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_msg(MSGT_AO,MSGL_ERR,
 151                     MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero);
 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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerOpenError, snd_strerror(err));
 172         return CONTROL_ERROR;
 173       }
 174
 175       if ((err = snd_mixer_attach(handle, card)) < 0) {
 176         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError,
 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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerRegisterError, 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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerLoadError, 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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToFindSimpleControl,
 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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingLeftChannel,
 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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingRightChannel,
 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_msg (MSGT_AO, MSGL_FATAL,
 268            MSGTR_AO_ALSA_CommandlineHelp);
 269 }
 270
 271 static int str_maxlen(strarg_t *str) {
 272   if (str->len > ALSA_DEVICE_SIZE)
 273     return 0;
 274   return 1;
 275 }
 276
 277 static int try_open_device(const char *device, int open_mode, int try_ac3)
 278 {
 279   int err, len;
 280   char *ac3_device, *args;
 281
 282   if (try_ac3) {
 283     /* to set the non-audio bit, use AES0=6 */
 284     len = strlen(device);
 285     ac3_device = malloc(len + 7 + 1);
 286     if (!ac3_device)
 287       return -ENOMEM;
 288     strcpy(ac3_device, device);
 289     args = strchr(ac3_device, ':');
 290     if (!args) {
 291       /* no existing parameters: add it behind device name */
 292       strcat(ac3_device, ":AES0=6");
 293     } else {
 294       do
 295         ++args;
 296       while (isspace(*args));
 297       if (*args == '\0') {
 298         /* ":" but no parameters */
 299         strcat(ac3_device, "AES0=6");
 300       } else if (*args != '{') {
 301         /* a simple list of parameters: add it at the end of the list */
 302         strcat(ac3_device, ",AES0=6");
 303       } else {
 304         /* parameters in config syntax: add it inside the { } block */
 305         do
 306           --len;
 307         while (len > 0 && isspace(ac3_device[len]));
 308         if (ac3_device[len] == '}')
 309           strcpy(ac3_device + len, " AES0=6}");
 310       }
 311     }
 312     err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
 313                        open_mode);
 314     free(ac3_device);
 315   }
 316   if (!try_ac3 || err < 0)
 317     err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
 318                        open_mode);
 319   return err;
 320 }
 321
 322 /*
 323     open & setup audio device
 324     return: 1=success 0=fail
 325 */
 326 static int init(int rate_hz, int channels, int format, int flags)
 327 {
 328     int err;
 329     int block;
 330     strarg_t device;
 331     snd_pcm_uframes_t bufsize;
 332     snd_pcm_uframes_t boundary;
 333     opt_t subopts[] = {
 334       {"block", OPT_ARG_BOOL, &block, NULL},
 335       {"device", OPT_ARG_STR, &device, (opt_test_f)str_maxlen},
 336       {NULL}
 337     };
 338
 339     char alsa_device[ALSA_DEVICE_SIZE + 1];
 340     // make sure alsa_device is null-terminated even when using strncpy etc.
 341     memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
 342
 343     mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
 344         channels, format);
 345     alsa_handler = NULL;
 346 #if SND_LIB_VERSION >= 0x010005
 347     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
 348 #else
 349     mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
 350 #endif
 351
 352     prepause_frames = 0;
 353
 354     snd_lib_error_set_handler(alsa_error_handler);
 355
 356     ao_data.samplerate = rate_hz;
 357     ao_data.format = format;
 358     ao_data.channels = channels;
 359
 360     switch (format)
 361       {
 362       case AF_FORMAT_S8:
 363         alsa_format = SND_PCM_FORMAT_S8;
 364         break;
 365       case AF_FORMAT_U8:
 366         alsa_format = SND_PCM_FORMAT_U8;
 367         break;
 368       case AF_FORMAT_U16_LE:
 369         alsa_format = SND_PCM_FORMAT_U16_LE;
 370         break;
 371       case AF_FORMAT_U16_BE:
 372         alsa_format = SND_PCM_FORMAT_U16_BE;
 373         break;
 374 #ifndef WORDS_BIGENDIAN
 375       case AF_FORMAT_AC3:
 376 #endif
 377       case AF_FORMAT_S16_LE:
 378         alsa_format = SND_PCM_FORMAT_S16_LE;
 379         break;
 380 #ifdef WORDS_BIGENDIAN
 381       case AF_FORMAT_AC3:
 382 #endif
 383       case AF_FORMAT_S16_BE:
 384         alsa_format = SND_PCM_FORMAT_S16_BE;
 385         break;
 386       case AF_FORMAT_U32_LE:
 387         alsa_format = SND_PCM_FORMAT_U32_LE;
 388         break;
 389       case AF_FORMAT_U32_BE:
 390         alsa_format = SND_PCM_FORMAT_U32_BE;
 391         break;
 392       case AF_FORMAT_S32_LE:
 393         alsa_format = SND_PCM_FORMAT_S32_LE;
 394         break;
 395       case AF_FORMAT_S32_BE:
 396         alsa_format = SND_PCM_FORMAT_S32_BE;
 397         break;
 398       case AF_FORMAT_FLOAT_LE:
 399         alsa_format = SND_PCM_FORMAT_FLOAT_LE;
 400         break;
 401       case AF_FORMAT_FLOAT_BE:
 402         alsa_format = SND_PCM_FORMAT_FLOAT_BE;
 403         break;
 404       case AF_FORMAT_MU_LAW:
 405         alsa_format = SND_PCM_FORMAT_MU_LAW;
 406         break;
 407       case AF_FORMAT_A_LAW:
 408         alsa_format = SND_PCM_FORMAT_A_LAW;
 409         break;
 410
 411       default:
 412         alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
 413         break;
 414       }
 415
 416     //subdevice parsing
 417     // set defaults
 418     block = 1;
 419     /* switch for spdif
 420      * sets opening sequence for SPDIF
 421      * sets also the playback and other switches 'on the fly'
 422      * while opening the abstract alias for the spdif subdevice
 423      * 'iec958'
 424      */
 425     if (format == AF_FORMAT_AC3) {
 426         device.str = "iec958";
 427         mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
 428     }
 429   else
 430         /* in any case for multichannel playback we should select
 431          * appropriate device
 432          */
 433         switch (channels) {
 434         case 1:
 435         case 2:
 436           device.str = "default";
 437           mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
 438           break;
 439         case 4:
 440           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 441             // hack - use the converter plugin
 442             device.str = "plug:surround40";
 443           else
 444             device.str = "surround40";
 445           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
 446           break;
 447         case 6:
 448           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 449             device.str = "plug:surround51";
 450           else
 451             device.str = "surround51";
 452           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
 453           break;
 454         default:
 455           device.str = "default";
 456           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels);
 457         }
 458     device.len = strlen(device.str);
 459     if (subopt_parse(ao_subdevice, subopts) != 0) {
 460         print_help();
 461         return 0;
 462     }
 463     ao_noblock = !block;
 464     parse_device(alsa_device, device.str, device.len);
 465
 466     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
 467
 468     //setting modes for block or nonblock-mode
 469     if (ao_noblock) {
 470       open_mode = SND_PCM_NONBLOCK;
 471     }
 472     else {
 473       open_mode = 0;
 474     }
 475
 476     //sets buff/chunksize if its set manually
 477     if (ao_data.buffersize) {
 478       switch (ao_data.buffersize)
 479         {
 480         case 1:
 481           alsa_fragcount = 16;
 482           chunk_size = 512;
 483             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
 484             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
 485           break;
 486         case 2:
 487           alsa_fragcount = 8;
 488           chunk_size = 1024;
 489             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
 490             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
 491           break;
 492         case 3:
 493           alsa_fragcount = 32;
 494           chunk_size = 512;
 495             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
 496             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
 497           break;
 498         case 4:
 499           alsa_fragcount = 16;
 500           chunk_size = 1024;
 501             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
 502             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
 503           break;
 504         default:
 505           alsa_fragcount = 16;
 506           chunk_size = 1024;
 507           break;
 508         }
 509     }
 510
 511     if (!alsa_handler) {
 512       //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
 513       if ((err = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0)
 514         {
 515           if (err != -EBUSY && ao_noblock) {
 516             mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
 517             if ((err = try_open_device(alsa_device, 0, format == AF_FORMAT_AC3)) < 0) {
 518               mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
 519               return 0;
 520             }
 521           } else {
 522             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
 523             return 0;
 524           }
 525         }
 526
 527       if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
 528          mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
 529       } else {
 530         mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
 531       }
 532
 533       snd_pcm_hw_params_alloca(&alsa_hwparams);
 534       snd_pcm_sw_params_alloca(&alsa_swparams);
 535
 536       // setting hw-parameters
 537       if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
 538         {
 539           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters,
 540                  snd_strerror(err));
 541           return 0;
 542         }
 543
 544       err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
 545                                          SND_PCM_ACCESS_RW_INTERLEAVED);
 546       if (err < 0) {
 547         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType,
 548                snd_strerror(err));
 549         return 0;
 550       }
 551
 552       /* workaround for nonsupported formats
 553          sets default format to S16_LE if the given formats aren't supported */
 554       if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
 555                                              alsa_format)) < 0)
 556       {
 557          mp_msg(MSGT_AO,MSGL_INFO,
 558                 MSGTR_AO_ALSA_FormatNotSupportedByHardware, af_fmt2str_short(format));
 559          alsa_format = SND_PCM_FORMAT_S16_LE;
 560          ao_data.format = AF_FORMAT_S16_LE;
 561       }
 562
 563       if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
 564                                               alsa_format)) < 0)
 565         {
 566           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat,
 567                  snd_strerror(err));
 568           return 0;
 569         }
 570
 571       if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
 572                                                      &ao_data.channels)) < 0)
 573         {
 574           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels,
 575                  snd_strerror(err));
 576           return 0;
 577         }
 578
 579       /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
 580          prefer our own resampler */
 581 #if SND_LIB_VERSION >= 0x010009
 582       if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
 583                                                      0)) < 0)
 584         {
 585           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling,
 586                  snd_strerror(err));
 587           return 0;
 588         }
 589 #endif
 590
 591       if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
 592                                                  &ao_data.samplerate, NULL)) < 0)
 593         {
 594           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2,
 595                  snd_strerror(err));
 596           return 0;
 597         }
 598
 599       bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8;
 600       bytes_per_sample *= ao_data.channels;
 601       ao_data.bps = ao_data.samplerate * bytes_per_sample;
 602
 603 #ifdef BUFFERTIME
 604       {
 605         int alsa_buffer_time = 500000; /* original 60 */
 606         int alsa_period_time;
 607         alsa_period_time = alsa_buffer_time/4;
 608         if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
 609                                                           &alsa_buffer_time, NULL)) < 0)
 610           {
 611             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
 612                    snd_strerror(err));
 613             return 0;
 614           } else
 615             alsa_buffer_time = err;
 616
 617         if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
 618                                                           &alsa_period_time, NULL)) < 0)
 619           /* original: alsa_buffer_time/ao_data.bps */
 620           {
 621             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodTime,
 622                    snd_strerror(err));
 623             return 0;
 624           }
 625         mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_BufferTimePeriodTime,
 626                alsa_buffer_time, err);
 627       }
 628 #endif//end SET_BUFFERTIME
 629
 630 #ifdef SET_CHUNKSIZE
 631       {
 632         //set chunksize
 633         if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
 634                                                           &chunk_size, NULL)) < 0)
 635           {
 636             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodSize,
 637                             chunk_size, snd_strerror(err));
 638             return 0;
 639           }
 640         else {
 641           mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %li\n", chunk_size);
 642         }
 643         if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
 644                                                       &alsa_fragcount, NULL)) < 0) {
 645           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods,
 646                  snd_strerror(err));
 647           return 0;
 648         }
 649         else {
 650           mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount);
 651         }
 652       }
 653 #endif//end SET_CHUNKSIZE
 654
 655       /* finally install hardware parameters */
 656       if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
 657         {
 658           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters,
 659                  snd_strerror(err));
 660           return 0;
 661         }
 662       // end setting hw-params
 663
 664
 665       // gets buffersize for control
 666       if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
 667         {
 668           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err));
 669           return 0;
 670         }
 671       else {
 672         ao_data.buffersize = bufsize * bytes_per_sample;
 673           mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
 674       }
 675
 676       if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
 677         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err));
 678         return 0;
 679       } else {
 680         mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
 681       }
 682       ao_data.outburst = chunk_size * bytes_per_sample;
 683
 684       /* setting software parameters */
 685       if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
 686         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
 687                snd_strerror(err));
 688         return 0;
 689       }
 690 #if SND_LIB_VERSION >= 0x000901
 691       if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
 692         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary,
 693                snd_strerror(err));
 694         return 0;
 695       }
 696 #else
 697       boundary = 0x7fffffff;
 698 #endif
 699       /* start playing when one period has been written */
 700       if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
 701         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold,
 702                snd_strerror(err));
 703         return 0;
 704       }
 705       /* disable underrun reporting */
 706       if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
 707         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold,
 708                snd_strerror(err));
 709         return 0;
 710       }
 711 #if SND_LIB_VERSION >= 0x000901
 712       /* play silence when there is an underrun */
 713       if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
 714         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize,
 715                snd_strerror(err));
 716         return 0;
 717       }
 718 #endif
 719       if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
 720         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
 721                snd_strerror(err));
 722         return 0;
 723       }
 724       /* end setting sw-params */
 725
 726       mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
 727              ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
 728              snd_pcm_format_description(alsa_format));
 729
 730     } // end switch alsa_handler (spdif)
 731     alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
 732     return 1;
 733 } // end init
 734
 735
 736 /* close audio device */
 737 static void uninit(int immed)
 738 {
 739
 740   if (alsa_handler) {
 741     int err;
 742
 743     if (!immed)
 744       snd_pcm_drain(alsa_handler);
 745
 746     if ((err = snd_pcm_close(alsa_handler)) < 0)
 747       {
 748         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmCloseError, snd_strerror(err));
 749         return;
 750       }
 751     else {
 752       alsa_handler = NULL;
 753       mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
 754     }
 755   }
 756   else {
 757     mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_NoHandlerDefined);
 758   }
 759 }
 760
 761 static void audio_pause(void)
 762 {
 763     int err;
 764
 765     if (alsa_can_pause) {
 766         if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
 767         {
 768             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPauseError, snd_strerror(err));
 769             return;
 770         }
 771           mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
 772     } else {
 773         if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
 774             || prepause_frames < 0)
 775             prepause_frames = 0;
 776
 777         if ((err = snd_pcm_drop(alsa_handler)) < 0)
 778         {
 779             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmDropError, snd_strerror(err));
 780             return;
 781         }
 782     }
 783 }
 784
 785 static void audio_resume(void)
 786 {
 787     int err;
 788
 789     if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
 790         mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
 791         while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
 792     }
 793     if (alsa_can_pause) {
 794         if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
 795         {
 796             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmResumeError, snd_strerror(err));
 797             return;
 798         }
 799           mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
 800     } else {
 801         if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 802         {
 803            mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
 804             return;
 805         }
 806         if (prepause_frames) {
 807             void *silence = calloc(prepause_frames, bytes_per_sample);
 808             play(silence, prepause_frames * bytes_per_sample, 0);
 809             free(silence);
 810         }
 811     }
 812 }
 813
 814 /* stop playing and empty buffers (for seeking/pause) */
 815 static void reset(void)
 816 {
 817     int err;
 818
 819     prepause_frames = 0;
 820     if ((err = snd_pcm_drop(alsa_handler)) < 0)
 821     {
 822         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
 823         return;
 824     }
 825     if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 826     {
 827         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
 828         return;
 829     }
 830     return;
 831 }
 832
 833 /*
 834     plays 'len' bytes of 'data'
 835     returns: number of bytes played
 836     modified last at 29.06.02 by jp
 837     thanxs for marius <marius@rospot.com> for giving us the light ;)
 838 */
 839
 840 static int play(void* data, int len, int flags)
 841 {
 842   int num_frames = len / bytes_per_sample;
 843   snd_pcm_sframes_t res = 0;
 844
 845   //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
 846
 847   if (!alsa_handler) {
 848     mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_DeviceConfigurationError);
 849     return 0;
 850   }
 851
 852   if (num_frames == 0)
 853     return 0;
 854
 855   do {
 856     res = snd_pcm_writei(alsa_handler, data, num_frames);
 857
 858       if (res == -EINTR) {
 859         /* nothing to do */
 860         res = 0;
 861       }
 862       else if (res == -ESTRPIPE) {      /* suspend */
 863         mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
 864         while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
 865           sleep(1);
 866       }
 867       if (res < 0) {
 868         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_WriteError, snd_strerror(res));
 869         mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_TryingToResetSoundcard);
 870         if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
 871           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(res));
 872           return 0;
 873           break;
 874         }
 875       }
 876   } while (res == 0);
 877
 878   return res < 0 ? res : res * bytes_per_sample;
 879 }
 880
 881 /* how many byes are free in the buffer */
 882 static int get_space(void)
 883 {
 884     snd_pcm_status_t *status;
 885     int ret;
 886
 887     snd_pcm_status_alloca(&status);
 888
 889     if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
 890     {
 891         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_CannotGetPcmStatus, snd_strerror(ret));
 892         return 0;
 893     }
 894
 895     unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
 896     if (space > ao_data.buffersize) // Buffer underrun?
 897         space = ao_data.buffersize;
 898     return space;
 899 }
 900
 901 /* delay in seconds between first and last sample in buffer */
 902 static float get_delay(void)
 903 {
 904   if (alsa_handler) {
 905     snd_pcm_sframes_t delay;
 906
 907     if (snd_pcm_delay(alsa_handler, &delay) < 0)
 908       return 0;
 909
 910     if (delay < 0) {
 911       /* underrun - move the application pointer forward to catch up */
 912 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
 913       snd_pcm_forward(alsa_handler, -delay);
 914 #endif
 915       delay = 0;
 916     }
 917     return (float)delay / (float)ao_data.samplerate;
 918   } else {
 919     return 0;
 920   }
 921 }