Try to auto-detect several vo_gl settings (ati-hack, force-pbo and rectangle).
[mplayer/glamo.git] / libao2 / ao_alsa.c
blobcd7b8627289b1da1513f454e15453341d478e087
1 /*
2 ao_alsa9/1.x - ALSA-0.9.x-1.x output plugin for MPlayer
4 (C) Alex Beregszaszi
6 modified for real alsa-0.9.0-support by Zsolt Barat <joy@streamminister.de>
7 additional AC3 passthrough support by Andy Lo A Foe <andy@alsaplayer.org>
8 08/22/2002 iec958-init rewritten and merged with common init, zsolt
9 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka
10 04/25/2004 printfs converted to mp_msg, Zsolt.
12 Any bugreports regarding to this driver are welcome.
15 #include <errno.h>
16 #include <sys/time.h>
17 #include <stdlib.h>
18 #include <stdarg.h>
19 #include <ctype.h>
20 #include <math.h>
21 #include <string.h>
22 #include <alloca.h>
24 #include "config.h"
25 #include "subopt-helper.h"
26 #include "mixer.h"
27 #include "mp_msg.h"
28 #include "help_mp.h"
30 #define ALSA_PCM_NEW_HW_PARAMS_API
31 #define ALSA_PCM_NEW_SW_PARAMS_API
33 #if HAVE_SYS_ASOUNDLIB_H
34 #include <sys/asoundlib.h>
35 #elif HAVE_ALSA_ASOUNDLIB_H
36 #include <alsa/asoundlib.h>
37 #else
38 #error "asoundlib.h is not in sys/ or alsa/ - please bugreport"
39 #endif
42 #include "audio_out.h"
43 #include "audio_out_internal.h"
44 #include "libaf/af_format.h"
46 static ao_info_t info =
48 "ALSA-0.9.x-1.x audio output",
49 "alsa",
50 "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
51 "under developement"
54 LIBAO_EXTERN(alsa)
56 static snd_pcm_t *alsa_handler;
57 static snd_pcm_format_t alsa_format;
58 static snd_pcm_hw_params_t *alsa_hwparams;
59 static snd_pcm_sw_params_t *alsa_swparams;
61 /* 16 sets buffersize to 16 * chunksize is as default 1024
62 * which seems to be good avarge for most situations
63 * so buffersize is 16384 frames by default */
64 static int alsa_fragcount = 16;
65 static snd_pcm_uframes_t chunk_size = 1024;
67 static size_t bytes_per_sample;
69 static int ao_noblock = 0;
71 static int open_mode;
72 static int alsa_can_pause = 0;
74 #define ALSA_DEVICE_SIZE 256
76 #undef BUFFERTIME
77 #define SET_CHUNKSIZE
79 static void alsa_error_handler(const char *file, int line, const char *function,
80 int err, const char *format, ...)
82 char tmp[0xc00];
83 va_list va;
85 va_start(va, format);
86 vsnprintf(tmp, sizeof tmp, format, va);
87 va_end(va);
88 tmp[sizeof tmp - 1] = '\0';
90 if (err)
91 mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
92 file, line, function, tmp, snd_strerror(err));
93 else
94 mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
95 file, line, function, tmp);
98 /* to set/get/query special features/parameters */
99 static int control(int cmd, void *arg)
101 switch(cmd) {
102 case AOCONTROL_QUERY_FORMAT:
103 return CONTROL_TRUE;
104 case AOCONTROL_GET_VOLUME:
105 case AOCONTROL_SET_VOLUME:
107 ao_control_vol_t *vol = (ao_control_vol_t *)arg;
109 int err;
110 snd_mixer_t *handle;
111 snd_mixer_elem_t *elem;
112 snd_mixer_selem_id_t *sid;
114 static char *mix_name = "PCM";
115 static char *card = "default";
116 static int mix_index = 0;
118 long pmin, pmax;
119 long get_vol, set_vol;
120 float f_multi;
122 if(mixer_channel) {
123 char *test_mix_index;
125 mix_name = strdup(mixer_channel);
126 if ((test_mix_index = strchr(mix_name, ','))){
127 *test_mix_index = 0;
128 test_mix_index++;
129 mix_index = strtol(test_mix_index, &test_mix_index, 0);
131 if (*test_mix_index){
132 mp_msg(MSGT_AO,MSGL_ERR,
133 MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero);
134 mix_index = 0 ;
138 if(mixer_device) card = mixer_device;
140 if(ao_data.format == AF_FORMAT_AC3)
141 return CONTROL_TRUE;
143 //allocate simple id
144 snd_mixer_selem_id_alloca(&sid);
146 //sets simple-mixer index and name
147 snd_mixer_selem_id_set_index(sid, mix_index);
148 snd_mixer_selem_id_set_name(sid, mix_name);
150 if (mixer_channel) {
151 free(mix_name);
152 mix_name = NULL;
155 if ((err = snd_mixer_open(&handle, 0)) < 0) {
156 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerOpenError, snd_strerror(err));
157 return CONTROL_ERROR;
160 if ((err = snd_mixer_attach(handle, card)) < 0) {
161 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError,
162 card, snd_strerror(err));
163 snd_mixer_close(handle);
164 return CONTROL_ERROR;
167 if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
168 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerRegisterError, snd_strerror(err));
169 snd_mixer_close(handle);
170 return CONTROL_ERROR;
172 err = snd_mixer_load(handle);
173 if (err < 0) {
174 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerLoadError, snd_strerror(err));
175 snd_mixer_close(handle);
176 return CONTROL_ERROR;
179 elem = snd_mixer_find_selem(handle, sid);
180 if (!elem) {
181 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToFindSimpleControl,
182 snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
183 snd_mixer_close(handle);
184 return CONTROL_ERROR;
187 snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
188 f_multi = (100 / (float)(pmax - pmin));
190 if (cmd == AOCONTROL_SET_VOLUME) {
192 set_vol = vol->left / f_multi + pmin + 0.5;
194 //setting channels
195 if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
196 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingLeftChannel,
197 snd_strerror(err));
198 return CONTROL_ERROR;
200 mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
202 set_vol = vol->right / f_multi + pmin + 0.5;
204 if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
205 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingRightChannel,
206 snd_strerror(err));
207 return CONTROL_ERROR;
209 mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
210 set_vol, pmin, pmax, f_multi);
212 if (snd_mixer_selem_has_playback_switch(elem)) {
213 int lmute = (vol->left == 0.0);
214 int rmute = (vol->right == 0.0);
215 if (snd_mixer_selem_has_playback_switch_joined(elem)) {
216 lmute = rmute = lmute && rmute;
217 } else {
218 snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
220 snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
223 else {
224 snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
225 vol->left = (get_vol - pmin) * f_multi;
226 snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
227 vol->right = (get_vol - pmin) * f_multi;
229 mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
231 snd_mixer_close(handle);
232 return CONTROL_OK;
235 } //end switch
236 return CONTROL_UNKNOWN;
239 static void parse_device (char *dest, const char *src, int len)
241 char *tmp;
242 memmove(dest, src, len);
243 dest[len] = 0;
244 while ((tmp = strrchr(dest, '.')))
245 tmp[0] = ',';
246 while ((tmp = strrchr(dest, '=')))
247 tmp[0] = ':';
250 static void print_help (void)
252 mp_msg (MSGT_AO, MSGL_FATAL,
253 MSGTR_AO_ALSA_CommandlineHelp);
256 static int str_maxlen(strarg_t *str) {
257 if (str->len > ALSA_DEVICE_SIZE)
258 return 0;
259 return 1;
262 static int try_open_device(const char *device, int open_mode, int try_ac3)
264 int err, len;
265 char *ac3_device, *args;
267 if (try_ac3) {
268 /* to set the non-audio bit, use AES0=6 */
269 len = strlen(device);
270 ac3_device = malloc(len + 7 + 1);
271 if (!ac3_device)
272 return -ENOMEM;
273 strcpy(ac3_device, device);
274 args = strchr(ac3_device, ':');
275 if (!args) {
276 /* no existing parameters: add it behind device name */
277 strcat(ac3_device, ":AES0=6");
278 } else {
280 ++args;
281 while (isspace(*args));
282 if (*args == '\0') {
283 /* ":" but no parameters */
284 strcat(ac3_device, "AES0=6");
285 } else if (*args != '{') {
286 /* a simple list of parameters: add it at the end of the list */
287 strcat(ac3_device, ",AES0=6");
288 } else {
289 /* parameters in config syntax: add it inside the { } block */
291 --len;
292 while (len > 0 && isspace(ac3_device[len]));
293 if (ac3_device[len] == '}')
294 strcpy(ac3_device + len, " AES0=6}");
297 err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
298 open_mode);
299 free(ac3_device);
301 if (!try_ac3 || err < 0)
302 err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
303 open_mode);
304 return err;
308 open & setup audio device
309 return: 1=success 0=fail
311 static int init(int rate_hz, int channels, int format, int flags)
313 int err;
314 int block;
315 strarg_t device;
316 snd_pcm_uframes_t bufsize;
317 snd_pcm_uframes_t boundary;
318 opt_t subopts[] = {
319 {"block", OPT_ARG_BOOL, &block, NULL},
320 {"device", OPT_ARG_STR, &device, (opt_test_f)str_maxlen},
321 {NULL}
324 char alsa_device[ALSA_DEVICE_SIZE + 1];
325 // make sure alsa_device is null-terminated even when using strncpy etc.
326 memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
328 mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
329 channels, format);
330 alsa_handler = NULL;
331 #if SND_LIB_VERSION >= 0x010005
332 mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
333 #else
334 mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
335 #endif
337 snd_lib_error_set_handler(alsa_error_handler);
339 ao_data.samplerate = rate_hz;
340 ao_data.format = format;
341 ao_data.channels = channels;
343 switch (format)
345 case AF_FORMAT_S8:
346 alsa_format = SND_PCM_FORMAT_S8;
347 break;
348 case AF_FORMAT_U8:
349 alsa_format = SND_PCM_FORMAT_U8;
350 break;
351 case AF_FORMAT_U16_LE:
352 alsa_format = SND_PCM_FORMAT_U16_LE;
353 break;
354 case AF_FORMAT_U16_BE:
355 alsa_format = SND_PCM_FORMAT_U16_BE;
356 break;
357 #ifndef WORDS_BIGENDIAN
358 case AF_FORMAT_AC3:
359 #endif
360 case AF_FORMAT_S16_LE:
361 alsa_format = SND_PCM_FORMAT_S16_LE;
362 break;
363 #ifdef WORDS_BIGENDIAN
364 case AF_FORMAT_AC3:
365 #endif
366 case AF_FORMAT_S16_BE:
367 alsa_format = SND_PCM_FORMAT_S16_BE;
368 break;
369 case AF_FORMAT_U32_LE:
370 alsa_format = SND_PCM_FORMAT_U32_LE;
371 break;
372 case AF_FORMAT_U32_BE:
373 alsa_format = SND_PCM_FORMAT_U32_BE;
374 break;
375 case AF_FORMAT_S32_LE:
376 alsa_format = SND_PCM_FORMAT_S32_LE;
377 break;
378 case AF_FORMAT_S32_BE:
379 alsa_format = SND_PCM_FORMAT_S32_BE;
380 break;
381 case AF_FORMAT_FLOAT_LE:
382 alsa_format = SND_PCM_FORMAT_FLOAT_LE;
383 break;
384 case AF_FORMAT_FLOAT_BE:
385 alsa_format = SND_PCM_FORMAT_FLOAT_BE;
386 break;
387 case AF_FORMAT_MU_LAW:
388 alsa_format = SND_PCM_FORMAT_MU_LAW;
389 break;
390 case AF_FORMAT_A_LAW:
391 alsa_format = SND_PCM_FORMAT_A_LAW;
392 break;
394 default:
395 alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
396 break;
399 //subdevice parsing
400 // set defaults
401 block = 1;
402 /* switch for spdif
403 * sets opening sequence for SPDIF
404 * sets also the playback and other switches 'on the fly'
405 * while opening the abstract alias for the spdif subdevice
406 * 'iec958'
408 if (format == AF_FORMAT_AC3) {
409 device.str = "iec958";
410 mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
412 else
413 /* in any case for multichannel playback we should select
414 * appropriate device
416 switch (channels) {
417 case 1:
418 case 2:
419 device.str = "default";
420 mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
421 break;
422 case 4:
423 if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
424 // hack - use the converter plugin
425 device.str = "plug:surround40";
426 else
427 device.str = "surround40";
428 mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
429 break;
430 case 6:
431 if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
432 device.str = "plug:surround51";
433 else
434 device.str = "surround51";
435 mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
436 break;
437 default:
438 device.str = "default";
439 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels);
441 device.len = strlen(device.str);
442 if (subopt_parse(ao_subdevice, subopts) != 0) {
443 print_help();
444 return 0;
446 ao_noblock = !block;
447 parse_device(alsa_device, device.str, device.len);
449 mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
451 //setting modes for block or nonblock-mode
452 if (ao_noblock) {
453 open_mode = SND_PCM_NONBLOCK;
455 else {
456 open_mode = 0;
459 //sets buff/chunksize if its set manually
460 if (ao_data.buffersize) {
461 switch (ao_data.buffersize)
463 case 1:
464 alsa_fragcount = 16;
465 chunk_size = 512;
466 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
467 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
468 break;
469 case 2:
470 alsa_fragcount = 8;
471 chunk_size = 1024;
472 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
473 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
474 break;
475 case 3:
476 alsa_fragcount = 32;
477 chunk_size = 512;
478 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
479 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
480 break;
481 case 4:
482 alsa_fragcount = 16;
483 chunk_size = 1024;
484 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
485 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
486 break;
487 default:
488 alsa_fragcount = 16;
489 chunk_size = 1024;
490 break;
494 if (!alsa_handler) {
495 //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
496 if ((err = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0)
498 if (err != -EBUSY && ao_noblock) {
499 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
500 if ((err = try_open_device(alsa_device, 0, format == AF_FORMAT_AC3)) < 0) {
501 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
502 return 0;
504 } else {
505 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
506 return 0;
510 if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
511 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
512 } else {
513 mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
516 snd_pcm_hw_params_alloca(&alsa_hwparams);
517 snd_pcm_sw_params_alloca(&alsa_swparams);
519 // setting hw-parameters
520 if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
522 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters,
523 snd_strerror(err));
524 return 0;
527 err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
528 SND_PCM_ACCESS_RW_INTERLEAVED);
529 if (err < 0) {
530 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType,
531 snd_strerror(err));
532 return 0;
535 /* workaround for nonsupported formats
536 sets default format to S16_LE if the given formats aren't supported */
537 if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
538 alsa_format)) < 0)
540 mp_msg(MSGT_AO,MSGL_INFO,
541 MSGTR_AO_ALSA_FormatNotSupportedByHardware, af_fmt2str_short(format));
542 alsa_format = SND_PCM_FORMAT_S16_LE;
543 ao_data.format = AF_FORMAT_S16_LE;
546 if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
547 alsa_format)) < 0)
549 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat,
550 snd_strerror(err));
551 return 0;
554 if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
555 &ao_data.channels)) < 0)
557 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels,
558 snd_strerror(err));
559 return 0;
562 /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
563 prefer our own resampler */
564 #if SND_LIB_VERSION >= 0x010009
565 if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
566 0)) < 0)
568 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling,
569 snd_strerror(err));
570 return 0;
572 #endif
574 if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
575 &ao_data.samplerate, NULL)) < 0)
577 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2,
578 snd_strerror(err));
579 return 0;
582 bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8;
583 bytes_per_sample *= ao_data.channels;
584 ao_data.bps = ao_data.samplerate * bytes_per_sample;
586 #ifdef BUFFERTIME
588 int alsa_buffer_time = 500000; /* original 60 */
589 int alsa_period_time;
590 alsa_period_time = alsa_buffer_time/4;
591 if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
592 &alsa_buffer_time, NULL)) < 0)
594 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
595 snd_strerror(err));
596 return 0;
597 } else
598 alsa_buffer_time = err;
600 if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
601 &alsa_period_time, NULL)) < 0)
602 /* original: alsa_buffer_time/ao_data.bps */
604 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodTime,
605 snd_strerror(err));
606 return 0;
608 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_BufferTimePeriodTime,
609 alsa_buffer_time, err);
611 #endif//end SET_BUFFERTIME
613 #ifdef SET_CHUNKSIZE
615 //set chunksize
616 if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
617 &chunk_size, NULL)) < 0)
619 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodSize,
620 chunk_size, snd_strerror(err));
621 return 0;
623 else {
624 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %li\n", chunk_size);
626 if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
627 &alsa_fragcount, NULL)) < 0) {
628 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods,
629 snd_strerror(err));
630 return 0;
632 else {
633 mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount);
636 #endif//end SET_CHUNKSIZE
638 /* finally install hardware parameters */
639 if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
641 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters,
642 snd_strerror(err));
643 return 0;
645 // end setting hw-params
648 // gets buffersize for control
649 if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
651 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err));
652 return 0;
654 else {
655 ao_data.buffersize = bufsize * bytes_per_sample;
656 mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
659 if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
660 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err));
661 return 0;
662 } else {
663 mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
665 ao_data.outburst = chunk_size * bytes_per_sample;
667 /* setting software parameters */
668 if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
669 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
670 snd_strerror(err));
671 return 0;
673 #if SND_LIB_VERSION >= 0x000901
674 if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
675 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary,
676 snd_strerror(err));
677 return 0;
679 #else
680 boundary = 0x7fffffff;
681 #endif
682 /* start playing when one period has been written */
683 if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
684 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold,
685 snd_strerror(err));
686 return 0;
688 /* disable underrun reporting */
689 if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
690 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold,
691 snd_strerror(err));
692 return 0;
694 #if SND_LIB_VERSION >= 0x000901
695 /* play silence when there is an underrun */
696 if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
697 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize,
698 snd_strerror(err));
699 return 0;
701 #endif
702 if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
703 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
704 snd_strerror(err));
705 return 0;
707 /* end setting sw-params */
709 mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
710 ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
711 snd_pcm_format_description(alsa_format));
713 } // end switch alsa_handler (spdif)
714 alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
715 return 1;
716 } // end init
719 /* close audio device */
720 static void uninit(int immed)
723 if (alsa_handler) {
724 int err;
726 if (!immed)
727 snd_pcm_drain(alsa_handler);
729 if ((err = snd_pcm_close(alsa_handler)) < 0)
731 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmCloseError, snd_strerror(err));
732 return;
734 else {
735 alsa_handler = NULL;
736 mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
739 else {
740 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_NoHandlerDefined);
744 static void audio_pause(void)
746 int err;
748 if (alsa_can_pause) {
749 if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
751 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPauseError, snd_strerror(err));
752 return;
754 mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
755 } else {
756 if ((err = snd_pcm_drop(alsa_handler)) < 0)
758 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmDropError, snd_strerror(err));
759 return;
764 static void audio_resume(void)
766 int err;
768 if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
769 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
770 while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
772 if (alsa_can_pause) {
773 if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
775 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmResumeError, snd_strerror(err));
776 return;
778 mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
779 } else {
780 if ((err = snd_pcm_prepare(alsa_handler)) < 0)
782 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
783 return;
788 /* stop playing and empty buffers (for seeking/pause) */
789 static void reset(void)
791 int err;
793 if ((err = snd_pcm_drop(alsa_handler)) < 0)
795 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
796 return;
798 if ((err = snd_pcm_prepare(alsa_handler)) < 0)
800 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
801 return;
803 return;
807 plays 'len' bytes of 'data'
808 returns: number of bytes played
809 modified last at 29.06.02 by jp
810 thanxs for marius <marius@rospot.com> for giving us the light ;)
813 static int play(void* data, int len, int flags)
815 int num_frames = len / bytes_per_sample;
816 snd_pcm_sframes_t res = 0;
818 //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
820 if (!alsa_handler) {
821 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_DeviceConfigurationError);
822 return 0;
825 if (num_frames == 0)
826 return 0;
828 do {
829 res = snd_pcm_writei(alsa_handler, data, num_frames);
831 if (res == -EINTR) {
832 /* nothing to do */
833 res = 0;
835 else if (res == -ESTRPIPE) { /* suspend */
836 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
837 while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
838 sleep(1);
840 if (res < 0) {
841 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_WriteError, snd_strerror(res));
842 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_TryingToResetSoundcard);
843 if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
844 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(res));
845 return 0;
846 break;
849 } while (res == 0);
851 return res < 0 ? res : res * bytes_per_sample;
854 /* how many byes are free in the buffer */
855 static int get_space(void)
857 snd_pcm_status_t *status;
858 int ret;
860 snd_pcm_status_alloca(&status);
862 if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
864 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_CannotGetPcmStatus, snd_strerror(ret));
865 return 0;
868 ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
869 if (ret > ao_data.buffersize) // Buffer underrun?
870 ret = ao_data.buffersize;
871 return ret;
874 /* delay in seconds between first and last sample in buffer */
875 static float get_delay(void)
877 if (alsa_handler) {
878 snd_pcm_sframes_t delay;
880 if (snd_pcm_delay(alsa_handler, &delay) < 0)
881 return 0;
883 if (delay < 0) {
884 /* underrun - move the application pointer forward to catch up */
885 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
886 snd_pcm_forward(alsa_handler, -delay);
887 #endif
888 delay = 0;
890 return (float)delay / (float)ao_data.samplerate;
891 } else {
892 return 0;