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           return CONTROL_ERROR;
 214         }
 215         mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
 216
 217         set_vol = vol->right / f_multi + pmin + 0.5;
 218
 219         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
 220           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
 221                  snd_strerror(err));
 222           return CONTROL_ERROR;
 223         }
 224         mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
 225                set_vol, pmin, pmax, f_multi);
 226
 227         if (snd_mixer_selem_has_playback_switch(elem)) {
 228           int lmute = (vol->left == 0.0);
 229           int rmute = (vol->right == 0.0);
 230           if (snd_mixer_selem_has_playback_switch_joined(elem)) {
 231             lmute = rmute = lmute && rmute;
 232           } else {
 233             snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
 234           }
 235           snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
 236         }
 237       }
 238       else {
 239         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
 240         vol->left = (get_vol - pmin) * f_multi;
 241         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
 242         vol->right = (get_vol - pmin) * f_multi;
 243
 244         mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
 245       }
 246       snd_mixer_close(handle);
 247       return CONTROL_OK;
 248     }
 249
 250   } //end switch
 251   return CONTROL_UNKNOWN;
 252 }
 253
 254 static void parse_device (char *dest, const char *src, int len)
 255 {
 256   char *tmp;
 257   memmove(dest, src, len);
 258   dest[len] = 0;
 259   while ((tmp = strrchr(dest, '.')))
 260     tmp[0] = ',';
 261   while ((tmp = strrchr(dest, '=')))
 262     tmp[0] = ':';
 263 }
 264
 265 static void print_help (void)
 266 {
 267   mp_tmsg (MSGT_AO, MSGL_FATAL,
 268     "\n[AO_ALSA] -ao alsa commandline help:\n"\
 269     "[AO_ALSA] Example: mplayer -ao alsa:device=hw=0.3\n"\
 270     "[AO_ALSA]   Sets first card fourth hardware device.\n\n"\
 271     "[AO_ALSA] Options:\n"\
 272     "[AO_ALSA]   noblock\n"\
 273     "[AO_ALSA]     Opens device in non-blocking mode.\n"\
 274     "[AO_ALSA]   device=<device-name>\n"\
 275     "[AO_ALSA]     Sets device (change , to . and : to =)\n");
 276 }
 277
 278 static int str_maxlen(strarg_t *str) {
 279   if (str->len > ALSA_DEVICE_SIZE)
 280     return 0;
 281   return 1;
 282 }
 283
 284 static int try_open_device(const char *device, int open_mode, int try_ac3)
 285 {
 286   int err, len;
 287   char *ac3_device, *args;
 288
 289   if (try_ac3) {
 290     /* to set the non-audio bit, use AES0=6 */
 291     len = strlen(device);
 292     ac3_device = malloc(len + 7 + 1);
 293     if (!ac3_device)
 294       return -ENOMEM;
 295     strcpy(ac3_device, device);
 296     args = strchr(ac3_device, ':');
 297     if (!args) {
 298       /* no existing parameters: add it behind device name */
 299       strcat(ac3_device, ":AES0=6");
 300     } else {
 301       do
 302         ++args;
 303       while (isspace(*args));
 304       if (*args == '\0') {
 305         /* ":" but no parameters */
 306         strcat(ac3_device, "AES0=6");
 307       } else if (*args != '{') {
 308         /* a simple list of parameters: add it at the end of the list */
 309         strcat(ac3_device, ",AES0=6");
 310       } else {
 311         /* parameters in config syntax: add it inside the { } block */
 312         do
 313           --len;
 314         while (len > 0 && isspace(ac3_device[len]));
 315         if (ac3_device[len] == '}')
 316           strcpy(ac3_device + len, " AES0=6}");
 317       }
 318     }
 319     err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
 320                        open_mode);
 321     free(ac3_device);
 322   }
 323   if (!try_ac3 || err < 0)
 324     err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
 325                        open_mode);
 326   return err;
 327 }
 328
 329 /*
 330     open & setup audio device
 331     return: 1=success 0=fail
 332 */
 333 static int init(int rate_hz, int channels, int format, int flags)
 334 {
 335     int err;
 336     int block;
 337     strarg_t device;
 338     snd_pcm_uframes_t bufsize;
 339     snd_pcm_uframes_t boundary;
 340     opt_t subopts[] = {
 341       {"block", OPT_ARG_BOOL, &block, NULL},
 342       {"device", OPT_ARG_STR, &device, (opt_test_f)str_maxlen},
 343       {NULL}
 344     };
 345
 346     char alsa_device[ALSA_DEVICE_SIZE + 1];
 347     // make sure alsa_device is null-terminated even when using strncpy etc.
 348     memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
 349
 350     mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
 351         channels, format);
 352     alsa_handler = NULL;
 353 #if SND_LIB_VERSION >= 0x010005
 354     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
 355 #else
 356     mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
 357 #endif
 358
 359     prepause_frames = 0;
 360
 361     snd_lib_error_set_handler(alsa_error_handler);
 362
 363     ao_data.samplerate = rate_hz;
 364     ao_data.format = format;
 365     ao_data.channels = channels;
 366
 367     switch (format)
 368       {
 369       case AF_FORMAT_S8:
 370         alsa_format = SND_PCM_FORMAT_S8;
 371         break;
 372       case AF_FORMAT_U8:
 373         alsa_format = SND_PCM_FORMAT_U8;
 374         break;
 375       case AF_FORMAT_U16_LE:
 376         alsa_format = SND_PCM_FORMAT_U16_LE;
 377         break;
 378       case AF_FORMAT_U16_BE:
 379         alsa_format = SND_PCM_FORMAT_U16_BE;
 380         break;
 381 #if !HAVE_BIGENDIAN
 382       case AF_FORMAT_AC3:
 383 #endif
 384       case AF_FORMAT_S16_LE:
 385         alsa_format = SND_PCM_FORMAT_S16_LE;
 386         break;
 387 #if HAVE_BIGENDIAN
 388       case AF_FORMAT_AC3:
 389 #endif
 390       case AF_FORMAT_S16_BE:
 391         alsa_format = SND_PCM_FORMAT_S16_BE;
 392         break;
 393       case AF_FORMAT_U32_LE:
 394         alsa_format = SND_PCM_FORMAT_U32_LE;
 395         break;
 396       case AF_FORMAT_U32_BE:
 397         alsa_format = SND_PCM_FORMAT_U32_BE;
 398         break;
 399       case AF_FORMAT_S32_LE:
 400         alsa_format = SND_PCM_FORMAT_S32_LE;
 401         break;
 402       case AF_FORMAT_S32_BE:
 403         alsa_format = SND_PCM_FORMAT_S32_BE;
 404         break;
 405       case AF_FORMAT_U24_LE:
 406         alsa_format = SND_PCM_FORMAT_U24_3LE;
 407         break;
 408       case AF_FORMAT_U24_BE:
 409         alsa_format = SND_PCM_FORMAT_U24_3BE;
 410         break;
 411       case AF_FORMAT_S24_LE:
 412         alsa_format = SND_PCM_FORMAT_S24_3LE;
 413         break;
 414       case AF_FORMAT_S24_BE:
 415         alsa_format = SND_PCM_FORMAT_S24_3BE;
 416         break;
 417       case AF_FORMAT_FLOAT_LE:
 418         alsa_format = SND_PCM_FORMAT_FLOAT_LE;
 419         break;
 420       case AF_FORMAT_FLOAT_BE:
 421         alsa_format = SND_PCM_FORMAT_FLOAT_BE;
 422         break;
 423       case AF_FORMAT_MU_LAW:
 424         alsa_format = SND_PCM_FORMAT_MU_LAW;
 425         break;
 426       case AF_FORMAT_A_LAW:
 427         alsa_format = SND_PCM_FORMAT_A_LAW;
 428         break;
 429
 430       default:
 431         alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
 432         break;
 433       }
 434
 435     //subdevice parsing
 436     // set defaults
 437     block = 1;
 438     /* switch for spdif
 439      * sets opening sequence for SPDIF
 440      * sets also the playback and other switches 'on the fly'
 441      * while opening the abstract alias for the spdif subdevice
 442      * 'iec958'
 443      */
 444     if (format == AF_FORMAT_AC3) {
 445         device.str = "iec958";
 446         mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
 447     }
 448   else
 449         /* in any case for multichannel playback we should select
 450          * appropriate device
 451          */
 452         switch (channels) {
 453         case 1:
 454         case 2:
 455           device.str = "default";
 456           mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
 457           break;
 458         case 4:
 459           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 460             // hack - use the converter plugin
 461             device.str = "plug:surround40";
 462           else
 463             device.str = "surround40";
 464           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
 465           break;
 466         case 6:
 467           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 468             device.str = "plug:surround51";
 469           else
 470             device.str = "surround51";
 471           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
 472           break;
 473         default:
 474           device.str = "default";
 475           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] %d channels are not supported.\n",channels);
 476         }
 477     device.len = strlen(device.str);
 478     if (subopt_parse(ao_subdevice, subopts) != 0) {
 479         print_help();
 480         return 0;
 481     }
 482     ao_noblock = !block;
 483     parse_device(alsa_device, device.str, device.len);
 484
 485     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
 486
 487     //setting modes for block or nonblock-mode
 488     if (ao_noblock) {
 489       open_mode = SND_PCM_NONBLOCK;
 490     }
 491     else {
 492       open_mode = 0;
 493     }
 494
 495     //sets buff/chunksize if its set manually
 496     if (ao_data.buffersize) {
 497       switch (ao_data.buffersize)
 498         {
 499         case 1:
 500           alsa_fragcount = 16;
 501           chunk_size = 512;
 502             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
 503             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
 504           break;
 505         case 2:
 506           alsa_fragcount = 8;
 507           chunk_size = 1024;
 508             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
 509             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
 510           break;
 511         case 3:
 512           alsa_fragcount = 32;
 513           chunk_size = 512;
 514             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
 515             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
 516           break;
 517         case 4:
 518           alsa_fragcount = 16;
 519           chunk_size = 1024;
 520             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
 521             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
 522           break;
 523         default:
 524           alsa_fragcount = 16;
 525           chunk_size = 1024;
 526           break;
 527         }
 528     }
 529
 530     if (!alsa_handler) {
 531       //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
 532       if ((err = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0)
 533         {
 534           if (err != -EBUSY && ao_noblock) {
 535             mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
 536             if ((err = try_open_device(alsa_device, 0, format == AF_FORMAT_AC3)) < 0) {
 537               mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 538               return 0;
 539             }
 540           } else {
 541             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 542             return 0;
 543           }
 544         }
 545
 546       if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
 547          mp_tmsg(MSGT_AO,MSGL_ERR,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err));
 548       } else {
 549         mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
 550       }
 551
 552       snd_pcm_hw_params_alloca(&alsa_hwparams);
 553       snd_pcm_sw_params_alloca(&alsa_swparams);
 554
 555       // setting hw-parameters
 556       if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
 557         {
 558           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get initial parameters: %s\n",
 559                  snd_strerror(err));
 560           return 0;
 561         }
 562
 563       err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
 564                                          SND_PCM_ACCESS_RW_INTERLEAVED);
 565       if (err < 0) {
 566         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set access type: %s\n",
 567                snd_strerror(err));
 568         return 0;
 569       }
 570
 571       /* workaround for nonsupported formats
 572          sets default format to S16_LE if the given formats aren't supported */
 573       if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
 574                                              alsa_format)) < 0)
 575       {
 576          mp_tmsg(MSGT_AO,MSGL_INFO,
 577                 "[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format));
 578          alsa_format = SND_PCM_FORMAT_S16_LE;
 579          ao_data.format = AF_FORMAT_S16_LE;
 580       }
 581
 582       if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
 583                                               alsa_format)) < 0)
 584         {
 585           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set format: %s\n",
 586                  snd_strerror(err));
 587           return 0;
 588         }
 589
 590       if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
 591                                                      &ao_data.channels)) < 0)
 592         {
 593           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set channels: %s\n",
 594                  snd_strerror(err));
 595           return 0;
 596         }
 597
 598       /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
 599          prefer our own resampler */
 600 #if SND_LIB_VERSION >= 0x010009
 601       if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
 602                                                      0)) < 0)
 603         {
 604           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to disable resampling: %s\n",
 605                  snd_strerror(err));
 606           return 0;
 607         }
 608 #endif
 609
 610       if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
 611                                                  &ao_data.samplerate, NULL)) < 0)
 612         {
 613           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set samplerate-2: %s\n",
 614                  snd_strerror(err));
 615           return 0;
 616         }
 617
 618       bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8;
 619       bytes_per_sample *= ao_data.channels;
 620       ao_data.bps = ao_data.samplerate * bytes_per_sample;
 621
 622 #ifdef BUFFERTIME
 623       {
 624         int alsa_buffer_time = 500000; /* original 60 */
 625         int alsa_period_time;
 626         alsa_period_time = alsa_buffer_time/4;
 627         if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
 628                                                           &alsa_buffer_time, NULL)) < 0)
 629           {
 630             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set buffer time near: %s\n",
 631                    snd_strerror(err));
 632             return 0;
 633           } else
 634             alsa_buffer_time = err;
 635
 636         if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
 637                                                           &alsa_period_time, NULL)) < 0)
 638           /* original: alsa_buffer_time/ao_data.bps */
 639           {
 640             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set period time: %s\n",
 641                    snd_strerror(err));
 642             return 0;
 643           }
 644         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] buffer_time: %d, period_time :%d\n",
 645                alsa_buffer_time, err);
 646       }
 647 #endif//end SET_BUFFERTIME
 648
 649 #ifdef SET_CHUNKSIZE
 650       {
 651         //set chunksize
 652         if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
 653                                                           &chunk_size, NULL)) < 0)
 654           {
 655             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to set period size(%ld): %s\n",
 656                             chunk_size, snd_strerror(err));
 657             return 0;
 658           }
 659         else {
 660           mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %li\n", chunk_size);
 661         }
 662         if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
 663                                                       &alsa_fragcount, NULL)) < 0) {
 664           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n",
 665                  snd_strerror(err));
 666           return 0;
 667         }
 668         else {
 669           mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount);
 670         }
 671       }
 672 #endif//end SET_CHUNKSIZE
 673
 674       /* finally install hardware parameters */
 675       if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
 676         {
 677           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set hw-parameters: %s\n",
 678                  snd_strerror(err));
 679           return 0;
 680         }
 681       // end setting hw-params
 682
 683
 684       // gets buffersize for control
 685       if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
 686         {
 687           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(err));
 688           return 0;
 689         }
 690       else {
 691         ao_data.buffersize = bufsize * bytes_per_sample;
 692           mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
 693       }
 694
 695       if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
 696         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to get period size: %s\n", snd_strerror(err));
 697         return 0;
 698       } else {
 699         mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
 700       }
 701       ao_data.outburst = chunk_size * bytes_per_sample;
 702
 703       /* setting software parameters */
 704       if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
 705         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
 706                snd_strerror(err));
 707         return 0;
 708       }
 709 #if SND_LIB_VERSION >= 0x000901
 710       if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
 711         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get boundary: %s\n",
 712                snd_strerror(err));
 713         return 0;
 714       }
 715 #else
 716       boundary = 0x7fffffff;
 717 #endif
 718       /* start playing when one period has been written */
 719       if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
 720         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set start threshold: %s\n",
 721                snd_strerror(err));
 722         return 0;
 723       }
 724       /* disable underrun reporting */
 725       if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
 726         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set stop threshold: %s\n",
 727                snd_strerror(err));
 728         return 0;
 729       }
 730 #if SND_LIB_VERSION >= 0x000901
 731       /* play silence when there is an underrun */
 732       if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
 733         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set silence size: %s\n",
 734                snd_strerror(err));
 735         return 0;
 736       }
 737 #endif
 738       if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
 739         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
 740                snd_strerror(err));
 741         return 0;
 742       }
 743       /* end setting sw-params */
 744
 745       mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
 746              ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
 747              snd_pcm_format_description(alsa_format));
 748
 749     } // end switch alsa_handler (spdif)
 750     alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
 751     return 1;
 752 } // end init
 753
 754
 755 /* close audio device */
 756 static void uninit(int immed)
 757 {
 758
 759   if (alsa_handler) {
 760     int err;
 761
 762     if (!immed)
 763       snd_pcm_drain(alsa_handler);
 764
 765     if ((err = snd_pcm_close(alsa_handler)) < 0)
 766       {
 767         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm close error: %s\n", snd_strerror(err));
 768         return;
 769       }
 770     else {
 771       alsa_handler = NULL;
 772       mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
 773     }
 774   }
 775   else {
 776     mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] No handler defined!\n");
 777   }
 778 }
 779
 780 static void audio_pause(void)
 781 {
 782     int err;
 783
 784     if (alsa_can_pause) {
 785         if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
 786         {
 787             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm pause error: %s\n", snd_strerror(err));
 788             return;
 789         }
 790           mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
 791     } else {
 792         if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
 793             || prepause_frames < 0)
 794             prepause_frames = 0;
 795
 796         if ((err = snd_pcm_drop(alsa_handler)) < 0)
 797         {
 798             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm drop error: %s\n", snd_strerror(err));
 799             return;
 800         }
 801     }
 802 }
 803
 804 static void audio_resume(void)
 805 {
 806     int err;
 807
 808     if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
 809         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
 810         while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
 811     }
 812     if (alsa_can_pause) {
 813         if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
 814         {
 815             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm resume error: %s\n", snd_strerror(err));
 816             return;
 817         }
 818           mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
 819     } else {
 820         if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 821         {
 822            mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 823             return;
 824         }
 825         if (prepause_frames) {
 826             void *silence = calloc(prepause_frames, bytes_per_sample);
 827             play(silence, prepause_frames * bytes_per_sample, 0);
 828             free(silence);
 829         }
 830     }
 831 }
 832
 833 /* stop playing and empty buffers (for seeking/pause) */
 834 static void reset(void)
 835 {
 836     int err;
 837
 838     prepause_frames = 0;
 839     if ((err = snd_pcm_drop(alsa_handler)) < 0)
 840     {
 841         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 842         return;
 843     }
 844     if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 845     {
 846         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 847         return;
 848     }
 849     return;
 850 }
 851
 852 /*
 853     plays 'len' bytes of 'data'
 854     returns: number of bytes played
 855     modified last at 29.06.02 by jp
 856     thanxs for marius <marius@rospot.com> for giving us the light ;)
 857 */
 858
 859 static int play(void* data, int len, int flags)
 860 {
 861   int num_frames = len / bytes_per_sample;
 862   snd_pcm_sframes_t res = 0;
 863
 864   //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
 865
 866   if (!alsa_handler) {
 867     mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Device configuration error.");
 868     return 0;
 869   }
 870
 871   if (num_frames == 0)
 872     return 0;
 873
 874   do {
 875     res = snd_pcm_writei(alsa_handler, data, num_frames);
 876
 877       if (res == -EINTR) {
 878         /* nothing to do */
 879         res = 0;
 880       }
 881       else if (res == -ESTRPIPE) {      /* suspend */
 882         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
 883         while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
 884           sleep(1);
 885       }
 886       if (res < 0) {
 887         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Write error: %s\n", snd_strerror(res));
 888         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Trying to reset soundcard.\n");
 889         if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
 890           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(res));
 891           return 0;
 892           break;
 893         }
 894       }
 895   } while (res == 0);
 896
 897   return res < 0 ? res : res * bytes_per_sample;
 898 }
 899
 900 /* how many byes are free in the buffer */
 901 static int get_space(void)
 902 {
 903     snd_pcm_status_t *status;
 904     int ret;
 905
 906     snd_pcm_status_alloca(&status);
 907
 908     if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
 909     {
 910         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Cannot get pcm status: %s\n", snd_strerror(ret));
 911         return 0;
 912     }
 913
 914     unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
 915     if (space > ao_data.buffersize) // Buffer underrun?
 916         space = ao_data.buffersize;
 917     return space;
 918 }
 919
 920 /* delay in seconds between first and last sample in buffer */
 921 static float get_delay(void)
 922 {
 923   if (alsa_handler) {
 924     snd_pcm_sframes_t delay;
 925
 926     if (snd_pcm_delay(alsa_handler, &delay) < 0)
 927       return 0;
 928
 929     if (delay < 0) {
 930       /* underrun - move the application pointer forward to catch up */
 931 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
 932       snd_pcm_forward(alsa_handler, -delay);
 933 #endif
 934       delay = 0;
 935     }
 936     return (float)delay / (float)ao_data.samplerate;
 937   } else {
 938     return 0;
 939   }
 940 }