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_GET_MUTE:
 101   case AOCONTROL_SET_MUTE:
 102   case AOCONTROL_GET_VOLUME:
 103   case AOCONTROL_SET_VOLUME:
 104     {
 105       int err;
 106       snd_mixer_t *handle;
 107       snd_mixer_elem_t *elem;
 108       snd_mixer_selem_id_t *sid;
 109
 110       char *mix_name = "Master";
 111       char *card = "default";
 112       int mix_index = 0;
 113
 114       long pmin, pmax;
 115       long get_vol, set_vol;
 116       float f_multi;
 117
 118       if(AF_FORMAT_IS_AC3(ao_data.format))
 119         return CONTROL_TRUE;
 120
 121       if(mixer_channel) {
 122          char *test_mix_index;
 123
 124          mix_name = strdup(mixer_channel);
 125          if ((test_mix_index = strchr(mix_name, ','))){
 126                 *test_mix_index = 0;
 127                 test_mix_index++;
 128                 mix_index = strtol(test_mix_index, &test_mix_index, 0);
 129
 130                 if (*test_mix_index){
 131                   mp_tmsg(MSGT_AO,MSGL_ERR,
 132                     "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
 133                   mix_index = 0 ;
 134                 }
 135          }
 136       }
 137       if(mixer_device) card = mixer_device;
 138
 139       //allocate simple id
 140       snd_mixer_selem_id_alloca(&sid);
 141
 142       //sets simple-mixer index and name
 143       snd_mixer_selem_id_set_index(sid, mix_index);
 144       snd_mixer_selem_id_set_name(sid, mix_name);
 145
 146       if (mixer_channel) {
 147         free(mix_name);
 148         mix_name = NULL;
 149       }
 150
 151       if ((err = snd_mixer_open(&handle, 0)) < 0) {
 152         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer open error: %s\n", snd_strerror(err));
 153         return CONTROL_ERROR;
 154       }
 155
 156       if ((err = snd_mixer_attach(handle, card)) < 0) {
 157         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer attach %s error: %s\n",
 158                card, snd_strerror(err));
 159         snd_mixer_close(handle);
 160         return CONTROL_ERROR;
 161       }
 162
 163       if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
 164         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer register error: %s\n", snd_strerror(err));
 165         snd_mixer_close(handle);
 166         return CONTROL_ERROR;
 167       }
 168       err = snd_mixer_load(handle);
 169       if (err < 0) {
 170         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer load error: %s\n", snd_strerror(err));
 171         snd_mixer_close(handle);
 172         return CONTROL_ERROR;
 173       }
 174
 175       elem = snd_mixer_find_selem(handle, sid);
 176       if (!elem) {
 177         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to find simple control '%s',%i.\n",
 178                snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
 179         snd_mixer_close(handle);
 180         return CONTROL_ERROR;
 181         }
 182
 183       snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
 184       f_multi = (100 / (float)(pmax - pmin));
 185
 186       switch (cmd) {
 187       case AOCONTROL_SET_VOLUME: {
 188         ao_control_vol_t *vol = arg;
 189         set_vol = vol->left / f_multi + pmin + 0.5;
 190
 191         //setting channels
 192         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
 193           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n",
 194                  snd_strerror(err));
 195           goto mixer_error;
 196         }
 197         mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
 198
 199         set_vol = vol->right / f_multi + pmin + 0.5;
 200
 201         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
 202           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
 203                  snd_strerror(err));
 204           goto mixer_error;
 205         }
 206         mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
 207                set_vol, pmin, pmax, f_multi);
 208         break;
 209       }
 210       case AOCONTROL_GET_VOLUME: {
 211         ao_control_vol_t *vol = arg;
 212         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
 213         vol->left = (get_vol - pmin) * f_multi;
 214         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
 215         vol->right = (get_vol - pmin) * f_multi;
 216         mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
 217         break;
 218       }
 219       case AOCONTROL_SET_MUTE: {
 220         bool *mute = arg;
 221         if (!snd_mixer_selem_has_playback_switch(elem))
 222           goto mixer_error;
 223         if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
 224             snd_mixer_selem_set_playback_switch(
 225                 elem, SND_MIXER_SCHN_FRONT_RIGHT, !*mute);
 226         }
 227         snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
 228                                             !*mute);
 229         break;
 230       }
 231       case AOCONTROL_GET_MUTE: {
 232         bool *mute = arg;
 233         if (!snd_mixer_selem_has_playback_switch(elem))
 234           goto mixer_error;
 235         int tmp = 1;
 236         snd_mixer_selem_get_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
 237                                             &tmp);
 238         *mute = !tmp;
 239         if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
 240             snd_mixer_selem_get_playback_switch(
 241                 elem, SND_MIXER_SCHN_FRONT_RIGHT, &tmp);
 242             *mute &= !tmp;
 243         }
 244         break;
 245       }
 246       }
 247       snd_mixer_close(handle);
 248       return CONTROL_OK;
 249     mixer_error:
 250       snd_mixer_close(handle);
 251       return CONTROL_ERROR;
 252     }
 253
 254   } //end switch
 255   return CONTROL_UNKNOWN;
 256 }
 257
 258 static void parse_device (char *dest, const char *src, int len)
 259 {
 260   char *tmp;
 261   memmove(dest, src, len);
 262   dest[len] = 0;
 263   while ((tmp = strrchr(dest, '.')))
 264     tmp[0] = ',';
 265   while ((tmp = strrchr(dest, '=')))
 266     tmp[0] = ':';
 267 }
 268
 269 static void print_help (void)
 270 {
 271   mp_tmsg (MSGT_AO, MSGL_FATAL,
 272     "\n[AO_ALSA] -ao alsa commandline help:\n"\
 273     "[AO_ALSA] Example: mplayer -ao alsa:device=hw=0.3\n"\
 274     "[AO_ALSA]   Sets first card fourth hardware device.\n\n"\
 275     "[AO_ALSA] Options:\n"\
 276     "[AO_ALSA]   noblock\n"\
 277     "[AO_ALSA]     Opens device in non-blocking mode.\n"\
 278     "[AO_ALSA]   device=<device-name>\n"\
 279     "[AO_ALSA]     Sets device (change , to . and : to =)\n");
 280 }
 281
 282 static int str_maxlen(void *strp) {
 283   strarg_t *str = strp;
 284   return str->len <= ALSA_DEVICE_SIZE;
 285 }
 286
 287 static int try_open_device(const char *device, int open_mode, int try_ac3)
 288 {
 289   int err, len;
 290   char *ac3_device, *args;
 291
 292   if (try_ac3) {
 293     /* to set the non-audio bit, use AES0=6 */
 294     len = strlen(device);
 295     ac3_device = malloc(len + 7 + 1);
 296     if (!ac3_device)
 297       return -ENOMEM;
 298     strcpy(ac3_device, device);
 299     args = strchr(ac3_device, ':');
 300     if (!args) {
 301       /* no existing parameters: add it behind device name */
 302       strcat(ac3_device, ":AES0=6");
 303     } else {
 304       do
 305         ++args;
 306       while (isspace(*args));
 307       if (*args == '\0') {
 308         /* ":" but no parameters */
 309         strcat(ac3_device, "AES0=6");
 310       } else if (*args != '{') {
 311         /* a simple list of parameters: add it at the end of the list */
 312         strcat(ac3_device, ",AES0=6");
 313       } else {
 314         /* parameters in config syntax: add it inside the { } block */
 315         do
 316           --len;
 317         while (len > 0 && isspace(ac3_device[len]));
 318         if (ac3_device[len] == '}')
 319           strcpy(ac3_device + len, " AES0=6}");
 320       }
 321     }
 322     err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
 323                        open_mode);
 324     free(ac3_device);
 325     if (!err)
 326       return 0;
 327   }
 328   return snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
 329                       open_mode);
 330 }
 331
 332 /*
 333     open & setup audio device
 334     return: 1=success 0=fail
 335 */
 336 static int init(int rate_hz, int channels, int format, int flags)
 337 {
 338     int err;
 339     int block;
 340     strarg_t device;
 341     snd_pcm_uframes_t chunk_size;
 342     snd_pcm_uframes_t bufsize;
 343     snd_pcm_uframes_t boundary;
 344     const opt_t subopts[] = {
 345       {"block", OPT_ARG_BOOL, &block, NULL},
 346       {"device", OPT_ARG_STR, &device, str_maxlen},
 347       {NULL}
 348     };
 349
 350     char alsa_device[ALSA_DEVICE_SIZE + 1];
 351     // make sure alsa_device is null-terminated even when using strncpy etc.
 352     memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
 353
 354     mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
 355         channels, format);
 356     alsa_handler = NULL;
 357     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
 358
 359     prepause_frames = 0;
 360
 361     snd_lib_error_set_handler(alsa_error_handler);
 362
 363     ao_data.samplerate = rate_hz;
 364     ao_data.format = format;
 365     ao_data.channels = channels;
 366
 367     switch (format)
 368       {
 369       case AF_FORMAT_S8:
 370         alsa_format = SND_PCM_FORMAT_S8;
 371         break;
 372       case AF_FORMAT_U8:
 373         alsa_format = SND_PCM_FORMAT_U8;
 374         break;
 375       case AF_FORMAT_U16_LE:
 376         alsa_format = SND_PCM_FORMAT_U16_LE;
 377         break;
 378       case AF_FORMAT_U16_BE:
 379         alsa_format = SND_PCM_FORMAT_U16_BE;
 380         break;
 381       case AF_FORMAT_AC3_LE:
 382       case AF_FORMAT_S16_LE:
 383         alsa_format = SND_PCM_FORMAT_S16_LE;
 384         break;
 385       case AF_FORMAT_AC3_BE:
 386       case AF_FORMAT_S16_BE:
 387         alsa_format = SND_PCM_FORMAT_S16_BE;
 388         break;
 389       case AF_FORMAT_U32_LE:
 390         alsa_format = SND_PCM_FORMAT_U32_LE;
 391         break;
 392       case AF_FORMAT_U32_BE:
 393         alsa_format = SND_PCM_FORMAT_U32_BE;
 394         break;
 395       case AF_FORMAT_S32_LE:
 396         alsa_format = SND_PCM_FORMAT_S32_LE;
 397         break;
 398       case AF_FORMAT_S32_BE:
 399         alsa_format = SND_PCM_FORMAT_S32_BE;
 400         break;
 401       case AF_FORMAT_U24_LE:
 402         alsa_format = SND_PCM_FORMAT_U24_3LE;
 403         break;
 404       case AF_FORMAT_U24_BE:
 405         alsa_format = SND_PCM_FORMAT_U24_3BE;
 406         break;
 407       case AF_FORMAT_S24_LE:
 408         alsa_format = SND_PCM_FORMAT_S24_3LE;
 409         break;
 410       case AF_FORMAT_S24_BE:
 411         alsa_format = SND_PCM_FORMAT_S24_3BE;
 412         break;
 413       case AF_FORMAT_FLOAT_LE:
 414         alsa_format = SND_PCM_FORMAT_FLOAT_LE;
 415         break;
 416       case AF_FORMAT_FLOAT_BE:
 417         alsa_format = SND_PCM_FORMAT_FLOAT_BE;
 418         break;
 419       case AF_FORMAT_MU_LAW:
 420         alsa_format = SND_PCM_FORMAT_MU_LAW;
 421         break;
 422       case AF_FORMAT_A_LAW:
 423         alsa_format = SND_PCM_FORMAT_A_LAW;
 424         break;
 425
 426       default:
 427         alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
 428         break;
 429       }
 430
 431     //subdevice parsing
 432     // set defaults
 433     block = 1;
 434     /* switch for spdif
 435      * sets opening sequence for SPDIF
 436      * sets also the playback and other switches 'on the fly'
 437      * while opening the abstract alias for the spdif subdevice
 438      * 'iec958'
 439      */
 440     if (AF_FORMAT_IS_AC3(format)) {
 441         device.str = "iec958";
 442         mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
 443     }
 444   else
 445         /* in any case for multichannel playback we should select
 446          * appropriate device
 447          */
 448         switch (channels) {
 449         case 1:
 450         case 2:
 451           device.str = "default";
 452           mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
 453           break;
 454         case 4:
 455           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 456             // hack - use the converter plugin
 457             device.str = "plug:surround40";
 458           else
 459             device.str = "surround40";
 460           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
 461           break;
 462         case 6:
 463           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 464             device.str = "plug:surround51";
 465           else
 466             device.str = "surround51";
 467           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
 468           break;
 469         case 8:
 470           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 471             device.str = "plug:surround71";
 472           else
 473             device.str = "surround71";
 474           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n");
 475           break;
 476         default:
 477           device.str = "default";
 478           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] %d channels are not supported.\n",channels);
 479         }
 480     device.len = strlen(device.str);
 481     if (subopt_parse(ao_subdevice, subopts) != 0) {
 482         print_help();
 483         return 0;
 484     }
 485     parse_device(alsa_device, device.str, device.len);
 486
 487     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
 488
 489     if (!alsa_handler) {
 490       int open_mode = block ? 0 : SND_PCM_NONBLOCK;
 491       int isac3 =  AF_FORMAT_IS_AC3(format);
 492       //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
 493       if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
 494         {
 495           if (err != -EBUSY && !block) {
 496             mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
 497             if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
 498               mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 499               return 0;
 500             }
 501           } else {
 502             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 503             return 0;
 504           }
 505         }
 506
 507       if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
 508          mp_tmsg(MSGT_AO,MSGL_ERR,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err));
 509       } else {
 510         mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
 511       }
 512
 513       snd_pcm_hw_params_alloca(&alsa_hwparams);
 514       snd_pcm_sw_params_alloca(&alsa_swparams);
 515
 516       // setting hw-parameters
 517       if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
 518         {
 519           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get initial parameters: %s\n",
 520                  snd_strerror(err));
 521           return 0;
 522         }
 523
 524       err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
 525                                          SND_PCM_ACCESS_RW_INTERLEAVED);
 526       if (err < 0) {
 527         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set access type: %s\n",
 528                snd_strerror(err));
 529         return 0;
 530       }
 531
 532       /* workaround for nonsupported formats
 533          sets default format to S16_LE if the given formats aren't supported */
 534       if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
 535                                              alsa_format)) < 0)
 536       {
 537          mp_tmsg(MSGT_AO,MSGL_INFO,
 538                 "[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format));
 539          alsa_format = SND_PCM_FORMAT_S16_LE;
 540          if (AF_FORMAT_IS_AC3(ao_data.format))
 541            ao_data.format = AF_FORMAT_AC3_LE;
 542          else
 543          ao_data.format = AF_FORMAT_S16_LE;
 544       }
 545
 546       if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
 547                                               alsa_format)) < 0)
 548         {
 549           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set format: %s\n",
 550                  snd_strerror(err));
 551           return 0;
 552         }
 553
 554       if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
 555                                                      &ao_data.channels)) < 0)
 556         {
 557           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set channels: %s\n",
 558                  snd_strerror(err));
 559           return 0;
 560         }
 561
 562       /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
 563          prefer our own resampler, since that allows users to choose the resampler,
 564          even per file if desired */
 565       if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
 566                                                      0)) < 0)
 567         {
 568           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to disable resampling: %s\n",
 569                  snd_strerror(err));
 570           return 0;
 571         }
 572
 573       if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
 574                                                  &ao_data.samplerate, NULL)) < 0)
 575         {
 576           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set samplerate-2: %s\n",
 577                  snd_strerror(err));
 578           return 0;
 579         }
 580
 581       bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
 582       bytes_per_sample *= ao_data.channels;
 583       ao_data.bps = ao_data.samplerate * bytes_per_sample;
 584
 585         if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
 586                                                           &(unsigned int){BUFFER_TIME}, NULL)) < 0)
 587           {
 588             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set buffer time near: %s\n",
 589                    snd_strerror(err));
 590             return 0;
 591           }
 592
 593         if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
 594                                                       &(unsigned int){FRAGCOUNT}, NULL)) < 0) {
 595           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n",
 596                  snd_strerror(err));
 597           return 0;
 598         }
 599
 600       /* finally install hardware parameters */
 601       if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
 602         {
 603           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set hw-parameters: %s\n",
 604                  snd_strerror(err));
 605           return 0;
 606         }
 607       // end setting hw-params
 608
 609
 610       // gets buffersize for control
 611       if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
 612         {
 613           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(err));
 614           return 0;
 615         }
 616       else {
 617         ao_data.buffersize = bufsize * bytes_per_sample;
 618           mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
 619       }
 620
 621       if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
 622         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to get period size: %s\n", snd_strerror(err));
 623         return 0;
 624       } else {
 625         mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
 626       }
 627       ao_data.outburst = chunk_size * bytes_per_sample;
 628
 629       /* setting software parameters */
 630       if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
 631         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
 632                snd_strerror(err));
 633         return 0;
 634       }
 635       if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
 636         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get boundary: %s\n",
 637                snd_strerror(err));
 638         return 0;
 639       }
 640       /* start playing when one period has been written */
 641       if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
 642         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set start threshold: %s\n",
 643                snd_strerror(err));
 644         return 0;
 645       }
 646       /* disable underrun reporting */
 647       if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
 648         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set stop threshold: %s\n",
 649                snd_strerror(err));
 650         return 0;
 651       }
 652       /* play silence when there is an underrun */
 653       if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
 654         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set silence size: %s\n",
 655                snd_strerror(err));
 656         return 0;
 657       }
 658       if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
 659         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
 660                snd_strerror(err));
 661         return 0;
 662       }
 663       /* end setting sw-params */
 664
 665       mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
 666              ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
 667              snd_pcm_format_description(alsa_format));
 668
 669     } // end switch alsa_handler (spdif)
 670     alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
 671     return 1;
 672 } // end init
 673
 674
 675 /* close audio device */
 676 static void uninit(int immed)
 677 {
 678
 679   if (alsa_handler) {
 680     int err;
 681
 682     if (!immed)
 683       snd_pcm_drain(alsa_handler);
 684
 685     if ((err = snd_pcm_close(alsa_handler)) < 0)
 686       {
 687         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm close error: %s\n", snd_strerror(err));
 688         return;
 689       }
 690     else {
 691       alsa_handler = NULL;
 692       mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
 693     }
 694   }
 695   else {
 696     mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] No handler defined!\n");
 697   }
 698 }
 699
 700 static void audio_pause(void)
 701 {
 702     int err;
 703
 704     if (alsa_can_pause) {
 705         if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
 706         {
 707             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm pause error: %s\n", snd_strerror(err));
 708             return;
 709         }
 710           mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
 711     } else {
 712         if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
 713             || prepause_frames < 0)
 714             prepause_frames = 0;
 715
 716         if ((err = snd_pcm_drop(alsa_handler)) < 0)
 717         {
 718             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm drop error: %s\n", snd_strerror(err));
 719             return;
 720         }
 721     }
 722 }
 723
 724 static void audio_resume(void)
 725 {
 726     int err;
 727
 728     if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
 729         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
 730         while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
 731     }
 732     if (alsa_can_pause) {
 733         if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
 734         {
 735             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm resume error: %s\n", snd_strerror(err));
 736             return;
 737         }
 738           mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
 739     } else {
 740         if ((err = snd_pcm_prepare(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 (prepause_frames) {
 746             void *silence = calloc(prepause_frames, bytes_per_sample);
 747             play(silence, prepause_frames * bytes_per_sample, 0);
 748             free(silence);
 749         }
 750     }
 751 }
 752
 753 /* stop playing and empty buffers (for seeking/pause) */
 754 static void reset(void)
 755 {
 756     int err;
 757
 758     prepause_frames = 0;
 759     if ((err = snd_pcm_drop(alsa_handler)) < 0)
 760     {
 761         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 762         return;
 763     }
 764     if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 765     {
 766         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 767         return;
 768     }
 769     return;
 770 }
 771
 772 /*
 773     plays 'len' bytes of 'data'
 774     returns: number of bytes played
 775     modified last at 29.06.02 by jp
 776     thanxs for marius <marius@rospot.com> for giving us the light ;)
 777 */
 778
 779 static int play(void* data, int len, int flags)
 780 {
 781   int num_frames;
 782   snd_pcm_sframes_t res = 0;
 783   if (!(flags & AOPLAY_FINAL_CHUNK))
 784       len = len / ao_data.outburst * ao_data.outburst;
 785   num_frames = len / bytes_per_sample;
 786
 787   //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
 788
 789   if (!alsa_handler) {
 790     mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Device configuration error.");
 791     return 0;
 792   }
 793
 794   if (num_frames == 0)
 795     return 0;
 796
 797   do {
 798     res = snd_pcm_writei(alsa_handler, data, num_frames);
 799
 800       if (res == -EINTR) {
 801         /* nothing to do */
 802         res = 0;
 803       }
 804       else if (res == -ESTRPIPE) {      /* suspend */
 805         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
 806         while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
 807           sleep(1);
 808       }
 809       if (res < 0) {
 810         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Write error: %s\n", snd_strerror(res));
 811         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Trying to reset soundcard.\n");
 812         if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
 813           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(res));
 814           return 0;
 815           break;
 816         }
 817       }
 818   } while (res == 0);
 819
 820   return res < 0 ? res : res * bytes_per_sample;
 821 }
 822
 823 /* how many byes are free in the buffer */
 824 static int get_space(void)
 825 {
 826     snd_pcm_status_t *status;
 827     int ret;
 828
 829     snd_pcm_status_alloca(&status);
 830
 831     if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
 832     {
 833         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Cannot get pcm status: %s\n", snd_strerror(ret));
 834         return 0;
 835     }
 836
 837     unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
 838     if (space > ao_data.buffersize) // Buffer underrun?
 839         space = ao_data.buffersize;
 840     return space;
 841 }
 842
 843 /* delay in seconds between first and last sample in buffer */
 844 static float get_delay(void)
 845 {
 846   if (alsa_handler) {
 847     snd_pcm_sframes_t delay;
 848
 849     if (snd_pcm_delay(alsa_handler, &delay) < 0)
 850       return 0;
 851
 852     if (delay < 0) {
 853       /* underrun - move the application pointer forward to catch up */
 854       snd_pcm_forward(alsa_handler, -delay);
 855       delay = 0;
 856     }
 857     return (float)delay / (float)ao_data.samplerate;
 858   } else {
 859     return 0;
 860   }
 861 }