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
  88 #define ALSA_DEVICE_SIZE 256
  89
  90 #undef BUFFERTIME
  91 #define SET_CHUNKSIZE
  92
  93 static void alsa_error_handler(const char *file, int line, const char *function,
  94                                int err, const char *format, ...)
  95 {
  96   char tmp[0xc00];
  97   va_list va;
  98
  99   va_start(va, format);
 100   vsnprintf(tmp, sizeof tmp, format, va);
 101   va_end(va);
 102   tmp[sizeof tmp - 1] = '\0';
 103
 104   if (err)
 105     mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
 106            file, line, function, tmp, snd_strerror(err));
 107   else
 108     mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
 109            file, line, function, tmp);
 110 }
 111
 112 /* to set/get/query special features/parameters */
 113 static int control(int cmd, void *arg)
 114 {
 115   switch(cmd) {
 116   case AOCONTROL_QUERY_FORMAT:
 117     return CONTROL_TRUE;
 118   case AOCONTROL_GET_VOLUME:
 119   case AOCONTROL_SET_VOLUME:
 120     {
 121       ao_control_vol_t *vol = (ao_control_vol_t *)arg;
 122
 123       int err;
 124       snd_mixer_t *handle;
 125       snd_mixer_elem_t *elem;
 126       snd_mixer_selem_id_t *sid;
 127
 128       static char *mix_name = "PCM";
 129       static char *card = "default";
 130       static int mix_index = 0;
 131
 132       long pmin, pmax;
 133       long get_vol, set_vol;
 134       float f_multi;
 135
 136       if(ao_data.format == AF_FORMAT_AC3)
 137         return CONTROL_TRUE;
 138
 139       if(mixer_channel) {
 140          char *test_mix_index;
 141
 142          mix_name = strdup(mixer_channel);
 143          if ((test_mix_index = strchr(mix_name, ','))){
 144                 *test_mix_index = 0;
 145                 test_mix_index++;
 146                 mix_index = strtol(test_mix_index, &test_mix_index, 0);
 147
 148                 if (*test_mix_index){
 149                   mp_msg(MSGT_AO,MSGL_ERR,
 150                     MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero);
 151                   mix_index = 0 ;
 152                 }
 153          }
 154       }
 155       if(mixer_device) card = mixer_device;
 156
 157       //allocate simple id
 158       snd_mixer_selem_id_alloca(&sid);
 159
 160       //sets simple-mixer index and name
 161       snd_mixer_selem_id_set_index(sid, mix_index);
 162       snd_mixer_selem_id_set_name(sid, mix_name);
 163
 164       if (mixer_channel) {
 165         free(mix_name);
 166         mix_name = NULL;
 167       }
 168
 169       if ((err = snd_mixer_open(&handle, 0)) < 0) {
 170         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerOpenError, snd_strerror(err));
 171         return CONTROL_ERROR;
 172       }
 173
 174       if ((err = snd_mixer_attach(handle, card)) < 0) {
 175         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError,
 176                card, snd_strerror(err));
 177         snd_mixer_close(handle);
 178         return CONTROL_ERROR;
 179       }
 180
 181       if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
 182         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerRegisterError, snd_strerror(err));
 183         snd_mixer_close(handle);
 184         return CONTROL_ERROR;
 185       }
 186       err = snd_mixer_load(handle);
 187       if (err < 0) {
 188         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerLoadError, snd_strerror(err));
 189         snd_mixer_close(handle);
 190         return CONTROL_ERROR;
 191       }
 192
 193       elem = snd_mixer_find_selem(handle, sid);
 194       if (!elem) {
 195         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToFindSimpleControl,
 196                snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
 197         snd_mixer_close(handle);
 198         return CONTROL_ERROR;
 199         }
 200
 201       snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
 202       f_multi = (100 / (float)(pmax - pmin));
 203
 204       if (cmd == AOCONTROL_SET_VOLUME) {
 205
 206         set_vol = vol->left / f_multi + pmin + 0.5;
 207
 208         //setting channels
 209         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
 210           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingLeftChannel,
 211                  snd_strerror(err));
 212           return CONTROL_ERROR;
 213         }
 214         mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
 215
 216         set_vol = vol->right / f_multi + pmin + 0.5;
 217
 218         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
 219           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingRightChannel,
 220                  snd_strerror(err));
 221           return CONTROL_ERROR;
 222         }
 223         mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
 224                set_vol, pmin, pmax, f_multi);
 225
 226         if (snd_mixer_selem_has_playback_switch(elem)) {
 227           int lmute = (vol->left == 0.0);
 228           int rmute = (vol->right == 0.0);
 229           if (snd_mixer_selem_has_playback_switch_joined(elem)) {
 230             lmute = rmute = lmute && rmute;
 231           } else {
 232             snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
 233           }
 234           snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
 235         }
 236       }
 237       else {
 238         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
 239         vol->left = (get_vol - pmin) * f_multi;
 240         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
 241         vol->right = (get_vol - pmin) * f_multi;
 242
 243         mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
 244       }
 245       snd_mixer_close(handle);
 246       return CONTROL_OK;
 247     }
 248
 249   } //end switch
 250   return CONTROL_UNKNOWN;
 251 }
 252
 253 static void parse_device (char *dest, const char *src, int len)
 254 {
 255   char *tmp;
 256   memmove(dest, src, len);
 257   dest[len] = 0;
 258   while ((tmp = strrchr(dest, '.')))
 259     tmp[0] = ',';
 260   while ((tmp = strrchr(dest, '=')))
 261     tmp[0] = ':';
 262 }
 263
 264 static void print_help (void)
 265 {
 266   mp_msg (MSGT_AO, MSGL_FATAL,
 267            MSGTR_AO_ALSA_CommandlineHelp);
 268 }
 269
 270 static int str_maxlen(strarg_t *str) {
 271   if (str->len > ALSA_DEVICE_SIZE)
 272     return 0;
 273   return 1;
 274 }
 275
 276 static int try_open_device(const char *device, int open_mode, int try_ac3)
 277 {
 278   int err, len;
 279   char *ac3_device, *args;
 280
 281   if (try_ac3) {
 282     /* to set the non-audio bit, use AES0=6 */
 283     len = strlen(device);
 284     ac3_device = malloc(len + 7 + 1);
 285     if (!ac3_device)
 286       return -ENOMEM;
 287     strcpy(ac3_device, device);
 288     args = strchr(ac3_device, ':');
 289     if (!args) {
 290       /* no existing parameters: add it behind device name */
 291       strcat(ac3_device, ":AES0=6");
 292     } else {
 293       do
 294         ++args;
 295       while (isspace(*args));
 296       if (*args == '\0') {
 297         /* ":" but no parameters */
 298         strcat(ac3_device, "AES0=6");
 299       } else if (*args != '{') {
 300         /* a simple list of parameters: add it at the end of the list */
 301         strcat(ac3_device, ",AES0=6");
 302       } else {
 303         /* parameters in config syntax: add it inside the { } block */
 304         do
 305           --len;
 306         while (len > 0 && isspace(ac3_device[len]));
 307         if (ac3_device[len] == '}')
 308           strcpy(ac3_device + len, " AES0=6}");
 309       }
 310     }
 311     err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
 312                        open_mode);
 313     free(ac3_device);
 314   }
 315   if (!try_ac3 || err < 0)
 316     err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
 317                        open_mode);
 318   return err;
 319 }
 320
 321 /*
 322     open & setup audio device
 323     return: 1=success 0=fail
 324 */
 325 static int init(int rate_hz, int channels, int format, int flags)
 326 {
 327     int err;
 328     int block;
 329     strarg_t device;
 330     snd_pcm_uframes_t bufsize;
 331     snd_pcm_uframes_t boundary;
 332     opt_t subopts[] = {
 333       {"block", OPT_ARG_BOOL, &block, NULL},
 334       {"device", OPT_ARG_STR, &device, (opt_test_f)str_maxlen},
 335       {NULL}
 336     };
 337
 338     char alsa_device[ALSA_DEVICE_SIZE + 1];
 339     // make sure alsa_device is null-terminated even when using strncpy etc.
 340     memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
 341
 342     mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
 343         channels, format);
 344     alsa_handler = NULL;
 345 #if SND_LIB_VERSION >= 0x010005
 346     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
 347 #else
 348     mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
 349 #endif
 350
 351     snd_lib_error_set_handler(alsa_error_handler);
 352
 353     ao_data.samplerate = rate_hz;
 354     ao_data.format = format;
 355     ao_data.channels = channels;
 356
 357     switch (format)
 358       {
 359       case AF_FORMAT_S8:
 360         alsa_format = SND_PCM_FORMAT_S8;
 361         break;
 362       case AF_FORMAT_U8:
 363         alsa_format = SND_PCM_FORMAT_U8;
 364         break;
 365       case AF_FORMAT_U16_LE:
 366         alsa_format = SND_PCM_FORMAT_U16_LE;
 367         break;
 368       case AF_FORMAT_U16_BE:
 369         alsa_format = SND_PCM_FORMAT_U16_BE;
 370         break;
 371 #ifndef WORDS_BIGENDIAN
 372       case AF_FORMAT_AC3:
 373 #endif
 374       case AF_FORMAT_S16_LE:
 375         alsa_format = SND_PCM_FORMAT_S16_LE;
 376         break;
 377 #ifdef WORDS_BIGENDIAN
 378       case AF_FORMAT_AC3:
 379 #endif
 380       case AF_FORMAT_S16_BE:
 381         alsa_format = SND_PCM_FORMAT_S16_BE;
 382         break;
 383       case AF_FORMAT_U32_LE:
 384         alsa_format = SND_PCM_FORMAT_U32_LE;
 385         break;
 386       case AF_FORMAT_U32_BE:
 387         alsa_format = SND_PCM_FORMAT_U32_BE;
 388         break;
 389       case AF_FORMAT_S32_LE:
 390         alsa_format = SND_PCM_FORMAT_S32_LE;
 391         break;
 392       case AF_FORMAT_S32_BE:
 393         alsa_format = SND_PCM_FORMAT_S32_BE;
 394         break;
 395       case AF_FORMAT_FLOAT_LE:
 396         alsa_format = SND_PCM_FORMAT_FLOAT_LE;
 397         break;
 398       case AF_FORMAT_FLOAT_BE:
 399         alsa_format = SND_PCM_FORMAT_FLOAT_BE;
 400         break;
 401       case AF_FORMAT_MU_LAW:
 402         alsa_format = SND_PCM_FORMAT_MU_LAW;
 403         break;
 404       case AF_FORMAT_A_LAW:
 405         alsa_format = SND_PCM_FORMAT_A_LAW;
 406         break;
 407
 408       default:
 409         alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
 410         break;
 411       }
 412
 413     //subdevice parsing
 414     // set defaults
 415     block = 1;
 416     /* switch for spdif
 417      * sets opening sequence for SPDIF
 418      * sets also the playback and other switches 'on the fly'
 419      * while opening the abstract alias for the spdif subdevice
 420      * 'iec958'
 421      */
 422     if (format == AF_FORMAT_AC3) {
 423         device.str = "iec958";
 424         mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
 425     }
 426   else
 427         /* in any case for multichannel playback we should select
 428          * appropriate device
 429          */
 430         switch (channels) {
 431         case 1:
 432         case 2:
 433           device.str = "default";
 434           mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
 435           break;
 436         case 4:
 437           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 438             // hack - use the converter plugin
 439             device.str = "plug:surround40";
 440           else
 441             device.str = "surround40";
 442           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
 443           break;
 444         case 6:
 445           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 446             device.str = "plug:surround51";
 447           else
 448             device.str = "surround51";
 449           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
 450           break;
 451         default:
 452           device.str = "default";
 453           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels);
 454         }
 455     device.len = strlen(device.str);
 456     if (subopt_parse(ao_subdevice, subopts) != 0) {
 457         print_help();
 458         return 0;
 459     }
 460     ao_noblock = !block;
 461     parse_device(alsa_device, device.str, device.len);
 462
 463     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
 464
 465     //setting modes for block or nonblock-mode
 466     if (ao_noblock) {
 467       open_mode = SND_PCM_NONBLOCK;
 468     }
 469     else {
 470       open_mode = 0;
 471     }
 472
 473     //sets buff/chunksize if its set manually
 474     if (ao_data.buffersize) {
 475       switch (ao_data.buffersize)
 476         {
 477         case 1:
 478           alsa_fragcount = 16;
 479           chunk_size = 512;
 480             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
 481             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
 482           break;
 483         case 2:
 484           alsa_fragcount = 8;
 485           chunk_size = 1024;
 486             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
 487             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
 488           break;
 489         case 3:
 490           alsa_fragcount = 32;
 491           chunk_size = 512;
 492             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
 493             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
 494           break;
 495         case 4:
 496           alsa_fragcount = 16;
 497           chunk_size = 1024;
 498             mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
 499             mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
 500           break;
 501         default:
 502           alsa_fragcount = 16;
 503           chunk_size = 1024;
 504           break;
 505         }
 506     }
 507
 508     if (!alsa_handler) {
 509       //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
 510       if ((err = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0)
 511         {
 512           if (err != -EBUSY && ao_noblock) {
 513             mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
 514             if ((err = try_open_device(alsa_device, 0, format == AF_FORMAT_AC3)) < 0) {
 515               mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
 516               return 0;
 517             }
 518           } else {
 519             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
 520             return 0;
 521           }
 522         }
 523
 524       if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
 525          mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
 526       } else {
 527         mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
 528       }
 529
 530       snd_pcm_hw_params_alloca(&alsa_hwparams);
 531       snd_pcm_sw_params_alloca(&alsa_swparams);
 532
 533       // setting hw-parameters
 534       if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
 535         {
 536           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters,
 537                  snd_strerror(err));
 538           return 0;
 539         }
 540
 541       err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
 542                                          SND_PCM_ACCESS_RW_INTERLEAVED);
 543       if (err < 0) {
 544         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType,
 545                snd_strerror(err));
 546         return 0;
 547       }
 548
 549       /* workaround for nonsupported formats
 550          sets default format to S16_LE if the given formats aren't supported */
 551       if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
 552                                              alsa_format)) < 0)
 553       {
 554          mp_msg(MSGT_AO,MSGL_INFO,
 555                 MSGTR_AO_ALSA_FormatNotSupportedByHardware, af_fmt2str_short(format));
 556          alsa_format = SND_PCM_FORMAT_S16_LE;
 557          ao_data.format = AF_FORMAT_S16_LE;
 558       }
 559
 560       if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
 561                                               alsa_format)) < 0)
 562         {
 563           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat,
 564                  snd_strerror(err));
 565           return 0;
 566         }
 567
 568       if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
 569                                                      &ao_data.channels)) < 0)
 570         {
 571           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels,
 572                  snd_strerror(err));
 573           return 0;
 574         }
 575
 576       /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
 577          prefer our own resampler */
 578 #if SND_LIB_VERSION >= 0x010009
 579       if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
 580                                                      0)) < 0)
 581         {
 582           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling,
 583                  snd_strerror(err));
 584           return 0;
 585         }
 586 #endif
 587
 588       if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
 589                                                  &ao_data.samplerate, NULL)) < 0)
 590         {
 591           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2,
 592                  snd_strerror(err));
 593           return 0;
 594         }
 595
 596       bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8;
 597       bytes_per_sample *= ao_data.channels;
 598       ao_data.bps = ao_data.samplerate * bytes_per_sample;
 599
 600 #ifdef BUFFERTIME
 601       {
 602         int alsa_buffer_time = 500000; /* original 60 */
 603         int alsa_period_time;
 604         alsa_period_time = alsa_buffer_time/4;
 605         if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
 606                                                           &alsa_buffer_time, NULL)) < 0)
 607           {
 608             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
 609                    snd_strerror(err));
 610             return 0;
 611           } else
 612             alsa_buffer_time = err;
 613
 614         if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
 615                                                           &alsa_period_time, NULL)) < 0)
 616           /* original: alsa_buffer_time/ao_data.bps */
 617           {
 618             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodTime,
 619                    snd_strerror(err));
 620             return 0;
 621           }
 622         mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_BufferTimePeriodTime,
 623                alsa_buffer_time, err);
 624       }
 625 #endif//end SET_BUFFERTIME
 626
 627 #ifdef SET_CHUNKSIZE
 628       {
 629         //set chunksize
 630         if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
 631                                                           &chunk_size, NULL)) < 0)
 632           {
 633             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodSize,
 634                             chunk_size, snd_strerror(err));
 635             return 0;
 636           }
 637         else {
 638           mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %li\n", chunk_size);
 639         }
 640         if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
 641                                                       &alsa_fragcount, NULL)) < 0) {
 642           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods,
 643                  snd_strerror(err));
 644           return 0;
 645         }
 646         else {
 647           mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount);
 648         }
 649       }
 650 #endif//end SET_CHUNKSIZE
 651
 652       /* finally install hardware parameters */
 653       if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
 654         {
 655           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters,
 656                  snd_strerror(err));
 657           return 0;
 658         }
 659       // end setting hw-params
 660
 661
 662       // gets buffersize for control
 663       if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
 664         {
 665           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err));
 666           return 0;
 667         }
 668       else {
 669         ao_data.buffersize = bufsize * bytes_per_sample;
 670           mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
 671       }
 672
 673       if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
 674         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err));
 675         return 0;
 676       } else {
 677         mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
 678       }
 679       ao_data.outburst = chunk_size * bytes_per_sample;
 680
 681       /* setting software parameters */
 682       if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
 683         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
 684                snd_strerror(err));
 685         return 0;
 686       }
 687 #if SND_LIB_VERSION >= 0x000901
 688       if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
 689         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary,
 690                snd_strerror(err));
 691         return 0;
 692       }
 693 #else
 694       boundary = 0x7fffffff;
 695 #endif
 696       /* start playing when one period has been written */
 697       if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
 698         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold,
 699                snd_strerror(err));
 700         return 0;
 701       }
 702       /* disable underrun reporting */
 703       if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
 704         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold,
 705                snd_strerror(err));
 706         return 0;
 707       }
 708 #if SND_LIB_VERSION >= 0x000901
 709       /* play silence when there is an underrun */
 710       if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
 711         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize,
 712                snd_strerror(err));
 713         return 0;
 714       }
 715 #endif
 716       if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
 717         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
 718                snd_strerror(err));
 719         return 0;
 720       }
 721       /* end setting sw-params */
 722
 723       mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
 724              ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
 725              snd_pcm_format_description(alsa_format));
 726
 727     } // end switch alsa_handler (spdif)
 728     alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
 729     return 1;
 730 } // end init
 731
 732
 733 /* close audio device */
 734 static void uninit(int immed)
 735 {
 736
 737   if (alsa_handler) {
 738     int err;
 739
 740     if (!immed)
 741       snd_pcm_drain(alsa_handler);
 742
 743     if ((err = snd_pcm_close(alsa_handler)) < 0)
 744       {
 745         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmCloseError, snd_strerror(err));
 746         return;
 747       }
 748     else {
 749       alsa_handler = NULL;
 750       mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
 751     }
 752   }
 753   else {
 754     mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_NoHandlerDefined);
 755   }
 756 }
 757
 758 static void audio_pause(void)
 759 {
 760     int err;
 761
 762     if (alsa_can_pause) {
 763         if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
 764         {
 765             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPauseError, snd_strerror(err));
 766             return;
 767         }
 768           mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
 769     } else {
 770         if ((err = snd_pcm_drop(alsa_handler)) < 0)
 771         {
 772             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmDropError, snd_strerror(err));
 773             return;
 774         }
 775     }
 776 }
 777
 778 static void audio_resume(void)
 779 {
 780     int err;
 781
 782     if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
 783         mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
 784         while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
 785     }
 786     if (alsa_can_pause) {
 787         if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
 788         {
 789             mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmResumeError, snd_strerror(err));
 790             return;
 791         }
 792           mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
 793     } else {
 794         if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 795         {
 796            mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
 797             return;
 798         }
 799     }
 800 }
 801
 802 /* stop playing and empty buffers (for seeking/pause) */
 803 static void reset(void)
 804 {
 805     int err;
 806
 807     if ((err = snd_pcm_drop(alsa_handler)) < 0)
 808     {
 809         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
 810         return;
 811     }
 812     if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 813     {
 814         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
 815         return;
 816     }
 817     return;
 818 }
 819
 820 /*
 821     plays 'len' bytes of 'data'
 822     returns: number of bytes played
 823     modified last at 29.06.02 by jp
 824     thanxs for marius <marius@rospot.com> for giving us the light ;)
 825 */
 826
 827 static int play(void* data, int len, int flags)
 828 {
 829   int num_frames = len / bytes_per_sample;
 830   snd_pcm_sframes_t res = 0;
 831
 832   //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
 833
 834   if (!alsa_handler) {
 835     mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_DeviceConfigurationError);
 836     return 0;
 837   }
 838
 839   if (num_frames == 0)
 840     return 0;
 841
 842   do {
 843     res = snd_pcm_writei(alsa_handler, data, num_frames);
 844
 845       if (res == -EINTR) {
 846         /* nothing to do */
 847         res = 0;
 848       }
 849       else if (res == -ESTRPIPE) {      /* suspend */
 850         mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
 851         while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
 852           sleep(1);
 853       }
 854       if (res < 0) {
 855         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_WriteError, snd_strerror(res));
 856         mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_TryingToResetSoundcard);
 857         if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
 858           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(res));
 859           return 0;
 860           break;
 861         }
 862       }
 863   } while (res == 0);
 864
 865   return res < 0 ? res : res * bytes_per_sample;
 866 }
 867
 868 /* how many byes are free in the buffer */
 869 static int get_space(void)
 870 {
 871     snd_pcm_status_t *status;
 872     int ret;
 873
 874     snd_pcm_status_alloca(&status);
 875
 876     if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
 877     {
 878         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_CannotGetPcmStatus, snd_strerror(ret));
 879         return 0;
 880     }
 881
 882     ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
 883     if (ret > ao_data.buffersize)  // Buffer underrun?
 884         ret = ao_data.buffersize;
 885     return ret;
 886 }
 887
 888 /* delay in seconds between first and last sample in buffer */
 889 static float get_delay(void)
 890 {
 891   if (alsa_handler) {
 892     snd_pcm_sframes_t delay;
 893
 894     if (snd_pcm_delay(alsa_handler, &delay) < 0)
 895       return 0;
 896
 897     if (delay < 0) {
 898       /* underrun - move the application pointer forward to catch up */
 899 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
 900       snd_pcm_forward(alsa_handler, -delay);
 901 #endif
 902       delay = 0;
 903     }
 904     return (float)delay / (float)ao_data.samplerate;
 905   } else {
 906     return 0;
 907   }
 908 }