Mercurial > sdl-ios-xcode
annotate src/audio/paudio/SDL_paudio.c @ 1312:c9b51268668f
Updated copyright information and removed rcs id lines (problematic in branch merges)
I batch edited these files, so please let me know if I've accidentally removed anybody's
credit here.
author | Sam Lantinga <slouken@libsdl.org> |
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date | Wed, 01 Feb 2006 06:32:25 +0000 |
parents | 74212992fb08 |
children | 3692456e7b0f |
rev | line source |
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0 | 1 /* |
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Sam Lantinga <slouken@libsdl.org>
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2 SDL - Simple DirectMedia Layer |
c9b51268668f
Updated copyright information and removed rcs id lines (problematic in branch merges)
Sam Lantinga <slouken@libsdl.org>
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3 Copyright (C) 1997-2006 Sam Lantinga |
0 | 4 |
5 This library is free software; you can redistribute it and/or | |
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Sam Lantinga <slouken@libsdl.org>
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6 modify it under the terms of the GNU Lesser General Public |
0 | 7 License as published by the Free Software Foundation; either |
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Sam Lantinga <slouken@libsdl.org>
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8 version 2.1 of the License, or (at your option) any later version. |
0 | 9 |
10 This library is distributed in the hope that it will be useful, | |
11 but WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
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Sam Lantinga <slouken@libsdl.org>
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13 Lesser General Public License for more details. |
0 | 14 |
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15 You should have received a copy of the GNU Lesser General Public |
c9b51268668f
Updated copyright information and removed rcs id lines (problematic in branch merges)
Sam Lantinga <slouken@libsdl.org>
parents:
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16 License along with this library; if not, write to the Free Software |
c9b51268668f
Updated copyright information and removed rcs id lines (problematic in branch merges)
Sam Lantinga <slouken@libsdl.org>
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17 Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA |
0 | 18 |
19 Carsten Griwodz | |
20 griff@kom.tu-darmstadt.de | |
21 | |
22 based on linux/SDL_dspaudio.c by Sam Lantinga | |
23 */ | |
24 | |
25 /* Allow access to a raw mixing buffer */ | |
26 | |
27 #include <stdlib.h> | |
28 #include <stdio.h> | |
29 #include <string.h> | |
30 #include <errno.h> | |
31 #include <unistd.h> | |
32 #include <fcntl.h> | |
33 #include <sys/time.h> | |
34 #include <sys/ioctl.h> | |
35 #include <sys/stat.h> | |
36 | |
37 #include "SDL_audio.h" | |
38 #include "SDL_error.h" | |
39 #include "SDL_audiomem.h" | |
40 #include "SDL_audio_c.h" | |
41 #include "SDL_timer.h" | |
42 #include "SDL_audiodev_c.h" | |
43 #include "SDL_paudio.h" | |
44 | |
45 #define DEBUG_AUDIO 1 | |
46 | |
47 /* A conflict within AIX 4.3.3 <sys/> headers and probably others as well. | |
48 * I guess nobody ever uses audio... Shame over AIX header files. */ | |
49 #include <sys/machine.h> | |
50 #undef BIG_ENDIAN | |
51 #include <sys/audio.h> | |
52 | |
53 /* The tag name used by paud audio */ | |
54 #define Paud_DRIVER_NAME "paud" | |
55 | |
56 /* Open the audio device for playback, and don't block if busy */ | |
57 /* #define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) */ | |
58 #define OPEN_FLAGS O_WRONLY | |
59 | |
60 /* Audio driver functions */ | |
61 static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec); | |
62 static void Paud_WaitAudio(_THIS); | |
63 static void Paud_PlayAudio(_THIS); | |
64 static Uint8 *Paud_GetAudioBuf(_THIS); | |
65 static void Paud_CloseAudio(_THIS); | |
66 | |
67 /* Audio driver bootstrap functions */ | |
68 | |
69 static int Audio_Available(void) | |
70 { | |
71 int fd; | |
72 int available; | |
73 | |
74 available = 0; | |
75 fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0); | |
76 if ( fd >= 0 ) { | |
77 available = 1; | |
78 close(fd); | |
79 } | |
80 return(available); | |
81 } | |
82 | |
83 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
84 { | |
85 free(device->hidden); | |
86 free(device); | |
87 } | |
88 | |
89 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
90 { | |
91 SDL_AudioDevice *this; | |
92 | |
93 /* Initialize all variables that we clean on shutdown */ | |
94 this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice)); | |
95 if ( this ) { | |
96 memset(this, 0, (sizeof *this)); | |
97 this->hidden = (struct SDL_PrivateAudioData *) | |
98 malloc((sizeof *this->hidden)); | |
99 } | |
100 if ( (this == NULL) || (this->hidden == NULL) ) { | |
101 SDL_OutOfMemory(); | |
102 if ( this ) { | |
103 free(this); | |
104 } | |
105 return(0); | |
106 } | |
107 memset(this->hidden, 0, (sizeof *this->hidden)); | |
108 audio_fd = -1; | |
109 | |
110 /* Set the function pointers */ | |
111 this->OpenAudio = Paud_OpenAudio; | |
112 this->WaitAudio = Paud_WaitAudio; | |
113 this->PlayAudio = Paud_PlayAudio; | |
114 this->GetAudioBuf = Paud_GetAudioBuf; | |
115 this->CloseAudio = Paud_CloseAudio; | |
116 | |
117 this->free = Audio_DeleteDevice; | |
118 | |
119 return this; | |
120 } | |
121 | |
122 AudioBootStrap Paud_bootstrap = { | |
123 Paud_DRIVER_NAME, "AIX Paudio", | |
124 Audio_Available, Audio_CreateDevice | |
125 }; | |
126 | |
127 /* This function waits until it is possible to write a full sound buffer */ | |
128 static void Paud_WaitAudio(_THIS) | |
129 { | |
130 fd_set fdset; | |
131 | |
132 /* See if we need to use timed audio synchronization */ | |
133 if ( frame_ticks ) { | |
134 /* Use timer for general audio synchronization */ | |
135 Sint32 ticks; | |
136 | |
137 ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS; | |
138 if ( ticks > 0 ) { | |
139 SDL_Delay(ticks); | |
140 } | |
141 } else { | |
142 audio_buffer paud_bufinfo; | |
143 | |
144 /* Use select() for audio synchronization */ | |
145 struct timeval timeout; | |
146 FD_ZERO(&fdset); | |
147 FD_SET(audio_fd, &fdset); | |
148 | |
149 if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) { | |
150 #ifdef DEBUG_AUDIO | |
151 fprintf(stderr, "Couldn't get audio buffer information\n"); | |
152 #endif | |
153 timeout.tv_sec = 10; | |
154 timeout.tv_usec = 0; | |
155 } else { | |
156 long ms_in_buf = paud_bufinfo.write_buf_time; | |
157 timeout.tv_sec = ms_in_buf/1000; | |
158 ms_in_buf = ms_in_buf - timeout.tv_sec*1000; | |
159 timeout.tv_usec = ms_in_buf*1000; | |
160 #ifdef DEBUG_AUDIO | |
161 fprintf( stderr, | |
162 "Waiting for write_buf_time=%ld,%ld\n", | |
163 timeout.tv_sec, | |
164 timeout.tv_usec ); | |
165 #endif | |
166 } | |
167 | |
168 #ifdef DEBUG_AUDIO | |
169 fprintf(stderr, "Waiting for audio to get ready\n"); | |
170 #endif | |
171 if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 ) { | |
172 const char *message = "Audio timeout - buggy audio driver? (disabled)"; | |
173 /* | |
174 * In general we should never print to the screen, | |
175 * but in this case we have no other way of letting | |
176 * the user know what happened. | |
177 */ | |
178 fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message); | |
179 this->enabled = 0; | |
180 /* Don't try to close - may hang */ | |
181 audio_fd = -1; | |
182 #ifdef DEBUG_AUDIO | |
183 fprintf(stderr, "Done disabling audio\n"); | |
184 #endif | |
185 } | |
186 #ifdef DEBUG_AUDIO | |
187 fprintf(stderr, "Ready!\n"); | |
188 #endif | |
189 } | |
190 } | |
191 | |
192 static void Paud_PlayAudio(_THIS) | |
193 { | |
194 int written; | |
195 | |
196 /* Write the audio data, checking for EAGAIN on broken audio drivers */ | |
197 do { | |
198 written = write(audio_fd, mixbuf, mixlen); | |
199 if ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) ) { | |
200 SDL_Delay(1); /* Let a little CPU time go by */ | |
201 } | |
202 } while ( (written < 0) && | |
203 ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)) ); | |
204 | |
205 /* If timer synchronization is enabled, set the next write frame */ | |
206 if ( frame_ticks ) { | |
207 next_frame += frame_ticks; | |
208 } | |
209 | |
210 /* If we couldn't write, assume fatal error for now */ | |
211 if ( written < 0 ) { | |
212 this->enabled = 0; | |
213 } | |
214 #ifdef DEBUG_AUDIO | |
215 fprintf(stderr, "Wrote %d bytes of audio data\n", written); | |
216 #endif | |
217 } | |
218 | |
219 static Uint8 *Paud_GetAudioBuf(_THIS) | |
220 { | |
221 return mixbuf; | |
222 } | |
223 | |
224 static void Paud_CloseAudio(_THIS) | |
225 { | |
226 if ( mixbuf != NULL ) { | |
227 SDL_FreeAudioMem(mixbuf); | |
228 mixbuf = NULL; | |
229 } | |
230 if ( audio_fd >= 0 ) { | |
231 close(audio_fd); | |
232 audio_fd = -1; | |
233 } | |
234 } | |
235 | |
236 static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec) | |
237 { | |
238 char audiodev[1024]; | |
239 int format; | |
240 int bytes_per_sample; | |
241 Uint16 test_format; | |
242 audio_init paud_init; | |
243 audio_buffer paud_bufinfo; | |
244 audio_status paud_status; | |
245 audio_control paud_control; | |
246 audio_change paud_change; | |
247 | |
248 /* Reset the timer synchronization flag */ | |
249 frame_ticks = 0.0; | |
250 | |
251 /* Open the audio device */ | |
252 audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0); | |
253 if ( audio_fd < 0 ) { | |
254 SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno)); | |
255 return -1; | |
256 } | |
257 | |
258 /* | |
259 * We can't set the buffer size - just ask the device for the maximum | |
260 * that we can have. | |
261 */ | |
262 if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) { | |
263 SDL_SetError("Couldn't get audio buffer information"); | |
264 return -1; | |
265 } | |
266 | |
267 mixbuf = NULL; | |
268 | |
269 if ( spec->channels > 1 ) | |
270 spec->channels = 2; | |
271 else | |
272 spec->channels = 1; | |
273 | |
274 /* | |
275 * Fields in the audio_init structure: | |
276 * | |
277 * Ignored by us: | |
278 * | |
279 * paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only? | |
280 * paud.slot_number; * slot number of the adapter | |
281 * paud.device_id; * adapter identification number | |
282 * | |
283 * Input: | |
284 * | |
285 * paud.srate; * the sampling rate in Hz | |
286 * paud.bits_per_sample; * 8, 16, 32, ... | |
287 * paud.bsize; * block size for this rate | |
288 * paud.mode; * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX | |
289 * paud.channels; * 1=mono, 2=stereo | |
290 * paud.flags; * FIXED - fixed length data | |
291 * * LEFT_ALIGNED, RIGHT_ALIGNED (var len only) | |
292 * * TWOS_COMPLEMENT - 2's complement data | |
293 * * SIGNED - signed? comment seems wrong in sys/audio.h | |
294 * * BIG_ENDIAN | |
295 * paud.operation; * PLAY, RECORD | |
296 * | |
297 * Output: | |
298 * | |
299 * paud.flags; * PITCH - pitch is supported | |
300 * * INPUT - input is supported | |
301 * * OUTPUT - output is supported | |
302 * * MONITOR - monitor is supported | |
303 * * VOLUME - volume is supported | |
304 * * VOLUME_DELAY - volume delay is supported | |
305 * * BALANCE - balance is supported | |
306 * * BALANCE_DELAY - balance delay is supported | |
307 * * TREBLE - treble control is supported | |
308 * * BASS - bass control is supported | |
309 * * BESTFIT_PROVIDED - best fit returned | |
310 * * LOAD_CODE - DSP load needed | |
311 * paud.rc; * NO_PLAY - DSP code can't do play requests | |
312 * * NO_RECORD - DSP code can't do record requests | |
313 * * INVALID_REQUEST - request was invalid | |
314 * * CONFLICT - conflict with open's flags | |
315 * * OVERLOADED - out of DSP MIPS or memory | |
316 * paud.position_resolution; * smallest increment for position | |
317 */ | |
318 | |
319 paud_init.srate = spec->freq; | |
320 paud_init.mode = PCM; | |
321 paud_init.operation = PLAY; | |
322 paud_init.channels = spec->channels; | |
323 | |
324 /* Try for a closest match on audio format */ | |
325 format = 0; | |
326 for ( test_format = SDL_FirstAudioFormat(spec->format); | |
327 ! format && test_format; ) { | |
328 #ifdef DEBUG_AUDIO | |
329 fprintf(stderr, "Trying format 0x%4.4x\n", test_format); | |
330 #endif | |
331 switch ( test_format ) { | |
332 case AUDIO_U8: | |
333 bytes_per_sample = 1; | |
334 paud_init.bits_per_sample = 8; | |
335 paud_init.flags = TWOS_COMPLEMENT | FIXED; | |
336 format = 1; | |
337 break; | |
338 case AUDIO_S8: | |
339 bytes_per_sample = 1; | |
340 paud_init.bits_per_sample = 8; | |
341 paud_init.flags = SIGNED | | |
342 TWOS_COMPLEMENT | FIXED; | |
343 format = 1; | |
344 break; | |
345 case AUDIO_S16LSB: | |
346 bytes_per_sample = 2; | |
347 paud_init.bits_per_sample = 16; | |
348 paud_init.flags = SIGNED | | |
349 TWOS_COMPLEMENT | FIXED; | |
350 format = 1; | |
351 break; | |
352 case AUDIO_S16MSB: | |
353 bytes_per_sample = 2; | |
354 paud_init.bits_per_sample = 16; | |
355 paud_init.flags = BIG_ENDIAN | | |
356 SIGNED | | |
357 TWOS_COMPLEMENT | FIXED; | |
358 format = 1; | |
359 break; | |
360 case AUDIO_U16LSB: | |
361 bytes_per_sample = 2; | |
362 paud_init.bits_per_sample = 16; | |
363 paud_init.flags = TWOS_COMPLEMENT | FIXED; | |
364 format = 1; | |
365 break; | |
366 case AUDIO_U16MSB: | |
367 bytes_per_sample = 2; | |
368 paud_init.bits_per_sample = 16; | |
369 paud_init.flags = BIG_ENDIAN | | |
370 TWOS_COMPLEMENT | FIXED; | |
371 format = 1; | |
372 break; | |
373 default: | |
374 break; | |
375 } | |
376 if ( ! format ) { | |
377 test_format = SDL_NextAudioFormat(); | |
378 } | |
379 } | |
380 if ( format == 0 ) { | |
381 #ifdef DEBUG_AUDIO | |
382 fprintf(stderr, "Couldn't find any hardware audio formats\n"); | |
383 #endif | |
384 SDL_SetError("Couldn't find any hardware audio formats"); | |
385 return -1; | |
386 } | |
387 spec->format = test_format; | |
388 | |
389 /* | |
390 * We know the buffer size and the max number of subsequent writes | |
391 * that can be pending. If more than one can pend, allow the application | |
392 * to do something like double buffering between our write buffer and | |
393 * the device's own buffer that we are filling with write() anyway. | |
394 * | |
395 * We calculate spec->samples like this because SDL_CalculateAudioSpec() | |
396 * will give put paud_bufinfo.write_buf_cap (or paud_bufinfo.write_buf_cap/2) | |
397 * into spec->size in return. | |
398 */ | |
399 if ( paud_bufinfo.request_buf_cap == 1 ) | |
400 { | |
401 spec->samples = paud_bufinfo.write_buf_cap | |
402 / bytes_per_sample | |
403 / spec->channels; | |
404 } | |
405 else | |
406 { | |
407 spec->samples = paud_bufinfo.write_buf_cap | |
408 / bytes_per_sample | |
409 / spec->channels | |
410 / 2; | |
411 } | |
412 paud_init.bsize = bytes_per_sample * spec->channels; | |
413 | |
414 SDL_CalculateAudioSpec(spec); | |
415 | |
416 /* | |
417 * The AIX paud device init can't modify the values of the audio_init | |
418 * structure that we pass to it. So we don't need any recalculation | |
419 * of this stuff and no reinit call as in linux dsp and dma code. | |
420 * | |
421 * /dev/paud supports all of the encoding formats, so we don't need | |
422 * to do anything like reopening the device, either. | |
423 */ | |
424 if ( ioctl(audio_fd, AUDIO_INIT, &paud_init) < 0 ) { | |
425 switch ( paud_init.rc ) | |
426 { | |
427 case 1 : | |
428 SDL_SetError("Couldn't set audio format: DSP can't do play requests"); | |
429 return -1; | |
430 break; | |
431 case 2 : | |
432 SDL_SetError("Couldn't set audio format: DSP can't do record requests"); | |
433 return -1; | |
434 break; | |
435 case 4 : | |
436 SDL_SetError("Couldn't set audio format: request was invalid"); | |
437 return -1; | |
438 break; | |
439 case 5 : | |
440 SDL_SetError("Couldn't set audio format: conflict with open's flags"); | |
441 return -1; | |
442 break; | |
443 case 6 : | |
444 SDL_SetError("Couldn't set audio format: out of DSP MIPS or memory"); | |
445 return -1; | |
446 break; | |
447 default : | |
448 SDL_SetError("Couldn't set audio format: not documented in sys/audio.h"); | |
449 return -1; | |
450 break; | |
451 } | |
452 } | |
453 | |
454 /* Allocate mixing buffer */ | |
455 mixlen = spec->size; | |
456 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); | |
457 if ( mixbuf == NULL ) { | |
458 return -1; | |
459 } | |
460 memset(mixbuf, spec->silence, spec->size); | |
461 | |
462 /* | |
463 * Set some paramters: full volume, first speaker that we can find. | |
464 * Ignore the other settings for now. | |
465 */ | |
466 paud_change.input = AUDIO_IGNORE; /* the new input source */ | |
467 paud_change.output = OUTPUT_1; /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */ | |
468 paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */ | |
469 paud_change.volume = 0x7fffffff; /* volume level [0-0x7fffffff] */ | |
470 paud_change.volume_delay = AUDIO_IGNORE; /* the new volume delay */ | |
471 paud_change.balance = 0x3fffffff; /* the new balance */ | |
472 paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */ | |
473 paud_change.treble = AUDIO_IGNORE; /* the new treble state */ | |
474 paud_change.bass = AUDIO_IGNORE; /* the new bass state */ | |
475 paud_change.pitch = AUDIO_IGNORE; /* the new pitch state */ | |
476 | |
477 paud_control.ioctl_request = AUDIO_CHANGE; | |
478 paud_control.request_info = (char*)&paud_change; | |
479 if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) { | |
480 #ifdef DEBUG_AUDIO | |
481 fprintf(stderr, "Can't change audio display settings\n" ); | |
482 #endif | |
483 } | |
484 | |
485 /* | |
486 * Tell the device to expect data. Actual start will wait for | |
487 * the first write() call. | |
488 */ | |
489 paud_control.ioctl_request = AUDIO_START; | |
490 paud_control.position = 0; | |
491 if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) { | |
492 #ifdef DEBUG_AUDIO | |
493 fprintf(stderr, "Can't start audio play\n" ); | |
494 #endif | |
495 SDL_SetError("Can't start audio play"); | |
496 return -1; | |
497 } | |
498 | |
499 /* Check to see if we need to use select() workaround */ | |
500 { char *workaround; | |
501 workaround = getenv("SDL_DSP_NOSELECT"); | |
502 if ( workaround ) { | |
503 frame_ticks = (float)(spec->samples*1000)/spec->freq; | |
504 next_frame = SDL_GetTicks()+frame_ticks; | |
505 } | |
506 } | |
507 | |
508 /* Get the parent process id (we're the parent of the audio thread) */ | |
509 parent = getpid(); | |
510 | |
511 /* We're ready to rock and roll. :-) */ | |
512 return 0; | |
513 } | |
514 |