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[mplayer/glamo.git] / libao2 / ao_oss.c
blob4666c400e919f935b244116c1bb2ad671c37b1c0
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_U24_LE
74 case AF_FORMAT_U24_LE: return AFMT_U24_LE;
75 #endif
76 #ifdef AFMT_U24_BE
77 case AF_FORMAT_U24_BE: return AFMT_U24_BE;
78 #endif
79 #ifdef AFMT_S24_LE
80 case AF_FORMAT_S24_LE: return AFMT_S24_LE;
81 #endif
82 #ifdef AFMT_S24_BE
83 case AF_FORMAT_S24_BE: return AFMT_S24_BE;
84 #endif
85 #ifdef AFMT_U32_LE
86 case AF_FORMAT_U32_LE: return AFMT_U32_LE;
87 #endif
88 #ifdef AFMT_U32_BE
89 case AF_FORMAT_U32_BE: return AFMT_U32_BE;
90 #endif
91 #ifdef AFMT_S32_LE
92 case AF_FORMAT_S32_LE: return AFMT_S32_LE;
93 #endif
94 #ifdef AFMT_S32_BE
95 case AF_FORMAT_S32_BE: return AFMT_S32_BE;
96 #endif
97 #ifdef AFMT_FLOAT
98 case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT;
99 #endif
100 // SPECIALS
101 case AF_FORMAT_MU_LAW: return AFMT_MU_LAW;
102 case AF_FORMAT_A_LAW: return AFMT_A_LAW;
103 case AF_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM;
104 #ifdef AFMT_MPEG
105 case AF_FORMAT_MPEG2: return AFMT_MPEG;
106 #endif
107 #ifdef AFMT_AC3
108 case AF_FORMAT_AC3: return AFMT_AC3;
109 #endif
111 mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", af_fmt2str_short(format));
112 return -1;
115 static int oss2format(int format)
117 switch(format)
119 case AFMT_U8: return AF_FORMAT_U8;
120 case AFMT_S8: return AF_FORMAT_S8;
121 case AFMT_U16_LE: return AF_FORMAT_U16_LE;
122 case AFMT_U16_BE: return AF_FORMAT_U16_BE;
123 case AFMT_S16_LE: return AF_FORMAT_S16_LE;
124 case AFMT_S16_BE: return AF_FORMAT_S16_BE;
125 #ifdef AFMT_U24_LE
126 case AFMT_U24_LE: return AF_FORMAT_U24_LE;
127 #endif
128 #ifdef AFMT_U24_BE
129 case AFMT_U24_BE: return AF_FORMAT_U24_BE;
130 #endif
131 #ifdef AFMT_S24_LE
132 case AFMT_S24_LE: return AF_FORMAT_S24_LE;
133 #endif
134 #ifdef AFMT_S24_BE
135 case AFMT_S24_BE: return AF_FORMAT_S24_BE;
136 #endif
137 #ifdef AFMT_U32_LE
138 case AFMT_U32_LE: return AF_FORMAT_U32_LE;
139 #endif
140 #ifdef AFMT_U32_BE
141 case AFMT_U32_BE: return AF_FORMAT_U32_BE;
142 #endif
143 #ifdef AFMT_S32_LE
144 case AFMT_S32_LE: return AF_FORMAT_S32_LE;
145 #endif
146 #ifdef AFMT_S32_BE
147 case AFMT_S32_BE: return AF_FORMAT_S32_BE;
148 #endif
149 #ifdef AFMT_FLOAT
150 case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE;
151 #endif
152 // SPECIALS
153 case AFMT_MU_LAW: return AF_FORMAT_MU_LAW;
154 case AFMT_A_LAW: return AF_FORMAT_A_LAW;
155 case AFMT_IMA_ADPCM: return AF_FORMAT_IMA_ADPCM;
156 #ifdef AFMT_MPEG
157 case AFMT_MPEG: return AF_FORMAT_MPEG2;
158 #endif
159 #ifdef AFMT_AC3
160 case AFMT_AC3: return AF_FORMAT_AC3;
161 #endif
163 mp_msg(MSGT_GLOBAL,MSGL_ERR,MSGTR_AO_OSS_UnknownUnsupportedFormat, format);
164 return -1;
167 static char *dsp=PATH_DEV_DSP;
168 static audio_buf_info zz;
169 static int audio_fd=-1;
170 static int prepause_space;
172 static const char *oss_mixer_device = PATH_DEV_MIXER;
173 static int oss_mixer_channel = SOUND_MIXER_PCM;
175 // to set/get/query special features/parameters
176 static int control(int cmd,void *arg){
177 switch(cmd){
178 case AOCONTROL_SET_DEVICE:
179 dsp=(char*)arg;
180 return CONTROL_OK;
181 case AOCONTROL_GET_DEVICE:
182 *(char**)arg=dsp;
183 return CONTROL_OK;
184 #ifdef SNDCTL_DSP_GETFMTS
185 case AOCONTROL_QUERY_FORMAT:
187 int format;
188 if (!ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format))
189 if ((unsigned int)format & (unsigned long)arg)
190 return CONTROL_TRUE;
191 return CONTROL_FALSE;
193 #endif
194 case AOCONTROL_GET_VOLUME:
195 case AOCONTROL_SET_VOLUME:
197 ao_control_vol_t *vol = (ao_control_vol_t *)arg;
198 int fd, v, devs;
200 if(ao_data.format == AF_FORMAT_AC3)
201 return CONTROL_TRUE;
203 if ((fd = open(oss_mixer_device, O_RDONLY)) > 0)
205 ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
206 if (devs & (1 << oss_mixer_channel))
208 if (cmd == AOCONTROL_GET_VOLUME)
210 ioctl(fd, MIXER_READ(oss_mixer_channel), &v);
211 vol->right = (v & 0xFF00) >> 8;
212 vol->left = v & 0x00FF;
214 else
216 v = ((int)vol->right << 8) | (int)vol->left;
217 ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v);
220 else
222 close(fd);
223 return CONTROL_ERROR;
225 close(fd);
226 return CONTROL_OK;
229 return CONTROL_ERROR;
231 return CONTROL_UNKNOWN;
234 // open & setup audio device
235 // return: 1=success 0=fail
236 static int init(int rate,int channels,int format,int flags){
237 char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
238 int oss_format;
239 char *mdev = mixer_device, *mchan = mixer_channel;
241 mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels,
242 af_fmt2str_short(format));
244 if (ao_subdevice) {
245 char *m,*c;
246 m = strchr(ao_subdevice,':');
247 if(m) {
248 c = strchr(m+1,':');
249 if(c) {
250 mchan = c+1;
251 c[0] = '\0';
253 mdev = m+1;
254 m[0] = '\0';
256 dsp = ao_subdevice;
259 if(mdev)
260 oss_mixer_device=mdev;
261 else
262 oss_mixer_device=PATH_DEV_MIXER;
264 if(mchan){
265 int fd, devs, i;
267 if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){
268 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenMixer,
269 oss_mixer_device, strerror(errno));
270 }else{
271 ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
272 close(fd);
274 for (i=0; i<SOUND_MIXER_NRDEVICES; i++){
275 if(!strcasecmp(mixer_channels[i], mchan)){
276 if(!(devs & (1 << i))){
277 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,mchan);
278 i = SOUND_MIXER_NRDEVICES+1;
279 break;
281 oss_mixer_channel = i;
282 break;
285 if(i==SOUND_MIXER_NRDEVICES){
286 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,mchan);
289 } else
290 oss_mixer_channel = SOUND_MIXER_PCM;
292 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' dsp device\n", dsp);
293 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", oss_mixer_device);
294 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", mixer_channels[oss_mixer_channel]);
296 #ifdef __linux__
297 audio_fd=open(dsp, O_WRONLY | O_NONBLOCK);
298 #else
299 audio_fd=open(dsp, O_WRONLY);
300 #endif
301 if(audio_fd<0){
302 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenDev, dsp, strerror(errno));
303 return 0;
306 #ifdef __linux__
307 /* Remove the non-blocking flag */
308 if(fcntl(audio_fd, F_SETFL, 0) < 0) {
309 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantMakeFd, strerror(errno));
310 return 0;
312 #endif
314 #if defined(FD_CLOEXEC) && defined(F_SETFD)
315 fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
316 #endif
318 if(format == AF_FORMAT_AC3) {
319 ao_data.samplerate=rate;
320 ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
323 ac3_retry:
324 ao_data.format=format;
325 oss_format=format2oss(format);
326 if (oss_format == -1) {
327 #ifdef WORDS_BIGENDIAN
328 oss_format=AFMT_S16_BE;
329 #else
330 oss_format=AFMT_S16_LE;
331 #endif
332 format=AF_FORMAT_S16_NE;
334 if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format)<0 ||
335 oss_format != format2oss(format)) {
336 mp_msg(MSGT_AO,MSGL_WARN, MSGTR_AO_OSS_CantSet, dsp,
337 af_fmt2str_short(format), af_fmt2str_short(AF_FORMAT_S16_NE) );
338 format=AF_FORMAT_S16_NE;
339 goto ac3_retry;
341 #if 0
342 if(oss_format!=format2oss(format))
343 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));
344 #endif
346 ao_data.format = oss2format(oss_format);
347 if (ao_data.format == -1) return 0;
349 mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
350 af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
352 ao_data.channels = channels;
353 if(format != AF_FORMAT_AC3) {
354 // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
355 if (ao_data.channels > 2) {
356 if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
357 ao_data.channels != channels ) {
358 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, channels);
359 return 0;
362 else {
363 int c = ao_data.channels-1;
364 if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
365 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, ao_data.channels);
366 return 0;
368 ao_data.channels=c+1;
370 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels);
371 // set rate
372 ao_data.samplerate=rate;
373 ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
374 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
377 if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
378 int r=0;
379 mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_OSS_CantUseGetospace);
380 if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
381 mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
382 } else {
383 ao_data.outburst=r;
384 mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
386 } else {
387 mp_msg(MSGT_AO,MSGL_V,"audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n",
388 zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
389 if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes;
390 ao_data.outburst=zz.fragsize;
393 if(ao_data.buffersize==-1){
394 // Measuring buffer size:
395 void* data;
396 ao_data.buffersize=0;
397 #ifdef HAVE_AUDIO_SELECT
398 data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst);
399 while(ao_data.buffersize<0x40000){
400 fd_set rfds;
401 struct timeval tv;
402 FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
403 tv.tv_sec=0; tv.tv_usec = 0;
404 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
405 write(audio_fd,data,ao_data.outburst);
406 ao_data.buffersize+=ao_data.outburst;
408 free(data);
409 if(ao_data.buffersize==0){
410 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantUseSelect);
411 return 0;
413 #endif
416 ao_data.bps=ao_data.channels;
417 if(ao_data.format != AF_FORMAT_U8 && ao_data.format != AF_FORMAT_S8)
418 ao_data.bps*=2;
420 ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down
421 ao_data.bps*=ao_data.samplerate;
423 return 1;
426 // close audio device
427 static void uninit(int immed){
428 if(audio_fd == -1) return;
429 #ifdef SNDCTL_DSP_SYNC
430 // to get the buffer played
431 if (!immed)
432 ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL);
433 #endif
434 #ifdef SNDCTL_DSP_RESET
435 if (immed)
436 ioctl(audio_fd, SNDCTL_DSP_RESET, NULL);
437 #endif
438 close(audio_fd);
439 audio_fd = -1;
442 // stop playing and empty buffers (for seeking/pause)
443 static void reset(void){
444 int oss_format;
445 uninit(1);
446 audio_fd=open(dsp, O_WRONLY);
447 if(audio_fd < 0){
448 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantReopen, strerror(errno));
449 return;
452 #if defined(FD_CLOEXEC) && defined(F_SETFD)
453 fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
454 #endif
456 oss_format = format2oss(ao_data.format);
457 if(ao_data.format == AF_FORMAT_AC3)
458 ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
459 ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
460 if(ao_data.format != AF_FORMAT_AC3) {
461 if (ao_data.channels > 2)
462 ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
463 else {
464 int c = ao_data.channels-1;
465 ioctl (audio_fd, SNDCTL_DSP_STEREO, &c);
467 ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
471 // stop playing, keep buffers (for pause)
472 static void audio_pause(void)
474 prepause_space = get_space();
475 uninit(1);
478 // resume playing, after audio_pause()
479 static void audio_resume(void)
481 int fillcnt;
482 reset();
483 fillcnt = get_space() - prepause_space;
484 if (fillcnt > 0 && !(ao_data.format & AF_FORMAT_SPECIAL_MASK)) {
485 void *silence = calloc(fillcnt, 1);
486 play(silence, fillcnt, 0);
487 free(silence);
492 // return: how many bytes can be played without blocking
493 static int get_space(void){
494 int playsize=ao_data.outburst;
496 #ifdef SNDCTL_DSP_GETOSPACE
497 if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
498 // calculate exact buffer space:
499 playsize = zz.fragments*zz.fragsize;
500 if (playsize > MAX_OUTBURST)
501 playsize = (MAX_OUTBURST / zz.fragsize) * zz.fragsize;
502 return playsize;
504 #endif
506 // check buffer
507 #ifdef HAVE_AUDIO_SELECT
508 { fd_set rfds;
509 struct timeval tv;
510 FD_ZERO(&rfds);
511 FD_SET(audio_fd, &rfds);
512 tv.tv_sec = 0;
513 tv.tv_usec = 0;
514 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block!
516 #endif
518 return ao_data.outburst;
521 // plays 'len' bytes of 'data'
522 // it should round it down to outburst*n
523 // return: number of bytes played
524 static int play(void* data,int len,int flags){
525 if(len==0)
526 return len;
527 if(len>ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
528 len/=ao_data.outburst;
529 len*=ao_data.outburst;
531 len=write(audio_fd,data,len);
532 return len;
535 static int audio_delay_method=2;
537 // return: delay in seconds between first and last sample in buffer
538 static float get_delay(void){
539 /* Calculate how many bytes/second is sent out */
540 if(audio_delay_method==2){
541 #ifdef SNDCTL_DSP_GETODELAY
542 int r=0;
543 if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
544 return ((float)r)/(float)ao_data.bps;
545 #endif
546 audio_delay_method=1; // fallback if not supported
548 if(audio_delay_method==1){
549 // SNDCTL_DSP_GETOSPACE
550 if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1)
551 return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps;
552 audio_delay_method=0; // fallback if not supported
554 return ((float)ao_data.buffersize)/(float)ao_data.bps;