/
mixer.c
1601 lines (1396 loc) · 44 KB
1
/*
2
SDL_mixer: An audio mixer library based on the SDL library
3
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
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.
20
21
*/
22
/* $Id$ */
23
24
25
26
27
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
28
#include "SDL.h"
29
30
#include "SDL_mixer.h"
31
#include "mixer.h"
32
#include "music.h"
33
34
35
#include "load_aiff.h"
#include "load_voc.h"
36
37
38
#define __MIX_INTERNAL_EFFECT__
#include "effects_internal.h"
39
/* Magic numbers for various audio file formats */
40
41
42
43
#define RIFF 0x46464952 /* "RIFF" */
#define WAVE 0x45564157 /* "WAVE" */
#define FORM 0x4d524f46 /* "FORM" */
#define CREA 0x61657243 /* "Crea" */
44
45
46
static int audio_opened = 0;
static SDL_AudioSpec mixer;
47
static SDL_AudioDeviceID audio_device;
48
49
50
typedef struct _Mix_effectinfo
{
51
52
53
54
Mix_EffectFunc_t callback;
Mix_EffectDone_t done_callback;
void *udata;
struct _Mix_effectinfo *next;
55
56
} effect_info;
57
static struct _Mix_Channel {
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
Mix_Chunk *chunk;
int playing;
int paused;
Uint8 *samples;
int volume;
int looping;
int tag;
Uint32 expire;
Uint32 start_time;
Mix_Fading fading;
int fade_volume;
int fade_volume_reset;
Uint32 fade_length;
Uint32 ticks_fade;
effect_info *effects;
73
} *mix_channel = NULL;
74
75
76
static effect_info *posteffects = NULL;
77
static int num_channels;
78
static int reserved_channels = 0;
79
80
81
/* Support for hooking into the mixer callback system */
82
static void (SDLCALL *mix_postmix)(void *udata, Uint8 *stream, int len) = NULL;
83
static void *mix_postmix_data = NULL;
84
85
/* rcg07062001 callback to alert when channels are done playing. */
86
static void (SDLCALL *channel_done_callback)(int channel) = NULL;
87
88
/* Support for user defined music functions */
89
static void (SDLCALL *mix_music)(void *udata, Uint8 *stream, int len) = music_mixer;
90
static void *music_data = NULL;
91
92
93
94
95
/* rcg06042009 report available decoders at runtime. */
static const char **chunk_decoders = NULL;
static int num_decoders = 0;
96
97
int Mix_GetNumChunkDecoders(void)
98
{
99
return(num_decoders);
100
101
102
103
}
const char *Mix_GetChunkDecoder(int index)
{
104
105
106
107
if ((index < 0) || (index >= num_decoders)) {
return NULL;
}
return(chunk_decoders[index]);
108
109
}
110
111
112
113
114
115
116
117
118
119
120
SDL_bool Mix_HasChunkDecoder(const char *name)
{
int index;
for (index = 0; index < num_decoders; ++index) {
if (SDL_strcasecmp(name, chunk_decoders[index]) == 0) {
return SDL_TRUE;
}
}
return SDL_FALSE;
}
121
122
static void add_chunk_decoder(const char *decoder)
{
123
void *ptr = SDL_realloc((void *)chunk_decoders, (num_decoders + 1) * sizeof (const char *));
124
125
126
127
128
if (ptr == NULL) {
return; /* oh well, go on without it. */
}
chunk_decoders = (const char **) ptr;
chunk_decoders[num_decoders++] = decoder;
129
}
130
131
132
133
/* rcg06192001 get linked library's version. */
const SDL_version *Mix_Linked_Version(void)
{
134
135
136
static SDL_version linked_version;
SDL_MIXER_VERSION(&linked_version);
return(&linked_version);
137
138
}
139
140
int Mix_Init(int flags)
{
141
int result = 0;
142
143
144
load_music();
145
if (flags & MIX_INIT_FLAC) {
146
if (has_music(MUS_FLAC)) {
147
result |= MIX_INIT_FLAC;
148
149
} else {
Mix_SetError("FLAC support not available");
150
151
152
}
}
if (flags & MIX_INIT_MOD) {
153
if (has_music(MUS_MOD)) {
154
result |= MIX_INIT_MOD;
155
156
} else {
Mix_SetError("MOD support not available");
157
158
159
}
}
if (flags & MIX_INIT_MP3) {
160
if (has_music(MUS_MP3)) {
161
result |= MIX_INIT_MP3;
162
163
} else {
Mix_SetError("MP3 support not available");
164
165
166
}
}
if (flags & MIX_INIT_OGG) {
167
if (has_music(MUS_OGG)) {
168
result |= MIX_INIT_OGG;
169
170
171
172
173
174
175
176
177
} else {
Mix_SetError("OGG support not available");
}
}
if (flags & MIX_INIT_MID) {
if (has_music(MUS_MID)) {
result |= MIX_INIT_MID;
} else {
Mix_SetError("MIDI support not available");
178
179
180
}
}
return (result);
181
182
183
184
}
void Mix_Quit()
{
185
unload_music();
186
187
}
188
static int _Mix_remove_all_effects(int channel, effect_info **e);
189
190
191
/*
* rcg06122001 Cleanup effect callbacks.
192
* MAKE SURE Mix_LockAudio() is called before this (or you're in the
193
194
195
* audio callback).
*/
static void _Mix_channel_done_playing(int channel)
196
{
197
198
199
200
201
202
203
204
205
if (channel_done_callback) {
channel_done_callback(channel);
}
/*
* Call internal function directly, to avoid locking audio from
* inside audio callback.
*/
_Mix_remove_all_effects(channel, &mix_channel[channel].effects);
206
207
208
209
210
}
static void *Mix_DoEffects(int chan, void *snd, int len)
{
211
212
213
214
215
216
217
218
219
220
221
int posteffect = (chan == MIX_CHANNEL_POST);
effect_info *e = ((posteffect) ? posteffects : mix_channel[chan].effects);
void *buf = snd;
if (e != NULL) { /* are there any registered effects? */
/* if this is the postmix, we can just overwrite the original. */
if (!posteffect) {
buf = SDL_malloc(len);
if (buf == NULL) {
return(snd);
}
222
SDL_memcpy(buf, snd, len);
223
224
225
226
227
228
229
230
231
232
233
}
for (; e != NULL; e = e->next) {
if (e->callback != NULL) {
e->callback(chan, buf, len, e->udata);
}
}
}
/* be sure to SDL_free() the return value if != snd ... */
return(buf);
234
235
236
}
237
/* Mixing function */
238
239
static void SDLCALL
mix_channels(void *udata, Uint8 *stream, int len)
240
{
241
Uint8 *mix_input;
242
int i, mixable, volume = MIX_MAX_VOLUME;
243
Uint32 sdl_ticks;
244
245
#if SDL_VERSION_ATLEAST(1, 3, 0)
246
/* Need to initialize the stream in SDL 1.3+ */
247
SDL_memset(stream, mixer.silence, len);
248
#endif
249
250
/* Mix the music (must be done before the channels are added) */
251
mix_music(music_data, stream, len);
252
253
254
/* Mix any playing channels... */
sdl_ticks = SDL_GetTicks();
255
256
257
for (i=0; i<num_channels; ++i) {
if (!mix_channel[i].paused) {
if (mix_channel[i].expire > 0 && mix_channel[i].expire < sdl_ticks) {
258
259
260
261
262
263
/* Expiration delay for that channel is reached */
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].fading = MIX_NO_FADING;
mix_channel[i].expire = 0;
_Mix_channel_done_playing(i);
264
} else if (mix_channel[i].fading != MIX_NO_FADING) {
265
Uint32 ticks = sdl_ticks - mix_channel[i].ticks_fade;
266
if (ticks >= mix_channel[i].fade_length) {
267
Mix_Volume(i, mix_channel[i].fade_volume_reset); /* Restore the volume */
268
if(mix_channel[i].fading == MIX_FADING_OUT) {
269
270
271
272
273
274
275
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].expire = 0;
_Mix_channel_done_playing(i);
}
mix_channel[i].fading = MIX_NO_FADING;
} else {
276
if (mix_channel[i].fading == MIX_FADING_OUT) {
277
Mix_Volume(i, (mix_channel[i].fade_volume * (mix_channel[i].fade_length-ticks))
278
/ mix_channel[i].fade_length);
279
} else {
280
Mix_Volume(i, (mix_channel[i].fade_volume * ticks) / mix_channel[i].fade_length);
281
282
283
}
}
}
284
if (mix_channel[i].playing > 0) {
285
286
287
288
289
290
int index = 0;
int remaining = len;
while (mix_channel[i].playing > 0 && index < len) {
remaining = len - index;
volume = (mix_channel[i].volume*mix_channel[i].chunk->volume) / MIX_MAX_VOLUME;
mixable = mix_channel[i].playing;
291
if (mixable > remaining) {
292
293
294
295
mixable = remaining;
}
mix_input = Mix_DoEffects(i, mix_channel[i].samples, mixable);
296
SDL_MixAudioFormat(stream+index,mix_input,mixer.format,mixable,volume);
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
if (mix_input != mix_channel[i].samples)
SDL_free(mix_input);
mix_channel[i].samples += mixable;
mix_channel[i].playing -= mixable;
index += mixable;
/* rcg06072001 Alert app if channel is done playing. */
if (!mix_channel[i].playing && !mix_channel[i].looping) {
_Mix_channel_done_playing(i);
}
}
/* If looping the sample and we are at its end, make sure
we will still return a full buffer */
312
while (mix_channel[i].looping && index < len) {
313
314
315
316
317
318
319
int alen = mix_channel[i].chunk->alen;
remaining = len - index;
if (remaining > alen) {
remaining = alen;
}
mix_input = Mix_DoEffects(i, mix_channel[i].chunk->abuf, remaining);
320
SDL_MixAudioFormat(stream+index, mix_input, mixer.format, remaining, volume);
321
322
323
if (mix_input != mix_channel[i].chunk->abuf)
SDL_free(mix_input);
324
325
326
if (mix_channel[i].looping > 0) {
--mix_channel[i].looping;
}
327
328
329
330
mix_channel[i].samples = mix_channel[i].chunk->abuf + remaining;
mix_channel[i].playing = mix_channel[i].chunk->alen - remaining;
index += remaining;
}
331
if (! mix_channel[i].playing && mix_channel[i].looping) {
332
333
334
if (mix_channel[i].looping > 0) {
--mix_channel[i].looping;
}
335
336
337
338
339
340
341
342
343
344
mix_channel[i].samples = mix_channel[i].chunk->abuf;
mix_channel[i].playing = mix_channel[i].chunk->alen;
}
}
}
}
/* rcg06122001 run posteffects... */
Mix_DoEffects(MIX_CHANNEL_POST, stream, len);
345
if (mix_postmix) {
346
347
mix_postmix(mix_postmix_data, stream, len);
}
348
349
}
350
#if 0
351
352
static void PrintFormat(char *title, SDL_AudioSpec *fmt)
{
353
354
355
356
printf("%s: %d bit %s audio (%s) at %u Hz\n", title, (fmt->format&0xFF),
(fmt->format&0x8000) ? "signed" : "unsigned",
(fmt->channels > 2) ? "surround" :
(fmt->channels > 1) ? "stereo" : "mono", fmt->freq);
357
}
358
#endif
359
360
/* Open the mixer with a certain desired audio format */
361
int Mix_OpenAudioDevice(int frequency, Uint16 format, int nchannels, int chunksize,
362
const char* device, int allowed_changes)
363
{
364
365
366
int i;
SDL_AudioSpec desired;
367
368
369
370
371
372
373
374
375
/* This used to call SDL_OpenAudio(), which initializes the audio
subsystem if necessary. Since SDL_OpenAudioDevice() doesn't,
we have to handle this case here. */
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) {
return -1;
}
}
376
/* If the mixer is already opened, increment open count */
377
378
if (audio_opened) {
if (format == mixer.format && nchannels == mixer.channels) {
379
380
381
++audio_opened;
return(0);
}
382
while (audio_opened) {
383
384
385
386
387
388
389
390
391
392
393
394
395
Mix_CloseAudio();
}
}
/* Set the desired format and frequency */
desired.freq = frequency;
desired.format = format;
desired.channels = nchannels;
desired.samples = chunksize;
desired.callback = mix_channels;
desired.userdata = NULL;
/* Accept nearly any audio format */
396
if ((audio_device = SDL_OpenAudioDevice(device, 0, &desired, &mixer, allowed_changes)) == 0) {
397
398
return(-1);
}
399
#if 0
400
PrintFormat("Audio device", &mixer);
401
402
#endif
403
/* Initialize the music players */
404
405
load_music();
if (open_music(&mixer) < 0) {
406
SDL_CloseAudioDevice(audio_device);
407
408
409
410
411
412
413
return(-1);
}
num_channels = MIX_CHANNELS;
mix_channel = (struct _Mix_Channel *) SDL_malloc(num_channels * sizeof(struct _Mix_Channel));
/* Clear out the audio channels */
414
for (i=0; i<num_channels; ++i) {
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
mix_channel[i].chunk = NULL;
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].volume = SDL_MIX_MAXVOLUME;
mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;
mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;
mix_channel[i].fading = MIX_NO_FADING;
mix_channel[i].tag = -1;
mix_channel[i].expire = 0;
mix_channel[i].effects = NULL;
mix_channel[i].paused = 0;
}
Mix_VolumeMusic(SDL_MIX_MAXVOLUME);
_Mix_InitEffects();
add_chunk_decoder("WAVE");
add_chunk_decoder("AIFF");
add_chunk_decoder("VOC");
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
if (has_music(MUS_MOD)) {
add_chunk_decoder("MOD");
}
if (has_music(MUS_MID)) {
add_chunk_decoder("MID");
}
if (has_music(MUS_OGG)) {
add_chunk_decoder("OGG");
}
if (has_music(MUS_MP3)) {
add_chunk_decoder("MP3");
}
if (has_music(MUS_FLAC)) {
add_chunk_decoder("FLAC");
}
449
450
audio_opened = 1;
451
SDL_PauseAudioDevice(audio_device, 0);
452
return(0);
453
454
}
455
456
457
/* Open the mixer with a certain desired audio format */
int Mix_OpenAudio(int frequency, Uint16 format, int nchannels, int chunksize)
{
458
459
460
return Mix_OpenAudioDevice(frequency, format, nchannels, chunksize, NULL,
SDL_AUDIO_ALLOW_FREQUENCY_CHANGE |
SDL_AUDIO_ALLOW_CHANNELS_CHANGE);
461
462
}
463
464
465
466
467
468
/* Dynamically change the number of channels managed by the mixer.
If decreasing the number of channels, the upper channels are
stopped.
*/
int Mix_AllocateChannels(int numchans)
{
469
if (numchans<0 || numchans==num_channels)
470
471
return(num_channels);
472
if (numchans < num_channels) {
473
474
475
476
477
478
479
/* Stop the affected channels */
int i;
for(i=numchans; i < num_channels; i++) {
Mix_UnregisterAllEffects(i);
Mix_HaltChannel(i);
}
}
480
Mix_LockAudio();
481
mix_channel = (struct _Mix_Channel *) SDL_realloc(mix_channel, numchans * sizeof(struct _Mix_Channel));
482
if (numchans > num_channels) {
483
484
485
486
487
488
/* Initialize the new channels */
int i;
for(i=num_channels; i < numchans; i++) {
mix_channel[i].chunk = NULL;
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
489
490
491
mix_channel[i].volume = MIX_MAX_VOLUME;
mix_channel[i].fade_volume = MIX_MAX_VOLUME;
mix_channel[i].fade_volume_reset = MIX_MAX_VOLUME;
492
493
494
495
496
497
498
499
mix_channel[i].fading = MIX_NO_FADING;
mix_channel[i].tag = -1;
mix_channel[i].expire = 0;
mix_channel[i].effects = NULL;
mix_channel[i].paused = 0;
}
}
num_channels = numchans;
500
Mix_UnlockAudio();
501
return(num_channels);
502
503
}
504
505
506
/* Return the actual mixer parameters */
int Mix_QuerySpec(int *frequency, Uint16 *format, int *channels)
{
507
508
if (audio_opened) {
if (frequency) {
509
510
*frequency = mixer.freq;
}
511
if (format) {
512
513
*format = mixer.format;
}
514
if (channels) {
515
516
517
518
*channels = mixer.channels;
}
}
return(audio_opened);
519
520
}
521
typedef struct _MusicFragment
522
{
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
Uint8 *data;
int size;
struct _MusicFragment *next;
} MusicFragment;
static SDL_AudioSpec *Mix_LoadMusic_RW(Mix_MusicType music_type, SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
{
int i;
Mix_MusicInterface *interface = NULL;
void *music = NULL;
Sint64 start;
SDL_bool playing;
MusicFragment *first = NULL, *last = NULL, *fragment = NULL;
int count = 0;
int fragment_size;
*spec = mixer;
/* Use fragments sized on full audio frame boundaries - this'll do */
fragment_size = spec->size;
start = SDL_RWtell(src);
for (i = 0; i < get_num_music_interfaces(); ++i) {
interface = get_music_interface(i);
if (interface->type != music_type) {
continue;
}
if (!interface->CreateFromRW || !interface->GetAudio) {
continue;
}
/* These music interfaces are not safe to use while music is playing */
if (interface->api == MIX_MUSIC_CMD ||
interface->api == MIX_MUSIC_MIKMOD ||
interface->api == MIX_MUSIC_NATIVEMIDI) {
continue;
}
music = interface->CreateFromRW(src, freesrc);
if (music) {
/* The interface owns the data source now */
freesrc = SDL_FALSE;
break;
}
/* Reset the stream for the next decoder */
SDL_RWseek(src, start, RW_SEEK_SET);
570
571
}
572
573
574
575
576
577
578
579
580
581
582
if (!music) {
if (freesrc) {
SDL_RWclose(src);
}
Mix_SetError("Unrecognized audio format");
return NULL;
}
Mix_LockAudio();
if (interface->Play) {
583
interface->Play(music, 1);
584
}
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
playing = SDL_TRUE;
while (playing) {
int left;
fragment = (MusicFragment *)SDL_malloc(sizeof(*fragment));
if (!fragment) {
/* Uh oh, out of memory, let's return what we have */
break;
}
fragment->data = (Uint8 *)SDL_malloc(fragment_size);
if (!fragment->data) {
/* Uh oh, out of memory, let's return what we have */
SDL_free(fragment);
break;
}
fragment->next = NULL;
left = interface->GetAudio(music, fragment->data, fragment_size);
if (left > 0) {
playing = SDL_FALSE;
} else if (interface->IsPlaying) {
playing = interface->IsPlaying(music);
}
fragment->size = (fragment_size - left);
if (!first) {
first = fragment;
}
if (last) {
last->next = fragment;
}
last = fragment;
++count;
}
if (interface->Stop) {
interface->Stop(music);
}
if (music) {
interface->Delete(music);
627
}
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
Mix_UnlockAudio();
if (count > 0) {
*audio_len = (count - 1) * fragment_size + fragment->size;
*audio_buf = (Uint8 *)SDL_malloc(*audio_len);
if (*audio_buf) {
Uint8 *dst = *audio_buf;
for (fragment = first; fragment; fragment = fragment->next) {
SDL_memcpy(dst, fragment->data, fragment->size);
dst += fragment->size;
}
} else {
SDL_OutOfMemory();
spec = NULL;
}
} else {
Mix_SetError("No audio data");
spec = NULL;
}
while (first) {
fragment = first;
first = first->next;
SDL_free(fragment->data);
SDL_free(fragment);
}
if (freesrc) {
SDL_RWclose(src);
}
return spec;
660
}
661
662
663
664
/* Load a wave file */
Mix_Chunk *Mix_LoadWAV_RW(SDL_RWops *src, int freesrc)
{
665
Uint8 magic[4];
666
667
668
669
670
671
Mix_Chunk *chunk;
SDL_AudioSpec wavespec, *loaded;
SDL_AudioCVT wavecvt;
int samplesize;
/* rcg06012001 Make sure src is valid */
672
if (!src) {
673
674
675
676
677
SDL_SetError("Mix_LoadWAV_RW with NULL src");
return(NULL);
}
/* Make sure audio has been opened */
678
if (!audio_opened) {
679
SDL_SetError("Audio device hasn't been opened");
680
if (freesrc) {
681
682
683
684
685
686
687
SDL_RWclose(src);
}
return(NULL);
}
/* Allocate the chunk memory */
chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk));
688
if (chunk == NULL) {
689
SDL_SetError("Out of memory");
690
if (freesrc) {
691
692
693
694
695
696
SDL_RWclose(src);
}
return(NULL);
}
/* Find out what kind of audio file this is */
697
698
699
700
701
702
703
if (SDL_RWread(src, magic, 1, 4) != 4) {
if (freesrc) {
SDL_RWclose(src);
}
Mix_SetError("Couldn't read first 4 bytes of audio data");
return NULL;
}
704
/* Seek backwards for compatibility with older loaders */
705
706
707
708
709
710
711
712
713
714
715
SDL_RWseek(src, -4, RW_SEEK_CUR);
if (SDL_memcmp(magic, "WAVE", 4) == 0 || SDL_memcmp(magic, "RIFF", 4) == 0) {
loaded = SDL_LoadWAV_RW(src, freesrc, &wavespec, (Uint8 **)&chunk->abuf, &chunk->alen);
} else if (SDL_memcmp(magic, "FORM", 4) == 0) {
loaded = Mix_LoadAIFF_RW(src, freesrc, &wavespec, (Uint8 **)&chunk->abuf, &chunk->alen);
} else if (SDL_memcmp(magic, "CREA", 4) == 0) {
loaded = Mix_LoadVOC_RW(src, freesrc, &wavespec, (Uint8 **)&chunk->abuf, &chunk->alen);
} else {
Mix_MusicType music_type = detect_music_type_from_magic(magic);
loaded = Mix_LoadMusic_RW(music_type, src, freesrc, &wavespec, (Uint8 **)&chunk->abuf, &chunk->alen);
716
}
717
if (!loaded) {
718
719
720
721
/* The individual loaders have closed src if needed */
SDL_free(chunk);
return(NULL);
}
722
723
#if 0
724
725
PrintFormat("Audio device", &mixer);
PrintFormat("-- Wave file", &wavespec);
726
727
#endif
728
/* Build the audio converter and create conversion buffers */
729
if (wavespec.format != mixer.format ||
730
wavespec.channels != mixer.channels ||
731
732
wavespec.freq != mixer.freq) {
if (SDL_BuildAudioCVT(&wavecvt,
733
wavespec.format, wavespec.channels, wavespec.freq,
734
mixer.format, mixer.channels, mixer.freq) < 0) {
735
736
737
738
739
740
741
SDL_free(chunk->abuf);
SDL_free(chunk);
return(NULL);
}
samplesize = ((wavespec.format & 0xFF)/8)*wavespec.channels;
wavecvt.len = chunk->alen & ~(samplesize-1);
wavecvt.buf = (Uint8 *)SDL_calloc(1, wavecvt.len*wavecvt.len_mult);
742
if (wavecvt.buf == NULL) {
743
744
745
746
747
SDL_SetError("Out of memory");
SDL_free(chunk->abuf);
SDL_free(chunk);
return(NULL);
}
748
SDL_memcpy(wavecvt.buf, chunk->abuf, wavecvt.len);
749
750
751
SDL_free(chunk->abuf);
/* Run the audio converter */
752
if (SDL_ConvertAudio(&wavecvt) < 0) {
753
754
755
756
757
758
759
760
761
762
763
764
765
SDL_free(wavecvt.buf);
SDL_free(chunk);
return(NULL);
}
chunk->abuf = wavecvt.buf;
chunk->alen = wavecvt.len_cvt;
}
chunk->allocated = 1;
chunk->volume = MIX_MAX_VOLUME;
return(chunk);
766
767
}
768
769
770
/* Load a wave file of the mixer format from a memory buffer */
Mix_Chunk *Mix_QuickLoad_WAV(Uint8 *mem)
{
771
772
773
774
Mix_Chunk *chunk;
Uint8 magic[4];
/* Make sure audio has been opened */
775
if (! audio_opened) {
776
777
778
779
780
781
SDL_SetError("Audio device hasn't been opened");
return(NULL);
}
/* Allocate the chunk memory */
chunk = (Mix_Chunk *)SDL_calloc(1,sizeof(Mix_Chunk));
782
if (chunk == NULL) {
783
784
785
786
787
788
789
790
SDL_SetError("Out of memory");
return(NULL);
}
/* Essentially just skip to the audio data (no error checking - fast) */
chunk->allocated = 0;
mem += 12; /* WAV header */
do {
791
SDL_memcpy(magic, mem, 4);
792
793
794
795
796
mem += 4;
chunk->alen = ((mem[3]<<24)|(mem[2]<<16)|(mem[1]<<8)|(mem[0]));
mem += 4;
chunk->abuf = mem;
mem += chunk->alen;
797
} while (memcmp(magic, "data", 4) != 0);
798
799
800
chunk->volume = MIX_MAX_VOLUME;
return(chunk);
801
802
}
803
804
805
/* Load raw audio data of the mixer format from a memory buffer */
Mix_Chunk *Mix_QuickLoad_RAW(Uint8 *mem, Uint32 len)
{
806
807
808
Mix_Chunk *chunk;
/* Make sure audio has been opened */
809
if (! audio_opened) {
810
811
812
813
814
815
SDL_SetError("Audio device hasn't been opened");
return(NULL);
}
/* Allocate the chunk memory */
chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk));
816
if (chunk == NULL) {
817
818
819
820
821
822
823
824
825
826
827
SDL_SetError("Out of memory");
return(NULL);
}
/* Essentially just point at the audio data (no error checking - fast) */
chunk->allocated = 0;
chunk->alen = len;
chunk->abuf = mem;
chunk->volume = MIX_MAX_VOLUME;
return(chunk);
828
829
}
830
831
832
/* Free an audio chunk previously loaded */
void Mix_FreeChunk(Mix_Chunk *chunk)
{
833
834
835
int i;
/* Caution -- if the chunk is playing, the mixer will crash */
836
if (chunk) {
837
/* Guarantee that this chunk isn't playing */
838
Mix_LockAudio();
839
840
841
if (mix_channel) {
for (i=0; i<num_channels; ++i) {
if (chunk == mix_channel[i].chunk) {
842
843
844
845
846
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
}
}
}
847
Mix_UnlockAudio();
848
/* Actually free the chunk */
849
if (chunk->allocated) {
850
851
852
853
SDL_free(chunk->abuf);
}
SDL_free(chunk);
}
854
855
}
856
857
858
859
/* Set a function that is called after all mixing is performed.
This can be used to provide real-time visual display of the audio stream
or add a custom mixer filter for the stream data.
*/
860
void Mix_SetPostMix(void (SDLCALL *mix_func)
861
862
(void *udata, Uint8 *stream, int len), void *arg)
{
863
Mix_LockAudio();
864
865
mix_postmix_data = arg;
mix_postmix = mix_func;
866
Mix_UnlockAudio();
867
868
}
869
870
871
/* Add your own music player or mixer function.
If 'mix_func' is NULL, the default music player is re-enabled.
*/
872
void Mix_HookMusic(void (SDLCALL *mix_func)(void *udata, Uint8 *stream, int len),
873
874
void *arg)
{
875
Mix_LockAudio();
876
if (mix_func != NULL) {
877
878
879
880
881
882
music_data = arg;
mix_music = mix_func;
} else {
music_data = NULL;
mix_music = music_mixer;
}
883
Mix_UnlockAudio();
884
885
886
887
}
void *Mix_GetMusicHookData(void)
{
888
return(music_data);
889
890
}
891
void Mix_ChannelFinished(void (SDLCALL *channel_finished)(int channel))
892
{
893
Mix_LockAudio();
894
channel_done_callback = channel_finished;
895
Mix_UnlockAudio();
896
897
898
}
899
900
901
902
903
904
/* Reserve the first channels (0 -> n-1) for the application, i.e. don't allocate
them dynamically to the next sample if requested with a -1 value below.
Returns the number of reserved channels.
*/
int Mix_ReserveChannels(int num)
{
905
906
907
908
if (num > num_channels)
num = num_channels;
reserved_channels = num;
return num;
909
910
}
911
912
static int checkchunkintegral(Mix_Chunk *chunk)
{
913
int frame_width = 1;
914
915
916
917
918
if ((mixer.format & 0xFF) == 16) frame_width = 2;
frame_width *= mixer.channels;
while (chunk->alen % frame_width) chunk->alen--;
return chunk->alen;
919
920
}
921
922
/* Play an audio chunk on a specific channel.
If the specified channel is -1, play on the first free channel.
923
924
'ticks' is the number of milliseconds at most to play the sample, or -1
if there is no limit.
925
926
Returns which channel was used to play the sound.
*/
927
int Mix_PlayChannelTimed(int which, Mix_Chunk *chunk, int loops, int ticks)
928
{
929
930
931
int i;
/* Don't play null pointers :-) */
932
if (chunk == NULL) {
933
934
935
Mix_SetError("Tried to play a NULL chunk");
return(-1);
}
936
if (!checkchunkintegral(chunk)) {
937
938
939
940
941
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
/* Lock the mixer while modifying the playing channels */
942
Mix_LockAudio();
943
944
{
/* If which is -1, play on the first free channel */
945
946
947
if (which == -1) {
for (i=reserved_channels; i<num_channels; ++i) {
if (mix_channel[i].playing <= 0)
948
949
break;
}
950
if (i == num_channels) {
951
952
953
954
955
956
957
958
Mix_SetError("No free channels available");
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
959
if (which >= 0 && which < num_channels) {
960
961
962
963
964
965
966
967
968
969
970
971
972
Uint32 sdl_ticks = SDL_GetTicks();
if (Mix_Playing(which))
_Mix_channel_done_playing(which);
mix_channel[which].samples = chunk->abuf;
mix_channel[which].playing = chunk->alen;
mix_channel[which].looping = loops;
mix_channel[which].chunk = chunk;
mix_channel[which].paused = 0;
mix_channel[which].fading = MIX_NO_FADING;
mix_channel[which].start_time = sdl_ticks;
mix_channel[which].expire = (ticks>0) ? (sdl_ticks + ticks) : 0;
}
}
973
Mix_UnlockAudio();
974
975
976
/* Return the channel on which the sound is being played */
return(which);
977
978
}
979
980
981
/* Change the expiration delay for a channel */
int Mix_ExpireChannel(int which, int ticks)
{
982
983
int status = 0;
984
if (which == -1) {
985
int i;
986
for (i=0; i < num_channels; ++ i) {
987
988
status += Mix_ExpireChannel(i, ticks);
}
989
} else if (which < num_channels) {
990
Mix_LockAudio();
991
mix_channel[which].expire = (ticks>0) ? (SDL_GetTicks() + ticks) : 0;
992
Mix_UnlockAudio();
993
994
995
++ status;
}
return(status);
996
997
}
998
/* Fade in a sound on a channel, over ms milliseconds */
999
int Mix_FadeInChannelTimed(int which, Mix_Chunk *chunk, int loops, int ms, int ticks)
1000
{