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 alsa_can_pause;
  81 static snd_pcm_sframes_t prepause_frames;
  82
  83 #define ALSA_DEVICE_SIZE 256
  84
  85 static void alsa_error_handler(const char *file, int line, const char *function,
  86                                int err, const char *format, ...)
  87 {
  88   char tmp[0xc00];
  89   va_list va;
  90
  91   va_start(va, format);
  92   vsnprintf(tmp, sizeof tmp, format, va);
  93   va_end(va);
  94   tmp[sizeof tmp - 1] = '\0';
  95
  96   if (err)
  97     mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
  98            file, line, function, tmp, snd_strerror(err));
  99   else
 100     mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
 101            file, line, function, tmp);
 102 }
 103
 104 /* to set/get/query special features/parameters */
 105 static int control(int cmd, void *arg)
 106 {
 107   switch(cmd) {
 108   case AOCONTROL_QUERY_FORMAT:
 109     return CONTROL_TRUE;
 110   case AOCONTROL_GET_VOLUME:
 111   case AOCONTROL_SET_VOLUME:
 112     {
 113       ao_control_vol_t *vol = (ao_control_vol_t *)arg;
 114
 115       int err;
 116       snd_mixer_t *handle;
 117       snd_mixer_elem_t *elem;
 118       snd_mixer_selem_id_t *sid;
 119
 120       char *mix_name = "PCM";
 121       char *card = "default";
 122       int mix_index = 0;
 123
 124       long pmin, pmax;
 125       long get_vol, set_vol;
 126       float f_multi;
 127
 128       if(AF_FORMAT_IS_AC3(ao_data.format))
 129         return CONTROL_TRUE;
 130
 131       if(mixer_channel) {
 132          char *test_mix_index;
 133
 134          mix_name = strdup(mixer_channel);
 135          if ((test_mix_index = strchr(mix_name, ','))){
 136                 *test_mix_index = 0;
 137                 test_mix_index++;
 138                 mix_index = strtol(test_mix_index, &test_mix_index, 0);
 139
 140                 if (*test_mix_index){
 141                   mp_tmsg(MSGT_AO,MSGL_ERR,
 142                     "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
 143                   mix_index = 0 ;
 144                 }
 145          }
 146       }
 147       if(mixer_device) card = mixer_device;
 148
 149       //allocate simple id
 150       snd_mixer_selem_id_alloca(&sid);
 151
 152       //sets simple-mixer index and name
 153       snd_mixer_selem_id_set_index(sid, mix_index);
 154       snd_mixer_selem_id_set_name(sid, mix_name);
 155
 156       if (mixer_channel) {
 157         free(mix_name);
 158         mix_name = NULL;
 159       }
 160
 161       if ((err = snd_mixer_open(&handle, 0)) < 0) {
 162         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer open error: %s\n", snd_strerror(err));
 163         return CONTROL_ERROR;
 164       }
 165
 166       if ((err = snd_mixer_attach(handle, card)) < 0) {
 167         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer attach %s error: %s\n",
 168                card, snd_strerror(err));
 169         snd_mixer_close(handle);
 170         return CONTROL_ERROR;
 171       }
 172
 173       if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
 174         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer register error: %s\n", snd_strerror(err));
 175         snd_mixer_close(handle);
 176         return CONTROL_ERROR;
 177       }
 178       err = snd_mixer_load(handle);
 179       if (err < 0) {
 180         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer load error: %s\n", snd_strerror(err));
 181         snd_mixer_close(handle);
 182         return CONTROL_ERROR;
 183       }
 184
 185       elem = snd_mixer_find_selem(handle, sid);
 186       if (!elem) {
 187         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to find simple control '%s',%i.\n",
 188                snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
 189         snd_mixer_close(handle);
 190         return CONTROL_ERROR;
 191         }
 192
 193       snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
 194       f_multi = (100 / (float)(pmax - pmin));
 195
 196       if (cmd == AOCONTROL_SET_VOLUME) {
 197
 198         set_vol = vol->left / f_multi + pmin + 0.5;
 199
 200         //setting channels
 201         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
 202           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n",
 203                  snd_strerror(err));
 204           snd_mixer_close(handle);
 205           return CONTROL_ERROR;
 206         }
 207         mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
 208
 209         set_vol = vol->right / f_multi + pmin + 0.5;
 210
 211         if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
 212           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
 213                  snd_strerror(err));
 214           snd_mixer_close(handle);
 215           return CONTROL_ERROR;
 216         }
 217         mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
 218                set_vol, pmin, pmax, f_multi);
 219
 220         if (snd_mixer_selem_has_playback_switch(elem)) {
 221           int lmute = (vol->left == 0.0);
 222           int rmute = (vol->right == 0.0);
 223           if (snd_mixer_selem_has_playback_switch_joined(elem)) {
 224             lmute = rmute = lmute && rmute;
 225           } else {
 226             snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
 227           }
 228           snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
 229         }
 230       }
 231       else {
 232         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
 233         vol->left = (get_vol - pmin) * f_multi;
 234         snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
 235         vol->right = (get_vol - pmin) * f_multi;
 236
 237         mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
 238       }
 239       snd_mixer_close(handle);
 240       return CONTROL_OK;
 241     }
 242
 243   } //end switch
 244   return CONTROL_UNKNOWN;
 245 }
 246
 247 static void parse_device (char *dest, const char *src, int len)
 248 {
 249   char *tmp;
 250   memmove(dest, src, len);
 251   dest[len] = 0;
 252   while ((tmp = strrchr(dest, '.')))
 253     tmp[0] = ',';
 254   while ((tmp = strrchr(dest, '=')))
 255     tmp[0] = ':';
 256 }
 257
 258 static void print_help (void)
 259 {
 260   mp_tmsg (MSGT_AO, MSGL_FATAL,
 261     "\n[AO_ALSA] -ao alsa commandline help:\n"\
 262     "[AO_ALSA] Example: mplayer -ao alsa:device=hw=0.3\n"\
 263     "[AO_ALSA]   Sets first card fourth hardware device.\n\n"\
 264     "[AO_ALSA] Options:\n"\
 265     "[AO_ALSA]   noblock\n"\
 266     "[AO_ALSA]     Opens device in non-blocking mode.\n"\
 267     "[AO_ALSA]   device=<device-name>\n"\
 268     "[AO_ALSA]     Sets device (change , to . and : to =)\n");
 269 }
 270
 271 static int str_maxlen(void *strp) {
 272   strarg_t *str = strp;
 273   return str->len <= ALSA_DEVICE_SIZE;
 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 chunk_size;
 331     snd_pcm_uframes_t bufsize;
 332     snd_pcm_uframes_t boundary;
 333     const opt_t subopts[] = {
 334       {"block", OPT_ARG_BOOL, &block, NULL},
 335       {"device", OPT_ARG_STR, &device, str_maxlen},
 336       {NULL}
 337     };
 338
 339     char alsa_device[ALSA_DEVICE_SIZE + 1];
 340     // make sure alsa_device is null-terminated even when using strncpy etc.
 341     memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
 342
 343     mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
 344         channels, format);
 345     alsa_handler = NULL;
 346 #if SND_LIB_VERSION >= 0x010005
 347     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
 348 #else
 349     mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
 350 #endif
 351
 352     prepause_frames = 0;
 353
 354     snd_lib_error_set_handler(alsa_error_handler);
 355
 356     ao_data.samplerate = rate_hz;
 357     ao_data.format = format;
 358     ao_data.channels = channels;
 359
 360     switch (format)
 361       {
 362       case AF_FORMAT_S8:
 363         alsa_format = SND_PCM_FORMAT_S8;
 364         break;
 365       case AF_FORMAT_U8:
 366         alsa_format = SND_PCM_FORMAT_U8;
 367         break;
 368       case AF_FORMAT_U16_LE:
 369         alsa_format = SND_PCM_FORMAT_U16_LE;
 370         break;
 371       case AF_FORMAT_U16_BE:
 372         alsa_format = SND_PCM_FORMAT_U16_BE;
 373         break;
 374       case AF_FORMAT_AC3_LE:
 375       case AF_FORMAT_S16_LE:
 376         alsa_format = SND_PCM_FORMAT_S16_LE;
 377         break;
 378       case AF_FORMAT_AC3_BE:
 379       case AF_FORMAT_S16_BE:
 380         alsa_format = SND_PCM_FORMAT_S16_BE;
 381         break;
 382       case AF_FORMAT_U32_LE:
 383         alsa_format = SND_PCM_FORMAT_U32_LE;
 384         break;
 385       case AF_FORMAT_U32_BE:
 386         alsa_format = SND_PCM_FORMAT_U32_BE;
 387         break;
 388       case AF_FORMAT_S32_LE:
 389         alsa_format = SND_PCM_FORMAT_S32_LE;
 390         break;
 391       case AF_FORMAT_S32_BE:
 392         alsa_format = SND_PCM_FORMAT_S32_BE;
 393         break;
 394       case AF_FORMAT_U24_LE:
 395         alsa_format = SND_PCM_FORMAT_U24_3LE;
 396         break;
 397       case AF_FORMAT_U24_BE:
 398         alsa_format = SND_PCM_FORMAT_U24_3BE;
 399         break;
 400       case AF_FORMAT_S24_LE:
 401         alsa_format = SND_PCM_FORMAT_S24_3LE;
 402         break;
 403       case AF_FORMAT_S24_BE:
 404         alsa_format = SND_PCM_FORMAT_S24_3BE;
 405         break;
 406       case AF_FORMAT_FLOAT_LE:
 407         alsa_format = SND_PCM_FORMAT_FLOAT_LE;
 408         break;
 409       case AF_FORMAT_FLOAT_BE:
 410         alsa_format = SND_PCM_FORMAT_FLOAT_BE;
 411         break;
 412       case AF_FORMAT_MU_LAW:
 413         alsa_format = SND_PCM_FORMAT_MU_LAW;
 414         break;
 415       case AF_FORMAT_A_LAW:
 416         alsa_format = SND_PCM_FORMAT_A_LAW;
 417         break;
 418
 419       default:
 420         alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
 421         break;
 422       }
 423
 424     //subdevice parsing
 425     // set defaults
 426     block = 1;
 427     /* switch for spdif
 428      * sets opening sequence for SPDIF
 429      * sets also the playback and other switches 'on the fly'
 430      * while opening the abstract alias for the spdif subdevice
 431      * 'iec958'
 432      */
 433     if (AF_FORMAT_IS_AC3(format)) {
 434         device.str = "iec958";
 435         mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
 436     }
 437   else
 438         /* in any case for multichannel playback we should select
 439          * appropriate device
 440          */
 441         switch (channels) {
 442         case 1:
 443         case 2:
 444           device.str = "default";
 445           mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
 446           break;
 447         case 4:
 448           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 449             // hack - use the converter plugin
 450             device.str = "plug:surround40";
 451           else
 452             device.str = "surround40";
 453           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
 454           break;
 455         case 6:
 456           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 457             device.str = "plug:surround51";
 458           else
 459             device.str = "surround51";
 460           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
 461           break;
 462         case 8:
 463           if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
 464             device.str = "plug:surround71";
 465           else
 466             device.str = "surround71";
 467           mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n");
 468           break;
 469         default:
 470           device.str = "default";
 471           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] %d channels are not supported.\n",channels);
 472         }
 473     device.len = strlen(device.str);
 474     if (subopt_parse(ao_subdevice, subopts) != 0) {
 475         print_help();
 476         return 0;
 477     }
 478     parse_device(alsa_device, device.str, device.len);
 479
 480     mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
 481
 482     if (!alsa_handler) {
 483       int open_mode = block ? 0 : SND_PCM_NONBLOCK;
 484       int isac3 =  AF_FORMAT_IS_AC3(format);
 485       //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
 486       if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
 487         {
 488           if (err != -EBUSY && !block) {
 489             mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
 490             if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
 491               mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 492               return 0;
 493             }
 494           } else {
 495             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 496             return 0;
 497           }
 498         }
 499
 500       if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
 501          mp_tmsg(MSGT_AO,MSGL_ERR,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err));
 502       } else {
 503         mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
 504       }
 505
 506       snd_pcm_hw_params_alloca(&alsa_hwparams);
 507       snd_pcm_sw_params_alloca(&alsa_swparams);
 508
 509       // setting hw-parameters
 510       if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
 511         {
 512           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get initial parameters: %s\n",
 513                  snd_strerror(err));
 514           return 0;
 515         }
 516
 517       err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
 518                                          SND_PCM_ACCESS_RW_INTERLEAVED);
 519       if (err < 0) {
 520         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set access type: %s\n",
 521                snd_strerror(err));
 522         return 0;
 523       }
 524
 525       /* workaround for nonsupported formats
 526          sets default format to S16_LE if the given formats aren't supported */
 527       if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
 528                                              alsa_format)) < 0)
 529       {
 530          mp_tmsg(MSGT_AO,MSGL_INFO,
 531                 "[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format));
 532          alsa_format = SND_PCM_FORMAT_S16_LE;
 533          if (AF_FORMAT_IS_AC3(ao_data.format))
 534            ao_data.format = AF_FORMAT_AC3_LE;
 535          else
 536          ao_data.format = AF_FORMAT_S16_LE;
 537       }
 538
 539       if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
 540                                               alsa_format)) < 0)
 541         {
 542           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set format: %s\n",
 543                  snd_strerror(err));
 544           return 0;
 545         }
 546
 547       if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
 548                                                      &ao_data.channels)) < 0)
 549         {
 550           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set channels: %s\n",
 551                  snd_strerror(err));
 552           return 0;
 553         }
 554
 555       /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
 556          prefer our own resampler */
 557 #if SND_LIB_VERSION >= 0x010009
 558       if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
 559                                                      0)) < 0)
 560         {
 561           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to disable resampling: %s\n",
 562                  snd_strerror(err));
 563           return 0;
 564         }
 565 #endif
 566
 567       if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
 568                                                  &ao_data.samplerate, NULL)) < 0)
 569         {
 570           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set samplerate-2: %s\n",
 571                  snd_strerror(err));
 572           return 0;
 573         }
 574
 575       bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
 576       bytes_per_sample *= ao_data.channels;
 577       ao_data.bps = ao_data.samplerate * bytes_per_sample;
 578
 579         if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
 580                                                           &(unsigned int){BUFFER_TIME}, NULL)) < 0)
 581           {
 582             mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set buffer time near: %s\n",
 583                    snd_strerror(err));
 584             return 0;
 585           }
 586
 587         if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
 588                                                       &(unsigned int){FRAGCOUNT}, NULL)) < 0) {
 589           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n",
 590                  snd_strerror(err));
 591           return 0;
 592         }
 593
 594       /* finally install hardware parameters */
 595       if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
 596         {
 597           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set hw-parameters: %s\n",
 598                  snd_strerror(err));
 599           return 0;
 600         }
 601       // end setting hw-params
 602
 603
 604       // gets buffersize for control
 605       if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
 606         {
 607           mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(err));
 608           return 0;
 609         }
 610       else {
 611         ao_data.buffersize = bufsize * bytes_per_sample;
 612           mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
 613       }
 614
 615       if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
 616         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to get period size: %s\n", snd_strerror(err));
 617         return 0;
 618       } else {
 619         mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
 620       }
 621       ao_data.outburst = chunk_size * bytes_per_sample;
 622
 623       /* setting software parameters */
 624       if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
 625         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
 626                snd_strerror(err));
 627         return 0;
 628       }
 629 #if SND_LIB_VERSION >= 0x000901
 630       if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
 631         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get boundary: %s\n",
 632                snd_strerror(err));
 633         return 0;
 634       }
 635 #else
 636       boundary = 0x7fffffff;
 637 #endif
 638       /* start playing when one period has been written */
 639       if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
 640         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set start threshold: %s\n",
 641                snd_strerror(err));
 642         return 0;
 643       }
 644       /* disable underrun reporting */
 645       if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
 646         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set stop threshold: %s\n",
 647                snd_strerror(err));
 648         return 0;
 649       }
 650 #if SND_LIB_VERSION >= 0x000901
 651       /* play silence when there is an underrun */
 652       if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
 653         mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set silence size: %s\n",
 654                snd_strerror(err));
 655         return 0;
 656       }
 657 #endif
 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 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
 855       snd_pcm_forward(alsa_handler, -delay);
 856 #endif
 857       delay = 0;
 858     }
 859     return (float)delay / (float)ao_data.samplerate;
 860   } else {
 861     return 0;
 862   }
 863 }