2 SDL - Simple DirectMedia Layer
3 Copyright (C) 1997-2006 Sam Lantinga
5 This library is free software; you can redistribute it and/or
6 modify it under the terms of the GNU Lesser General Public
7 License as published by the Free Software Foundation; either
8 version 2.1 of the License, or (at your option) any later version.
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
13 Lesser General Public License for more details.
15 You should have received a copy of the GNU Lesser General Public
16 License along with this library; if not, write to the Free Software
17 Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "SDL_config.h"
24 /* Allow access to a raw mixing buffer */
29 #include <sys/ioctl.h>
30 #include <sys/audioio.h>
33 #include <sys/audioio.h>
36 #include <sys/types.h>
40 #include "SDL_timer.h"
41 #include "SDL_audio.h"
42 #include "../SDL_audiomem.h"
43 #include "../SDL_audio_c.h"
44 #include "../SDL_audiodev_c.h"
45 #include "SDL_sunaudio.h"
47 /* Open the audio device for playback, and don't block if busy */
48 #define OPEN_FLAGS (O_WRONLY|O_NONBLOCK)
50 /* Audio driver functions */
51 static int DSP_OpenAudio(_THIS
, SDL_AudioSpec
*spec
);
52 static void DSP_WaitAudio(_THIS
);
53 static void DSP_PlayAudio(_THIS
);
54 static Uint8
*DSP_GetAudioBuf(_THIS
);
55 static void DSP_CloseAudio(_THIS
);
57 static Uint8
snd2au(int sample
);
59 /* Audio driver bootstrap functions */
61 static int Audio_Available(void)
67 fd
= SDL_OpenAudioPath(NULL
, 0, OPEN_FLAGS
, 1);
75 static void Audio_DeleteDevice(SDL_AudioDevice
*device
)
77 SDL_free(device
->hidden
);
81 static SDL_AudioDevice
*Audio_CreateDevice(int devindex
)
83 SDL_AudioDevice
*this;
85 /* Initialize all variables that we clean on shutdown */
86 this = (SDL_AudioDevice
*)SDL_malloc(sizeof(SDL_AudioDevice
));
88 SDL_memset(this, 0, (sizeof *this));
89 this->hidden
= (struct SDL_PrivateAudioData
*)
90 SDL_malloc((sizeof *this->hidden
));
92 if ( (this == NULL
) || (this->hidden
== NULL
) ) {
99 SDL_memset(this->hidden
, 0, (sizeof *this->hidden
));
102 /* Set the function pointers */
103 this->OpenAudio
= DSP_OpenAudio
;
104 this->WaitAudio
= DSP_WaitAudio
;
105 this->PlayAudio
= DSP_PlayAudio
;
106 this->GetAudioBuf
= DSP_GetAudioBuf
;
107 this->CloseAudio
= DSP_CloseAudio
;
109 this->free
= Audio_DeleteDevice
;
114 AudioBootStrap SUNAUDIO_bootstrap
= {
115 "audio", "UNIX /dev/audio interface",
116 Audio_Available
, Audio_CreateDevice
120 void CheckUnderflow(_THIS
)
126 ioctl(audio_fd
, AUDIO_GETINFO
, &info
);
127 left
= (written
- info
.play
.samples
);
128 if ( written
&& (left
== 0) ) {
129 fprintf(stderr
, "audio underflow!\n");
135 void DSP_WaitAudio(_THIS
)
138 #define SLEEP_FUDGE 10 /* 10 ms scheduling fudge factor */
142 ioctl(audio_fd
, AUDIO_GETINFO
, &info
);
143 left
= (written
- info
.play
.samples
);
144 if ( left
> fragsize
) {
147 sleepy
= ((left
- fragsize
)/frequency
);
148 sleepy
-= SLEEP_FUDGE
;
157 FD_SET(audio_fd
, &fdset
);
158 select(audio_fd
+1, NULL
, &fdset
, NULL
, NULL
);
162 void DSP_PlayAudio(_THIS
)
164 /* Write the audio data */
166 /* Assuming that this->spec.freq >= 8000 Hz */
167 int accum
, incr
, pos
;
171 incr
= this->spec
.freq
/8;
173 switch (audio_fmt
& 0xFF) {
178 for ( pos
=0; pos
< fragsize
; ++pos
) {
179 *aubuf
= snd2au((0x80-*sndbuf
)*64);
181 while ( accum
> 0 ) {
192 sndbuf
= (Sint16
*)mixbuf
;
193 for ( pos
=0; pos
< fragsize
; ++pos
) {
194 *aubuf
= snd2au(*sndbuf
/4);
196 while ( accum
> 0 ) {
206 CheckUnderflow(this);
208 if ( write(audio_fd
, ulaw_buf
, fragsize
) < 0 ) {
209 /* Assume fatal error, for now */
215 CheckUnderflow(this);
217 if ( write(audio_fd
, mixbuf
, this->spec
.size
) < 0 ) {
218 /* Assume fatal error, for now */
225 Uint8
*DSP_GetAudioBuf(_THIS
)
230 void DSP_CloseAudio(_THIS
)
232 if ( mixbuf
!= NULL
) {
233 SDL_FreeAudioMem(mixbuf
);
236 if ( ulaw_buf
!= NULL
) {
243 int DSP_OpenAudio(_THIS
, SDL_AudioSpec
*spec
)
249 int desired_freq
= spec
->freq
;
251 /* Initialize our freeable variables, in case we fail*/
256 /* Determine the audio parameters from the AudioSpec */
257 switch ( spec
->format
& 0xFF ) {
259 case 8: { /* Unsigned 8 bit audio data */
260 spec
->format
= AUDIO_U8
;
262 enc
= AUDIO_ENCODING_LINEAR8
;
267 case 16: { /* Signed 16 bit audio data */
268 spec
->format
= AUDIO_S16SYS
;
270 enc
= AUDIO_ENCODING_LINEAR
;
276 SDL_SetError("Unsupported audio format");
280 audio_fmt
= spec
->format
;
282 /* Open the audio device */
283 audio_fd
= SDL_OpenAudioPath(audiodev
, sizeof(audiodev
), OPEN_FLAGS
, 1);
284 if ( audio_fd
< 0 ) {
285 SDL_SetError("Couldn't open %s: %s", audiodev
,
290 ulaw_only
= 0; /* modern Suns do support linear audio */
294 AUDIO_INITINFO(&info
); /* init all fields to "no change" */
296 /* Try to set the requested settings */
297 info
.play
.sample_rate
= spec
->freq
;
298 info
.play
.channels
= spec
->channels
;
299 info
.play
.precision
= (enc
== AUDIO_ENCODING_ULAW
)
300 ? 8 : spec
->format
& 0xff;
301 info
.play
.encoding
= enc
;
302 if( ioctl(audio_fd
, AUDIO_SETINFO
, &info
) == 0 ) {
304 /* Check to be sure we got what we wanted */
305 if(ioctl(audio_fd
, AUDIO_GETINFO
, &info
) < 0) {
306 SDL_SetError("Error getting audio parameters: %s",
310 if(info
.play
.encoding
== enc
311 && info
.play
.precision
== (spec
->format
& 0xff)
312 && info
.play
.channels
== spec
->channels
) {
313 /* Yow! All seems to be well! */
314 spec
->freq
= info
.play
.sample_rate
;
320 case AUDIO_ENCODING_LINEAR8
:
321 /* unsigned 8bit apparently not supported here */
322 enc
= AUDIO_ENCODING_LINEAR
;
323 spec
->format
= AUDIO_S16SYS
;
324 break; /* try again */
326 case AUDIO_ENCODING_LINEAR
:
327 /* linear 16bit didn't work either, resort to ยต-law */
328 enc
= AUDIO_ENCODING_ULAW
;
331 spec
->format
= AUDIO_U8
;
337 SDL_SetError("Error setting audio parameters: %s",
342 #endif /* AUDIO_SETINFO */
345 /* We can actually convert on-the-fly to U-Law */
347 spec
->freq
= desired_freq
;
348 fragsize
= (spec
->samples
*1000)/(spec
->freq
/8);
350 ulaw_buf
= (Uint8
*)SDL_malloc(fragsize
);
351 if ( ulaw_buf
== NULL
) {
357 fragsize
= spec
->samples
;
358 frequency
= spec
->freq
/1000;
361 fprintf(stderr
, "Audio device %s U-Law only\n",
362 ulaw_only
? "is" : "is not");
363 fprintf(stderr
, "format=0x%x chan=%d freq=%d\n",
364 spec
->format
, spec
->channels
, spec
->freq
);
367 /* Update the fragment size as size in bytes */
368 SDL_CalculateAudioSpec(spec
);
370 /* Allocate mixing buffer */
371 mixbuf
= (Uint8
*)SDL_AllocAudioMem(spec
->size
);
372 if ( mixbuf
== NULL
) {
376 SDL_memset(mixbuf
, spec
->silence
, spec
->size
);
378 /* We're ready to rock and roll. :-) */
382 /************************************************************************/
383 /* This function (snd2au()) copyrighted: */
384 /************************************************************************/
385 /* Copyright 1989 by Rich Gopstein and Harris Corporation */
387 /* Permission to use, copy, modify, and distribute this software */
388 /* and its documentation for any purpose and without fee is */
389 /* hereby granted, provided that the above copyright notice */
390 /* appears in all copies and that both that copyright notice and */
391 /* this permission notice appear in supporting documentation, and */
392 /* that the name of Rich Gopstein and Harris Corporation not be */
393 /* used in advertising or publicity pertaining to distribution */
394 /* of the software without specific, written prior permission. */
395 /* Rich Gopstein and Harris Corporation make no representations */
396 /* about the suitability of this software for any purpose. It */
397 /* provided "as is" without express or implied warranty. */
398 /************************************************************************/
400 static Uint8
snd2au(int sample
)
413 sample
= 0xF0 | (15 - sample
/ 2);
414 } else if (sample
< 96) {
415 sample
= 0xE0 | (15 - (sample
- 32) / 4);
416 } else if (sample
< 224) {
417 sample
= 0xD0 | (15 - (sample
- 96) / 8);
418 } else if (sample
< 480) {
419 sample
= 0xC0 | (15 - (sample
- 224) / 16);
420 } else if (sample
< 992) {
421 sample
= 0xB0 | (15 - (sample
- 480) / 32);
422 } else if (sample
< 2016) {
423 sample
= 0xA0 | (15 - (sample
- 992) / 64);
424 } else if (sample
< 4064) {
425 sample
= 0x90 | (15 - (sample
- 2016) / 128);
426 } else if (sample
< 8160) {
427 sample
= 0x80 | (15 - (sample
- 4064) / 256);
431 return (mask
& sample
);