Merge svn changes up to r28366
[mplayer/glamo.git] / libao2 / ao_alsa.c
blob1ea974f6c12786fbdeed2f35383e280885f2f5e0
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;
73 static snd_pcm_sframes_t prepause_frames;
75 #define ALSA_DEVICE_SIZE 256
77 #undef BUFFERTIME
78 #define SET_CHUNKSIZE
80 static void alsa_error_handler(const char *file, int line, const char *function,
81 int err, const char *format, ...)
83 char tmp[0xc00];
84 va_list va;
86 va_start(va, format);
87 vsnprintf(tmp, sizeof tmp, format, va);
88 va_end(va);
89 tmp[sizeof tmp - 1] = '\0';
91 if (err)
92 mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
93 file, line, function, tmp, snd_strerror(err));
94 else
95 mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
96 file, line, function, tmp);
99 /* to set/get/query special features/parameters */
100 static int control(int cmd, void *arg)
102 switch(cmd) {
103 case AOCONTROL_QUERY_FORMAT:
104 return CONTROL_TRUE;
105 case AOCONTROL_GET_VOLUME:
106 case AOCONTROL_SET_VOLUME:
108 ao_control_vol_t *vol = (ao_control_vol_t *)arg;
110 int err;
111 snd_mixer_t *handle;
112 snd_mixer_elem_t *elem;
113 snd_mixer_selem_id_t *sid;
115 static char *mix_name = "PCM";
116 static char *card = "default";
117 static int mix_index = 0;
119 long pmin, pmax;
120 long get_vol, set_vol;
121 float f_multi;
123 if(ao_data.format == AF_FORMAT_AC3)
124 return CONTROL_TRUE;
126 if(mixer_channel) {
127 char *test_mix_index;
129 mix_name = strdup(mixer_channel);
130 if ((test_mix_index = strchr(mix_name, ','))){
131 *test_mix_index = 0;
132 test_mix_index++;
133 mix_index = strtol(test_mix_index, &test_mix_index, 0);
135 if (*test_mix_index){
136 mp_msg(MSGT_AO,MSGL_ERR,
137 MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero);
138 mix_index = 0 ;
142 if(mixer_device) card = mixer_device;
144 //allocate simple id
145 snd_mixer_selem_id_alloca(&sid);
147 //sets simple-mixer index and name
148 snd_mixer_selem_id_set_index(sid, mix_index);
149 snd_mixer_selem_id_set_name(sid, mix_name);
151 if (mixer_channel) {
152 free(mix_name);
153 mix_name = NULL;
156 if ((err = snd_mixer_open(&handle, 0)) < 0) {
157 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerOpenError, snd_strerror(err));
158 return CONTROL_ERROR;
161 if ((err = snd_mixer_attach(handle, card)) < 0) {
162 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError,
163 card, snd_strerror(err));
164 snd_mixer_close(handle);
165 return CONTROL_ERROR;
168 if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
169 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerRegisterError, snd_strerror(err));
170 snd_mixer_close(handle);
171 return CONTROL_ERROR;
173 err = snd_mixer_load(handle);
174 if (err < 0) {
175 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerLoadError, snd_strerror(err));
176 snd_mixer_close(handle);
177 return CONTROL_ERROR;
180 elem = snd_mixer_find_selem(handle, sid);
181 if (!elem) {
182 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToFindSimpleControl,
183 snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
184 snd_mixer_close(handle);
185 return CONTROL_ERROR;
188 snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
189 f_multi = (100 / (float)(pmax - pmin));
191 if (cmd == AOCONTROL_SET_VOLUME) {
193 set_vol = vol->left / f_multi + pmin + 0.5;
195 //setting channels
196 if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
197 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingLeftChannel,
198 snd_strerror(err));
199 return CONTROL_ERROR;
201 mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
203 set_vol = vol->right / f_multi + pmin + 0.5;
205 if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
206 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingRightChannel,
207 snd_strerror(err));
208 return CONTROL_ERROR;
210 mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
211 set_vol, pmin, pmax, f_multi);
213 if (snd_mixer_selem_has_playback_switch(elem)) {
214 int lmute = (vol->left == 0.0);
215 int rmute = (vol->right == 0.0);
216 if (snd_mixer_selem_has_playback_switch_joined(elem)) {
217 lmute = rmute = lmute && rmute;
218 } else {
219 snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
221 snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
224 else {
225 snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
226 vol->left = (get_vol - pmin) * f_multi;
227 snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
228 vol->right = (get_vol - pmin) * f_multi;
230 mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
232 snd_mixer_close(handle);
233 return CONTROL_OK;
236 } //end switch
237 return CONTROL_UNKNOWN;
240 static void parse_device (char *dest, const char *src, int len)
242 char *tmp;
243 memmove(dest, src, len);
244 dest[len] = 0;
245 while ((tmp = strrchr(dest, '.')))
246 tmp[0] = ',';
247 while ((tmp = strrchr(dest, '=')))
248 tmp[0] = ':';
251 static void print_help (void)
253 mp_msg (MSGT_AO, MSGL_FATAL,
254 MSGTR_AO_ALSA_CommandlineHelp);
257 static int str_maxlen(strarg_t *str) {
258 if (str->len > ALSA_DEVICE_SIZE)
259 return 0;
260 return 1;
263 static int try_open_device(const char *device, int open_mode, int try_ac3)
265 int err, len;
266 char *ac3_device, *args;
268 if (try_ac3) {
269 /* to set the non-audio bit, use AES0=6 */
270 len = strlen(device);
271 ac3_device = malloc(len + 7 + 1);
272 if (!ac3_device)
273 return -ENOMEM;
274 strcpy(ac3_device, device);
275 args = strchr(ac3_device, ':');
276 if (!args) {
277 /* no existing parameters: add it behind device name */
278 strcat(ac3_device, ":AES0=6");
279 } else {
281 ++args;
282 while (isspace(*args));
283 if (*args == '\0') {
284 /* ":" but no parameters */
285 strcat(ac3_device, "AES0=6");
286 } else if (*args != '{') {
287 /* a simple list of parameters: add it at the end of the list */
288 strcat(ac3_device, ",AES0=6");
289 } else {
290 /* parameters in config syntax: add it inside the { } block */
292 --len;
293 while (len > 0 && isspace(ac3_device[len]));
294 if (ac3_device[len] == '}')
295 strcpy(ac3_device + len, " AES0=6}");
298 err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
299 open_mode);
300 free(ac3_device);
302 if (!try_ac3 || err < 0)
303 err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
304 open_mode);
305 return err;
309 open & setup audio device
310 return: 1=success 0=fail
312 static int init(int rate_hz, int channels, int format, int flags)
314 int err;
315 int block;
316 strarg_t device;
317 snd_pcm_uframes_t bufsize;
318 snd_pcm_uframes_t boundary;
319 opt_t subopts[] = {
320 {"block", OPT_ARG_BOOL, &block, NULL},
321 {"device", OPT_ARG_STR, &device, (opt_test_f)str_maxlen},
322 {NULL}
325 char alsa_device[ALSA_DEVICE_SIZE + 1];
326 // make sure alsa_device is null-terminated even when using strncpy etc.
327 memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
329 mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
330 channels, format);
331 alsa_handler = NULL;
332 #if SND_LIB_VERSION >= 0x010005
333 mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
334 #else
335 mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
336 #endif
338 prepause_frames = 0;
340 snd_lib_error_set_handler(alsa_error_handler);
342 ao_data.samplerate = rate_hz;
343 ao_data.format = format;
344 ao_data.channels = channels;
346 switch (format)
348 case AF_FORMAT_S8:
349 alsa_format = SND_PCM_FORMAT_S8;
350 break;
351 case AF_FORMAT_U8:
352 alsa_format = SND_PCM_FORMAT_U8;
353 break;
354 case AF_FORMAT_U16_LE:
355 alsa_format = SND_PCM_FORMAT_U16_LE;
356 break;
357 case AF_FORMAT_U16_BE:
358 alsa_format = SND_PCM_FORMAT_U16_BE;
359 break;
360 #ifndef WORDS_BIGENDIAN
361 case AF_FORMAT_AC3:
362 #endif
363 case AF_FORMAT_S16_LE:
364 alsa_format = SND_PCM_FORMAT_S16_LE;
365 break;
366 #ifdef WORDS_BIGENDIAN
367 case AF_FORMAT_AC3:
368 #endif
369 case AF_FORMAT_S16_BE:
370 alsa_format = SND_PCM_FORMAT_S16_BE;
371 break;
372 case AF_FORMAT_U32_LE:
373 alsa_format = SND_PCM_FORMAT_U32_LE;
374 break;
375 case AF_FORMAT_U32_BE:
376 alsa_format = SND_PCM_FORMAT_U32_BE;
377 break;
378 case AF_FORMAT_S32_LE:
379 alsa_format = SND_PCM_FORMAT_S32_LE;
380 break;
381 case AF_FORMAT_S32_BE:
382 alsa_format = SND_PCM_FORMAT_S32_BE;
383 break;
384 case AF_FORMAT_FLOAT_LE:
385 alsa_format = SND_PCM_FORMAT_FLOAT_LE;
386 break;
387 case AF_FORMAT_FLOAT_BE:
388 alsa_format = SND_PCM_FORMAT_FLOAT_BE;
389 break;
390 case AF_FORMAT_MU_LAW:
391 alsa_format = SND_PCM_FORMAT_MU_LAW;
392 break;
393 case AF_FORMAT_A_LAW:
394 alsa_format = SND_PCM_FORMAT_A_LAW;
395 break;
397 default:
398 alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
399 break;
402 //subdevice parsing
403 // set defaults
404 block = 1;
405 /* switch for spdif
406 * sets opening sequence for SPDIF
407 * sets also the playback and other switches 'on the fly'
408 * while opening the abstract alias for the spdif subdevice
409 * 'iec958'
411 if (format == AF_FORMAT_AC3) {
412 device.str = "iec958";
413 mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
415 else
416 /* in any case for multichannel playback we should select
417 * appropriate device
419 switch (channels) {
420 case 1:
421 case 2:
422 device.str = "default";
423 mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
424 break;
425 case 4:
426 if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
427 // hack - use the converter plugin
428 device.str = "plug:surround40";
429 else
430 device.str = "surround40";
431 mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
432 break;
433 case 6:
434 if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
435 device.str = "plug:surround51";
436 else
437 device.str = "surround51";
438 mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
439 break;
440 default:
441 device.str = "default";
442 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels);
444 device.len = strlen(device.str);
445 if (subopt_parse(ao_subdevice, subopts) != 0) {
446 print_help();
447 return 0;
449 ao_noblock = !block;
450 parse_device(alsa_device, device.str, device.len);
452 mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
454 //setting modes for block or nonblock-mode
455 if (ao_noblock) {
456 open_mode = SND_PCM_NONBLOCK;
458 else {
459 open_mode = 0;
462 //sets buff/chunksize if its set manually
463 if (ao_data.buffersize) {
464 switch (ao_data.buffersize)
466 case 1:
467 alsa_fragcount = 16;
468 chunk_size = 512;
469 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
470 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
471 break;
472 case 2:
473 alsa_fragcount = 8;
474 chunk_size = 1024;
475 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
476 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
477 break;
478 case 3:
479 alsa_fragcount = 32;
480 chunk_size = 512;
481 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
482 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
483 break;
484 case 4:
485 alsa_fragcount = 16;
486 chunk_size = 1024;
487 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
488 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
489 break;
490 default:
491 alsa_fragcount = 16;
492 chunk_size = 1024;
493 break;
497 if (!alsa_handler) {
498 //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
499 if ((err = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0)
501 if (err != -EBUSY && ao_noblock) {
502 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
503 if ((err = try_open_device(alsa_device, 0, format == AF_FORMAT_AC3)) < 0) {
504 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
505 return 0;
507 } else {
508 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
509 return 0;
513 if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
514 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
515 } else {
516 mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
519 snd_pcm_hw_params_alloca(&alsa_hwparams);
520 snd_pcm_sw_params_alloca(&alsa_swparams);
522 // setting hw-parameters
523 if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
525 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters,
526 snd_strerror(err));
527 return 0;
530 err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
531 SND_PCM_ACCESS_RW_INTERLEAVED);
532 if (err < 0) {
533 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType,
534 snd_strerror(err));
535 return 0;
538 /* workaround for nonsupported formats
539 sets default format to S16_LE if the given formats aren't supported */
540 if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
541 alsa_format)) < 0)
543 mp_msg(MSGT_AO,MSGL_INFO,
544 MSGTR_AO_ALSA_FormatNotSupportedByHardware, af_fmt2str_short(format));
545 alsa_format = SND_PCM_FORMAT_S16_LE;
546 ao_data.format = AF_FORMAT_S16_LE;
549 if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
550 alsa_format)) < 0)
552 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat,
553 snd_strerror(err));
554 return 0;
557 if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
558 &ao_data.channels)) < 0)
560 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels,
561 snd_strerror(err));
562 return 0;
565 /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
566 prefer our own resampler */
567 #if SND_LIB_VERSION >= 0x010009
568 if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
569 0)) < 0)
571 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling,
572 snd_strerror(err));
573 return 0;
575 #endif
577 if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
578 &ao_data.samplerate, NULL)) < 0)
580 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2,
581 snd_strerror(err));
582 return 0;
585 bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8;
586 bytes_per_sample *= ao_data.channels;
587 ao_data.bps = ao_data.samplerate * bytes_per_sample;
589 #ifdef BUFFERTIME
591 int alsa_buffer_time = 500000; /* original 60 */
592 int alsa_period_time;
593 alsa_period_time = alsa_buffer_time/4;
594 if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
595 &alsa_buffer_time, NULL)) < 0)
597 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
598 snd_strerror(err));
599 return 0;
600 } else
601 alsa_buffer_time = err;
603 if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
604 &alsa_period_time, NULL)) < 0)
605 /* original: alsa_buffer_time/ao_data.bps */
607 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodTime,
608 snd_strerror(err));
609 return 0;
611 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_BufferTimePeriodTime,
612 alsa_buffer_time, err);
614 #endif//end SET_BUFFERTIME
616 #ifdef SET_CHUNKSIZE
618 //set chunksize
619 if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
620 &chunk_size, NULL)) < 0)
622 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodSize,
623 chunk_size, snd_strerror(err));
624 return 0;
626 else {
627 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %li\n", chunk_size);
629 if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
630 &alsa_fragcount, NULL)) < 0) {
631 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods,
632 snd_strerror(err));
633 return 0;
635 else {
636 mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount);
639 #endif//end SET_CHUNKSIZE
641 /* finally install hardware parameters */
642 if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
644 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters,
645 snd_strerror(err));
646 return 0;
648 // end setting hw-params
651 // gets buffersize for control
652 if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
654 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err));
655 return 0;
657 else {
658 ao_data.buffersize = bufsize * bytes_per_sample;
659 mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
662 if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
663 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err));
664 return 0;
665 } else {
666 mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
668 ao_data.outburst = chunk_size * bytes_per_sample;
670 /* setting software parameters */
671 if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
672 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
673 snd_strerror(err));
674 return 0;
676 #if SND_LIB_VERSION >= 0x000901
677 if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
678 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary,
679 snd_strerror(err));
680 return 0;
682 #else
683 boundary = 0x7fffffff;
684 #endif
685 /* start playing when one period has been written */
686 if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
687 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold,
688 snd_strerror(err));
689 return 0;
691 /* disable underrun reporting */
692 if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
693 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold,
694 snd_strerror(err));
695 return 0;
697 #if SND_LIB_VERSION >= 0x000901
698 /* play silence when there is an underrun */
699 if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
700 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize,
701 snd_strerror(err));
702 return 0;
704 #endif
705 if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
706 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
707 snd_strerror(err));
708 return 0;
710 /* end setting sw-params */
712 mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
713 ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
714 snd_pcm_format_description(alsa_format));
716 } // end switch alsa_handler (spdif)
717 alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
718 return 1;
719 } // end init
722 /* close audio device */
723 static void uninit(int immed)
726 if (alsa_handler) {
727 int err;
729 if (!immed)
730 snd_pcm_drain(alsa_handler);
732 if ((err = snd_pcm_close(alsa_handler)) < 0)
734 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmCloseError, snd_strerror(err));
735 return;
737 else {
738 alsa_handler = NULL;
739 mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
742 else {
743 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_NoHandlerDefined);
747 static void audio_pause(void)
749 int err;
751 if (alsa_can_pause) {
752 if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
754 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPauseError, snd_strerror(err));
755 return;
757 mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
758 } else {
759 if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
760 || prepause_frames < 0)
761 prepause_frames = 0;
763 if ((err = snd_pcm_drop(alsa_handler)) < 0)
765 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmDropError, snd_strerror(err));
766 return;
771 static void audio_resume(void)
773 int err;
775 if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
776 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
777 while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
779 if (alsa_can_pause) {
780 if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
782 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmResumeError, snd_strerror(err));
783 return;
785 mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
786 } else {
787 if ((err = snd_pcm_prepare(alsa_handler)) < 0)
789 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
790 return;
792 if (prepause_frames) {
793 void *silence = calloc(prepause_frames, bytes_per_sample);
794 play(silence, prepause_frames * bytes_per_sample, 0);
795 free(silence);
800 /* stop playing and empty buffers (for seeking/pause) */
801 static void reset(void)
803 int err;
805 prepause_frames = 0;
806 if ((err = snd_pcm_drop(alsa_handler)) < 0)
808 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
809 return;
811 if ((err = snd_pcm_prepare(alsa_handler)) < 0)
813 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
814 return;
816 return;
820 plays 'len' bytes of 'data'
821 returns: number of bytes played
822 modified last at 29.06.02 by jp
823 thanxs for marius <marius@rospot.com> for giving us the light ;)
826 static int play(void* data, int len, int flags)
828 int num_frames = len / bytes_per_sample;
829 snd_pcm_sframes_t res = 0;
831 //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
833 if (!alsa_handler) {
834 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_DeviceConfigurationError);
835 return 0;
838 if (num_frames == 0)
839 return 0;
841 do {
842 res = snd_pcm_writei(alsa_handler, data, num_frames);
844 if (res == -EINTR) {
845 /* nothing to do */
846 res = 0;
848 else if (res == -ESTRPIPE) { /* suspend */
849 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
850 while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
851 sleep(1);
853 if (res < 0) {
854 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_WriteError, snd_strerror(res));
855 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_TryingToResetSoundcard);
856 if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
857 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(res));
858 return 0;
859 break;
862 } while (res == 0);
864 return res < 0 ? res : res * bytes_per_sample;
867 /* how many byes are free in the buffer */
868 static int get_space(void)
870 snd_pcm_status_t *status;
871 int ret;
873 snd_pcm_status_alloca(&status);
875 if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
877 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_CannotGetPcmStatus, snd_strerror(ret));
878 return 0;
881 unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
882 if (space > ao_data.buffersize) // Buffer underrun?
883 space = ao_data.buffersize;
884 return space;
887 /* delay in seconds between first and last sample in buffer */
888 static float get_delay(void)
890 if (alsa_handler) {
891 snd_pcm_sframes_t delay;
893 if (snd_pcm_delay(alsa_handler, &delay) < 0)
894 return 0;
896 if (delay < 0) {
897 /* underrun - move the application pointer forward to catch up */
898 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
899 snd_pcm_forward(alsa_handler, -delay);
900 #endif
901 delay = 0;
903 return (float)delay / (float)ao_data.samplerate;
904 } else {
905 return 0;