sync with en/mplayer.1 rev. 30822
[mplayer/glamo.git] / libao2 / ao_oss.c
blob6c23026e87936026b310254cca04e3f4adba3ff4
1 /*
2 * OSS audio output driver
4 * This file is part of MPlayer.
6 * MPlayer is free software; you can redistribute it and/or modify
7 * it under the terms of the GNU General Public License as published by
8 * the Free Software Foundation; either version 2 of the License, or
9 * (at your option) any later version.
11 * MPlayer is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14 * GNU General Public License for more details.
16 * You should have received a copy of the GNU General Public License along
17 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
18 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
21 #include <stdio.h>
22 #include <stdlib.h>
24 #include <sys/ioctl.h>
25 #include <unistd.h>
26 #include <sys/time.h>
27 #include <sys/types.h>
28 #include <sys/stat.h>
29 #include <fcntl.h>
30 #include <errno.h>
31 #include <string.h>
33 #include "config.h"
34 #include "mp_msg.h"
35 #include "mixer.h"
36 #include "help_mp.h"
38 #ifdef HAVE_SYS_SOUNDCARD_H
39 #include <sys/soundcard.h>
40 #else
41 #ifdef HAVE_SOUNDCARD_H
42 #include <soundcard.h>
43 #endif
44 #endif
46 #include "libaf/af_format.h"
48 #include "audio_out.h"
49 #include "audio_out_internal.h"
51 static const ao_info_t info =
53 "OSS/ioctl audio output",
54 "oss",
55 "A'rpi",
59 /* Support for >2 output channels added 2001-11-25 - Steve Davies <steve@daviesfam.org> */
61 LIBAO_EXTERN(oss)
63 static int format2oss(int format)
65 switch(format)
67 case AF_FORMAT_U8: return AFMT_U8;
68 case AF_FORMAT_S8: return AFMT_S8;
69 case AF_FORMAT_U16_LE: return AFMT_U16_LE;
70 case AF_FORMAT_U16_BE: return AFMT_U16_BE;
71 case AF_FORMAT_S16_LE: return AFMT_S16_LE;
72 case AF_FORMAT_S16_BE: return AFMT_S16_BE;
73 #ifdef AFMT_S24_PACKED
74 case AF_FORMAT_S24_LE: return AFMT_S24_PACKED;
75 #endif
76 #ifdef AFMT_U32_LE
77 case AF_FORMAT_U32_LE: return AFMT_U32_LE;
78 #endif
79 #ifdef AFMT_U32_BE
80 case AF_FORMAT_U32_BE: return AFMT_U32_BE;
81 #endif
82 #ifdef AFMT_S32_LE
83 case AF_FORMAT_S32_LE: return AFMT_S32_LE;
84 #endif
85 #ifdef AFMT_S32_BE
86 case AF_FORMAT_S32_BE: return AFMT_S32_BE;
87 #endif
88 #ifdef AFMT_FLOAT
89 case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT;
90 #endif
91 // SPECIALS
92 case AF_FORMAT_MU_LAW: return AFMT_MU_LAW;
93 case AF_FORMAT_A_LAW: return AFMT_A_LAW;
94 case AF_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM;
95 #ifdef AFMT_MPEG
96 case AF_FORMAT_MPEG2: return AFMT_MPEG;
97 #endif
98 #ifdef AFMT_AC3
99 case AF_FORMAT_AC3_NE: return AFMT_AC3;
100 #endif
102 mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", af_fmt2str_short(format));
103 return -1;
106 static int oss2format(int format)
108 switch(format)
110 case AFMT_U8: return AF_FORMAT_U8;
111 case AFMT_S8: return AF_FORMAT_S8;
112 case AFMT_U16_LE: return AF_FORMAT_U16_LE;
113 case AFMT_U16_BE: return AF_FORMAT_U16_BE;
114 case AFMT_S16_LE: return AF_FORMAT_S16_LE;
115 case AFMT_S16_BE: return AF_FORMAT_S16_BE;
116 #ifdef AFMT_S24_PACKED
117 case AFMT_S24_PACKED: return AF_FORMAT_S24_LE;
118 #endif
119 #ifdef AFMT_U32_LE
120 case AFMT_U32_LE: return AF_FORMAT_U32_LE;
121 #endif
122 #ifdef AFMT_U32_BE
123 case AFMT_U32_BE: return AF_FORMAT_U32_BE;
124 #endif
125 #ifdef AFMT_S32_LE
126 case AFMT_S32_LE: return AF_FORMAT_S32_LE;
127 #endif
128 #ifdef AFMT_S32_BE
129 case AFMT_S32_BE: return AF_FORMAT_S32_BE;
130 #endif
131 #ifdef AFMT_FLOAT
132 case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE;
133 #endif
134 // SPECIALS
135 case AFMT_MU_LAW: return AF_FORMAT_MU_LAW;
136 case AFMT_A_LAW: return AF_FORMAT_A_LAW;
137 case AFMT_IMA_ADPCM: return AF_FORMAT_IMA_ADPCM;
138 #ifdef AFMT_MPEG
139 case AFMT_MPEG: return AF_FORMAT_MPEG2;
140 #endif
141 #ifdef AFMT_AC3
142 case AFMT_AC3: return AF_FORMAT_AC3_NE;
143 #endif
145 mp_msg(MSGT_GLOBAL,MSGL_ERR,MSGTR_AO_OSS_UnknownUnsupportedFormat, format);
146 return -1;
149 static char *dsp=PATH_DEV_DSP;
150 static audio_buf_info zz;
151 static int audio_fd=-1;
152 static int prepause_space;
154 static const char *oss_mixer_device = PATH_DEV_MIXER;
155 static int oss_mixer_channel = SOUND_MIXER_PCM;
157 // to set/get/query special features/parameters
158 static int control(int cmd,void *arg){
159 switch(cmd){
160 case AOCONTROL_SET_DEVICE:
161 dsp=(char*)arg;
162 return CONTROL_OK;
163 case AOCONTROL_GET_DEVICE:
164 *(char**)arg=dsp;
165 return CONTROL_OK;
166 #ifdef SNDCTL_DSP_GETFMTS
167 case AOCONTROL_QUERY_FORMAT:
169 int format;
170 if (!ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format))
171 if ((unsigned int)format & (unsigned long)arg)
172 return CONTROL_TRUE;
173 return CONTROL_FALSE;
175 #endif
176 case AOCONTROL_GET_VOLUME:
177 case AOCONTROL_SET_VOLUME:
179 ao_control_vol_t *vol = (ao_control_vol_t *)arg;
180 int fd, v, devs;
182 if(AF_FORMAT_IS_AC3(ao_data.format))
183 return CONTROL_TRUE;
185 if ((fd = open(oss_mixer_device, O_RDONLY)) > 0)
187 ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
188 if (devs & (1 << oss_mixer_channel))
190 if (cmd == AOCONTROL_GET_VOLUME)
192 ioctl(fd, MIXER_READ(oss_mixer_channel), &v);
193 vol->right = (v & 0xFF00) >> 8;
194 vol->left = v & 0x00FF;
196 else
198 v = ((int)vol->right << 8) | (int)vol->left;
199 ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v);
202 else
204 close(fd);
205 return CONTROL_ERROR;
207 close(fd);
208 return CONTROL_OK;
211 return CONTROL_ERROR;
213 return CONTROL_UNKNOWN;
216 // open & setup audio device
217 // return: 1=success 0=fail
218 static int init(int rate,int channels,int format,int flags){
219 char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
220 int oss_format;
221 char *mdev = mixer_device, *mchan = mixer_channel;
223 mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels,
224 af_fmt2str_short(format));
226 if (ao_subdevice) {
227 char *m,*c;
228 m = strchr(ao_subdevice,':');
229 if(m) {
230 c = strchr(m+1,':');
231 if(c) {
232 mchan = c+1;
233 c[0] = '\0';
235 mdev = m+1;
236 m[0] = '\0';
238 dsp = ao_subdevice;
241 if(mdev)
242 oss_mixer_device=mdev;
243 else
244 oss_mixer_device=PATH_DEV_MIXER;
246 if(mchan){
247 int fd, devs, i;
249 if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){
250 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenMixer,
251 oss_mixer_device, strerror(errno));
252 }else{
253 ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
254 close(fd);
256 for (i=0; i<SOUND_MIXER_NRDEVICES; i++){
257 if(!strcasecmp(mixer_channels[i], mchan)){
258 if(!(devs & (1 << i))){
259 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,mchan);
260 i = SOUND_MIXER_NRDEVICES+1;
261 break;
263 oss_mixer_channel = i;
264 break;
267 if(i==SOUND_MIXER_NRDEVICES){
268 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,mchan);
271 } else
272 oss_mixer_channel = SOUND_MIXER_PCM;
274 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' dsp device\n", dsp);
275 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", oss_mixer_device);
276 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", mixer_channels[oss_mixer_channel]);
278 #ifdef __linux__
279 audio_fd=open(dsp, O_WRONLY | O_NONBLOCK);
280 #else
281 audio_fd=open(dsp, O_WRONLY);
282 #endif
283 if(audio_fd<0){
284 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenDev, dsp, strerror(errno));
285 return 0;
288 #ifdef __linux__
289 /* Remove the non-blocking flag */
290 if(fcntl(audio_fd, F_SETFL, 0) < 0) {
291 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantMakeFd, strerror(errno));
292 return 0;
294 #endif
296 #if defined(FD_CLOEXEC) && defined(F_SETFD)
297 fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
298 #endif
300 if(AF_FORMAT_IS_AC3(format)) {
301 ao_data.samplerate=rate;
302 ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
305 ac3_retry:
306 if (AF_FORMAT_IS_AC3(format))
307 format = AF_FORMAT_AC3_NE;
308 ao_data.format=format;
309 oss_format=format2oss(format);
310 if (oss_format == -1) {
311 #if HAVE_BIGENDIAN
312 oss_format=AFMT_S16_BE;
313 #else
314 oss_format=AFMT_S16_LE;
315 #endif
316 format=AF_FORMAT_S16_NE;
318 if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format)<0 ||
319 oss_format != format2oss(format)) {
320 mp_msg(MSGT_AO,MSGL_WARN, MSGTR_AO_OSS_CantSet, dsp,
321 af_fmt2str_short(format), af_fmt2str_short(AF_FORMAT_S16_NE) );
322 format=AF_FORMAT_S16_NE;
323 goto ac3_retry;
325 #if 0
326 if(oss_format!=format2oss(format))
327 mp_msg(MSGT_AO,MSGL_WARN,"WARNING! Your soundcard does NOT support %s sample format! Broken audio or bad playback speed are possible! Try with '-af format'\n",audio_out_format_name(format));
328 #endif
330 ao_data.format = oss2format(oss_format);
331 if (ao_data.format == -1) return 0;
333 mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
334 af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
336 ao_data.channels = channels;
337 if(!AF_FORMAT_IS_AC3(format)) {
338 // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
339 if (ao_data.channels > 2) {
340 if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
341 ao_data.channels != channels ) {
342 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, channels);
343 return 0;
346 else {
347 int c = ao_data.channels-1;
348 if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
349 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, ao_data.channels);
350 return 0;
352 ao_data.channels=c+1;
354 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels);
355 // set rate
356 ao_data.samplerate=rate;
357 ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
358 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
361 if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
362 int r=0;
363 mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_OSS_CantUseGetospace);
364 if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
365 mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
366 } else {
367 ao_data.outburst=r;
368 mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
370 } else {
371 mp_msg(MSGT_AO,MSGL_V,"audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n",
372 zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
373 if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes;
374 ao_data.outburst=zz.fragsize;
377 if(ao_data.buffersize==-1){
378 // Measuring buffer size:
379 void* data;
380 ao_data.buffersize=0;
381 #ifdef HAVE_AUDIO_SELECT
382 data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst);
383 while(ao_data.buffersize<0x40000){
384 fd_set rfds;
385 struct timeval tv;
386 FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
387 tv.tv_sec=0; tv.tv_usec = 0;
388 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
389 write(audio_fd,data,ao_data.outburst);
390 ao_data.buffersize+=ao_data.outburst;
392 free(data);
393 if(ao_data.buffersize==0){
394 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantUseSelect);
395 return 0;
397 #endif
400 ao_data.bps=ao_data.channels;
401 switch (ao_data.format & AF_FORMAT_BITS_MASK) {
402 case AF_FORMAT_8BIT:
403 break;
404 case AF_FORMAT_16BIT:
405 ao_data.bps*=2;
406 break;
407 case AF_FORMAT_24BIT:
408 ao_data.bps*=3;
409 break;
410 case AF_FORMAT_32BIT:
411 ao_data.bps*=4;
412 break;
415 ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down
416 ao_data.bps*=ao_data.samplerate;
418 return 1;
421 // close audio device
422 static void uninit(int immed){
423 if(audio_fd == -1) return;
424 #ifdef SNDCTL_DSP_SYNC
425 // to get the buffer played
426 if (!immed)
427 ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL);
428 #endif
429 #ifdef SNDCTL_DSP_RESET
430 if (immed)
431 ioctl(audio_fd, SNDCTL_DSP_RESET, NULL);
432 #endif
433 close(audio_fd);
434 audio_fd = -1;
437 // stop playing and empty buffers (for seeking/pause)
438 static void reset(void){
439 int oss_format;
440 uninit(1);
441 audio_fd=open(dsp, O_WRONLY);
442 if(audio_fd < 0){
443 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantReopen, strerror(errno));
444 return;
447 #if defined(FD_CLOEXEC) && defined(F_SETFD)
448 fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
449 #endif
451 oss_format = format2oss(ao_data.format);
452 if(AF_FORMAT_IS_AC3(ao_data.format))
453 ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
454 ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
455 if(!AF_FORMAT_IS_AC3(ao_data.format)) {
456 if (ao_data.channels > 2)
457 ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
458 else {
459 int c = ao_data.channels-1;
460 ioctl (audio_fd, SNDCTL_DSP_STEREO, &c);
462 ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
466 // stop playing, keep buffers (for pause)
467 static void audio_pause(void)
469 prepause_space = get_space();
470 uninit(1);
473 // resume playing, after audio_pause()
474 static void audio_resume(void)
476 int fillcnt;
477 reset();
478 fillcnt = get_space() - prepause_space;
479 if (fillcnt > 0 && !(ao_data.format & AF_FORMAT_SPECIAL_MASK)) {
480 void *silence = calloc(fillcnt, 1);
481 play(silence, fillcnt, 0);
482 free(silence);
487 // return: how many bytes can be played without blocking
488 static int get_space(void){
489 int playsize=ao_data.outburst;
491 #ifdef SNDCTL_DSP_GETOSPACE
492 if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
493 // calculate exact buffer space:
494 playsize = zz.fragments*zz.fragsize;
495 if (playsize > MAX_OUTBURST)
496 playsize = (MAX_OUTBURST / zz.fragsize) * zz.fragsize;
497 return playsize;
499 #endif
501 // check buffer
502 #ifdef HAVE_AUDIO_SELECT
503 { fd_set rfds;
504 struct timeval tv;
505 FD_ZERO(&rfds);
506 FD_SET(audio_fd, &rfds);
507 tv.tv_sec = 0;
508 tv.tv_usec = 0;
509 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block!
511 #endif
513 return ao_data.outburst;
516 // plays 'len' bytes of 'data'
517 // it should round it down to outburst*n
518 // return: number of bytes played
519 static int play(void* data,int len,int flags){
520 if(len==0)
521 return len;
522 if(len>ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
523 len/=ao_data.outburst;
524 len*=ao_data.outburst;
526 len=write(audio_fd,data,len);
527 return len;
530 static int audio_delay_method=2;
532 // return: delay in seconds between first and last sample in buffer
533 static float get_delay(void){
534 /* Calculate how many bytes/second is sent out */
535 if(audio_delay_method==2){
536 #ifdef SNDCTL_DSP_GETODELAY
537 int r=0;
538 if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
539 return ((float)r)/(float)ao_data.bps;
540 #endif
541 audio_delay_method=1; // fallback if not supported
543 if(audio_delay_method==1){
544 // SNDCTL_DSP_GETOSPACE
545 if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1)
546 return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps;
547 audio_delay_method=0; // fallback if not supported
549 return ((float)ao_data.buffersize)/(float)ao_data.bps;