lib/service/servicemp3.cpp

   1 #ifdef HAVE_GSTREAMER
   2
   3         /* note: this requires gstreamer 0.10.x and a big list of plugins. */
   4         /* it's currently hardcoded to use a big-endian alsasink as sink. */
   5 #include <lib/base/eerror.h>
   6 #include <lib/base/object.h>
   7 #include <lib/base/ebase.h>
   8 #include <string>
   9 #include <lib/service/servicemp3.h>
  10 #include <lib/service/service.h>
  11 #include <lib/base/init_num.h>
  12 #include <lib/base/init.h>
  13 #include <gst/gst.h>
  14
  15 // eServiceFactoryMP3
  16
  17 eServiceFactoryMP3::eServiceFactoryMP3()
  18 {
  19         ePtr<eServiceCenter> sc;
  20
  21         eServiceCenter::getPrivInstance(sc);
  22         if (sc)
  23         {
  24                 std::list<std::string> extensions;
  25                 extensions.push_back("mp3");
  26                 extensions.push_back("ogg");
  27                 extensions.push_back("mpg");
  28                 extensions.push_back("vob");
  29                 extensions.push_back("wav");
  30                 extensions.push_back("wave");
  31                 extensions.push_back("mkv");
  32                 extensions.push_back("avi");
  33                 sc->addServiceFactory(eServiceFactoryMP3::id, this, extensions);
  34         }
  35
  36         m_service_info = new eStaticServiceMP3Info();
  37 }
  38
  39 eServiceFactoryMP3::~eServiceFactoryMP3()
  40 {
  41         ePtr<eServiceCenter> sc;
  42
  43         eServiceCenter::getPrivInstance(sc);
  44         if (sc)
  45                 sc->removeServiceFactory(eServiceFactoryMP3::id);
  46 }
  47
  48 DEFINE_REF(eServiceFactoryMP3)
  49
  50         // iServiceHandler
  51 RESULT eServiceFactoryMP3::play(const eServiceReference &ref, ePtr<iPlayableService> &ptr)
  52 {
  53                 // check resources...
  54         ptr = new eServiceMP3(ref.path.c_str());
  55         return 0;
  56 }
  57
  58 RESULT eServiceFactoryMP3::record(const eServiceReference &ref, ePtr<iRecordableService> &ptr)
  59 {
  60         ptr=0;
  61         return -1;
  62 }
  63
  64 RESULT eServiceFactoryMP3::list(const eServiceReference &, ePtr<iListableService> &ptr)
  65 {
  66         ptr=0;
  67         return -1;
  68 }
  69
  70 RESULT eServiceFactoryMP3::info(const eServiceReference &ref, ePtr<iStaticServiceInformation> &ptr)
  71 {
  72         ptr = m_service_info;
  73         return 0;
  74 }
  75
  76 RESULT eServiceFactoryMP3::offlineOperations(const eServiceReference &, ePtr<iServiceOfflineOperations> &ptr)
  77 {
  78         ptr = 0;
  79         return -1;
  80 }
  81
  82
  83 // eStaticServiceMP3Info
  84
  85
  86 // eStaticServiceMP3Info is seperated from eServiceMP3 to give information
  87 // about unopened files.
  88
  89 // probably eServiceMP3 should use this class as well, and eStaticServiceMP3Info
  90 // should have a database backend where ID3-files etc. are cached.
  91 // this would allow listing the mp3 database based on certain filters.
  92
  93 DEFINE_REF(eStaticServiceMP3Info)
  94
  95 eStaticServiceMP3Info::eStaticServiceMP3Info()
  96 {
  97 }
  98
  99 RESULT eStaticServiceMP3Info::getName(const eServiceReference &ref, std::string &name)
 100 {
 101         size_t last = ref.path.rfind('/');
 102         if (last != std::string::npos)
 103                 name = ref.path.substr(last+1);
 104         else
 105                 name = ref.path;
 106         return 0;
 107 }
 108
 109 int eStaticServiceMP3Info::getLength(const eServiceReference &ref)
 110 {
 111         return -1;
 112 }
 113
 114 // eServiceMP3
 115
 116 eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1)
 117 {
 118         m_stream_tags = 0;
 119         m_currentAudioStream = 0;
 120         CONNECT(m_pump.recv_msg, eServiceMP3::gstPoll);
 121         GstElement *source = 0;
 122
 123         GstElement *filter = 0, *decoder = 0, *conv = 0, *flt = 0, *sink = 0; /* for audio */
 124
 125         GstElement *audio = 0, *switch_audio = 0, *queue_audio = 0, *video = 0, *queue_video = 0, *videodemux = 0;
 126
 127         m_state = stIdle;
 128         eDebug("SERVICEMP3 construct!");
 129
 130                 /* FIXME: currently, decodebin isn't possible for
 131                    video streams. in that case, make a manual pipeline. */
 132
 133         const char *ext = strrchr(filename, '.');
 134         if (!ext)
 135                 ext = filename;
 136
 137         int is_mpeg_ps = !(strcasecmp(ext, ".mpeg") && strcasecmp(ext, ".mpg") && strcasecmp(ext, ".vob") && strcasecmp(ext, ".bin"));
 138         int is_mpeg_ts = !strcasecmp(ext, ".ts");
 139         int is_matroska = !strcasecmp(ext, ".mkv");
 140         int is_avi = !strcasecmp(ext, ".avi");
 141         int is_mp3 = !strcasecmp(ext, ".mp3"); /* force mp3 instead of decodebin */
 142         int is_video = is_mpeg_ps || is_mpeg_ts || is_matroska || is_avi;
 143         int is_streaming = !strncmp(filename, "http://", 7);
 144         int is_AudioCD = !(strncmp(filename, "/autofs/", 8) || strncmp(filename+strlen(filename)-13, "/track-", 7) || strcasecmp(ext, ".wav"));
 145
 146         eDebug("filename: %s, is_mpeg_ps: %d, is_mpeg_ts: %d, is_video: %d, is_streaming: %d, is_mp3: %d, is_matroska: %d, is_avi: %d, is_AudioCD: %d", filename, is_mpeg_ps, is_mpeg_ts, is_video, is_streaming, is_mp3, is_matroska, is_avi, is_AudioCD);
 147
 148         int is_audio = !is_video;
 149
 150         int all_ok = 0;
 151
 152 //      GError *blubb = NULL;
 153 //      GstElement* m_gst_pipeline = gst_parse_launch("filesrc location=/media/hdd/movie/artehd_2lang.mkv ! matroskademux name=demux  demux.audio_00 ! input-selector name=a demux.audio_01 ! a.   a. ! queue ! dvbaudiosink", &blubb);
 154 //      return;
 155
 156         m_gst_pipeline = gst_pipeline_new ("mediaplayer");
 157         if (!m_gst_pipeline)
 158                 eWarning("failed to create pipeline");
 159
 160         if (is_AudioCD)
 161         {
 162                 source = gst_element_factory_make ("cdiocddasrc", "cda-source");
 163                 if (source)
 164                         g_object_set (G_OBJECT (source), "device", "/dev/cdroms/cdrom0", NULL);
 165                 else
 166                         is_AudioCD = 0;
 167         }
 168         if ( !is_streaming && !is_AudioCD )
 169                 source = gst_element_factory_make ("filesrc", "file-source");
 170         else if ( is_streaming )
 171         {
 172                 source = gst_element_factory_make ("neonhttpsrc", "http-source");
 173                 if (source)
 174                         g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
 175         }
 176
 177         if (!source)
 178                 eWarning("failed to create %s", is_streaming ? "neonhttpsrc" : "filesrc");
 179                                 /* configure source */
 180         else if (!is_AudioCD)
 181                 g_object_set (G_OBJECT (source), "location", filename, NULL);
 182         else
 183         {
 184                 int track = atoi(filename+18);
 185                 eDebug("play audio CD track #%i",track);
 186                 if (track > 0)
 187                         g_object_set (G_OBJECT (source), "track", track, NULL);
 188         }
 189
 190         if (is_audio)
 191         {
 192                         /* filesrc -> decodebin -> audioconvert -> capsfilter -> alsasink */
 193                 const char *decodertype = is_mp3 ? "mad" : "decodebin";
 194
 195                 decoder = gst_element_factory_make (decodertype, "decoder");
 196                 if (!decoder)
 197                         eWarning("failed to create %s decoder", decodertype);
 198
 199                         /* mp3 decoding needs id3demux to extract ID3 data. 'decodebin' would do that internally. */
 200                 if (is_mp3)
 201                 {
 202                         filter = gst_element_factory_make ("id3demux", "filter");
 203                         if (!filter)
 204                                 eWarning("failed to create id3demux");
 205                 }
 206
 207                 conv = gst_element_factory_make ("audioconvert", "converter");
 208                 if (!conv)
 209                         eWarning("failed to create audioconvert");
 210
 211                 flt = gst_element_factory_make ("capsfilter", "flt");
 212                 if (!flt)
 213                         eWarning("failed to create capsfilter");
 214
 215                         /* for some reasons, we need to set the sample format to depth/width=16, because auto negotiation doesn't work. */
 216                         /* endianness, however, is not required to be set anymore. */
 217                 if (flt)
 218                 {
 219                         GstCaps *caps = gst_caps_new_simple("audio/x-raw-int", /* "endianness", G_TYPE_INT, 4321, */ "depth", G_TYPE_INT, 16, "width", G_TYPE_INT, 16, /*"channels", G_TYPE_INT, 2, */(char*)0);
 220                         g_object_set (G_OBJECT (flt), "caps", caps, (char*)0);
 221                         gst_caps_unref(caps);
 222                 }
 223
 224                 sink = gst_element_factory_make ("alsasink", "alsa-output");
 225                 if (!sink)
 226                         eWarning("failed to create osssink");
 227
 228                 if (source && decoder && conv && sink)
 229                         all_ok = 1;
 230         } else /* is_video */
 231         {
 232                         /* filesrc -> mpegdemux -> | queue_audio -> dvbaudiosink
 233                                                    | queue_video -> dvbvideosink */
 234
 235                 audio = gst_element_factory_make("dvbaudiosink", "audiosink");
 236                 queue_audio = gst_element_factory_make("queue", "queue_audio");
 237
 238                 video = gst_element_factory_make("dvbvideosink", "videosink");
 239                 queue_video = gst_element_factory_make("queue", "queue_video");
 240
 241                 if (is_mpeg_ps)
 242                         videodemux = gst_element_factory_make("flupsdemux", "videodemux");
 243                 else if (is_mpeg_ts)
 244                         videodemux = gst_element_factory_make("flutsdemux", "videodemux");
 245                 else if (is_matroska)
 246                         videodemux = gst_element_factory_make("matroskademux", "videodemux");
 247                 else if (is_avi)
 248                         videodemux = gst_element_factory_make("avidemux", "videodemux");
 249
 250                 if (!videodemux)
 251                 {
 252                         eDebug("fluendo mpegdemux not available, falling back to mpegdemux\n");
 253                         videodemux = gst_element_factory_make("mpegdemux", "videodemux");
 254                 }
 255
 256                 eDebug("audio: %p, queue_audio %p, video %p, queue_video %p, videodemux %p", audio, queue_audio, video, queue_video, videodemux);
 257                 if (audio && queue_audio && video && queue_video && videodemux)
 258                 {
 259                         g_object_set (G_OBJECT (queue_audio), "max-size-bytes", 256*1024, NULL);
 260                         g_object_set (G_OBJECT (queue_audio), "max-size-buffers", 0, NULL);
 261                         g_object_set (G_OBJECT (queue_audio), "max-size-time", (guint64)0, NULL);
 262                         g_object_set (G_OBJECT (queue_video), "max-size-buffers", 0, NULL);
 263                         g_object_set (G_OBJECT (queue_video), "max-size-bytes", 2*1024*1024, NULL);
 264                         g_object_set (G_OBJECT (queue_video), "max-size-time", (guint64)0, NULL);
 265                         all_ok = 1;
 266                 }
 267         }
 268
 269         if (m_gst_pipeline && all_ok)
 270         {
 271                 gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline)), gstBusSyncHandler, this);
 272
 273                 if (is_AudioCD)
 274                 {
 275                         queue_audio = gst_element_factory_make("queue", "queue_audio");
 276                         g_object_set (G_OBJECT (sink), "preroll-queue-len", 80, NULL);
 277                         gst_bin_add_many (GST_BIN (m_gst_pipeline), source, queue_audio, conv, sink, NULL);
 278                         gst_element_link_many(source, queue_audio, conv, sink, NULL);
 279                 }
 280                 else if (is_audio)
 281                 {
 282                         queue_audio = gst_element_factory_make("queue", "queue_audio");
 283
 284                         if (!is_mp3)
 285                         {
 286                                         /* decodebin has dynamic pads. When they get created, we connect them to the audio bin */
 287                                 g_signal_connect (decoder, "new-decoded-pad", G_CALLBACK(gstCBnewPad), this);
 288                                 g_signal_connect (decoder, "unknown-type", G_CALLBACK(gstCBunknownType), this);
 289                                 g_object_set (G_OBJECT (sink), "preroll-queue-len", 80, NULL);
 290                         }
 291
 292                                 /* gst_bin will take the 'floating references' */
 293                         gst_bin_add_many (GST_BIN (m_gst_pipeline),
 294                                                 source, queue_audio, decoder, NULL);
 295
 296                         if (filter)
 297                         {
 298                                         /* id3demux also has dynamic pads, which need to be connected to the decoder (this is done in the 'gstCBfilterPadAdded' CB) */
 299                                 gst_bin_add(GST_BIN(m_gst_pipeline), filter);
 300                                 gst_element_link(source, filter);
 301                                 g_signal_connect (filter, "pad-added", G_CALLBACK(gstCBfilterPadAdded), this);
 302                         } else
 303                                         /* in decodebin's case we can just connect the source with the decodebin, and decodebin will take care about id3demux (or whatever is required) */
 304                                 gst_element_link_many(source, queue_audio, decoder, NULL);
 305
 306                                 /* create audio bin with the audioconverter, the capsfilter and the audiosink */
 307                         audio = gst_bin_new ("audiobin");
 308
 309                         GstPad *audiopad = gst_element_get_static_pad (conv, "sink");
 310                         gst_bin_add_many(GST_BIN(audio), conv, flt, sink, (char*)0);
 311                         gst_element_link_many(conv, flt, sink, (char*)0);
 312                         gst_element_add_pad(audio, gst_ghost_pad_new ("sink", audiopad));
 313                         gst_object_unref(audiopad);
 314                         gst_bin_add (GST_BIN(m_gst_pipeline), audio);
 315
 316                                 /* in mad's case, we can directly connect the decoder to the audiobin. otherwise, we do this in gstCBnewPad */
 317                         if (is_mp3)
 318                                 gst_element_link(decoder, audio);
 319                 } else /* is_video */
 320                 {
 321                         switch_audio = gst_element_factory_make ("input-selector", "switch_audio");
 322                         g_object_set (G_OBJECT (switch_audio), "select-all", TRUE, NULL);
 323                         gst_bin_add_many(GST_BIN(m_gst_pipeline), source, videodemux, switch_audio, audio, queue_audio, video, queue_video, NULL);
 324                         gst_element_link(source, videodemux);
 325                         gst_element_link(switch_audio, queue_audio);
 326                         gst_element_link(queue_audio, audio);
 327                         gst_element_link(queue_video, video);
 328                         g_signal_connect(videodemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
 329                 }
 330         } else
 331         {
 332                 if (m_gst_pipeline)
 333                         gst_object_unref(GST_OBJECT(m_gst_pipeline));
 334                 if (source)
 335                         gst_object_unref(GST_OBJECT(source));
 336                 if (decoder)
 337                         gst_object_unref(GST_OBJECT(decoder));
 338                 if (conv)
 339                         gst_object_unref(GST_OBJECT(conv));
 340                 if (sink)
 341                         gst_object_unref(GST_OBJECT(sink));
 342
 343                 if (audio)
 344                         gst_object_unref(GST_OBJECT(audio));
 345                 if (queue_audio)
 346                         gst_object_unref(GST_OBJECT(queue_audio));
 347                 if (video)
 348                         gst_object_unref(GST_OBJECT(video));
 349                 if (queue_video)
 350                         gst_object_unref(GST_OBJECT(queue_video));
 351                 if (videodemux)
 352                         gst_object_unref(GST_OBJECT(videodemux));
 353
 354                 eDebug("sorry, can't play.");
 355                 m_gst_pipeline = 0;
 356         }
 357
 358         gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
 359 }
 360
 361 eServiceMP3::~eServiceMP3()
 362 {
 363         if (m_state == stRunning)
 364                 stop();
 365
 366         if (m_stream_tags)
 367                 gst_tag_list_free(m_stream_tags);
 368
 369         if (m_gst_pipeline)
 370         {
 371                 gst_object_unref (GST_OBJECT (m_gst_pipeline));
 372                 eDebug("SERVICEMP3 destruct!");
 373         }
 374 }
 375
 376 DEFINE_REF(eServiceMP3);
 377
 378 RESULT eServiceMP3::connectEvent(const Slot2<void,iPlayableService*,int> &event, ePtr<eConnection> &connection)
 379 {
 380         connection = new eConnection((iPlayableService*)this, m_event.connect(event));
 381         return 0;
 382 }
 383
 384 RESULT eServiceMP3::start()
 385 {
 386         assert(m_state == stIdle);
 387
 388         m_state = stRunning;
 389         if (m_gst_pipeline)
 390         {
 391                 eDebug("starting pipeline");
 392                 gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
 393         }
 394         m_event(this, evStart);
 395         return 0;
 396 }
 397
 398 RESULT eServiceMP3::stop()
 399 {
 400         assert(m_state != stIdle);
 401         if (m_state == stStopped)
 402                 return -1;
 403         eDebug("MP3: %s stop\n", m_filename.c_str());
 404         gst_element_set_state(m_gst_pipeline, GST_STATE_NULL);
 405         m_state = stStopped;
 406         return 0;
 407 }
 408
 409 RESULT eServiceMP3::setTarget(int target)
 410 {
 411         return -1;
 412 }
 413
 414 RESULT eServiceMP3::pause(ePtr<iPauseableService> &ptr)
 415 {
 416         ptr=this;
 417         return 0;
 418 }
 419
 420 RESULT eServiceMP3::setSlowMotion(int ratio)
 421 {
 422         return -1;
 423 }
 424
 425 RESULT eServiceMP3::setFastForward(int ratio)
 426 {
 427         return -1;
 428 }
 429
 430                 // iPausableService
 431 RESULT eServiceMP3::pause()
 432 {
 433         if (!m_gst_pipeline)
 434                 return -1;
 435         gst_element_set_state(m_gst_pipeline, GST_STATE_PAUSED);
 436         return 0;
 437 }
 438
 439 RESULT eServiceMP3::unpause()
 440 {
 441         if (!m_gst_pipeline)
 442                 return -1;
 443         gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING);
 444         return 0;
 445 }
 446
 447         /* iSeekableService */
 448 RESULT eServiceMP3::seek(ePtr<iSeekableService> &ptr)
 449 {
 450         ptr = this;
 451         return 0;
 452 }
 453
 454 RESULT eServiceMP3::getLength(pts_t &pts)
 455 {
 456         if (!m_gst_pipeline)
 457                 return -1;
 458         if (m_state != stRunning)
 459                 return -1;
 460
 461         GstFormat fmt = GST_FORMAT_TIME;
 462         gint64 len;
 463
 464         if (!gst_element_query_duration(m_gst_pipeline, &fmt, &len))
 465                 return -1;
 466
 467                 /* len is in nanoseconds. we have 90 000 pts per second. */
 468
 469         pts = len / 11111;
 470         return 0;
 471 }
 472
 473 RESULT eServiceMP3::seekTo(pts_t to)
 474 {
 475         if (!m_gst_pipeline)
 476                 return -1;
 477
 478                 /* convert pts to nanoseconds */
 479         gint64 time_nanoseconds = to * 11111LL;
 480         if (!gst_element_seek (m_gst_pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH,
 481                 GST_SEEK_TYPE_SET, time_nanoseconds,
 482                 GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE))
 483         {
 484                 eDebug("SEEK failed");
 485                 return -1;
 486         }
 487         return 0;
 488 }
 489
 490 RESULT eServiceMP3::seekRelative(int direction, pts_t to)
 491 {
 492         if (!m_gst_pipeline)
 493                 return -1;
 494
 495         pause();
 496
 497         pts_t ppos;
 498         getPlayPosition(ppos);
 499         ppos += to * direction;
 500         if (ppos < 0)
 501                 ppos = 0;
 502         seekTo(ppos);
 503
 504         unpause();
 505
 506         return 0;
 507 }
 508
 509 RESULT eServiceMP3::getPlayPosition(pts_t &pts)
 510 {
 511         if (!m_gst_pipeline)
 512                 return -1;
 513         if (m_state != stRunning)
 514                 return -1;
 515
 516         GstFormat fmt = GST_FORMAT_TIME;
 517         gint64 len;
 518
 519         if (!gst_element_query_position(m_gst_pipeline, &fmt, &len))
 520                 return -1;
 521
 522                 /* len is in nanoseconds. we have 90 000 pts per second. */
 523         pts = len / 11111;
 524         return 0;
 525 }
 526
 527 RESULT eServiceMP3::setTrickmode(int trick)
 528 {
 529                 /* trickmode currently doesn't make any sense for us. */
 530         return -1;
 531 }
 532
 533 RESULT eServiceMP3::isCurrentlySeekable()
 534 {
 535         return 1;
 536 }
 537
 538 RESULT eServiceMP3::info(ePtr<iServiceInformation>&i)
 539 {
 540         i = this;
 541         return 0;
 542 }
 543
 544 RESULT eServiceMP3::getName(std::string &name)
 545 {
 546         name = m_filename;
 547         size_t n = name.rfind('/');
 548         if (n != std::string::npos)
 549                 name = name.substr(n + 1);
 550         return 0;
 551 }
 552
 553 int eServiceMP3::getInfo(int w)
 554 {
 555         gchar *tag = 0;
 556
 557         switch (w)
 558         {
 559         case sTitle:
 560         case sArtist:
 561         case sAlbum:
 562         case sComment:
 563         case sTracknumber:
 564         case sGenre:
 565         case sVideoType:
 566                 return resIsString;
 567         case sCurrentTitle:
 568                 tag = GST_TAG_TRACK_NUMBER;
 569                 break;
 570         case sTotalTitles:
 571                 tag = GST_TAG_TRACK_COUNT;
 572                 break;
 573         default:
 574                 return resNA;
 575         }
 576
 577         if (!m_stream_tags || !tag)
 578                 return 0;
 579
 580         guint value;
 581         if (gst_tag_list_get_uint(m_stream_tags, tag, &value))
 582                 return (int) value;
 583
 584         return 0;
 585
 586 }
 587
 588 std::string eServiceMP3::getInfoString(int w)
 589 {
 590         gchar *tag = 0;
 591         switch (w)
 592         {
 593         case sTitle:
 594                 tag = GST_TAG_TITLE;
 595                 break;
 596         case sArtist:
 597                 tag = GST_TAG_ARTIST;
 598                 break;
 599         case sAlbum:
 600                 tag = GST_TAG_ALBUM;
 601                 break;
 602         case sComment:
 603                 tag = GST_TAG_COMMENT;
 604                 break;
 605         case sTracknumber:
 606                 tag = GST_TAG_TRACK_NUMBER;
 607                 break;
 608         case sGenre:
 609                 tag = GST_TAG_GENRE;
 610                 break;
 611         case sVideoType:
 612                 tag = GST_TAG_VIDEO_CODEC;
 613                 break;
 614         default:
 615                 return "";
 616         }
 617
 618         if (!m_stream_tags || !tag)
 619                 return "";
 620
 621         gchar *value;
 622
 623         if (gst_tag_list_get_string(m_stream_tags, tag, &value))
 624         {
 625                 std::string res = value;
 626                 g_free(value);
 627                 return res;
 628         }
 629
 630         return "";
 631 }
 632
 633 RESULT eServiceMP3::audioChannel(ePtr<iAudioChannelSelection> &ptr)
 634 {
 635         ptr = this;
 636         return 0;
 637 }
 638
 639 RESULT eServiceMP3::audioTracks(ePtr<iAudioTrackSelection> &ptr)
 640 {
 641         ptr = this;
 642         return 0;
 643 }
 644
 645 int eServiceMP3::getNumberOfTracks()
 646 {
 647         eDebug("eServiceMP3::getNumberOfTracks()=%i",m_audioStreams.size());
 648         return m_audioStreams.size();
 649 }
 650
 651 int eServiceMP3::getCurrentTrack()
 652 {
 653         eDebug("eServiceMP3::getCurrentTrack()=%i",m_currentAudioStream);
 654         return m_currentAudioStream;
 655 }
 656
 657 RESULT eServiceMP3::selectTrack(unsigned int i)
 658 {
 659         eDebug("eServiceMP3::selectTrack(%i)",i);
 660
 661         GstPadLinkReturn ret;
 662         gint nb_sources;
 663         GstPad *active_pad;
 664
 665         GstElement *selector = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_audio");
 666         g_object_get (G_OBJECT (selector), "n-pads", &nb_sources, NULL);
 667         g_object_get (G_OBJECT (selector), "active-pad", &active_pad, NULL);
 668
 669         if ( i >= m_audioStreams.size() || i >= nb_sources || m_currentAudioStream >= m_audioStreams.size() )
 670                 return -2;
 671
 672         char sinkpad[8];
 673         sprintf(sinkpad, "sink%d", i);
 674
 675         g_object_set (G_OBJECT (selector), "active-pad", gst_element_get_pad (selector, sinkpad), NULL);
 676         g_object_get (G_OBJECT (selector), "active-pad", &active_pad, NULL);
 677
 678         gchar *name;
 679         name = gst_pad_get_name (active_pad);
 680         eDebug ("switched audio to (%s)", name);
 681
 682         m_currentAudioStream = i;
 683         return 0;
 684 }
 685
 686 int eServiceMP3::getCurrentChannel()
 687 {
 688         return STEREO;
 689 }
 690
 691 RESULT eServiceMP3::selectChannel(int i)
 692 {
 693         eDebug("eServiceMP3::selectChannel(%i)",i);
 694         return 0;
 695 }
 696
 697 RESULT eServiceMP3::getTrackInfo(struct iAudioTrackInfo &info, unsigned int i)
 698 {
 699 //      eDebug("eServiceMP3::getTrackInfo(&info, %i)",i);
 700         if (i >= m_audioStreams.size())
 701                 return -2;
 702         if (m_audioStreams[i].type == audioStream::atMP2)
 703                 info.m_description = "MP2";
 704         else if (m_audioStreams[i].type == audioStream::atMP3)
 705                 info.m_description = "MP3";
 706         else if (m_audioStreams[i].type == audioStream::atAC3)
 707                 info.m_description = "AC3";
 708         else if (m_audioStreams[i].type == audioStream::atAAC)
 709                 info.m_description = "AAC";
 710         else  if (m_audioStreams[i].type == audioStream::atDTS)
 711                 info.m_description = "DTS";
 712         else
 713                 info.m_description = "???";
 714         if (info.m_language.empty())
 715                 info.m_language = m_audioStreams[i].language_code;
 716         return 0;
 717 }
 718
 719 void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
 720 {
 721         if (!msg)
 722                 return;
 723         gchar *sourceName;
 724         GstObject *source;
 725
 726         source = GST_MESSAGE_SRC(msg);
 727         sourceName = gst_object_get_name(source);
 728
 729         if (gst_message_get_structure(msg))
 730         {
 731                 gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
 732                 eDebug("gst_message from %s: %s", sourceName, string);
 733                 g_free(string);
 734         }
 735         else
 736                 eDebug("gst_message from %s: %s (without structure)", sourceName, GST_MESSAGE_TYPE_NAME(msg));
 737
 738         switch (GST_MESSAGE_TYPE (msg))
 739         {
 740         case GST_MESSAGE_EOS:
 741                 m_event((iPlayableService*)this, evEOF);
 742                 break;
 743         case GST_MESSAGE_ERROR:
 744         {
 745                 gchar *debug;
 746                 GError *err;
 747
 748                 gst_message_parse_error (msg, &err, &debug);
 749                 g_free (debug);
 750                 eWarning("Gstreamer error: %s (%i)", err->message, err->code );
 751                 if ( err->domain == GST_STREAM_ERROR && err->code == GST_STREAM_ERROR_DECODE )
 752                 {
 753                         if ( g_strrstr(sourceName, "videosink") )
 754                                 m_event((iPlayableService*)this, evUser+11);
 755                 }
 756                 g_error_free(err);
 757                         /* TODO: signal error condition to user */
 758                 break;
 759         }
 760         case GST_MESSAGE_TAG:
 761         {
 762                 GstTagList *tags, *result;
 763                 gst_message_parse_tag(msg, &tags);
 764
 765                 result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_PREPEND);
 766                 if (result)
 767                 {
 768                         if (m_stream_tags)
 769                                 gst_tag_list_free(m_stream_tags);
 770                         m_stream_tags = result;
 771                 }
 772
 773                 gchar *g_audiocodec;
 774                 if (gst_tag_list_get_string(m_stream_tags, GST_TAG_AUDIO_CODEC, &g_audiocodec))
 775                 {
 776                         static int a_str_cnt = 0;
 777                         if ( g_strrstr(g_audiocodec, "MPEG-1 layer 2") )
 778                                 m_audioStreams[a_str_cnt].type = audioStream::atMP2;
 779                         else if ( g_strrstr(g_audiocodec, "AC-3 audio") )
 780                                 m_audioStreams[a_str_cnt].type = audioStream::atAC3;
 781                         gchar *g_language;
 782                         if ( gst_tag_list_get_string(m_stream_tags, GST_TAG_LANGUAGE_CODE, &g_language) )
 783                         {
 784                                 m_audioStreams[a_str_cnt].language_code = std::string(g_language);
 785                                 g_free (g_language);
 786                         }
 787                         g_free (g_audiocodec);
 788                         a_str_cnt++;
 789                 }
 790                 break;
 791         }
 792         default:
 793                 break;
 794         }
 795         g_free (sourceName);
 796 }
 797
 798 GstBusSyncReply eServiceMP3::gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data)
 799 {
 800         eServiceMP3 *_this = (eServiceMP3*)user_data;
 801         _this->m_pump.send(1);
 802                 /* wake */
 803         return GST_BUS_PASS;
 804 }
 805
 806 void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
 807 {
 808         eServiceMP3 *_this = (eServiceMP3*)user_data;
 809         gchar *name;
 810         name = gst_pad_get_name (pad);
 811         eDebug ("A new pad %s was created", name);
 812         if (g_strrstr(name,"audio")) // mpegdemux, matroskademux, avidemux use video_nn with n=0,1,.., flupsdemux uses stream id
 813         {
 814                 audioStream audio;
 815                 audio.pad = pad;
 816                 _this->m_audioStreams.push_back(audio);
 817                 GstElement *selector = gst_bin_get_by_name( GST_BIN(_this->m_gst_pipeline), "switch_audio" );
 818                 GstPadLinkReturn ret = gst_pad_link(pad, gst_element_get_request_pad (selector, "sink%d"));
 819                 if ( _this->m_audioStreams.size() == 1 )
 820                 {
 821                         _this->selectTrack(_this->m_audioStreams.size()-1);
 822                         gst_element_set_state (_this->m_gst_pipeline, GST_STATE_PLAYING);
 823                 }
 824                 else
 825                         g_object_set (G_OBJECT (selector), "select-all", FALSE, NULL);
 826         }
 827         if (g_strrstr(name,"video"))
 828         {
 829                 GstElement *video = gst_bin_get_by_name(GST_BIN (_this->m_gst_pipeline),"queue_video");
 830                 gst_pad_link(pad, gst_element_get_static_pad (video, "sink"));
 831         }
 832         g_free (name);
 833 }
 834
 835 void eServiceMP3::gstCBfilterPadAdded(GstElement *filter, GstPad *pad, gpointer user_data)
 836 {
 837         eServiceMP3 *_this = (eServiceMP3*)user_data;
 838         GstElement *decoder = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"decoder");
 839         gst_pad_link(pad, gst_element_get_static_pad (decoder, "sink"));
 840 }
 841
 842 void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last, gpointer user_data)
 843 {
 844         eServiceMP3 *_this = (eServiceMP3*)user_data;
 845         GstCaps *caps;
 846         GstStructure *str;
 847         GstPad *audiopad;
 848
 849         /* only link once */
 850         GstElement *audio = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"audiobin");
 851         audiopad = gst_element_get_static_pad (audio, "sink");
 852         if ( !audiopad || GST_PAD_IS_LINKED (audiopad)) {
 853                 eDebug("audio already linked!");
 854                 g_object_unref (audiopad);
 855                 return;
 856         }
 857
 858         /* check media type */
 859         caps = gst_pad_get_caps (pad);
 860         str = gst_caps_get_structure (caps, 0);
 861         eDebug("gst new pad! %s", gst_structure_get_name (str));
 862
 863         if (!g_strrstr (gst_structure_get_name (str), "audio")) {
 864                 gst_caps_unref (caps);
 865                 gst_object_unref (audiopad);
 866                 return;
 867         }
 868
 869         gst_caps_unref (caps);
 870         gst_pad_link (pad, audiopad);
 871 }
 872
 873 void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
 874 {
 875         GstStructure *str;
 876
 877         /* check media type */
 878         caps = gst_pad_get_caps (pad);
 879         str = gst_caps_get_structure (caps, 0);
 880         eDebug("unknown type: %s - this can't be decoded.", gst_structure_get_name (str));
 881         gst_caps_unref (caps);
 882 }
 883
 884 void eServiceMP3::gstPoll(const int&)
 885 {
 886                 /* ok, we have a serious problem here. gstBusSyncHandler sends
 887                    us the wakup signal, but likely before it was posted.
 888                    the usleep, an EVIL HACK (DON'T DO THAT!!!) works around this.
 889
 890                    I need to understand the API a bit more to make this work
 891                    proplerly. */
 892         usleep(1);
 893
 894         GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline));
 895         GstMessage *message;
 896         while ((message = gst_bus_pop (bus)))
 897         {
 898                 gstBusCall(bus, message);
 899                 gst_message_unref (message);
 900         }
 901 }
 902
 903 eAutoInitPtr<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
 904 #else
 905 #warning gstreamer not available, not building media player
 906 #endif