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 #define BUFFER_TIME 500000  // 0.5 s
  76 #define FRAGCOUNT 16
  77
  78 static size_t bytes_per_sample;
  79
  80 static int ao_noblock = 0;
  81
  82 static int open_mode;
  83 static int alsa_can_pause = 0;
  84 static snd_pcm_sframes_t prepause_frames;
  85
  86 #define ALSA_DEVICE_SIZE 256
  87
  88 static void alsa_error_handler(const char *file, int line, const char *function,
  89                                int err, const char *format, ...)
  90 {
  91   char tmp[0xc00];
  92   va_list va;
  93
  94   va_start(va, format);
  95   vsnprintf(tmp, sizeof tmp, format, va);
  96   va_end(va);
  97   tmp[sizeof tmp - 1] = '\0';
  98
  99   if (err)
 100     mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
 101            file, line, function, tmp, snd_strerror(err));
 102   else
 103     mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
 104            file, line, function, tmp);
 105 }
 106
 107 /* to set/get/query special features/parameters */
 108 static int control(int cmd, void *arg)
 109 {
 110   switch(cmd) {
 111   case AOCONTROL_QUERY_FORMAT:
 112     return CONTROL_TRUE;
 113   case AOCONTROL_GET_VOLUME:
 114   case AOCONTROL_SET_VOLUME:
 115     {
 116       ao_control_vol_t *vol = (ao_control_vol_t *)arg;
 117
 118       int err;
 119       snd_mixer_t *handle;
 120       snd_mixer_elem_t *elem;
 121       snd_mixer_selem_id_t *sid;
 122
 123       static char *mix_name = "PCM";
 124       static char *card = "default";
 125       static int mix_index = 0;
 126
 127       long pmin, pmax;
 128       long get_vol, set_vol;
 129       float f_multi;
 130
 131       if(ao_data.format == AF_FORMAT_AC3)
 132         return CONTROL_TRUE;
 133
 134       if(mixer_channel) {
 135          char *test_mix_index;
 136
 137          mix_name = strdup(mixer_channel);
 138          if ((test_mix_index = strchr(mix_name, ','))){
 139                 *test_mix_index = 0;
 140                 test_mix_index++;
 141                 mix_index = strtol(test_mix_index, &test_mix_index, 0);
 142
 143                 if (*test_mix_index){
 144                   mp_tmsg(MSGT_AO,MSGL_ERR,
 145                     "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
 146                   mix_index = 0 ;
 147                 }
 148          }
 149       }
 150       if(mixer_device) card = mixer_device;
 151
 152       //allocate simple id
 153       snd_mixer_selem_id_alloca(&sid);
 154
 155       //sets simple-mixer index and name
 156       snd_mixer_selem_id_set_index(sid, mix_index);
 157       snd_mixer_selem_id_set_name(sid, mix_name);
 158
 159       if (mixer_channel) {
 160         free(mix_name);
 161         mix_name = NULL;
 162       }
 163
 164       if ((err = snd_mixer_open(&handle, 0)) < 0) {
 165         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer open error: %s\n", snd_strerror(err));
 166         return CONTROL_ERROR;
 167       }
 168
 169       if ((err = snd_mixer_attach(handle, card)) < 0) {
 170         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer attach %s error: %s\n",
 171                card, snd_strerror(err));
 172         snd_mixer_close(handle);
 173         return CONTROL_ERROR;
 174       }
 175
 176       if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
 177         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer register error: %s\n", snd_strerror(err));
 178         snd_mixer_close(handle);
 179         return CONTROL_ERROR;
 180       }
 181       err = snd_mixer_load(handle);
 182       if (err < 0) {
 183         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer load error: %s\n", snd_strerror(err));
 184         snd_mixer_close(handle);
 185         return CONTROL_ERROR;
 186       }
 187
 188       elem = snd_mixer_find_selem(handle, sid);
 189       if (!elem) {
 190         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to find simple control '%s',%i.\n",
 191                snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
 192         snd_mixer_close(handle);
 193         return CONTROL_ERROR;
 194         }
 195
 196       snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
 197       f_multi = (100 / (float)(pmax - pmin));
 198
 199       if (cmd == AOCONTROL_SET_VOLUME) {
 200
 201         set_vol = vol->left / f_multi + pmin + 0.5;
 202
 203         //setting channels
 204         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
 205           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n",
 206                  snd_strerror(err));
 207           snd_mixer_close(handle);
 208           return CONTROL_ERROR;
 209         }
 210         mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
 211
 212         set_vol = vol->right / f_multi + pmin + 0.5;
 213
 214         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
 215           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
 216                  snd_strerror(err));
 217           snd_mixer_close(handle);
 218           return CONTROL_ERROR;
 219         }
 220         mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
 221                set_vol, pmin, pmax, f_multi);
 222
 223         if (snd_mixer_selem_has_playback_switch(elem)) {
 224           int lmute = (vol->left == 0.0);
 225           int rmute = (vol->right == 0.0);
 226           if (snd_mixer_selem_has_playback_switch_joined(elem)) {
 227             lmute = rmute = lmute && rmute;
 228           } else {
 229             snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
 230           }
 231           snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
 232         }
 233       }
 234       else {
 235         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
 236         vol->left = (get_vol - pmin) * f_multi;
 237         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
 238         vol->right = (get_vol - pmin) * f_multi;
 239
 240         mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
 241       }
 242       snd_mixer_close(handle);
 243       return CONTROL_OK;
 244     }
 245
 246   } //end switch
 247   return CONTROL_UNKNOWN;
 248 }
 249
 250 static void parse_device (char *dest, const char *src, int len)
 251 {
 252   char *tmp;
 253   memmove(dest, src, len);
 254   dest[len] = 0;
 255   while ((tmp = strrchr(dest, '.')))
 256     tmp[0] = ',';
 257   while ((tmp = strrchr(dest, '=')))
 258     tmp[0] = ':';
 259 }
 260
 261 static void print_help (void)
 262 {
 263   mp_tmsg (MSGT_AO, MSGL_FATAL,
 264     "\n[AO_ALSA] -ao alsa commandline help:\n"\
 265     "[AO_ALSA] Example: mplayer -ao alsa:device=hw=0.3\n"\
 266     "[AO_ALSA]   Sets first card fourth hardware device.\n\n"\
 267     "[AO_ALSA] Options:\n"\
 268     "[AO_ALSA]   noblock\n"\
 269     "[AO_ALSA]     Opens device in non-blocking mode.\n"\
 270     "[AO_ALSA]   device=<device-name>\n"\
 271     "[AO_ALSA]     Sets device (change , to . and : to =)\n");
 272 }
 273
 274 static int str_maxlen(strarg_t *str) {
 275   if (str->len > ALSA_DEVICE_SIZE)
 276     return 0;
 277   return 1;
 278 }
 279
 280 static int try_open_device(const char *device, int open_mode, int try_ac3)
 281 {
 282   int err, len;
 283   char *ac3_device, *args;
 284
 285   if (try_ac3) {
 286     /* to set the non-audio bit, use AES0=6 */
 287     len = strlen(device);
 288     ac3_device = malloc(len + 7 + 1);
 289     if (!ac3_device)
 290       return -ENOMEM;
 291     strcpy(ac3_device, device);
 292     args = strchr(ac3_device, ':');
 293     if (!args) {
 294       /* no existing parameters: add it behind device name */
 295       strcat(ac3_device, ":AES0=6");
 296     } else {
 297       do
 298         ++args;
 299       while (isspace(*args));
 300       if (*args == '\0') {
 301         /* ":" but no parameters */
 302         strcat(ac3_device, "AES0=6");
 303       } else if (*args != '{') {
 304         /* a simple list of parameters: add it at the end of the list */
 305         strcat(ac3_device, ",AES0=6");
 306       } else {
 307         /* parameters in config syntax: add it inside the { } block */
 308         do
 309           --len;
 310         while (len > 0 && isspace(ac3_device[len]));
 311         if (ac3_device[len] == '}')
 312           strcpy(ac3_device + len, " AES0=6}");
 313       }
 314     }
 315     err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
 316                        open_mode);
 317     free(ac3_device);
 318   }
 319   if (!try_ac3 || err < 0)
 320     err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
 321                        open_mode);
 322   return err;
 323 }
 324
 325 /*
 326     open & setup audio device
 327     return: 1=success 0=fail
 328 */
 329 static int init(int rate_hz, int channels, int format, int flags)
 330 {
 331     int err;
 332     int block;
 333     strarg_t device;
 334     snd_pcm_uframes_t chunk_size;
 335     snd_pcm_uframes_t bufsize;
 336     snd_pcm_uframes_t boundary;
 337     const opt_t subopts[] = {
 338       {"block", OPT_ARG_BOOL, &block, NULL},
 339       {"device", OPT_ARG_STR, &device, (opt_test_f)str_maxlen},
 340       {NULL}
 341     };
 342
 343     char alsa_device[ALSA_DEVICE_SIZE + 1];
 344     // make sure alsa_device is null-terminated even when using strncpy etc.
 345     memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
 346
 347     mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
 348         channels, format);
 349     alsa_handler = NULL;
 350 #if SND_LIB_VERSION >= 0x010005
 351     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
 352 #else
 353     mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
 354 #endif
 355
 356     prepause_frames = 0;
 357
 358     snd_lib_error_set_handler(alsa_error_handler);
 359
 360     ao_data.samplerate = rate_hz;
 361     ao_data.format = format;
 362     ao_data.channels = channels;
 363
 364     switch (format)
 365       {
 366       case AF_FORMAT_S8:
 367         alsa_format = SND_PCM_FORMAT_S8;
 368         break;
 369       case AF_FORMAT_U8:
 370         alsa_format = SND_PCM_FORMAT_U8;
 371         break;
 372       case AF_FORMAT_U16_LE:
 373         alsa_format = SND_PCM_FORMAT_U16_LE;
 374         break;
 375       case AF_FORMAT_U16_BE:
 376         alsa_format = SND_PCM_FORMAT_U16_BE;
 377         break;
 378 #if !HAVE_BIGENDIAN
 379       case AF_FORMAT_AC3:
 380 #endif
 381       case AF_FORMAT_S16_LE:
 382         alsa_format = SND_PCM_FORMAT_S16_LE;
 383         break;
 384 #if HAVE_BIGENDIAN
 385       case AF_FORMAT_AC3:
 386 #endif
 387       case AF_FORMAT_S16_BE:
 388         alsa_format = SND_PCM_FORMAT_S16_BE;
 389         break;
 390       case AF_FORMAT_U32_LE:
 391         alsa_format = SND_PCM_FORMAT_U32_LE;
 392         break;
 393       case AF_FORMAT_U32_BE:
 394         alsa_format = SND_PCM_FORMAT_U32_BE;
 395         break;
 396       case AF_FORMAT_S32_LE:
 397         alsa_format = SND_PCM_FORMAT_S32_LE;
 398         break;
 399       case AF_FORMAT_S32_BE:
 400         alsa_format = SND_PCM_FORMAT_S32_BE;
 401         break;
 402       case AF_FORMAT_U24_LE:
 403         alsa_format = SND_PCM_FORMAT_U24_3LE;
 404         break;
 405       case AF_FORMAT_U24_BE:
 406         alsa_format = SND_PCM_FORMAT_U24_3BE;
 407         break;
 408       case AF_FORMAT_S24_LE:
 409         alsa_format = SND_PCM_FORMAT_S24_3LE;
 410         break;
 411       case AF_FORMAT_S24_BE:
 412         alsa_format = SND_PCM_FORMAT_S24_3BE;
 413         break;
 414       case AF_FORMAT_FLOAT_LE:
 415         alsa_format = SND_PCM_FORMAT_FLOAT_LE;
 416         break;
 417       case AF_FORMAT_FLOAT_BE:
 418         alsa_format = SND_PCM_FORMAT_FLOAT_BE;
 419         break;
 420       case AF_FORMAT_MU_LAW:
 421         alsa_format = SND_PCM_FORMAT_MU_LAW;
 422         break;
 423       case AF_FORMAT_A_LAW:
 424         alsa_format = SND_PCM_FORMAT_A_LAW;
 425         break;
 426
 427       default:
 428         alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
 429         break;
 430       }
 431
 432     //subdevice parsing
 433     // set defaults
 434     block = 1;
 435     /* switch for spdif
 436      * sets opening sequence for SPDIF
 437      * sets also the playback and other switches 'on the fly'
 438      * while opening the abstract alias for the spdif subdevice
 439      * 'iec958'
 440      */
 441     if (format == AF_FORMAT_AC3) {
 442         device.str = "iec958";
 443         mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
 444     }
 445   else
 446         /* in any case for multichannel playback we should select
 447          * appropriate device
 448          */
 449         switch (channels) {
 450         case 1:
 451         case 2:
 452           device.str = "default";
 453           mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
 454           break;
 455         case 4:
 456           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 457             // hack - use the converter plugin
 458             device.str = "plug:surround40";
 459           else
 460             device.str = "surround40";
 461           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
 462           break;
 463         case 6:
 464           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 465             device.str = "plug:surround51";
 466           else
 467             device.str = "surround51";
 468           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
 469           break;
 470         case 8:
 471           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 472             device.str = "plug:surround71";
 473           else
 474             device.str = "surround71";
 475           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n");
 476           break;
 477         default:
 478           device.str = "default";
 479           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] %d channels are not supported.\n",channels);
 480         }
 481     device.len = strlen(device.str);
 482     if (subopt_parse(ao_subdevice, subopts) != 0) {
 483         print_help();
 484         return 0;
 485     }
 486     ao_noblock = !block;
 487     parse_device(alsa_device, device.str, device.len);
 488
 489     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
 490
 491     //setting modes for block or nonblock-mode
 492     if (ao_noblock) {
 493       open_mode = SND_PCM_NONBLOCK;
 494     }
 495     else {
 496       open_mode = 0;
 497     }
 498
 499     if (!alsa_handler) {
 500       //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
 501       if ((err = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0)
 502         {
 503           if (err != -EBUSY && ao_noblock) {
 504             mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
 505             if ((err = try_open_device(alsa_device, 0, format == AF_FORMAT_AC3)) < 0) {
 506               mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 507               return 0;
 508             }
 509           } else {
 510             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 511             return 0;
 512           }
 513         }
 514
 515       if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
 516          mp_tmsg(MSGT_AO,MSGL_ERR,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err));
 517       } else {
 518         mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
 519       }
 520
 521       snd_pcm_hw_params_alloca(&alsa_hwparams);
 522       snd_pcm_sw_params_alloca(&alsa_swparams);
 523
 524       // setting hw-parameters
 525       if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
 526         {
 527           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get initial parameters: %s\n",
 528                  snd_strerror(err));
 529           return 0;
 530         }
 531
 532       err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
 533                                          SND_PCM_ACCESS_RW_INTERLEAVED);
 534       if (err < 0) {
 535         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set access type: %s\n",
 536                snd_strerror(err));
 537         return 0;
 538       }
 539
 540       /* workaround for nonsupported formats
 541          sets default format to S16_LE if the given formats aren't supported */
 542       if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
 543                                              alsa_format)) < 0)
 544       {
 545          mp_tmsg(MSGT_AO,MSGL_INFO,
 546                 "[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format));
 547          alsa_format = SND_PCM_FORMAT_S16_LE;
 548          ao_data.format = AF_FORMAT_S16_LE;
 549       }
 550
 551       if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
 552                                               alsa_format)) < 0)
 553         {
 554           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set format: %s\n",
 555                  snd_strerror(err));
 556           return 0;
 557         }
 558
 559       if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
 560                                                      &ao_data.channels)) < 0)
 561         {
 562           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set channels: %s\n",
 563                  snd_strerror(err));
 564           return 0;
 565         }
 566
 567       /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
 568          prefer our own resampler */
 569 #if SND_LIB_VERSION >= 0x010009
 570       if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
 571                                                      0)) < 0)
 572         {
 573           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to disable resampling: %s\n",
 574                  snd_strerror(err));
 575           return 0;
 576         }
 577 #endif
 578
 579       if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
 580                                                  &ao_data.samplerate, NULL)) < 0)
 581         {
 582           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set samplerate-2: %s\n",
 583                  snd_strerror(err));
 584           return 0;
 585         }
 586
 587       bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8;
 588       bytes_per_sample *= ao_data.channels;
 589       ao_data.bps = ao_data.samplerate * bytes_per_sample;
 590
 591         if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
 592                                                           &(unsigned int){BUFFER_TIME}, NULL)) < 0)
 593           {
 594             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set buffer time near: %s\n",
 595                    snd_strerror(err));
 596             return 0;
 597           }
 598
 599         if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
 600                                                       &(unsigned int){FRAGCOUNT}, NULL)) < 0) {
 601           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n",
 602                  snd_strerror(err));
 603           return 0;
 604         }
 605
 606       /* finally install hardware parameters */
 607       if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
 608         {
 609           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set hw-parameters: %s\n",
 610                  snd_strerror(err));
 611           return 0;
 612         }
 613       // end setting hw-params
 614
 615
 616       // gets buffersize for control
 617       if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
 618         {
 619           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(err));
 620           return 0;
 621         }
 622       else {
 623         ao_data.buffersize = bufsize * bytes_per_sample;
 624           mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
 625       }
 626
 627       if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
 628         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to get period size: %s\n", snd_strerror(err));
 629         return 0;
 630       } else {
 631         mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
 632       }
 633       ao_data.outburst = chunk_size * bytes_per_sample;
 634
 635       /* setting software parameters */
 636       if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
 637         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
 638                snd_strerror(err));
 639         return 0;
 640       }
 641 #if SND_LIB_VERSION >= 0x000901
 642       if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
 643         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get boundary: %s\n",
 644                snd_strerror(err));
 645         return 0;
 646       }
 647 #else
 648       boundary = 0x7fffffff;
 649 #endif
 650       /* start playing when one period has been written */
 651       if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
 652         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set start threshold: %s\n",
 653                snd_strerror(err));
 654         return 0;
 655       }
 656       /* disable underrun reporting */
 657       if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
 658         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set stop threshold: %s\n",
 659                snd_strerror(err));
 660         return 0;
 661       }
 662 #if SND_LIB_VERSION >= 0x000901
 663       /* play silence when there is an underrun */
 664       if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
 665         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set silence size: %s\n",
 666                snd_strerror(err));
 667         return 0;
 668       }
 669 #endif
 670       if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
 671         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
 672                snd_strerror(err));
 673         return 0;
 674       }
 675       /* end setting sw-params */
 676
 677       mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
 678              ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
 679              snd_pcm_format_description(alsa_format));
 680
 681     } // end switch alsa_handler (spdif)
 682     alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
 683     return 1;
 684 } // end init
 685
 686
 687 /* close audio device */
 688 static void uninit(int immed)
 689 {
 690
 691   if (alsa_handler) {
 692     int err;
 693
 694     if (!immed)
 695       snd_pcm_drain(alsa_handler);
 696
 697     if ((err = snd_pcm_close(alsa_handler)) < 0)
 698       {
 699         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm close error: %s\n", snd_strerror(err));
 700         return;
 701       }
 702     else {
 703       alsa_handler = NULL;
 704       mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
 705     }
 706   }
 707   else {
 708     mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] No handler defined!\n");
 709   }
 710 }
 711
 712 static void audio_pause(void)
 713 {
 714     int err;
 715
 716     if (alsa_can_pause) {
 717         if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
 718         {
 719             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm pause error: %s\n", snd_strerror(err));
 720             return;
 721         }
 722           mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
 723     } else {
 724         if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
 725             || prepause_frames < 0)
 726             prepause_frames = 0;
 727
 728         if ((err = snd_pcm_drop(alsa_handler)) < 0)
 729         {
 730             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm drop error: %s\n", snd_strerror(err));
 731             return;
 732         }
 733     }
 734 }
 735
 736 static void audio_resume(void)
 737 {
 738     int err;
 739
 740     if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
 741         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
 742         while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
 743     }
 744     if (alsa_can_pause) {
 745         if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
 746         {
 747             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm resume error: %s\n", snd_strerror(err));
 748             return;
 749         }
 750           mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
 751     } else {
 752         if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 753         {
 754            mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 755             return;
 756         }
 757         if (prepause_frames) {
 758             void *silence = calloc(prepause_frames, bytes_per_sample);
 759             play(silence, prepause_frames * bytes_per_sample, 0);
 760             free(silence);
 761         }
 762     }
 763 }
 764
 765 /* stop playing and empty buffers (for seeking/pause) */
 766 static void reset(void)
 767 {
 768     int err;
 769
 770     prepause_frames = 0;
 771     if ((err = snd_pcm_drop(alsa_handler)) < 0)
 772     {
 773         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 774         return;
 775     }
 776     if ((err = snd_pcm_prepare(alsa_handler)) < 0)
 777     {
 778         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
 779         return;
 780     }
 781     return;
 782 }
 783
 784 /*
 785     plays 'len' bytes of 'data'
 786     returns: number of bytes played
 787     modified last at 29.06.02 by jp
 788     thanxs for marius <marius@rospot.com> for giving us the light ;)
 789 */
 790
 791 static int play(void* data, int len, int flags)
 792 {
 793   int num_frames;
 794   snd_pcm_sframes_t res = 0;
 795   if (!(flags & AOPLAY_FINAL_CHUNK))
 796       len = len / ao_data.outburst * ao_data.outburst;
 797   num_frames = len / bytes_per_sample;
 798
 799   //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
 800
 801   if (!alsa_handler) {
 802     mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Device configuration error.");
 803     return 0;
 804   }
 805
 806   if (num_frames == 0)
 807     return 0;
 808
 809   do {
 810     res = snd_pcm_writei(alsa_handler, data, num_frames);
 811
 812       if (res == -EINTR) {
 813         /* nothing to do */
 814         res = 0;
 815       }
 816       else if (res == -ESTRPIPE) {      /* suspend */
 817         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
 818         while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
 819           sleep(1);
 820       }
 821       if (res < 0) {
 822         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Write error: %s\n", snd_strerror(res));
 823         mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Trying to reset soundcard.\n");
 824         if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
 825           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(res));
 826           return 0;
 827           break;
 828         }
 829       }
 830   } while (res == 0);
 831
 832   return res < 0 ? res : res * bytes_per_sample;
 833 }
 834
 835 /* how many byes are free in the buffer */
 836 static int get_space(void)
 837 {
 838     snd_pcm_status_t *status;
 839     int ret;
 840
 841     snd_pcm_status_alloca(&status);
 842
 843     if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
 844     {
 845         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Cannot get pcm status: %s\n", snd_strerror(ret));
 846         return 0;
 847     }
 848
 849     unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
 850     if (space > ao_data.buffersize) // Buffer underrun?
 851         space = ao_data.buffersize;
 852     return space;
 853 }
 854
 855 /* delay in seconds between first and last sample in buffer */
 856 static float get_delay(void)
 857 {
 858   if (alsa_handler) {
 859     snd_pcm_sframes_t delay;
 860
 861     if (snd_pcm_delay(alsa_handler, &delay) < 0)
 862       return 0;
 863
 864     if (delay < 0) {
 865       /* underrun - move the application pointer forward to catch up */
 866 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
 867       snd_pcm_forward(alsa_handler, -delay);
 868 #endif
 869       delay = 0;
 870     }
 871     return (float)delay / (float)ao_data.samplerate;
 872   } else {
 873     return 0;
 874   }
 875 }