIO.MultiChannel.pool[index]); |
883 | } |
884 | |
885 | // Stop any sound played with PlaySoundMulti() |
886 | void StopSoundMulti(void) |
887 | { |
888 | for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) StopAudioBuffer(AUDIO.MultiChannel.pool[i]); |
889 | } |
890 | |
891 | // Get number of sounds playing in the multichannel buffer pool |
892 | int GetSoundsPlaying(void) |
893 | { |
894 | int counter = 0; |
895 | |
896 | for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) |
897 | { |
898 | if (IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i])) counter++; |
899 | } |
900 | |
901 | return counter; |
902 | } |
903 | |
904 | // Pause a sound |
905 | void PauseSound(Sound sound) |
906 | { |
907 | PauseAudioBuffer(sound.stream.buffer); |
908 | } |
909 | |
910 | // Resume a paused sound |
911 | void ResumeSound(Sound sound) |
912 | { |
913 | ResumeAudioBuffer(sound.stream.buffer); |
914 | } |
915 | |
916 | // Stop reproducing a sound |
917 | void StopSound(Sound sound) |
918 | { |
919 | StopAudioBuffer(sound.stream.buffer); |
920 | } |
921 | |
922 | // Check if a sound is playing |
923 | bool IsSoundPlaying(Sound sound) |
924 | { |
925 | return IsAudioBufferPlaying(sound.stream.buffer); |
926 | } |
927 | |
928 | // Set volume for a sound |
929 | void SetSoundVolume(Sound sound, float volume) |
930 | { |
931 | SetAudioBufferVolume(sound.stream.buffer, volume); |
932 | } |
933 | |
934 | // Set pitch for a sound |
935 | void SetSoundPitch(Sound sound, float pitch) |
936 | { |
937 | SetAudioBufferPitch(sound.stream.buffer, pitch); |
938 | } |
939 | |
940 | // Convert wave data to desired format |
941 | void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) |
942 | { |
943 | ma_format formatIn = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32)); |
944 | ma_format formatOut = (( sampleSize == 8)? ma_format_u8 : (( sampleSize == 16)? ma_format_s16 : ma_format_f32)); |
945 | |
946 | ma_uint32 frameCountIn = wave->sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so. |
947 | |
948 | ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate); |
949 | if (frameCount == 0) |
950 | { |
951 | TRACELOG(LOG_WARNING, "WAVE: Failed to get frame count for format conversion" ); |
952 | return; |
953 | } |
954 | |
955 | void *data = RL_MALLOC(frameCount*channels*(sampleSize/8)); |
956 | |
957 | frameCount = (ma_uint32)ma_convert_frames(data, frameCount, formatOut, channels, sampleRate, wave->data, frameCountIn, formatIn, wave->channels, wave->sampleRate); |
958 | if (frameCount == 0) |
959 | { |
960 | TRACELOG(LOG_WARNING, "WAVE: Failed format conversion" ); |
961 | return; |
962 | } |
963 | |
964 | wave->sampleCount = frameCount; |
965 | wave->sampleSize = sampleSize; |
966 | wave->sampleRate = sampleRate; |
967 | wave->channels = channels; |
968 | RL_FREE(wave->data); |
969 | wave->data = data; |
970 | } |
971 | |
972 | // Copy a wave to a new wave |
973 | Wave WaveCopy(Wave wave) |
974 | { |
975 | Wave newWave = { 0 }; |
976 | |
977 | newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8*wave.channels); |
978 | |
979 | if (newWave.data != NULL) |
980 | { |
981 | // NOTE: Size must be provided in bytes |
982 | memcpy(newWave.data, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8); |
983 | |
984 | newWave.sampleCount = wave.sampleCount; |
985 | newWave.sampleRate = wave.sampleRate; |
986 | newWave.sampleSize = wave.sampleSize; |
987 | newWave.channels = wave.channels; |
988 | } |
989 | |
990 | return newWave; |
991 | } |
992 | |
993 | // Crop a wave to defined samples range |
994 | // NOTE: Security check in case of out-of-range |
995 | void WaveCrop(Wave *wave, int initSample, int finalSample) |
996 | { |
997 | if ((initSample >= 0) && (initSample < finalSample) && |
998 | (finalSample > 0) && ((unsigned int)finalSample < wave->sampleCount)) |
999 | { |
1000 | int sampleCount = finalSample - initSample; |
1001 | |
1002 | void *data = RL_MALLOC(sampleCount*wave->sampleSize/8*wave->channels); |
1003 | |
1004 | memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8); |
1005 | |
1006 | RL_FREE(wave->data); |
1007 | wave->data = data; |
1008 | } |
1009 | else TRACELOG(LOG_WARNING, "WAVE: Crop range out of bounds" ); |
1010 | } |
1011 | |
1012 | // Get samples data from wave as a floats array |
1013 | // NOTE: Returned sample values are normalized to range [-1..1] |
1014 | float *GetWaveData(Wave wave) |
1015 | { |
1016 | float *samples = (float *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(float)); |
1017 | |
1018 | for (unsigned int i = 0; i < wave.sampleCount; i++) |
1019 | { |
1020 | for (unsigned int j = 0; j < wave.channels; j++) |
1021 | { |
1022 | if (wave.sampleSize == 8) samples[wave.channels*i + j] = (float)(((unsigned char *)wave.data)[wave.channels*i + j] - 127)/256.0f; |
1023 | else if (wave.sampleSize == 16) samples[wave.channels*i + j] = (float)((short *)wave.data)[wave.channels*i + j]/32767.0f; |
1024 | else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j]; |
1025 | } |
1026 | } |
1027 | |
1028 | return samples; |
1029 | } |
1030 | |
1031 | //---------------------------------------------------------------------------------- |
1032 | // Module Functions Definition - Music loading and stream playing (.OGG) |
1033 | //---------------------------------------------------------------------------------- |
1034 | |
1035 | // Load music stream from file |
1036 | Music LoadMusicStream(const char *fileName) |
1037 | { |
1038 | Music music = { 0 }; |
1039 | bool musicLoaded = false; |
1040 | |
1041 | if (false) { } |
1042 | #if defined(SUPPORT_FILEFORMAT_OGG) |
1043 | else if (IsFileExtension(fileName, ".ogg" )) |
1044 | { |
1045 | // Open ogg audio stream |
1046 | music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL); |
1047 | |
1048 | if (music.ctxData != NULL) |
1049 | { |
1050 | music.ctxType = MUSIC_AUDIO_OGG; |
1051 | stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData); // Get Ogg file info |
1052 | |
1053 | // OGG bit rate defaults to 16 bit, it's enough for compressed format |
1054 | music.stream = InitAudioStream(info.sample_rate, 16, info.channels); |
1055 | music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels; |
1056 | music.loopCount = 0; // Infinite loop by default |
1057 | musicLoaded = true; |
1058 | } |
1059 | } |
1060 | #endif |
1061 | #if defined(SUPPORT_FILEFORMAT_FLAC) |
1062 | else if (IsFileExtension(fileName, ".flac" )) |
1063 | { |
1064 | music.ctxData = drflac_open_file(fileName); |
1065 | |
1066 | if (music.ctxData != NULL) |
1067 | { |
1068 | music.ctxType = MUSIC_AUDIO_FLAC; |
1069 | drflac *ctxFlac = (drflac *)music.ctxData; |
1070 | |
1071 | music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels); |
1072 | music.sampleCount = (unsigned int)ctxFlac->totalSampleCount; |
1073 | music.loopCount = 0; // Infinite loop by default |
1074 | musicLoaded = true; |
1075 | } |
1076 | } |
1077 | #endif |
1078 | #if defined(SUPPORT_FILEFORMAT_MP3) |
1079 | else if (IsFileExtension(fileName, ".mp3" )) |
1080 | { |
1081 | drmp3 *ctxMp3 = RL_MALLOC(sizeof(drmp3)); |
1082 | music.ctxData = ctxMp3; |
1083 | |
1084 | int result = drmp3_init_file(ctxMp3, fileName, NULL); |
1085 | |
1086 | if (result > 0) |
1087 | { |
1088 | music.ctxType = MUSIC_AUDIO_MP3; |
1089 | |
1090 | music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels); |
1091 | music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels; |
1092 | music.loopCount = 0; // Infinite loop by default |
1093 | musicLoaded = true; |
1094 | } |
1095 | } |
1096 | #endif |
1097 | #if defined(SUPPORT_FILEFORMAT_XM) |
1098 | else if (IsFileExtension(fileName, ".xm" )) |
1099 | { |
1100 | jar_xm_context_t *ctxXm = NULL; |
1101 | |
1102 | int result = jar_xm_create_context_from_file(&ctxXm, 48000, fileName); |
1103 | |
1104 | if (result == 0) // XM AUDIO.System.context created successfully |
1105 | { |
1106 | music.ctxType = MUSIC_MODULE_XM; |
1107 | jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops |
1108 | |
1109 | // NOTE: Only stereo is supported for XM |
1110 | music.stream = InitAudioStream(48000, 16, 2); |
1111 | music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; |
1112 | music.loopCount = 0; // Infinite loop by default |
1113 | jar_xm_reset(ctxXm); // make sure we start at the beginning of the song |
1114 | musicLoaded = true; |
1115 | |
1116 | music.ctxData = ctxXm; |
1117 | } |
1118 | } |
1119 | #endif |
1120 | #if defined(SUPPORT_FILEFORMAT_MOD) |
1121 | else if (IsFileExtension(fileName, ".mod" )) |
1122 | { |
1123 | jar_mod_context_t *ctxMod = RL_MALLOC(sizeof(jar_mod_context_t)); |
1124 | |
1125 | jar_mod_init(ctxMod); |
1126 | int result = jar_mod_load_file(ctxMod, fileName); |
1127 | |
1128 | if (result > 0) |
1129 | { |
1130 | music.ctxType = MUSIC_MODULE_MOD; |
1131 | |
1132 | // NOTE: Only stereo is supported for MOD |
1133 | music.stream = InitAudioStream(48000, 16, 2); |
1134 | music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; |
1135 | music.loopCount = 0; // Infinite loop by default |
1136 | musicLoaded = true; |
1137 | |
1138 | music.ctxData = ctxMod; |
1139 | } |
1140 | } |
1141 | #endif |
1142 | else TRACELOG(LOG_WARNING, "STREAM: [%s] Fileformat not supported" , fileName); |
1143 | |
1144 | if (!musicLoaded) |
1145 | { |
1146 | if (false) { } |
1147 | #if defined(SUPPORT_FILEFORMAT_OGG) |
1148 | else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); |
1149 | #endif |
1150 | #if defined(SUPPORT_FILEFORMAT_FLAC) |
1151 | else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData); |
1152 | #endif |
1153 | #if defined(SUPPORT_FILEFORMAT_MP3) |
1154 | else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); } |
1155 | #endif |
1156 | #if defined(SUPPORT_FILEFORMAT_XM) |
1157 | else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData); |
1158 | #endif |
1159 | #if defined(SUPPORT_FILEFORMAT_MOD) |
1160 | else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); } |
1161 | #endif |
1162 | |
1163 | TRACELOG(LOG_WARNING, "FILEIO: [%s] Music file could not be opened" , fileName); |
1164 | } |
1165 | else |
1166 | { |
1167 | // Show some music stream info |
1168 | TRACELOG(LOG_INFO, "FILEIO: [%s] Music file successfully loaded:" , fileName); |
1169 | TRACELOG(LOG_INFO, " > Total samples: %i" , music.sampleCount); |
1170 | TRACELOG(LOG_INFO, " > Sample rate: %i Hz" , music.stream.sampleRate); |
1171 | TRACELOG(LOG_INFO, " > Sample size: %i bits" , music.stream.sampleSize); |
1172 | TRACELOG(LOG_INFO, " > Channels: %i (%s)" , music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi" ); |
1173 | } |
1174 | |
1175 | return music; |
1176 | } |
1177 | |
1178 | // Unload music stream |
1179 | void UnloadMusicStream(Music music) |
1180 | { |
1181 | CloseAudioStream(music.stream); |
1182 | |
1183 | if (false) { } |
1184 | #if defined(SUPPORT_FILEFORMAT_OGG) |
1185 | else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); |
1186 | #endif |
1187 | #if defined(SUPPORT_FILEFORMAT_FLAC) |
1188 | else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData); |
1189 | #endif |
1190 | #if defined(SUPPORT_FILEFORMAT_MP3) |
1191 | else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); } |
1192 | #endif |
1193 | #if defined(SUPPORT_FILEFORMAT_XM) |
1194 | else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData); |
1195 | #endif |
1196 | #if defined(SUPPORT_FILEFORMAT_MOD) |
1197 | else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); } |
1198 | #endif |
1199 | } |
1200 | |
1201 | // Start music playing (open stream) |
1202 | void PlayMusicStream(Music music) |
1203 | { |
1204 | if (music.stream.buffer != NULL) |
1205 | { |
1206 | // For music streams, we need to make sure we maintain the frame cursor position |
1207 | // This is a hack for this section of code in UpdateMusicStream() |
1208 | // NOTE: In case window is minimized, music stream is stopped, just make sure to |
1209 | // play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music); |
1210 | ma_uint32 frameCursorPos = music.stream.buffer->frameCursorPos; |
1211 | PlayAudioStream(music.stream); // WARNING: This resets the cursor position. |
1212 | music.stream.buffer->frameCursorPos = frameCursorPos; |
1213 | } |
1214 | } |
1215 | |
1216 | // Pause music playing |
1217 | void PauseMusicStream(Music music) |
1218 | { |
1219 | PauseAudioStream(music.stream); |
1220 | } |
1221 | |
1222 | // Resume music playing |
1223 | void ResumeMusicStream(Music music) |
1224 | { |
1225 | ResumeAudioStream(music.stream); |
1226 | } |
1227 | |
1228 | // Stop music playing (close stream) |
1229 | void StopMusicStream(Music music) |
1230 | { |
1231 | StopAudioStream(music.stream); |
1232 | |
1233 | switch (music.ctxType) |
1234 | { |
1235 | #if defined(SUPPORT_FILEFORMAT_OGG) |
1236 | case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break; |
1237 | #endif |
1238 | #if defined(SUPPORT_FILEFORMAT_FLAC) |
1239 | case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, 0); break; |
1240 | #endif |
1241 | #if defined(SUPPORT_FILEFORMAT_MP3) |
1242 | case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break; |
1243 | #endif |
1244 | #if defined(SUPPORT_FILEFORMAT_XM) |
1245 | case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break; |
1246 | #endif |
1247 | #if defined(SUPPORT_FILEFORMAT_MOD) |
1248 | case MUSIC_MODULE_MOD: jar_mod_seek_start((jar_mod_context_t *)music.ctxData); break; |
1249 | #endif |
1250 | default: break; |
1251 | } |
1252 | } |
1253 | |
1254 | // Update (re-fill) music buffers if data already processed |
1255 | void UpdateMusicStream(Music music) |
1256 | { |
1257 | if (music.stream.buffer == NULL) return; |
1258 | |
1259 | bool streamEnding = false; |
1260 | |
1261 | unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2; |
1262 | |
1263 | // NOTE: Using dynamic allocation because it could require more than 16KB |
1264 | void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1); |
1265 | |
1266 | int samplesCount = 0; // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts |
1267 | |
1268 | // TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly... |
1269 | //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels; |
1270 | int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels); |
1271 | |
1272 | while (IsAudioStreamProcessed(music.stream)) |
1273 | { |
1274 | if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels; |
1275 | else samplesCount = sampleLeft; |
1276 | |
1277 | switch (music.ctxType) |
1278 | { |
1279 | #if defined(SUPPORT_FILEFORMAT_OGG) |
1280 | case MUSIC_AUDIO_OGG: |
1281 | { |
1282 | // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!) |
1283 | stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, samplesCount); |
1284 | |
1285 | } break; |
1286 | #endif |
1287 | #if defined(SUPPORT_FILEFORMAT_FLAC) |
1288 | case MUSIC_AUDIO_FLAC: |
1289 | { |
1290 | // NOTE: Returns the number of samples to process (not required) |
1291 | drflac_read_pcm_frames_s16((drflac *)music.ctxData, samplesCount, (short *)pcm); |
1292 | |
1293 | } break; |
1294 | #endif |
1295 | #if defined(SUPPORT_FILEFORMAT_MP3) |
1296 | case MUSIC_AUDIO_MP3: |
1297 | { |
1298 | // NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed |
1299 | drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm); |
1300 | |
1301 | } break; |
1302 | #endif |
1303 | #if defined(SUPPORT_FILEFORMAT_XM) |
1304 | case MUSIC_MODULE_XM: |
1305 | { |
1306 | // NOTE: Internally this function considers 2 channels generation, so samplesCount/2 |
1307 | jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, samplesCount/2); |
1308 | } break; |
1309 | #endif |
1310 | #if defined(SUPPORT_FILEFORMAT_MOD) |
1311 | case MUSIC_MODULE_MOD: |
1312 | { |
1313 | // NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2 |
1314 | jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, samplesCount/2, 0); |
1315 | } break; |
1316 | #endif |
1317 | default: break; |
1318 | } |
1319 | |
1320 | UpdateAudioStream(music.stream, pcm, samplesCount); |
1321 | |
1322 | if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD)) |
1323 | { |
1324 | if (samplesCount > 1) sampleLeft -= samplesCount/2; |
1325 | else sampleLeft -= samplesCount; |
1326 | } |
1327 | else sampleLeft -= samplesCount; |
1328 | |
1329 | if (sampleLeft <= 0) |
1330 | { |
1331 | streamEnding = true; |
1332 | break; |
1333 | } |
1334 | } |
1335 | |
1336 | // Free allocated pcm data |
1337 | RL_FREE(pcm); |
1338 | |
1339 | // Reset audio stream for looping |
1340 | if (streamEnding) |
1341 | { |
1342 | StopMusicStream(music); // Stop music (and reset) |
1343 | |
1344 | // Decrease loopCount to stop when required |
1345 | if (music.loopCount > 1) |
1346 | { |
1347 | music.loopCount--; // Decrease loop count |
1348 | PlayMusicStream(music); // Play again |
1349 | } |
1350 | else if (music.loopCount == 0) PlayMusicStream(music); |
1351 | } |
1352 | else |
1353 | { |
1354 | // NOTE: In case window is minimized, music stream is stopped, |
1355 | // just make sure to play again on window restore |
1356 | if (IsMusicPlaying(music)) PlayMusicStream(music); |
1357 | } |
1358 | } |
1359 | |
1360 | // Check if any music is playing |
1361 | bool IsMusicPlaying(Music music) |
1362 | { |
1363 | return IsAudioStreamPlaying(music.stream); |
1364 | } |
1365 | |
1366 | // Set volume for music |
1367 | void SetMusicVolume(Music music, float volume) |
1368 | { |
1369 | SetAudioStreamVolume(music.stream, volume); |
1370 | } |
1371 | |
1372 | // Set pitch for music |
1373 | void SetMusicPitch(Music music, float pitch) |
1374 | { |
1375 | SetAudioStreamPitch(music.stream, pitch); |
1376 | } |
1377 | |
1378 | // Set music loop count (loop repeats) |
1379 | // NOTE: If set to 0, means infinite loop |
1380 | void SetMusicLoopCount(Music music, int count) |
1381 | { |
1382 | music.loopCount = count; |
1383 | } |
1384 | |
1385 | // Get music time length (in seconds) |
1386 | float GetMusicTimeLength(Music music) |
1387 | { |
1388 | float totalSeconds = 0.0f; |
1389 | |
1390 | totalSeconds = (float)music.sampleCount/(music.stream.sampleRate*music.stream.channels); |
1391 | |
1392 | return totalSeconds; |
1393 | } |
1394 | |
1395 | // Get current music time played (in seconds) |
1396 | float GetMusicTimePlayed(Music music) |
1397 | { |
1398 | float secondsPlayed = 0.0f; |
1399 | |
1400 | if (music.stream.buffer != NULL) |
1401 | { |
1402 | //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels; |
1403 | unsigned int samplesPlayed = music.stream.buffer->totalFramesProcessed*music.stream.channels; |
1404 | secondsPlayed = (float)samplesPlayed / (music.stream.sampleRate*music.stream.channels); |
1405 | } |
1406 | |
1407 | return secondsPlayed; |
1408 | } |
1409 | |
1410 | // Init audio stream (to stream audio pcm data) |
1411 | AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels) |
1412 | { |
1413 | AudioStream stream = { 0 }; |
1414 | |
1415 | stream.sampleRate = sampleRate; |
1416 | stream.sampleSize = sampleSize; |
1417 | stream.channels = channels; |
1418 | |
1419 | ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32)); |
1420 | |
1421 | // The size of a streaming buffer must be at least double the size of a period |
1422 | unsigned int periodSize = AUDIO.System.device.playback.internalPeriodSizeInFrames; |
1423 | unsigned int subBufferSize = AUDIO.Buffer.defaultSize; // Default buffer size (audio stream) |
1424 | |
1425 | if (subBufferSize < periodSize) subBufferSize = periodSize; |
1426 | |
1427 | // Create a double audio buffer of defined size |
1428 | stream.buffer = LoadAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM); |
1429 | |
1430 | if (stream.buffer != NULL) |
1431 | { |
1432 | stream.buffer->looping = true; // Always loop for streaming buffers |
1433 | TRACELOG(LOG_INFO, "STREAM: Initialized successfully (%i Hz, %i bit, %s)" , stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo" ); |
1434 | } |
1435 | else TRACELOG(LOG_WARNING, "STREAM: Failed to load audio buffer, stream could not be created" ); |
1436 | |
1437 | return stream; |
1438 | } |
1439 | |
1440 | // Close audio stream and free memory |
1441 | void CloseAudioStream(AudioStream stream) |
1442 | { |
1443 | UnloadAudioBuffer(stream.buffer); |
1444 | |
1445 | TRACELOG(LOG_INFO, "STREAM: Unloaded audio stream data from RAM" ); |
1446 | } |
1447 | |
1448 | // Update audio stream buffers with data |
1449 | // NOTE 1: Only updates one buffer of the stream source: unqueue -> update -> queue |
1450 | // NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed() |
1451 | void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) |
1452 | { |
1453 | if (stream.buffer != NULL) |
1454 | { |
1455 | if (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]) |
1456 | { |
1457 | ma_uint32 subBufferToUpdate = 0; |
1458 | |
1459 | if (stream.buffer->isSubBufferProcessed[0] && stream.buffer->isSubBufferProcessed[1]) |
1460 | { |
1461 | // Both buffers are available for updating. |
1462 | // Update the first one and make sure the cursor is moved back to the front. |
1463 | subBufferToUpdate = 0; |
1464 | stream.buffer->frameCursorPos = 0; |
1465 | } |
1466 | else |
1467 | { |
1468 | // Just update whichever sub-buffer is processed. |
1469 | subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1; |
1470 | } |
1471 | |
1472 | ma_uint32 subBufferSizeInFrames = stream.buffer->sizeInFrames/2; |
1473 | unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate); |
1474 | |
1475 | // TODO: Get total frames processed on this buffer... DOES NOT WORK. |
1476 | stream.buffer->totalFramesProcessed += subBufferSizeInFrames; |
1477 | |
1478 | // Does this API expect a whole buffer to be updated in one go? |
1479 | // Assuming so, but if not will need to change this logic. |
1480 | if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels) |
1481 | { |
1482 | ma_uint32 framesToWrite = subBufferSizeInFrames; |
1483 | |
1484 | if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels; |
1485 | |
1486 | ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8); |
1487 | memcpy(subBuffer, data, bytesToWrite); |
1488 | |
1489 | // Any leftover frames should be filled with zeros. |
1490 | ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite; |
1491 | |
1492 | if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8)); |
1493 | |
1494 | stream.buffer->isSubBufferProcessed[subBufferToUpdate] = false; |
1495 | } |
1496 | else TRACELOG(LOG_WARNING, "STREAM: Attempting to write too many frames to buffer" ); |
1497 | } |
1498 | else TRACELOG(LOG_WARNING, "STREAM: Buffer not available for updating" ); |
1499 | } |
1500 | } |
1501 | |
1502 | // Check if any audio stream buffers requires refill |
1503 | bool IsAudioStreamProcessed(AudioStream stream) |
1504 | { |
1505 | if (stream.buffer == NULL) return false; |
1506 | |
1507 | return (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]); |
1508 | } |
1509 | |
1510 | // Play audio stream |
1511 | void PlayAudioStream(AudioStream stream) |
1512 | { |
1513 | PlayAudioBuffer(stream.buffer); |
1514 | } |
1515 | |
1516 | // Play audio stream |
1517 | void PauseAudioStream(AudioStream stream) |
1518 | { |
1519 | PauseAudioBuffer(stream.buffer); |
1520 | } |
1521 | |
1522 | // Resume audio stream playing |
1523 | void ResumeAudioStream(AudioStream stream) |
1524 | { |
1525 | ResumeAudioBuffer(stream.buffer); |
1526 | } |
1527 | |
1528 | // Check if audio stream is playing. |
1529 | bool IsAudioStreamPlaying(AudioStream stream) |
1530 | { |
1531 | return IsAudioBufferPlaying(stream.buffer); |
1532 | } |
1533 | |
1534 | // Stop audio stream |
1535 | void StopAudioStream(AudioStream stream) |
1536 | { |
1537 | StopAudioBuffer(stream.buffer); |
1538 | } |
1539 | |
1540 | // Set volume for audio stream (1.0 is max level) |
1541 | void SetAudioStreamVolume(AudioStream stream, float volume) |
1542 | { |
1543 | SetAudioBufferVolume(stream.buffer, volume); |
1544 | } |
1545 | |
1546 | // Set pitch for audio stream (1.0 is base level) |
1547 | void SetAudioStreamPitch(AudioStream stream, float pitch) |
1548 | { |
1549 | SetAudioBufferPitch(stream.buffer, pitch); |
1550 | } |
1551 | |
1552 | // Default size for new audio streams |
1553 | void SetAudioStreamBufferSizeDefault(int size) |
1554 | { |
1555 | AUDIO.Buffer.defaultSize = size; |
1556 | } |
1557 | |
1558 | //---------------------------------------------------------------------------------- |
1559 | // Module specific Functions Definition |
1560 | //---------------------------------------------------------------------------------- |
1561 | |
1562 | // Log callback function |
1563 | static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message) |
1564 | { |
1565 | (void)pContext; |
1566 | (void)pDevice; |
1567 | |
1568 | TRACELOG(LOG_ERROR, "miniaudio: %s" , message); // All log messages from miniaudio are errors |
1569 | } |
1570 | |
1571 | // Reads audio data from an AudioBuffer object in internal format. |
1572 | static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount) |
1573 | { |
1574 | ma_uint32 subBufferSizeInFrames = (audioBuffer->sizeInFrames > 1)? audioBuffer->sizeInFrames/2 : audioBuffer->sizeInFrames; |
1575 | ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames; |
1576 | |
1577 | if (currentSubBufferIndex > 1) return 0; |
1578 | |
1579 | // Another thread can update the processed state of buffers so |
1580 | // we just take a copy here to try and avoid potential synchronization problems |
1581 | bool isSubBufferProcessed[2]; |
1582 | isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0]; |
1583 | isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1]; |
1584 | |
1585 | ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn); |
1586 | |
1587 | // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0 |
1588 | ma_uint32 framesRead = 0; |
1589 | while (1) |
1590 | { |
1591 | // We break from this loop differently depending on the buffer's usage |
1592 | // - For static buffers, we simply fill as much data as we can |
1593 | // - For streaming buffers we only fill the halves of the buffer that are processed |
1594 | // Unprocessed halves must keep their audio data in-tact |
1595 | if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) |
1596 | { |
1597 | if (framesRead >= frameCount) break; |
1598 | } |
1599 | else |
1600 | { |
1601 | if (isSubBufferProcessed[currentSubBufferIndex]) break; |
1602 | } |
1603 | |
1604 | ma_uint32 totalFramesRemaining = (frameCount - framesRead); |
1605 | if (totalFramesRemaining == 0) break; |
1606 | |
1607 | ma_uint32 framesRemainingInOutputBuffer; |
1608 | if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) |
1609 | { |
1610 | framesRemainingInOutputBuffer = audioBuffer->sizeInFrames - audioBuffer->frameCursorPos; |
1611 | } |
1612 | else |
1613 | { |
1614 | ma_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames*currentSubBufferIndex; |
1615 | framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer); |
1616 | } |
1617 | |
1618 | ma_uint32 framesToRead = totalFramesRemaining; |
1619 | if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer; |
1620 | |
1621 | memcpy((unsigned char *)framesOut + (framesRead*frameSizeInBytes), audioBuffer->data + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes); |
1622 | audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->sizeInFrames; |
1623 | framesRead += framesToRead; |
1624 | |
1625 | // If we've read to the end of the buffer, mark it as processed |
1626 | if (framesToRead == framesRemainingInOutputBuffer) |
1627 | { |
1628 | audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true; |
1629 | isSubBufferProcessed[currentSubBufferIndex] = true; |
1630 | |
1631 | currentSubBufferIndex = (currentSubBufferIndex + 1)%2; |
1632 | |
1633 | // We need to break from this loop if we're not looping |
1634 | if (!audioBuffer->looping) |
1635 | { |
1636 | StopAudioBuffer(audioBuffer); |
1637 | break; |
1638 | } |
1639 | } |
1640 | } |
1641 | |
1642 | // Zero-fill excess |
1643 | ma_uint32 totalFramesRemaining = (frameCount - framesRead); |
1644 | if (totalFramesRemaining > 0) |
1645 | { |
1646 | memset((unsigned char *)framesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes); |
1647 | |
1648 | // For static buffers we can fill the remaining frames with silence for safety, but we don't want |
1649 | // to report those frames as "read". The reason for this is that the caller uses the return value |
1650 | // to know whether or not a non-looping sound has finished playback. |
1651 | if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining; |
1652 | } |
1653 | |
1654 | return framesRead; |
1655 | } |
1656 | |
1657 | // Reads audio data from an AudioBuffer object in device format. Returned data will be in a format appropriate for mixing. |
1658 | static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount) |
1659 | { |
1660 | // What's going on here is that we're continuously converting data from the AudioBuffer's internal format to the mixing format, which |
1661 | // should be defined by the output format of the data converter. We do this until frameCount frames have been output. The important |
1662 | // detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output |
1663 | // frames. This can be achieved with ma_data_converter_get_required_input_frame_count(). |
1664 | ma_uint8 inputBuffer[4096]; |
1665 | ma_uint32 inputBufferFrameCap = sizeof(inputBuffer) / ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn); |
1666 | |
1667 | ma_uint32 totalOutputFramesProcessed = 0; |
1668 | while (totalOutputFramesProcessed < frameCount) |
1669 | { |
1670 | ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed; |
1671 | |
1672 | ma_uint64 inputFramesToProcessThisIteration = ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration); |
1673 | if (inputFramesToProcessThisIteration > inputBufferFrameCap) |
1674 | { |
1675 | inputFramesToProcessThisIteration = inputBufferFrameCap; |
1676 | } |
1677 | |
1678 | float *runningFramesOut = framesOut + (totalOutputFramesProcessed * audioBuffer->converter.config.channelsOut); |
1679 | |
1680 | /* At this point we can convert the data to our mixing format. */ |
1681 | ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration); /* Safe cast. */ |
1682 | ma_uint64 outputFramesProcessedThisIteration = outputFramesToProcessThisIteration; |
1683 | ma_data_converter_process_pcm_frames(&audioBuffer->converter, inputBuffer, &inputFramesProcessedThisIteration, runningFramesOut, &outputFramesProcessedThisIteration); |
1684 | |
1685 | totalOutputFramesProcessed += (ma_uint32)outputFramesProcessedThisIteration; /* Safe cast. */ |
1686 | |
1687 | if (inputFramesProcessedThisIteration < inputFramesToProcessThisIteration) |
1688 | { |
1689 | break; /* Ran out of input data. */ |
1690 | } |
1691 | |
1692 | /* This should never be hit, but will add it here for safety. Ensures we get out of the loop when no input nor output frames are processed. */ |
1693 | if (inputFramesProcessedThisIteration == 0 && outputFramesProcessedThisIteration == 0) |
1694 | { |
1695 | break; |
1696 | } |
1697 | } |
1698 | |
1699 | return totalOutputFramesProcessed; |
1700 | } |
1701 | |
1702 | |
1703 | // Sending audio data to device callback function |
1704 | // NOTE: All the mixing takes place here |
1705 | static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount) |
1706 | { |
1707 | (void)pDevice; |
1708 | |
1709 | // Mixing is basically just an accumulation, we need to initialize the output buffer to 0 |
1710 | memset(pFramesOut, 0, frameCount*pDevice->playback.channels*ma_get_bytes_per_sample(pDevice->playback.format)); |
1711 | |
1712 | // Using a mutex here for thread-safety which makes things not real-time |
1713 | // This is unlikely to be necessary for this project, but may want to consider how you might want to avoid this |
1714 | ma_mutex_lock(&AUDIO.System.lock); |
1715 | { |
1716 | for (AudioBuffer *audioBuffer = AUDIO.Buffer.first; audioBuffer != NULL; audioBuffer = audioBuffer->next) |
1717 | { |
1718 | // Ignore stopped or paused sounds |
1719 | if (!audioBuffer->playing || audioBuffer->paused) continue; |
1720 | |
1721 | ma_uint32 framesRead = 0; |
1722 | |
1723 | while (1) |
1724 | { |
1725 | if (framesRead >= frameCount) break; |
1726 | |
1727 | // Just read as much data as we can from the stream |
1728 | ma_uint32 framesToRead = (frameCount - framesRead); |
1729 | |
1730 | while (framesToRead > 0) |
1731 | { |
1732 | float tempBuffer[1024]; // 512 frames for stereo |
1733 | |
1734 | ma_uint32 framesToReadRightNow = framesToRead; |
1735 | if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS) |
1736 | { |
1737 | framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS; |
1738 | } |
1739 | |
1740 | ma_uint32 framesJustRead = ReadAudioBufferFramesInMixingFormat(audioBuffer, tempBuffer, framesToReadRightNow); |
1741 | if (framesJustRead > 0) |
1742 | { |
1743 | float *framesOut = (float *)pFramesOut + (framesRead*AUDIO.System.device.playback.channels); |
1744 | float *framesIn = tempBuffer; |
1745 | |
1746 | MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume); |
1747 | |
1748 | framesToRead -= framesJustRead; |
1749 | framesRead += framesJustRead; |
1750 | } |
1751 | |
1752 | if (!audioBuffer->playing) |
1753 | { |
1754 | framesRead = frameCount; |
1755 | break; |
1756 | } |
1757 | |
1758 | // If we weren't able to read all the frames we requested, break |
1759 | if (framesJustRead < framesToReadRightNow) |
1760 | { |
1761 | if (!audioBuffer->looping) |
1762 | { |
1763 | StopAudioBuffer(audioBuffer); |
1764 | break; |
1765 | } |
1766 | else |
1767 | { |
1768 | // Should never get here, but just for safety, |
1769 | // move the cursor position back to the start and continue the loop |
1770 | audioBuffer->frameCursorPos = 0; |
1771 | continue; |
1772 | } |
1773 | } |
1774 | } |
1775 | |
1776 | // If for some reason we weren't able to read every frame we'll need to break from the loop |
1777 | // Not doing this could theoretically put us into an infinite loop |
1778 | if (framesToRead > 0) break; |
1779 | } |
1780 | } |
1781 | } |
1782 | |
1783 | ma_mutex_unlock(&AUDIO.System.lock); |
1784 | } |
1785 | |
1786 | // This is the main mixing function. Mixing is pretty simple in this project - it's just an accumulation. |
1787 | // NOTE: framesOut is both an input and an output. It will be initially filled with zeros outside of this function. |
1788 | static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume) |
1789 | { |
1790 | for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame) |
1791 | { |
1792 | for (ma_uint32 iChannel = 0; iChannel < AUDIO.System.device.playback.channels; ++iChannel) |
1793 | { |
1794 | float *frameOut = framesOut + (iFrame*AUDIO.System.device.playback.channels); |
1795 | const float *frameIn = framesIn + (iFrame*AUDIO.System.device.playback.channels); |
1796 | |
1797 | frameOut[iChannel] += (frameIn[iChannel]*localVolume); |
1798 | } |
1799 | } |
1800 | } |
1801 | |
1802 | // Initialise the multichannel buffer pool |
1803 | static void InitAudioBufferPool(void) |
1804 | { |
1805 | // Dummy buffers |
1806 | for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) |
1807 | { |
1808 | AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC); |
1809 | } |
1810 | |
1811 | // TODO: Verification required for log |
1812 | TRACELOG(LOG_INFO, "AUDIO: Multichannel pool size: %i" , MAX_AUDIO_BUFFER_POOL_CHANNELS); |
1813 | } |
1814 | |
1815 | // Close the audio buffers pool |
1816 | static void CloseAudioBufferPool(void) |
1817 | { |
1818 | for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) |
1819 | { |
1820 | RL_FREE(AUDIO.MultiChannel.pool[i]->data); |
1821 | RL_FREE(AUDIO.MultiChannel.pool[i]); |
1822 | } |
1823 | } |
1824 | |
1825 | #if defined(SUPPORT_FILEFORMAT_WAV) |
1826 | // Load WAV file into Wave structure |
1827 | static Wave LoadWAV(const char *fileName) |
1828 | { |
1829 | // Basic WAV headers structs |
1830 | typedef struct { |
1831 | char chunkID[4]; |
1832 | int chunkSize; |
1833 | char format[4]; |
1834 | } ; |
1835 | |
1836 | typedef struct { |
1837 | char subChunkID[4]; |
1838 | int subChunkSize; |
1839 | short audioFormat; |
1840 | short numChannels; |
1841 | int sampleRate; |
1842 | int byteRate; |
1843 | short blockAlign; |
1844 | short bitsPerSample; |
1845 | } WAVFormat; |
1846 | |
1847 | typedef struct { |
1848 | char subChunkID[4]; |
1849 | int subChunkSize; |
1850 | } WAVData; |
1851 | |
1852 | WAVRiffHeader = { 0 }; |
1853 | WAVFormat wavFormat = { 0 }; |
1854 | WAVData wavData = { 0 }; |
1855 | |
1856 | Wave wave = { 0 }; |
1857 | FILE *wavFile = NULL; |
1858 | |
1859 | wavFile = fopen(fileName, "rb" ); |
1860 | |
1861 | if (wavFile == NULL) |
1862 | { |
1863 | TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open WAV file" , fileName); |
1864 | wave.data = NULL; |
1865 | } |
1866 | else |
1867 | { |
1868 | // Read in the first chunk into the struct |
1869 | fread(&wavRiffHeader, sizeof(WAVRiffHeader), 1, wavFile); |
1870 | |
1871 | // Check for RIFF and WAVE tags |
1872 | if ((wavRiffHeader.chunkID[0] != 'R') || |
1873 | (wavRiffHeader.chunkID[1] != 'I') || |
1874 | (wavRiffHeader.chunkID[2] != 'F') || |
1875 | (wavRiffHeader.chunkID[3] != 'F') || |
1876 | (wavRiffHeader.format[0] != 'W') || |
1877 | (wavRiffHeader.format[1] != 'A') || |
1878 | (wavRiffHeader.format[2] != 'V') || |
1879 | (wavRiffHeader.format[3] != 'E')) |
1880 | { |
1881 | TRACELOG(LOG_WARNING, "WAVE: [%s] RIFF or WAVE header are not valid" , fileName); |
1882 | } |
1883 | else |
1884 | { |
1885 | // Read in the 2nd chunk for the wave info |
1886 | fread(&wavFormat, sizeof(WAVFormat), 1, wavFile); |
1887 | |
1888 | // Check for fmt tag |
1889 | if ((wavFormat.subChunkID[0] != 'f') || (wavFormat.subChunkID[1] != 'm') || |
1890 | (wavFormat.subChunkID[2] != 't') || (wavFormat.subChunkID[3] != ' ')) |
1891 | { |
1892 | TRACELOG(LOG_WARNING, "WAVE: [%s] Wave format header is not valid" , fileName); |
1893 | } |
1894 | else |
1895 | { |
1896 | // Check for extra parameters; |
1897 | if (wavFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR); |
1898 | |
1899 | // Read in the the last byte of data before the sound file |
1900 | fread(&wavData, sizeof(WAVData), 1, wavFile); |
1901 | |
1902 | // Check for data tag |
1903 | if ((wavData.subChunkID[0] != 'd') || (wavData.subChunkID[1] != 'a') || |
1904 | (wavData.subChunkID[2] != 't') || (wavData.subChunkID[3] != 'a')) |
1905 | { |
1906 | TRACELOG(LOG_WARNING, "WAVE: [%s] Data header is not valid" , fileName); |
1907 | } |
1908 | else |
1909 | { |
1910 | // Allocate memory for data |
1911 | wave.data = RL_MALLOC(wavData.subChunkSize); |
1912 | |
1913 | // Read in the sound data into the soundData variable |
1914 | fread(wave.data, wavData.subChunkSize, 1, wavFile); |
1915 | |
1916 | // Store wave parameters |
1917 | wave.sampleRate = wavFormat.sampleRate; |
1918 | wave.sampleSize = wavFormat.bitsPerSample; |
1919 | wave.channels = wavFormat.numChannels; |
1920 | |
1921 | // NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes |
1922 | if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32)) |
1923 | { |
1924 | TRACELOG(LOG_WARNING, "WAVE: [%s] Sample size (%ibit) not supported, converted to 16bit" , fileName, wave.sampleSize); |
1925 | WaveFormat(&wave, wave.sampleRate, 16, wave.channels); |
1926 | } |
1927 | |
1928 | // NOTE: Only support up to 2 channels (mono, stereo) |
1929 | if (wave.channels > 2) |
1930 | { |
1931 | WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2); |
1932 | TRACELOG(LOG_WARNING, "WAVE: [%s] Channels number (%i) not supported, converted to 2 channels" , fileName, wave.channels); |
1933 | } |
1934 | |
1935 | // NOTE: subChunkSize comes in bytes, we need to translate it to number of samples |
1936 | wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels; |
1937 | |
1938 | TRACELOG(LOG_INFO, "WAVE: [%s] File loaded successfully (%i Hz, %i bit, %s)" , fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo" ); |
1939 | } |
1940 | } |
1941 | } |
1942 | |
1943 | fclose(wavFile); |
1944 | } |
1945 | |
1946 | return wave; |
1947 | } |
1948 | |
1949 | // Save wave data as WAV file |
1950 | static int SaveWAV(Wave wave, const char *fileName) |
1951 | { |
1952 | int success = 0; |
1953 | int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; |
1954 | |
1955 | // Basic WAV headers structs |
1956 | typedef struct { |
1957 | char chunkID[4]; |
1958 | int chunkSize; |
1959 | char format[4]; |
1960 | } ; |
1961 | |
1962 | typedef struct { |
1963 | char subChunkID[4]; |
1964 | int subChunkSize; |
1965 | short audioFormat; |
1966 | short numChannels; |
1967 | int sampleRate; |
1968 | int byteRate; |
1969 | short blockAlign; |
1970 | short bitsPerSample; |
1971 | } WaveFormat; |
1972 | |
1973 | typedef struct { |
1974 | char subChunkID[4]; |
1975 | int subChunkSize; |
1976 | } WaveData; |
1977 | |
1978 | FILE *wavFile = fopen(fileName, "wb" ); |
1979 | |
1980 | if (wavFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open audio file" , fileName); |
1981 | else |
1982 | { |
1983 | RiffHeader ; |
1984 | WaveFormat waveFormat; |
1985 | WaveData waveData; |
1986 | |
1987 | // Fill structs with data |
1988 | riffHeader.chunkID[0] = 'R'; |
1989 | riffHeader.chunkID[1] = 'I'; |
1990 | riffHeader.chunkID[2] = 'F'; |
1991 | riffHeader.chunkID[3] = 'F'; |
1992 | riffHeader.chunkSize = 44 - 4 + wave.sampleCount*wave.sampleSize/8; |
1993 | riffHeader.format[0] = 'W'; |
1994 | riffHeader.format[1] = 'A'; |
1995 | riffHeader.format[2] = 'V'; |
1996 | riffHeader.format[3] = 'E'; |
1997 | |
1998 | waveFormat.subChunkID[0] = 'f'; |
1999 | waveFormat.subChunkID[1] = 'm'; |
2000 | waveFormat.subChunkID[2] = 't'; |
2001 | waveFormat.subChunkID[3] = ' '; |
2002 | waveFormat.subChunkSize = 16; |
2003 | waveFormat.audioFormat = 1; |
2004 | waveFormat.numChannels = wave.channels; |
2005 | waveFormat.sampleRate = wave.sampleRate; |
2006 | waveFormat.byteRate = wave.sampleRate*wave.sampleSize/8; |
2007 | waveFormat.blockAlign = wave.sampleSize/8; |
2008 | waveFormat.bitsPerSample = wave.sampleSize; |
2009 | |
2010 | waveData.subChunkID[0] = 'd'; |
2011 | waveData.subChunkID[1] = 'a'; |
2012 | waveData.subChunkID[2] = 't'; |
2013 | waveData.subChunkID[3] = 'a'; |
2014 | waveData.subChunkSize = dataSize; |
2015 | |
2016 | fwrite(&riffHeader, sizeof(RiffHeader), 1, wavFile); |
2017 | fwrite(&waveFormat, sizeof(WaveFormat), 1, wavFile); |
2018 | fwrite(&waveData, sizeof(WaveData), 1, wavFile); |
2019 | |
2020 | success = fwrite(wave.data, dataSize, 1, wavFile); |
2021 | |
2022 | fclose(wavFile); |
2023 | } |
2024 | |
2025 | // If all data has been written correctly to file, success = 1 |
2026 | return success; |
2027 | } |
2028 | #endif |
2029 | |
2030 | #if defined(SUPPORT_FILEFORMAT_OGG) |
2031 | // Load OGG file into Wave structure |
2032 | // NOTE: Using stb_vorbis library |
2033 | static Wave LoadOGG(const char *fileName) |
2034 | { |
2035 | Wave wave = { 0 }; |
2036 | |
2037 | stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL); |
2038 | |
2039 | if (oggFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open OGG file" , fileName); |
2040 | else |
2041 | { |
2042 | stb_vorbis_info info = stb_vorbis_get_info(oggFile); |
2043 | |
2044 | wave.sampleRate = info.sample_rate; |
2045 | wave.sampleSize = 16; // 16 bit per sample (short) |
2046 | wave.channels = info.channels; |
2047 | wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggFile)*info.channels; // Independent by channel |
2048 | |
2049 | float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile); |
2050 | if (totalSeconds > 10) TRACELOG(LOG_WARNING, "WAVE: [%s] Ogg audio length larger than 10 seconds (%f), that's a big file in memory, consider music streaming" , fileName, totalSeconds); |
2051 | |
2052 | wave.data = (short *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(short)); |
2053 | |
2054 | // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!) |
2055 | stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels); |
2056 | TRACELOG(LOG_INFO, "WAVE: [%s] OGG file loaded successfully (%i Hz, %i bit, %s)" , fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo" ); |
2057 | |
2058 | stb_vorbis_close(oggFile); |
2059 | } |
2060 | |
2061 | return wave; |
2062 | } |
2063 | #endif |
2064 | |
2065 | #if defined(SUPPORT_FILEFORMAT_FLAC) |
2066 | // Load FLAC file into Wave structure |
2067 | // NOTE: Using dr_flac library |
2068 | static Wave LoadFLAC(const char *fileName) |
2069 | { |
2070 | Wave wave = { 0 }; |
2071 | |
2072 | // Decode an entire FLAC file in one go |
2073 | unsigned long long int totalSampleCount = 0; |
2074 | wave.data = drflac_open_file_and_read_pcm_frames_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount); |
2075 | |
2076 | if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load FLAC data" , fileName); |
2077 | else |
2078 | { |
2079 | wave.sampleCount = (unsigned int)totalSampleCount; |
2080 | wave.sampleSize = 16; |
2081 | |
2082 | // NOTE: Only support up to 2 channels (mono, stereo) |
2083 | if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] FLAC channels number (%i) not supported" , fileName, wave.channels); |
2084 | |
2085 | TRACELOG(LOG_INFO, "WAVE: [%s] FLAC file loaded successfully (%i Hz, %i bit, %s)" , fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo" ); |
2086 | } |
2087 | |
2088 | return wave; |
2089 | } |
2090 | #endif |
2091 | |
2092 | #if defined(SUPPORT_FILEFORMAT_MP3) |
2093 | // Load MP3 file into Wave structure |
2094 | // NOTE: Using dr_mp3 library |
2095 | static Wave LoadMP3(const char *fileName) |
2096 | { |
2097 | Wave wave = { 0 }; |
2098 | |
2099 | // Decode an entire MP3 file in one go |
2100 | unsigned long long int totalFrameCount = 0; |
2101 | drmp3_config config = { 0 }; |
2102 | wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount); |
2103 | |
2104 | if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load MP3 data" , fileName); |
2105 | else |
2106 | { |
2107 | wave.channels = config.outputChannels; |
2108 | wave.sampleRate = config.outputSampleRate; |
2109 | wave.sampleCount = (int)totalFrameCount*wave.channels; |
2110 | wave.sampleSize = 32; |
2111 | |
2112 | // NOTE: Only support up to 2 channels (mono, stereo) |
2113 | if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] MP3 channels number (%i) not supported" , fileName, wave.channels); |
2114 | |
2115 | TRACELOG(LOG_INFO, "WAVE: [%s] MP3 file loaded successfully (%i Hz, %i bit, %s)" , fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo" ); |
2116 | } |
2117 | |
2118 | return wave; |
2119 | } |
2120 | #endif |
2121 | |
2122 | // Some required functions for audio standalone module version |
2123 | #if defined(RAUDIO_STANDALONE) |
2124 | // Check file extension |
2125 | bool IsFileExtension(const char *fileName, const char *ext) |
2126 | { |
2127 | bool result = false; |
2128 | const char *fileExt; |
2129 | |
2130 | if ((fileExt = strrchr(fileName, '.')) != NULL) |
2131 | { |
2132 | if (strcmp(fileExt, ext) == 0) result = true; |
2133 | } |
2134 | |
2135 | return result; |
2136 | } |
2137 | #endif |
2138 | |
2139 | #undef AudioBuffer |
2140 | |