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
  43 #define ALSA_PCM_NEW_HW_PARAMS_API
  44 #define ALSA_PCM_NEW_SW_PARAMS_API
  45
  46 #include <alsa/asoundlib.h>
  47
  48 #include "audio_out.h"
  49 #include "audio_out_internal.h"
  50 #include "libaf/af_format.h"
  51
  52 static const ao_info_t info =
  53 {
  54     "ALSA-0.9.x-1.x audio output",
  55     "alsa",
  56     "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
  57     "under development"
  58 };
  59
  60 LIBAO_EXTERN(alsa)
  61
  62 static snd_pcm_t *alsa_handler;
  63 static snd_pcm_format_t alsa_format;
  64 static snd_pcm_hw_params_t *alsa_hwparams;
  65 static snd_pcm_sw_params_t *alsa_swparams;
  66
  67 #define BUFFER_TIME 500000  // 0.5 s
  68 #define FRAGCOUNT 16
  69
  70 static size_t bytes_per_sample;
  71
  72 static int alsa_can_pause;
  73 static snd_pcm_sframes_t prepause_frames;
  74
  75 #define ALSA_DEVICE_SIZE 256
  76
  77 static void alsa_error_handler(const char *file, int line, const char *function,
  78                                int err, const char *format, ...)
  79 {
  80   char tmp[0xc00];
  81   va_list va;
  82
  83   va_start(va, format);
  84   vsnprintf(tmp, sizeof tmp, format, va);
  85   va_end(va);
  86   tmp[sizeof tmp - 1] = '\0';
  87
  88   if (err)
  89     mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
  90            file, line, function, tmp, snd_strerror(err));
  91   else
  92     mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
  93            file, line, function, tmp);
  94 }
  95
  96 /* to set/get/query special features/parameters */
  97 static int control(int cmd, void *arg)
  98 {
  99   switch(cmd) {
 100   case AOCONTROL_QUERY_FORMAT:
 101     return CONTROL_TRUE;
 102   case AOCONTROL_GET_VOLUME:
 103   case AOCONTROL_SET_VOLUME:
 104     {
 105       ao_control_vol_t *vol = (ao_control_vol_t *)arg;
 106
 107       int err;
 108       snd_mixer_t *handle;
 109       snd_mixer_elem_t *elem;
 110       snd_mixer_selem_id_t *sid;
 111
 112       char *mix_name = "Master";
 113       char *card = "default";
 114       int mix_index = 0;
 115
 116       long pmin, pmax;
 117       long get_vol, set_vol;
 118       float f_multi;
 119
 120       if(AF_FORMAT_IS_AC3(ao_data.format))
 121         return CONTROL_TRUE;
 122
 123       if(mixer_channel) {
 124          char *test_mix_index;
 125
 126          mix_name = strdup(mixer_channel);
 127          if ((test_mix_index = strchr(mix_name, ','))){
 128                 *test_mix_index = 0;
 129                 test_mix_index++;
 130                 mix_index = strtol(test_mix_index, &test_mix_index, 0);
 131
 132                 if (*test_mix_index){
 133                   mp_tmsg(MSGT_AO,MSGL_ERR,
 134                     "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
 135                   mix_index = 0 ;
 136                 }
 137          }
 138       }
 139       if(mixer_device) card = mixer_device;
 140
 141       //allocate simple id
 142       snd_mixer_selem_id_alloca(&sid);
 143
 144       //sets simple-mixer index and name
 145       snd_mixer_selem_id_set_index(sid, mix_index);
 146       snd_mixer_selem_id_set_name(sid, mix_name);
 147
 148       if (mixer_channel) {
 149         free(mix_name);
 150         mix_name = NULL;
 151       }
 152
 153       if ((err = snd_mixer_open(&handle, 0)) < 0) {
 154         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer open error: %s\n", snd_strerror(err));
 155         return CONTROL_ERROR;
 156       }
 157
 158       if ((err = snd_mixer_attach(handle, card)) < 0) {
 159         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer attach %s error: %s\n",
 160                card, snd_strerror(err));
 161         snd_mixer_close(handle);
 162         return CONTROL_ERROR;
 163       }
 164
 165       if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
 166         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer register error: %s\n", snd_strerror(err));
 167         snd_mixer_close(handle);
 168         return CONTROL_ERROR;
 169       }
 170       err = snd_mixer_load(handle);
 171       if (err < 0) {
 172         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer load error: %s\n", snd_strerror(err));
 173         snd_mixer_close(handle);
 174         return CONTROL_ERROR;
 175       }
 176
 177       elem = snd_mixer_find_selem(handle, sid);
 178       if (!elem) {
 179         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to find simple control '%s',%i.\n",
 180                snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
 181         snd_mixer_close(handle);
 182         return CONTROL_ERROR;
 183         }
 184
 185       snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
 186       f_multi = (100 / (float)(pmax - pmin));
 187
 188       if (cmd == AOCONTROL_SET_VOLUME) {
 189
 190         set_vol = vol->left / f_multi + pmin + 0.5;
 191
 192         //setting channels
 193         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
 194           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n",
 195                  snd_strerror(err));
 196           snd_mixer_close(handle);
 197           return CONTROL_ERROR;
 198         }
 199         mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
 200
 201         set_vol = vol->right / f_multi + pmin + 0.5;
 202
 203         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
 204           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
 205                  snd_strerror(err));
 206           snd_mixer_close(handle);
 207           return CONTROL_ERROR;
 208         }
 209         mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
 210                set_vol, pmin, pmax, f_multi);
 211
 212         if (snd_mixer_selem_has_playback_switch(elem)) {
 213           int lmute = (vol->left == 0.0);
 214           int rmute = (vol->right == 0.0);
 215           if (snd_mixer_selem_has_playback_switch_joined(elem)) {
 216             lmute = rmute = lmute && rmute;
 217           } else {
 218             snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
 219           }
 220           snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
 221         }
 222       }
 223       else {
 224         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
 225         vol->left = (get_vol - pmin) * f_multi;
 226         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
 227         vol->right = (get_vol - pmin) * f_multi;
 228
 229         mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
 230       }
 231       snd_mixer_close(handle);
 232       return CONTROL_OK;
 233     }
 234
 235   } //end switch
 236   return CONTROL_UNKNOWN;
 237 }
 238
 239 static void parse_device (char *dest, const char *src, int len)
 240 {
 241   char *tmp;
 242   memmove(dest, src, len);
 243   dest[len] = 0;
 244   while ((tmp = strrchr(dest, '.')))
 245     tmp[0] = ',';
 246   while ((tmp = strrchr(dest, '=')))
 247     tmp[0] = ':';
 248 }
 249
 250 static void print_help (void)
 251 {
 252   mp_tmsg (MSGT_AO, MSGL_FATAL,
 253     "\n[AO_ALSA] -ao alsa commandline help:\n"\
 254     "[AO_ALSA] Example: mplayer -ao alsa:device=hw=0.3\n"\
 255     "[AO_ALSA]   Sets first card fourth hardware device.\n\n"\
 256     "[AO_ALSA] Options:\n"\
 257     "[AO_ALSA]   noblock\n"\
 258     "[AO_ALSA]     Opens device in non-blocking mode.\n"\
 259     "[AO_ALSA]   device=<device-name>\n"\
 260     "[AO_ALSA]     Sets device (change , to . and : to =)\n");
 261 }
 262
 263 static int str_maxlen(void *strp) {
 264   strarg_t *str = strp;
 265   return str->len <= ALSA_DEVICE_SIZE;
 266 }
 267
 268 static int try_open_device(const char *device, int open_mode, int try_ac3)
 269 {
 270   int err, len;
 271   char *ac3_device, *args;
 272
 273   if (try_ac3) {
 274     /* to set the non-audio bit, use AES0=6 */
 275     len = strlen(device);
 276     ac3_device = malloc(len + 7 + 1);
 277     if (!ac3_device)
 278       return -ENOMEM;
 279     strcpy(ac3_device, device);
 280     args = strchr(ac3_device, ':');
 281     if (!args) {
 282       /* no existing parameters: add it behind device name */
 283       strcat(ac3_device, ":AES0=6");
 284     } else {
 285       do
 286         ++args;
 287       while (isspace(*args));
 288       if (*args == '\0') {
 289         /* ":" but no parameters */
 290         strcat(ac3_device, "AES0=6");
 291       } else if (*args != '{') {
 292         /* a simple list of parameters: add it at the end of the list */
 293         strcat(ac3_device, ",AES0=6");
 294       } else {
 295         /* parameters in config syntax: add it inside the { } block */
 296         do
 297           --len;
 298         while (len > 0 && isspace(ac3_device[len]));
 299         if (ac3_device[len] == '}')
 300           strcpy(ac3_device + len, " AES0=6}");
 301       }
 302     }
 303     err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
 304                        open_mode);
 305     free(ac3_device);
 306     if (!err)
 307       return 0;
 308   }
 309   return snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
 310                       open_mode);
 311 }
 312
 313 /*
 314     open & setup audio device
 315     return: 1=success 0=fail
 316 */
 317 static int init(int rate_hz, int channels, int format, int flags)
 318 {
 319     int err;
 320     int block;
 321     strarg_t device;
 322     snd_pcm_uframes_t chunk_size;
 323     snd_pcm_uframes_t bufsize;
 324     snd_pcm_uframes_t boundary;
 325     const opt_t subopts[] = {
 326       {"block", OPT_ARG_BOOL, &block, NULL},
 327       {"device", OPT_ARG_STR, &device, str_maxlen},
 328       {NULL}
 329     };
 330
 331     char alsa_device[ALSA_DEVICE_SIZE + 1];
 332     // make sure alsa_device is null-terminated even when using strncpy etc.
 333     memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
 334
 335     mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
 336         channels, format);
 337     alsa_handler = NULL;
 338     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
 339
 340     prepause_frames = 0;
 341
 342     snd_lib_error_set_handler(alsa_error_handler);
 343
 344     ao_data.samplerate = rate_hz;
 345     ao_data.format = format;
 346     ao_data.channels = channels;
 347
 348     switch (format)
 349       {
 350       case AF_FORMAT_S8:
 351         alsa_format = SND_PCM_FORMAT_S8;
 352         break;
 353       case AF_FORMAT_U8:
 354         alsa_format = SND_PCM_FORMAT_U8;
 355         break;
 356       case AF_FORMAT_U16_LE:
 357         alsa_format = SND_PCM_FORMAT_U16_LE;
 358         break;
 359       case AF_FORMAT_U16_BE:
 360         alsa_format = SND_PCM_FORMAT_U16_BE;
 361         break;
 362       case AF_FORMAT_AC3_LE:
 363       case AF_FORMAT_S16_LE:
 364         alsa_format = SND_PCM_FORMAT_S16_LE;
 365         break;
 366       case AF_FORMAT_AC3_BE:
 367       case AF_FORMAT_S16_BE:
 368         alsa_format = SND_PCM_FORMAT_S16_BE;
 369         break;
 370       case AF_FORMAT_U32_LE:
 371         alsa_format = SND_PCM_FORMAT_U32_LE;
 372         break;
 373       case AF_FORMAT_U32_BE:
 374         alsa_format = SND_PCM_FORMAT_U32_BE;
 375         break;
 376       case AF_FORMAT_S32_LE:
 377         alsa_format = SND_PCM_FORMAT_S32_LE;
 378         break;
 379       case AF_FORMAT_S32_BE:
 380         alsa_format = SND_PCM_FORMAT_S32_BE;
 381         break;
 382       case AF_FORMAT_U24_LE:
 383         alsa_format = SND_PCM_FORMAT_U24_3LE;
 384         break;
 385       case AF_FORMAT_U24_BE:
 386         alsa_format = SND_PCM_FORMAT_U24_3BE;
 387         break;
 388       case AF_FORMAT_S24_LE:
 389         alsa_format = SND_PCM_FORMAT_S24_3LE;
 390         break;
 391       case AF_FORMAT_S24_BE:
 392         alsa_format = SND_PCM_FORMAT_S24_3BE;
 393         break;
 394       case AF_FORMAT_FLOAT_LE:
 395         alsa_format = SND_PCM_FORMAT_FLOAT_LE;
 396         break;
 397       case AF_FORMAT_FLOAT_BE:
 398         alsa_format = SND_PCM_FORMAT_FLOAT_BE;
 399         break;
 400       case AF_FORMAT_MU_LAW:
 401         alsa_format = SND_PCM_FORMAT_MU_LAW;
 402         break;
 403       case AF_FORMAT_A_LAW:
 404         alsa_format = SND_PCM_FORMAT_A_LAW;
 405         break;
 406
 407       default:
 408         alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
 409         break;
 410       }
 411
 412     //subdevice parsing
 413     // set defaults
 414     block = 1;
 415     /* switch for spdif
 416      * sets opening sequence for SPDIF
 417      * sets also the playback and other switches 'on the fly'
 418      * while opening the abstract alias for the spdif subdevice
 419      * 'iec958'
 420      */
 421     if (AF_FORMAT_IS_AC3(format)) {
 422         device.str = "iec958";
 423         mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
 424     }
 425   else
 426         /* in any case for multichannel playback we should select
 427          * appropriate device
 428          */
 429         switch (channels) {
 430         case 1:
 431         case 2:
 432           device.str = "default";
 433           mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
 434           break;
 435         case 4:
 436           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 437             // hack - use the converter plugin
 438             device.str = "plug:surround40";
 439           else
 440             device.str = "surround40";
 441           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
 442           break;
 443         case 6:
 444           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 445             device.str = "plug:surround51";
 446           else
 447             device.str = "surround51";
 448           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
 449           break;
 450         case 8:
 451           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 452             device.str = "plug:surround71";
 453           else
 454             device.str = "surround71";
 455           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n");
 456           break;
 457         default:
 458           device.str = "default";
 459           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] %d channels are not supported.\n",channels);
 460         }
 461     device.len = strlen(device.str);
 462     if (subopt_parse(ao_subdevice, subopts) != 0) {
 463         print_help();
 464         return 0;
 465     }
 466     parse_device(alsa_device, device.str, device.len);
 467
 468     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
 469
 470     if (!alsa_handler) {
 471       int open_mode = block ? 0 : SND_PCM_NONBLOCK;
 472       int isac3 =  AF_FORMAT_IS_AC3(format);
 473       //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
 474       if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
 475         {
 476           if (err != -EBUSY && !block) {
 477             mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
 478             if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
 479               mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 480               return 0;
 481             }
 482           } else {
 483             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 484             return 0;
 485           }
 486         }
 487
 488       if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
 489          mp_tmsg(MSGT_AO,MSGL_ERR,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err));
 490       } else {
 491         mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
 492       }
 493
 494       snd_pcm_hw_params_alloca(&alsa_hwparams);
 495       snd_pcm_sw_params_alloca(&alsa_swparams);
 496
 497       // setting hw-parameters
 498       if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
 499         {
 500           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get initial parameters: %s\n",
 501                  snd_strerror(err));
 502           return 0;
 503         }
 504
 505       err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
 506                                          SND_PCM_ACCESS_RW_INTERLEAVED);
 507       if (err < 0) {
 508         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set access type: %s\n",
 509                snd_strerror(err));
 510         return 0;
 511       }
 512
 513       /* workaround for nonsupported formats
 514          sets default format to S16_LE if the given formats aren't supported */
 515       if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
 516                                              alsa_format)) < 0)
 517       {
 518          mp_tmsg(MSGT_AO,MSGL_INFO,
 519                 "[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format));
 520          alsa_format = SND_PCM_FORMAT_S16_LE;
 521          if (AF_FORMAT_IS_AC3(ao_data.format))
 522            ao_data.format = AF_FORMAT_AC3_LE;
 523          else
 524          ao_data.format = AF_FORMAT_S16_LE;
 525       }
 526
 527       if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
 528                                               alsa_format)) < 0)
 529         {
 530           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set format: %s\n",
 531                  snd_strerror(err));
 532           return 0;
 533         }
 534
 535       if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
 536                                                      &ao_data.channels)) < 0)
 537         {
 538           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set channels: %s\n",
 539                  snd_strerror(err));
 540           return 0;
 541         }
 542
 543       /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
 544          prefer our own resampler, since that allows users to choose the resampler,
 545          even per file if desired */
 546       if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
 547                                                      0)) < 0)
 548         {
 549           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to disable resampling: %s\n",
 550                  snd_strerror(err));
 551           return 0;
 552         }
 553
 554       if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
 555                                                  &ao_data.samplerate, NULL)) < 0)
 556         {
 557           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set samplerate-2: %s\n",
 558                  snd_strerror(err));
 559           return 0;
 560         }
 561
 562       bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
 563       bytes_per_sample *= ao_data.channels;
 564       ao_data.bps = ao_data.samplerate * bytes_per_sample;
 565
 566         if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
 567                                                           &(unsigned int){BUFFER_TIME}, NULL)) < 0)
 568           {
 569             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set buffer time near: %s\n",
 570                    snd_strerror(err));
 571             return 0;
 572           }
 573
 574         if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
 575                                                       &(unsigned int){FRAGCOUNT}, NULL)) < 0) {
 576           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n",
 577                  snd_strerror(err));
 578           return 0;
 579         }
 580
 581       /* finally install hardware parameters */
 582       if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
 583         {
 584           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set hw-parameters: %s\n",
 585                  snd_strerror(err));
 586           return 0;
 587         }
 588       // end setting hw-params
 589
 590
 591       // gets buffersize for control
 592       if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
 593         {
 594           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(err));
 595           return 0;
 596         }
 597       else {
 598         ao_data.buffersize = bufsize * bytes_per_sample;
 599           mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
 600       }
 601
 602       if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
 603         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to get period size: %s\n", snd_strerror(err));
 604         return 0;
 605       } else {
 606         mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
 607       }
 608       ao_data.outburst = chunk_size * bytes_per_sample;
 609
 610       /* setting software parameters */
 611       if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
 612         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
 613                snd_strerror(err));
 614         return 0;
 615       }
 616       if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
 617         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get boundary: %s\n",
 618                snd_strerror(err));
 619         return 0;
 620       }
 621       /* start playing when one period has been written */
 622       if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
 623         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set start threshold: %s\n",
 624                snd_strerror(err));
 625         return 0;
 626       }
 627       /* disable underrun reporting */
 628       if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
 629         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set stop threshold: %s\n",
 630                snd_strerror(err));
 631         return 0;
 632       }
 633       /* play silence when there is an underrun */
 634       if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
 635         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set silence size: %s\n",
 636                snd_strerror(err));
 637         return 0;
 638       }
 639       if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
 640         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
 641                snd_strerror(err));
 642         return 0;
 643       }
 644       /* end setting sw-params */
 645
 646       mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
 647              ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
 648              snd_pcm_format_description(alsa_format));
 649
 650     } // end switch alsa_handler (spdif)
 651     alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
 652     return 1;
 653 } // end init
 654
 655
 656 /* close audio device */
 657 static void uninit(int immed)
 658 {
 659
 660   if (alsa_handler) {
 661     int err;
 662
 663     if (!immed)
 664       snd_pcm_drain(alsa_handler);
 665
 666     if ((err = snd_pcm_close(alsa_handler)) < 0)
 667       {
 668         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm close error: %s\n", snd_strerror(err));
 669         return;
 670       }
 671     else {
 672       alsa_handler = NULL;
 673       mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
 674     }
 675   }
 676   else {
 677     mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] No handler defined!\n");
 678   }
 679 }
 680
 681 static void audio_pause(void)
 682 {
 683     int err;
 684
 685     if (alsa_can_pause) {
 686         if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
 687         {
 688             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm pause error: %s\n", snd_strerror(err));
 689             return;
 690         }
 691           mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
 692     } else {
 693         if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
 694             || prepause_frames < 0)
 695             prepause_frames = 0;
 696
 697         if ((err = snd_pcm_drop(alsa_handler)) < 0)
 698         {
 699             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm drop error: %s\n", snd_strerror(err));
 700             return;
 701         }
 702     }
 703 }
 704
 705 static void audio_resume(void)
 706 {
 707     int err;
 708
 709     if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
 710         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
 711         while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
 712     }
 713     if (alsa_can_pause) {
 714         if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
 715         {
 716             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm resume error: %s\n", snd_strerror(err));
 717             return;
 718         }
 719           mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
 720     } else {
 721         if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 722         {
 723            mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 724             return;
 725         }
 726         if (prepause_frames) {
 727             void *silence = calloc(prepause_frames, bytes_per_sample);
 728             play(silence, prepause_frames * bytes_per_sample, 0);
 729             free(silence);
 730         }
 731     }
 732 }
 733
 734 /* stop playing and empty buffers (for seeking/pause) */
 735 static void reset(void)
 736 {
 737     int err;
 738
 739     prepause_frames = 0;
 740     if ((err = snd_pcm_drop(alsa_handler)) < 0)
 741     {
 742         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 743         return;
 744     }
 745     if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 746     {
 747         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 748         return;
 749     }
 750     return;
 751 }
 752
 753 /*
 754     plays 'len' bytes of 'data'
 755     returns: number of bytes played
 756     modified last at 29.06.02 by jp
 757     thanxs for marius <marius@rospot.com> for giving us the light ;)
 758 */
 759
 760 static int play(void* data, int len, int flags)
 761 {
 762   int num_frames;
 763   snd_pcm_sframes_t res = 0;
 764   if (!(flags & AOPLAY_FINAL_CHUNK))
 765       len = len / ao_data.outburst * ao_data.outburst;
 766   num_frames = len / bytes_per_sample;
 767
 768   //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
 769
 770   if (!alsa_handler) {
 771     mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Device configuration error.");
 772     return 0;
 773   }
 774
 775   if (num_frames == 0)
 776     return 0;
 777
 778   do {
 779     res = snd_pcm_writei(alsa_handler, data, num_frames);
 780
 781       if (res == -EINTR) {
 782         /* nothing to do */
 783         res = 0;
 784       }
 785       else if (res == -ESTRPIPE) {      /* suspend */
 786         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
 787         while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
 788           sleep(1);
 789       }
 790       if (res < 0) {
 791         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Write error: %s\n", snd_strerror(res));
 792         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Trying to reset soundcard.\n");
 793         if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
 794           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(res));
 795           return 0;
 796           break;
 797         }
 798       }
 799   } while (res == 0);
 800
 801   return res < 0 ? res : res * bytes_per_sample;
 802 }
 803
 804 /* how many byes are free in the buffer */
 805 static int get_space(void)
 806 {
 807     snd_pcm_status_t *status;
 808     int ret;
 809
 810     snd_pcm_status_alloca(&status);
 811
 812     if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
 813     {
 814         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Cannot get pcm status: %s\n", snd_strerror(ret));
 815         return 0;
 816     }
 817
 818     unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
 819     if (space > ao_data.buffersize) // Buffer underrun?
 820         space = ao_data.buffersize;
 821     return space;
 822 }
 823
 824 /* delay in seconds between first and last sample in buffer */
 825 static float get_delay(void)
 826 {
 827   if (alsa_handler) {
 828     snd_pcm_sframes_t delay;
 829
 830     if (snd_pcm_delay(alsa_handler, &delay) < 0)
 831       return 0;
 832
 833     if (delay < 0) {
 834       /* underrun - move the application pointer forward to catch up */
 835       snd_pcm_forward(alsa_handler, -delay);
 836       delay = 0;
 837     }
 838     return (float)delay / (float)ao_data.samplerate;
 839   } else {
 840     return 0;
 841   }
 842 }