libao2/ao_alsa.c

   1 /*
   2  * ALSA 0.9.x-1.x audio output driver
   3  *
   4  * Copyright (C) 2004 Alex Beregszaszi
   5  *
   6  * modified for real ALSA 0.9.0 support by Zsolt Barat <joy@streamminister.de>
   7  * additional AC-3 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.
  11  *
  12  * This file is part of MPlayer.
  13  *
  14  * MPlayer is free software; you can redistribute it and/or modify
  15  * it under the terms of the GNU General Public License as published by
  16  * the Free Software Foundation; either version 2 of the License, or
  17  * (at your option) any later version.
  18  *
  19  * MPlayer is distributed in the hope that it will be useful,
  20  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  21  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  22  * GNU General Public License for more details.
  23  *
  24  * You should have received a copy of the GNU General Public License along
  25  * with MPlayer; if not, write to the Free Software Foundation, Inc.,
  26  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  27  */
  28
  29 #include <errno.h>
  30 #include <sys/time.h>
  31 #include <stdlib.h>
  32 #include <stdarg.h>
  33 #include <ctype.h>
  34 #include <math.h>
  35 #include <string.h>
  36 #include <alloca.h>
  37
  38 #include "config.h"
  39 #include "subopt-helper.h"
  40 #include "mixer.h"
  41 #include "mp_msg.h"
  42 #include "help_mp.h"
  43
  44 #define ALSA_PCM_NEW_HW_PARAMS_API
  45 #define ALSA_PCM_NEW_SW_PARAMS_API
  46
  47 #ifdef HAVE_SYS_ASOUNDLIB_H
  48 #include <sys/asoundlib.h>
  49 #elif defined(HAVE_ALSA_ASOUNDLIB_H)
  50 #include <alsa/asoundlib.h>
  51 #else
  52 #error "asoundlib.h is not in sys/ or alsa/ - please bugreport"
  53 #endif
  54
  55
  56 #include "audio_out.h"
  57 #include "audio_out_internal.h"
  58 #include "libaf/af_format.h"
  59
  60 static const ao_info_t info =
  61 {
  62     "ALSA-0.9.x-1.x audio output",
  63     "alsa",
  64     "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
  65     "under developement"
  66 };
  67
  68 LIBAO_EXTERN(alsa)
  69
  70 static snd_pcm_t *alsa_handler;
  71 static snd_pcm_format_t alsa_format;
  72 static snd_pcm_hw_params_t *alsa_hwparams;
  73 static snd_pcm_sw_params_t *alsa_swparams;
  74
  75 /* 16 sets buffersize to 16 * chunksize is as default 1024
  76  * which seems to be good avarge for most situations
  77  * so buffersize is 16384 frames by default */
  78 static int alsa_fragcount = 16;
  79 static snd_pcm_uframes_t chunk_size = 1024;
  80
  81 static size_t bytes_per_sample;
  82
  83 static int ao_noblock = 0;
  84
  85 static int open_mode;
  86 static int alsa_can_pause = 0;
  87 static snd_pcm_sframes_t prepause_frames;
  88
  89 #define ALSA_DEVICE_SIZE 256
  90
  91 #undef BUFFERTIME
  92 #define SET_CHUNKSIZE
  93
  94 static void alsa_error_handler(const char *file, int line, const char *function,
  95                                int err, const char *format, ...)
  96 {
  97   char tmp[0xc00];
  98   va_list va;
  99
 100   va_start(va, format);
 101   vsnprintf(tmp, sizeof tmp, format, va);
 102   va_end(va);
 103   tmp[sizeof tmp - 1] = '\0';
 104
 105   if (err)
 106     mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
 107            file, line, function, tmp, snd_strerror(err));
 108   else
 109     mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
 110            file, line, function, tmp);
 111 }
 112
 113 /* to set/get/query special features/parameters */
 114 static int control(int cmd, void *arg)
 115 {
 116   switch(cmd) {
 117   case AOCONTROL_QUERY_FORMAT:
 118     return CONTROL_TRUE;
 119   case AOCONTROL_GET_VOLUME:
 120   case AOCONTROL_SET_VOLUME:
 121     {
 122       ao_control_vol_t *vol = (ao_control_vol_t *)arg;
 123
 124       int err;
 125       snd_mixer_t *handle;
 126       snd_mixer_elem_t *elem;
 127       snd_mixer_selem_id_t *sid;
 128
 129       static char *mix_name = "PCM";
 130       static char *card = "default";
 131       static int mix_index = 0;
 132
 133       long pmin, pmax;
 134       long get_vol, set_vol;
 135       float f_multi;
 136
 137       if(ao_data.format == AF_FORMAT_AC3)
 138         return CONTROL_TRUE;
 139
 140       if(mixer_channel) {
 141          char *test_mix_index;
 142
 143          mix_name = strdup(mixer_channel);
 144          if ((test_mix_index = strchr(mix_name, ','))){
 145                 *test_mix_index = 0;
 146                 test_mix_index++;
 147                 mix_index = strtol(test_mix_index, &test_mix_index, 0);
 148
 149                 if (*test_mix_index){
 150                   mp_tmsg(MSGT_AO,MSGL_ERR,
 151                     "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
 152                   mix_index = 0 ;
 153                 }
 154          }
 155       }
 156       if(mixer_device) card = mixer_device;
 157
 158       //allocate simple id
 159       snd_mixer_selem_id_alloca(&sid);
 160
 161       //sets simple-mixer index and name
 162       snd_mixer_selem_id_set_index(sid, mix_index);
 163       snd_mixer_selem_id_set_name(sid, mix_name);
 164
 165       if (mixer_channel) {
 166         free(mix_name);
 167         mix_name = NULL;
 168       }
 169
 170       if ((err = snd_mixer_open(&handle, 0)) < 0) {
 171         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer open error: %s\n", snd_strerror(err));
 172         return CONTROL_ERROR;
 173       }
 174
 175       if ((err = snd_mixer_attach(handle, card)) < 0) {
 176         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer attach %s error: %s\n",
 177                card, snd_strerror(err));
 178         snd_mixer_close(handle);
 179         return CONTROL_ERROR;
 180       }
 181
 182       if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
 183         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer register error: %s\n", snd_strerror(err));
 184         snd_mixer_close(handle);
 185         return CONTROL_ERROR;
 186       }
 187       err = snd_mixer_load(handle);
 188       if (err < 0) {
 189         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer load error: %s\n", snd_strerror(err));
 190         snd_mixer_close(handle);
 191         return CONTROL_ERROR;
 192       }
 193
 194       elem = snd_mixer_find_selem(handle, sid);
 195       if (!elem) {
 196         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to find simple control '%s',%i.\n",
 197                snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
 198         snd_mixer_close(handle);
 199         return CONTROL_ERROR;
 200         }
 201
 202       snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
 203       f_multi = (100 / (float)(pmax - pmin));
 204
 205       if (cmd == AOCONTROL_SET_VOLUME) {
 206
 207         set_vol = vol->left / f_multi + pmin + 0.5;
 208
 209         //setting channels
 210         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
 211           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n",
 212                  snd_strerror(err));
 213           snd_mixer_close(handle);
 214           return CONTROL_ERROR;
 215         }
 216         mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
 217
 218         set_vol = vol->right / f_multi + pmin + 0.5;
 219
 220         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
 221           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
 222                  snd_strerror(err));
 223           snd_mixer_close(handle);
 224           return CONTROL_ERROR;
 225         }
 226         mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
 227                set_vol, pmin, pmax, f_multi);
 228
 229         if (snd_mixer_selem_has_playback_switch(elem)) {
 230           int lmute = (vol->left == 0.0);
 231           int rmute = (vol->right == 0.0);
 232           if (snd_mixer_selem_has_playback_switch_joined(elem)) {
 233             lmute = rmute = lmute && rmute;
 234           } else {
 235             snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
 236           }
 237           snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
 238         }
 239       }
 240       else {
 241         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
 242         vol->left = (get_vol - pmin) * f_multi;
 243         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
 244         vol->right = (get_vol - pmin) * f_multi;
 245
 246         mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
 247       }
 248       snd_mixer_close(handle);
 249       return CONTROL_OK;
 250     }
 251
 252   } //end switch
 253   return CONTROL_UNKNOWN;
 254 }
 255
 256 static void parse_device (char *dest, const char *src, int len)
 257 {
 258   char *tmp;
 259   memmove(dest, src, len);
 260   dest[len] = 0;
 261   while ((tmp = strrchr(dest, '.')))
 262     tmp[0] = ',';
 263   while ((tmp = strrchr(dest, '=')))
 264     tmp[0] = ':';
 265 }
 266
 267 static void print_help (void)
 268 {
 269   mp_tmsg (MSGT_AO, MSGL_FATAL,
 270     "\n[AO_ALSA] -ao alsa commandline help:\n"\
 271     "[AO_ALSA] Example: mplayer -ao alsa:device=hw=0.3\n"\
 272     "[AO_ALSA]   Sets first card fourth hardware device.\n\n"\
 273     "[AO_ALSA] Options:\n"\
 274     "[AO_ALSA]   noblock\n"\
 275     "[AO_ALSA]     Opens device in non-blocking mode.\n"\
 276     "[AO_ALSA]   device=<device-name>\n"\
 277     "[AO_ALSA]     Sets device (change , to . and : to =)\n");
 278 }
 279
 280 static int str_maxlen(strarg_t *str) {
 281   if (str->len > ALSA_DEVICE_SIZE)
 282     return 0;
 283   return 1;
 284 }
 285
 286 static int try_open_device(const char *device, int open_mode, int try_ac3)
 287 {
 288   int err, len;
 289   char *ac3_device, *args;
 290
 291   if (try_ac3) {
 292     /* to set the non-audio bit, use AES0=6 */
 293     len = strlen(device);
 294     ac3_device = malloc(len + 7 + 1);
 295     if (!ac3_device)
 296       return -ENOMEM;
 297     strcpy(ac3_device, device);
 298     args = strchr(ac3_device, ':');
 299     if (!args) {
 300       /* no existing parameters: add it behind device name */
 301       strcat(ac3_device, ":AES0=6");
 302     } else {
 303       do
 304         ++args;
 305       while (isspace(*args));
 306       if (*args == '\0') {
 307         /* ":" but no parameters */
 308         strcat(ac3_device, "AES0=6");
 309       } else if (*args != '{') {
 310         /* a simple list of parameters: add it at the end of the list */
 311         strcat(ac3_device, ",AES0=6");
 312       } else {
 313         /* parameters in config syntax: add it inside the { } block */
 314         do
 315           --len;
 316         while (len > 0 && isspace(ac3_device[len]));
 317         if (ac3_device[len] == '}')
 318           strcpy(ac3_device + len, " AES0=6}");
 319       }
 320     }
 321     err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
 322                        open_mode);
 323     free(ac3_device);
 324   }
 325   if (!try_ac3 || err < 0)
 326     err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
 327                        open_mode);
 328   return err;
 329 }
 330
 331 /*
 332     open & setup audio device
 333     return: 1=success 0=fail
 334 */
 335 static int init(int rate_hz, int channels, int format, int flags)
 336 {
 337     int err;
 338     int block;
 339     strarg_t device;
 340     snd_pcm_uframes_t bufsize;
 341     snd_pcm_uframes_t boundary;
 342     opt_t subopts[] = {
 343       {"block", OPT_ARG_BOOL, &block, NULL},
 344       {"device", OPT_ARG_STR, &device, (opt_test_f)str_maxlen},
 345       {NULL}
 346     };
 347
 348     char alsa_device[ALSA_DEVICE_SIZE + 1];
 349     // make sure alsa_device is null-terminated even when using strncpy etc.
 350     memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
 351
 352     mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
 353         channels, format);
 354     alsa_handler = NULL;
 355 #if SND_LIB_VERSION >= 0x010005
 356     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
 357 #else
 358     mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
 359 #endif
 360
 361     prepause_frames = 0;
 362
 363     snd_lib_error_set_handler(alsa_error_handler);
 364
 365     ao_data.samplerate = rate_hz;
 366     ao_data.format = format;
 367     ao_data.channels = channels;
 368
 369     switch (format)
 370       {
 371       case AF_FORMAT_S8:
 372         alsa_format = SND_PCM_FORMAT_S8;
 373         break;
 374       case AF_FORMAT_U8:
 375         alsa_format = SND_PCM_FORMAT_U8;
 376         break;
 377       case AF_FORMAT_U16_LE:
 378         alsa_format = SND_PCM_FORMAT_U16_LE;
 379         break;
 380       case AF_FORMAT_U16_BE:
 381         alsa_format = SND_PCM_FORMAT_U16_BE;
 382         break;
 383 #if !HAVE_BIGENDIAN
 384       case AF_FORMAT_AC3:
 385 #endif
 386       case AF_FORMAT_S16_LE:
 387         alsa_format = SND_PCM_FORMAT_S16_LE;
 388         break;
 389 #if HAVE_BIGENDIAN
 390       case AF_FORMAT_AC3:
 391 #endif
 392       case AF_FORMAT_S16_BE:
 393         alsa_format = SND_PCM_FORMAT_S16_BE;
 394         break;
 395       case AF_FORMAT_U32_LE:
 396         alsa_format = SND_PCM_FORMAT_U32_LE;
 397         break;
 398       case AF_FORMAT_U32_BE:
 399         alsa_format = SND_PCM_FORMAT_U32_BE;
 400         break;
 401       case AF_FORMAT_S32_LE:
 402         alsa_format = SND_PCM_FORMAT_S32_LE;
 403         break;
 404       case AF_FORMAT_S32_BE:
 405         alsa_format = SND_PCM_FORMAT_S32_BE;
 406         break;
 407       case AF_FORMAT_U24_LE:
 408         alsa_format = SND_PCM_FORMAT_U24_3LE;
 409         break;
 410       case AF_FORMAT_U24_BE:
 411         alsa_format = SND_PCM_FORMAT_U24_3BE;
 412         break;
 413       case AF_FORMAT_S24_LE:
 414         alsa_format = SND_PCM_FORMAT_S24_3LE;
 415         break;
 416       case AF_FORMAT_S24_BE:
 417         alsa_format = SND_PCM_FORMAT_S24_3BE;
 418         break;
 419       case AF_FORMAT_FLOAT_LE:
 420         alsa_format = SND_PCM_FORMAT_FLOAT_LE;
 421         break;
 422       case AF_FORMAT_FLOAT_BE:
 423         alsa_format = SND_PCM_FORMAT_FLOAT_BE;
 424         break;
 425       case AF_FORMAT_MU_LAW:
 426         alsa_format = SND_PCM_FORMAT_MU_LAW;
 427         break;
 428       case AF_FORMAT_A_LAW:
 429         alsa_format = SND_PCM_FORMAT_A_LAW;
 430         break;
 431
 432       default:
 433         alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
 434         break;
 435       }
 436
 437     //subdevice parsing
 438     // set defaults
 439     block = 1;
 440     /* switch for spdif
 441      * sets opening sequence for SPDIF
 442      * sets also the playback and other switches 'on the fly'
 443      * while opening the abstract alias for the spdif subdevice
 444      * 'iec958'
 445      */
 446     if (format == AF_FORMAT_AC3) {
 447         device.str = "iec958";
 448         mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
 449     }
 450   else
 451         /* in any case for multichannel playback we should select
 452          * appropriate device
 453          */
 454         switch (channels) {
 455         case 1:
 456         case 2:
 457           device.str = "default";
 458           mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
 459           break;
 460         case 4:
 461           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 462             // hack - use the converter plugin
 463             device.str = "plug:surround40";
 464           else
 465             device.str = "surround40";
 466           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
 467           break;
 468         case 6:
 469           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 470             device.str = "plug:surround51";
 471           else
 472             device.str = "surround51";
 473           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
 474           break;
 475         default:
 476           device.str = "default";
 477           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] %d channels are not supported.\n",channels);
 478         }
 479     device.len = strlen(device.str);
 480     if (subopt_parse(ao_subdevice, subopts) != 0) {
 481         print_help();
 482         return 0;
 483     }
 484     ao_noblock = !block;
 485     parse_device(alsa_device, device.str, device.len);
 486
 487     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
 488
 489     //setting modes for block or nonblock-mode
 490     if (ao_noblock) {
 491       open_mode = SND_PCM_NONBLOCK;
 492     }
 493     else {
 494       open_mode = 0;
 495     }
 496
 497     //sets buff/chunksize if its set manually
 498     if (ao_data.buffersize) {
 499       switch (ao_data.buffersize)
 500         {
 501         case 1:
 502           alsa_fragcount = 16;
 503           chunk_size = 512;
 504             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
 505             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
 506           break;
 507         case 2:
 508           alsa_fragcount = 8;
 509           chunk_size = 1024;
 510             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
 511             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
 512           break;
 513         case 3:
 514           alsa_fragcount = 32;
 515           chunk_size = 512;
 516             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
 517             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
 518           break;
 519         case 4:
 520           alsa_fragcount = 16;
 521           chunk_size = 1024;
 522             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
 523             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
 524           break;
 525         default:
 526           alsa_fragcount = 16;
 527           chunk_size = 1024;
 528           break;
 529         }
 530     }
 531
 532     if (!alsa_handler) {
 533       //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
 534       if ((err = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0)
 535         {
 536           if (err != -EBUSY && ao_noblock) {
 537             mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
 538             if ((err = try_open_device(alsa_device, 0, format == AF_FORMAT_AC3)) < 0) {
 539               mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 540               return 0;
 541             }
 542           } else {
 543             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 544             return 0;
 545           }
 546         }
 547
 548       if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
 549          mp_tmsg(MSGT_AO,MSGL_ERR,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err));
 550       } else {
 551         mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
 552       }
 553
 554       snd_pcm_hw_params_alloca(&alsa_hwparams);
 555       snd_pcm_sw_params_alloca(&alsa_swparams);
 556
 557       // setting hw-parameters
 558       if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
 559         {
 560           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get initial parameters: %s\n",
 561                  snd_strerror(err));
 562           return 0;
 563         }
 564
 565       err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
 566                                          SND_PCM_ACCESS_RW_INTERLEAVED);
 567       if (err < 0) {
 568         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set access type: %s\n",
 569                snd_strerror(err));
 570         return 0;
 571       }
 572
 573       /* workaround for nonsupported formats
 574          sets default format to S16_LE if the given formats aren't supported */
 575       if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
 576                                              alsa_format)) < 0)
 577       {
 578          mp_tmsg(MSGT_AO,MSGL_INFO,
 579                 "[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format));
 580          alsa_format = SND_PCM_FORMAT_S16_LE;
 581          ao_data.format = AF_FORMAT_S16_LE;
 582       }
 583
 584       if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
 585                                               alsa_format)) < 0)
 586         {
 587           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set format: %s\n",
 588                  snd_strerror(err));
 589           return 0;
 590         }
 591
 592       if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
 593                                                      &ao_data.channels)) < 0)
 594         {
 595           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set channels: %s\n",
 596                  snd_strerror(err));
 597           return 0;
 598         }
 599
 600       /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
 601          prefer our own resampler */
 602 #if SND_LIB_VERSION >= 0x010009
 603       if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
 604                                                      0)) < 0)
 605         {
 606           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to disable resampling: %s\n",
 607                  snd_strerror(err));
 608           return 0;
 609         }
 610 #endif
 611
 612       if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
 613                                                  &ao_data.samplerate, NULL)) < 0)
 614         {
 615           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set samplerate-2: %s\n",
 616                  snd_strerror(err));
 617           return 0;
 618         }
 619
 620       bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8;
 621       bytes_per_sample *= ao_data.channels;
 622       ao_data.bps = ao_data.samplerate * bytes_per_sample;
 623
 624 #ifdef BUFFERTIME
 625       {
 626         int alsa_buffer_time = 500000; /* original 60 */
 627         int alsa_period_time;
 628         alsa_period_time = alsa_buffer_time/4;
 629         if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
 630                                                           &alsa_buffer_time, NULL)) < 0)
 631           {
 632             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set buffer time near: %s\n",
 633                    snd_strerror(err));
 634             return 0;
 635           } else
 636             alsa_buffer_time = err;
 637
 638         if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
 639                                                           &alsa_period_time, NULL)) < 0)
 640           /* original: alsa_buffer_time/ao_data.bps */
 641           {
 642             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set period time: %s\n",
 643                    snd_strerror(err));
 644             return 0;
 645           }
 646         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] buffer_time: %d, period_time :%d\n",
 647                alsa_buffer_time, err);
 648       }
 649 #endif//end SET_BUFFERTIME
 650
 651 #ifdef SET_CHUNKSIZE
 652       {
 653         //set chunksize
 654         if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
 655                                                           &chunk_size, NULL)) < 0)
 656           {
 657             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to set period size(%ld): %s\n",
 658                             chunk_size, snd_strerror(err));
 659             return 0;
 660           }
 661         else {
 662           mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %li\n", chunk_size);
 663         }
 664         if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
 665                                                       &alsa_fragcount, NULL)) < 0) {
 666           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n",
 667                  snd_strerror(err));
 668           return 0;
 669         }
 670         else {
 671           mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount);
 672         }
 673       }
 674 #endif//end SET_CHUNKSIZE
 675
 676       /* finally install hardware parameters */
 677       if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
 678         {
 679           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set hw-parameters: %s\n",
 680                  snd_strerror(err));
 681           return 0;
 682         }
 683       // end setting hw-params
 684
 685
 686       // gets buffersize for control
 687       if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
 688         {
 689           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(err));
 690           return 0;
 691         }
 692       else {
 693         ao_data.buffersize = bufsize * bytes_per_sample;
 694           mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
 695       }
 696
 697       if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
 698         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to get period size: %s\n", snd_strerror(err));
 699         return 0;
 700       } else {
 701         mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
 702       }
 703       ao_data.outburst = chunk_size * bytes_per_sample;
 704
 705       /* setting software parameters */
 706       if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
 707         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
 708                snd_strerror(err));
 709         return 0;
 710       }
 711 #if SND_LIB_VERSION >= 0x000901
 712       if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
 713         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get boundary: %s\n",
 714                snd_strerror(err));
 715         return 0;
 716       }
 717 #else
 718       boundary = 0x7fffffff;
 719 #endif
 720       /* start playing when one period has been written */
 721       if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
 722         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set start threshold: %s\n",
 723                snd_strerror(err));
 724         return 0;
 725       }
 726       /* disable underrun reporting */
 727       if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
 728         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set stop threshold: %s\n",
 729                snd_strerror(err));
 730         return 0;
 731       }
 732 #if SND_LIB_VERSION >= 0x000901
 733       /* play silence when there is an underrun */
 734       if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
 735         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set silence size: %s\n",
 736                snd_strerror(err));
 737         return 0;
 738       }
 739 #endif
 740       if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
 741         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
 742                snd_strerror(err));
 743         return 0;
 744       }
 745       /* end setting sw-params */
 746
 747       mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
 748              ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
 749              snd_pcm_format_description(alsa_format));
 750
 751     } // end switch alsa_handler (spdif)
 752     alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
 753     return 1;
 754 } // end init
 755
 756
 757 /* close audio device */
 758 static void uninit(int immed)
 759 {
 760
 761   if (alsa_handler) {
 762     int err;
 763
 764     if (!immed)
 765       snd_pcm_drain(alsa_handler);
 766
 767     if ((err = snd_pcm_close(alsa_handler)) < 0)
 768       {
 769         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm close error: %s\n", snd_strerror(err));
 770         return;
 771       }
 772     else {
 773       alsa_handler = NULL;
 774       mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
 775     }
 776   }
 777   else {
 778     mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] No handler defined!\n");
 779   }
 780 }
 781
 782 static void audio_pause(void)
 783 {
 784     int err;
 785
 786     if (alsa_can_pause) {
 787         if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
 788         {
 789             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm pause error: %s\n", snd_strerror(err));
 790             return;
 791         }
 792           mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
 793     } else {
 794         if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
 795             || prepause_frames < 0)
 796             prepause_frames = 0;
 797
 798         if ((err = snd_pcm_drop(alsa_handler)) < 0)
 799         {
 800             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm drop error: %s\n", snd_strerror(err));
 801             return;
 802         }
 803     }
 804 }
 805
 806 static void audio_resume(void)
 807 {
 808     int err;
 809
 810     if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
 811         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
 812         while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
 813     }
 814     if (alsa_can_pause) {
 815         if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
 816         {
 817             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm resume error: %s\n", snd_strerror(err));
 818             return;
 819         }
 820           mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
 821     } else {
 822         if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 823         {
 824            mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 825             return;
 826         }
 827         if (prepause_frames) {
 828             void *silence = calloc(prepause_frames, bytes_per_sample);
 829             play(silence, prepause_frames * bytes_per_sample, 0);
 830             free(silence);
 831         }
 832     }
 833 }
 834
 835 /* stop playing and empty buffers (for seeking/pause) */
 836 static void reset(void)
 837 {
 838     int err;
 839
 840     prepause_frames = 0;
 841     if ((err = snd_pcm_drop(alsa_handler)) < 0)
 842     {
 843         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 844         return;
 845     }
 846     if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 847     {
 848         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 849         return;
 850     }
 851     return;
 852 }
 853
 854 /*
 855     plays 'len' bytes of 'data'
 856     returns: number of bytes played
 857     modified last at 29.06.02 by jp
 858     thanxs for marius <marius@rospot.com> for giving us the light ;)
 859 */
 860
 861 static int play(void* data, int len, int flags)
 862 {
 863   int num_frames = len / bytes_per_sample;
 864   snd_pcm_sframes_t res = 0;
 865
 866   //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
 867
 868   if (!alsa_handler) {
 869     mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Device configuration error.");
 870     return 0;
 871   }
 872
 873   if (num_frames == 0)
 874     return 0;
 875
 876   do {
 877     res = snd_pcm_writei(alsa_handler, data, num_frames);
 878
 879       if (res == -EINTR) {
 880         /* nothing to do */
 881         res = 0;
 882       }
 883       else if (res == -ESTRPIPE) {      /* suspend */
 884         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
 885         while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
 886           sleep(1);
 887       }
 888       if (res < 0) {
 889         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Write error: %s\n", snd_strerror(res));
 890         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Trying to reset soundcard.\n");
 891         if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
 892           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(res));
 893           return 0;
 894           break;
 895         }
 896       }
 897   } while (res == 0);
 898
 899   return res < 0 ? res : res * bytes_per_sample;
 900 }
 901
 902 /* how many byes are free in the buffer */
 903 static int get_space(void)
 904 {
 905     snd_pcm_status_t *status;
 906     int ret;
 907
 908     snd_pcm_status_alloca(&status);
 909
 910     if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
 911     {
 912         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Cannot get pcm status: %s\n", snd_strerror(ret));
 913         return 0;
 914     }
 915
 916     unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
 917     if (space > ao_data.buffersize) // Buffer underrun?
 918         space = ao_data.buffersize;
 919     return space;
 920 }
 921
 922 /* delay in seconds between first and last sample in buffer */
 923 static float get_delay(void)
 924 {
 925   if (alsa_handler) {
 926     snd_pcm_sframes_t delay;
 927
 928     if (snd_pcm_delay(alsa_handler, &delay) < 0)
 929       return 0;
 930
 931     if (delay < 0) {
 932       /* underrun - move the application pointer forward to catch up */
 933 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
 934       snd_pcm_forward(alsa_handler, -delay);
 935 #endif
 936       delay = 0;
 937     }
 938     return (float)delay / (float)ao_data.samplerate;
 939   } else {
 940     return 0;
 941   }
 942 }