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_audioStreams.clear();
 120         m_currentAudioStream = 0;
 121         CONNECT(m_pump.recv_msg, eServiceMP3::gstPoll);
 122         GstElement *source = 0;
 123
 124         GstElement *filter = 0, *decoder = 0, *conv = 0, *flt = 0, *sink = 0; /* for audio */
 125
 126         GstElement *audio = 0, *switch_audio = 0, *queue_audio = 0, *video = 0, *queue_video = 0, *videodemux = 0;
 127
 128         m_state = stIdle;
 129         eDebug("SERVICEMP3 construct!");
 130
 131                 /* FIXME: currently, decodebin isn't possible for
 132                    video streams. in that case, make a manual pipeline. */
 133
 134         const char *ext = strrchr(filename, '.');
 135         if (!ext)
 136                 ext = filename;
 137
 138         int is_mpeg_ps = !(strcasecmp(ext, ".mpeg") && strcasecmp(ext, ".mpg") && strcasecmp(ext, ".vob") && strcasecmp(ext, ".bin"));
 139         int is_mpeg_ts = !strcasecmp(ext, ".ts");
 140         int is_matroska = !strcasecmp(ext, ".mkv");
 141         int is_avi = !strcasecmp(ext, ".avi");
 142         int is_mp3 = !strcasecmp(ext, ".mp3"); /* force mp3 instead of decodebin */
 143         int is_video = is_mpeg_ps || is_mpeg_ts || is_matroska || is_avi;
 144         int is_streaming = !strncmp(filename, "http://", 7);
 145         int is_AudioCD = !(strncmp(filename, "/autofs/", 8) || strncmp(filename+strlen(filename)-13, "/track-", 7) || strcasecmp(ext, ".wav"));
 146
 147         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);
 148
 149         int is_audio = !is_video;
 150
 151         int all_ok = 0;
 152
 153         m_gst_pipeline = gst_pipeline_new ("mediaplayer");
 154         if (!m_gst_pipeline)
 155                 eWarning("failed to create pipeline");
 156
 157         if (is_AudioCD)
 158         {
 159                 source = gst_element_factory_make ("cdiocddasrc", "cda-source");
 160                 if (source)
 161                         g_object_set (G_OBJECT (source), "device", "/dev/cdroms/cdrom0", NULL);
 162                 else
 163                         is_AudioCD = 0;
 164         }
 165         if ( !is_streaming && !is_AudioCD )
 166                 source = gst_element_factory_make ("filesrc", "file-source");
 167         else if ( is_streaming )
 168         {
 169                 source = gst_element_factory_make ("neonhttpsrc", "http-source");
 170                 if (source)
 171                         g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
 172         }
 173
 174         if (!source)
 175                 eWarning("failed to create %s", is_streaming ? "neonhttpsrc" : "filesrc");
 176                                 /* configure source */
 177         else if (!is_AudioCD)
 178                 g_object_set (G_OBJECT (source), "location", filename, NULL);
 179         else
 180         {
 181                 int track = atoi(filename+18);
 182                 eDebug("play audio CD track #%i",track);
 183                 if (track > 0)
 184                         g_object_set (G_OBJECT (source), "track", track, NULL);
 185         }
 186
 187         if (is_audio)
 188         {
 189                         /* filesrc -> decodebin -> audioconvert -> capsfilter -> alsasink */
 190                 const char *decodertype = is_mp3 ? "mad" : "decodebin";
 191
 192                 decoder = gst_element_factory_make (decodertype, "decoder");
 193                 if (!decoder)
 194                         eWarning("failed to create %s decoder", decodertype);
 195
 196                         /* mp3 decoding needs id3demux to extract ID3 data. 'decodebin' would do that internally. */
 197                 if (is_mp3)
 198                 {
 199                         filter = gst_element_factory_make ("id3demux", "filter");
 200                         if (!filter)
 201                                 eWarning("failed to create id3demux");
 202                 }
 203
 204                 conv = gst_element_factory_make ("audioconvert", "converter");
 205                 if (!conv)
 206                         eWarning("failed to create audioconvert");
 207
 208                 flt = gst_element_factory_make ("capsfilter", "flt");
 209                 if (!flt)
 210                         eWarning("failed to create capsfilter");
 211
 212                         /* for some reasons, we need to set the sample format to depth/width=16, because auto negotiation doesn't work. */
 213                         /* endianness, however, is not required to be set anymore. */
 214                 if (flt)
 215                 {
 216                         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);
 217                         g_object_set (G_OBJECT (flt), "caps", caps, (char*)0);
 218                         gst_caps_unref(caps);
 219                 }
 220
 221                 sink = gst_element_factory_make ("alsasink", "alsa-output");
 222                 if (!sink)
 223                         eWarning("failed to create osssink");
 224
 225                 if (source && decoder && conv && sink)
 226                         all_ok = 1;
 227         } else /* is_video */
 228         {
 229                         /* filesrc -> mpegdemux -> | queue_audio -> dvbaudiosink
 230                                                    | queue_video -> dvbvideosink */
 231
 232                 audio = gst_element_factory_make("dvbaudiosink", "audiosink");
 233                 queue_audio = gst_element_factory_make("queue", "queue_audio");
 234
 235                 video = gst_element_factory_make("dvbvideosink", "videosink");
 236                 queue_video = gst_element_factory_make("queue", "queue_video");
 237
 238                 if (is_mpeg_ps)
 239                         videodemux = gst_element_factory_make("flupsdemux", "videodemux");
 240                 else if (is_mpeg_ts)
 241                         videodemux = gst_element_factory_make("flutsdemux", "videodemux");
 242                 else if (is_matroska)
 243                         videodemux = gst_element_factory_make("matroskademux", "videodemux");
 244                 else if (is_avi)
 245                         videodemux = gst_element_factory_make("avidemux", "videodemux");
 246
 247                 if (!videodemux)
 248                 {
 249                         eDebug("fluendo mpegdemux not available, falling back to mpegdemux\n");
 250                         videodemux = gst_element_factory_make("mpegdemux", "videodemux");
 251                 }
 252
 253                 eDebug("audio: %p, queue_audio %p, video %p, queue_video %p, videodemux %p", audio, queue_audio, video, queue_video, videodemux);
 254                 if (audio && queue_audio && video && queue_video && videodemux)
 255                 {
 256                         g_object_set (G_OBJECT (queue_audio), "max-size-bytes", 256*1024, NULL);
 257                         g_object_set (G_OBJECT (queue_audio), "max-size-buffers", 0, NULL);
 258                         g_object_set (G_OBJECT (queue_audio), "max-size-time", (guint64)0, NULL);
 259                         g_object_set (G_OBJECT (queue_video), "max-size-buffers", 0, NULL);
 260                         g_object_set (G_OBJECT (queue_video), "max-size-bytes", 2*1024*1024, NULL);
 261                         g_object_set (G_OBJECT (queue_video), "max-size-time", (guint64)0, NULL);
 262                         all_ok = 1;
 263                 }
 264         }
 265
 266         if (m_gst_pipeline && all_ok)
 267         {
 268                 gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline)), gstBusSyncHandler, this);
 269
 270                 if (is_AudioCD)
 271                 {
 272                         queue_audio = gst_element_factory_make("queue", "queue_audio");
 273                         g_object_set (G_OBJECT (sink), "preroll-queue-len", 80, NULL);
 274                         gst_bin_add_many (GST_BIN (m_gst_pipeline), source, queue_audio, conv, sink, NULL);
 275                         gst_element_link_many(source, queue_audio, conv, sink, NULL);
 276                 }
 277                 else if (is_audio)
 278                 {
 279                         queue_audio = gst_element_factory_make("queue", "queue_audio");
 280
 281                         if (!is_mp3)
 282                         {
 283                                         /* decodebin has dynamic pads. When they get created, we connect them to the audio bin */
 284                                 g_signal_connect (decoder, "new-decoded-pad", G_CALLBACK(gstCBnewPad), this);
 285                                 g_signal_connect (decoder, "unknown-type", G_CALLBACK(gstCBunknownType), this);
 286                                 g_object_set (G_OBJECT (sink), "preroll-queue-len", 80, NULL);
 287                         }
 288
 289                                 /* gst_bin will take the 'floating references' */
 290                         gst_bin_add_many (GST_BIN (m_gst_pipeline),
 291                                                 source, queue_audio, decoder, NULL);
 292
 293                         if (filter)
 294                         {
 295                                         /* id3demux also has dynamic pads, which need to be connected to the decoder (this is done in the 'gstCBfilterPadAdded' CB) */
 296                                 gst_bin_add(GST_BIN(m_gst_pipeline), filter);
 297                                 gst_element_link(source, filter);
 298                                 g_signal_connect (filter, "pad-added", G_CALLBACK(gstCBfilterPadAdded), this);
 299                         } else
 300                                         /* in decodebin's case we can just connect the source with the decodebin, and decodebin will take care about id3demux (or whatever is required) */
 301                                 gst_element_link_many(source, queue_audio, decoder, NULL);
 302
 303                                 /* create audio bin with the audioconverter, the capsfilter and the audiosink */
 304                         audio = gst_bin_new ("audiobin");
 305
 306                         GstPad *audiopad = gst_element_get_static_pad (conv, "sink");
 307                         gst_bin_add_many(GST_BIN(audio), conv, flt, sink, (char*)0);
 308                         gst_element_link_many(conv, flt, sink, (char*)0);
 309                         gst_element_add_pad(audio, gst_ghost_pad_new ("sink", audiopad));
 310                         gst_object_unref(audiopad);
 311                         gst_bin_add (GST_BIN(m_gst_pipeline), audio);
 312                                 /* in mad's case, we can directly connect the decoder to the audiobin. otherwise, we do this in gstCBnewPad */
 313                         if (is_mp3)
 314                                 gst_element_link(decoder, audio);
 315                         audioStream audioStreamElem;
 316                         m_audioStreams.push_back(audioStreamElem);
 317                 } else /* is_video */
 318                 {
 319                         gst_bin_add_many(GST_BIN(m_gst_pipeline), source, videodemux, audio, queue_audio, video, queue_video, NULL);
 320                         switch_audio = gst_element_factory_make ("input-selector", "switch_audio");
 321                         if (switch_audio)
 322                         {
 323                                 g_object_set (G_OBJECT (switch_audio), "select-all", TRUE, NULL);
 324                                 gst_bin_add(GST_BIN(m_gst_pipeline), switch_audio);
 325                                 gst_element_link(switch_audio, queue_audio);
 326                         }
 327                         gst_element_link(source, videodemux);
 328                         gst_element_link(queue_audio, audio);
 329                         gst_element_link(queue_video, video);
 330                         g_signal_connect(videodemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
 331                 }
 332         } else
 333         {
 334                 if (m_gst_pipeline)
 335                         gst_object_unref(GST_OBJECT(m_gst_pipeline));
 336                 if (source)
 337                         gst_object_unref(GST_OBJECT(source));
 338                 if (decoder)
 339                         gst_object_unref(GST_OBJECT(decoder));
 340                 if (conv)
 341                         gst_object_unref(GST_OBJECT(conv));
 342                 if (sink)
 343                         gst_object_unref(GST_OBJECT(sink));
 344
 345                 if (audio)
 346                         gst_object_unref(GST_OBJECT(audio));
 347                 if (queue_audio)
 348                         gst_object_unref(GST_OBJECT(queue_audio));
 349                 if (video)
 350                         gst_object_unref(GST_OBJECT(video));
 351                 if (queue_video)
 352                         gst_object_unref(GST_OBJECT(queue_video));
 353                 if (videodemux)
 354                         gst_object_unref(GST_OBJECT(videodemux));
 355
 356                 eDebug("sorry, can't play.");
 357                 m_gst_pipeline = 0;
 358         }
 359
 360         gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
 361 }
 362
 363 eServiceMP3::~eServiceMP3()
 364 {
 365         if (m_state == stRunning)
 366                 stop();
 367
 368         if (m_stream_tags)
 369                 gst_tag_list_free(m_stream_tags);
 370
 371         if (m_gst_pipeline)
 372         {
 373                 gst_object_unref (GST_OBJECT (m_gst_pipeline));
 374                 eDebug("SERVICEMP3 destruct!");
 375         }
 376 }
 377
 378 DEFINE_REF(eServiceMP3);
 379
 380 RESULT eServiceMP3::connectEvent(const Slot2<void,iPlayableService*,int> &event, ePtr<eConnection> &connection)
 381 {
 382         connection = new eConnection((iPlayableService*)this, m_event.connect(event));
 383         return 0;
 384 }
 385
 386 RESULT eServiceMP3::start()
 387 {
 388         assert(m_state == stIdle);
 389
 390         m_state = stRunning;
 391         if (m_gst_pipeline)
 392         {
 393                 eDebug("starting pipeline");
 394                 gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
 395         }
 396         m_event(this, evStart);
 397         return 0;
 398 }
 399
 400 RESULT eServiceMP3::stop()
 401 {
 402         assert(m_state != stIdle);
 403         if (m_state == stStopped)
 404                 return -1;
 405         eDebug("MP3: %s stop\n", m_filename.c_str());
 406         gst_element_set_state(m_gst_pipeline, GST_STATE_NULL);
 407         m_state = stStopped;
 408         return 0;
 409 }
 410
 411 RESULT eServiceMP3::setTarget(int target)
 412 {
 413         return -1;
 414 }
 415
 416 RESULT eServiceMP3::pause(ePtr<iPauseableService> &ptr)
 417 {
 418         ptr=this;
 419         return 0;
 420 }
 421
 422 RESULT eServiceMP3::setSlowMotion(int ratio)
 423 {
 424         return -1;
 425 }
 426
 427 RESULT eServiceMP3::setFastForward(int ratio)
 428 {
 429         return -1;
 430 }
 431
 432                 // iPausableService
 433 RESULT eServiceMP3::pause()
 434 {
 435         if (!m_gst_pipeline)
 436                 return -1;
 437         gst_element_set_state(m_gst_pipeline, GST_STATE_PAUSED);
 438         return 0;
 439 }
 440
 441 RESULT eServiceMP3::unpause()
 442 {
 443         if (!m_gst_pipeline)
 444                 return -1;
 445         gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING);
 446         return 0;
 447 }
 448
 449         /* iSeekableService */
 450 RESULT eServiceMP3::seek(ePtr<iSeekableService> &ptr)
 451 {
 452         ptr = this;
 453         return 0;
 454 }
 455
 456 RESULT eServiceMP3::getLength(pts_t &pts)
 457 {
 458         if (!m_gst_pipeline)
 459                 return -1;
 460         if (m_state != stRunning)
 461                 return -1;
 462
 463         GstFormat fmt = GST_FORMAT_TIME;
 464         gint64 len;
 465
 466         if (!gst_element_query_duration(m_gst_pipeline, &fmt, &len))
 467                 return -1;
 468
 469                 /* len is in nanoseconds. we have 90 000 pts per second. */
 470
 471         pts = len / 11111;
 472         return 0;
 473 }
 474
 475 RESULT eServiceMP3::seekTo(pts_t to)
 476 {
 477         if (!m_gst_pipeline)
 478                 return -1;
 479
 480                 /* convert pts to nanoseconds */
 481         gint64 time_nanoseconds = to * 11111LL;
 482         if (!gst_element_seek (m_gst_pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH,
 483                 GST_SEEK_TYPE_SET, time_nanoseconds,
 484                 GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE))
 485         {
 486                 eDebug("SEEK failed");
 487                 return -1;
 488         }
 489         return 0;
 490 }
 491
 492 RESULT eServiceMP3::seekRelative(int direction, pts_t to)
 493 {
 494         if (!m_gst_pipeline)
 495                 return -1;
 496
 497         pause();
 498
 499         pts_t ppos;
 500         getPlayPosition(ppos);
 501         ppos += to * direction;
 502         if (ppos < 0)
 503                 ppos = 0;
 504         seekTo(ppos);
 505
 506         unpause();
 507
 508         return 0;
 509 }
 510
 511 RESULT eServiceMP3::getPlayPosition(pts_t &pts)
 512 {
 513         if (!m_gst_pipeline)
 514                 return -1;
 515         if (m_state != stRunning)
 516                 return -1;
 517
 518         GstFormat fmt = GST_FORMAT_TIME;
 519         gint64 len;
 520
 521         if (!gst_element_query_position(m_gst_pipeline, &fmt, &len))
 522                 return -1;
 523
 524                 /* len is in nanoseconds. we have 90 000 pts per second. */
 525         pts = len / 11111;
 526         return 0;
 527 }
 528
 529 RESULT eServiceMP3::setTrickmode(int trick)
 530 {
 531                 /* trickmode currently doesn't make any sense for us. */
 532         return -1;
 533 }
 534
 535 RESULT eServiceMP3::isCurrentlySeekable()
 536 {
 537         return 1;
 538 }
 539
 540 RESULT eServiceMP3::info(ePtr<iServiceInformation>&i)
 541 {
 542         i = this;
 543         return 0;
 544 }
 545
 546 RESULT eServiceMP3::getName(std::string &name)
 547 {
 548         name = m_filename;
 549         size_t n = name.rfind('/');
 550         if (n != std::string::npos)
 551                 name = name.substr(n + 1);
 552         return 0;
 553 }
 554
 555 int eServiceMP3::getInfo(int w)
 556 {
 557         gchar *tag = 0;
 558
 559         switch (w)
 560         {
 561         case sTitle:
 562         case sArtist:
 563         case sAlbum:
 564         case sComment:
 565         case sTracknumber:
 566         case sGenre:
 567         case sVideoType:
 568                 return resIsString;
 569         case sCurrentTitle:
 570                 tag = GST_TAG_TRACK_NUMBER;
 571                 break;
 572         case sTotalTitles:
 573                 tag = GST_TAG_TRACK_COUNT;
 574                 break;
 575         default:
 576                 return resNA;
 577         }
 578
 579         if (!m_stream_tags || !tag)
 580                 return 0;
 581
 582         guint value;
 583         if (gst_tag_list_get_uint(m_stream_tags, tag, &value))
 584                 return (int) value;
 585
 586         return 0;
 587
 588 }
 589
 590 std::string eServiceMP3::getInfoString(int w)
 591 {
 592         gchar *tag = 0;
 593         switch (w)
 594         {
 595         case sTitle:
 596                 tag = GST_TAG_TITLE;
 597                 break;
 598         case sArtist:
 599                 tag = GST_TAG_ARTIST;
 600                 break;
 601         case sAlbum:
 602                 tag = GST_TAG_ALBUM;
 603                 break;
 604         case sComment:
 605                 tag = GST_TAG_COMMENT;
 606                 break;
 607         case sTracknumber:
 608                 tag = GST_TAG_TRACK_NUMBER;
 609                 break;
 610         case sGenre:
 611                 tag = GST_TAG_GENRE;
 612                 break;
 613         case sVideoType:
 614                 tag = GST_TAG_VIDEO_CODEC;
 615                 break;
 616         default:
 617                 return "";
 618         }
 619
 620         if (!m_stream_tags || !tag)
 621                 return "";
 622
 623         gchar *value;
 624
 625         if (gst_tag_list_get_string(m_stream_tags, tag, &value))
 626         {
 627                 std::string res = value;
 628                 g_free(value);
 629                 return res;
 630         }
 631
 632         return "";
 633 }
 634
 635 RESULT eServiceMP3::audioChannel(ePtr<iAudioChannelSelection> &ptr)
 636 {
 637         ptr = this;
 638         return 0;
 639 }
 640
 641 RESULT eServiceMP3::audioTracks(ePtr<iAudioTrackSelection> &ptr)
 642 {
 643         ptr = this;
 644         return 0;
 645 }
 646
 647 int eServiceMP3::getNumberOfTracks()
 648 {
 649         return m_audioStreams.size();
 650 }
 651
 652 int eServiceMP3::getCurrentTrack()
 653 {
 654         return m_currentAudioStream;
 655 }
 656
 657 RESULT eServiceMP3::selectTrack(unsigned int i)
 658 {
 659         gint nb_sources;
 660         GstPad *active_pad;
 661         GstElement *selector = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_audio");
 662         if ( !selector)
 663         {
 664                 eDebug("can't switch audio tracks! gst-plugin-selector needed");
 665                 return -1;
 666         }
 667         g_object_get (G_OBJECT (selector), "n-pads", &nb_sources, NULL);
 668         g_object_get (G_OBJECT (selector), "active-pad", &active_pad, NULL);
 669         if ( i >= m_audioStreams.size() || i >= nb_sources || m_currentAudioStream >= m_audioStreams.size() )
 670                 return -2;
 671         char sinkpad[8];
 672         sprintf(sinkpad, "sink%d", i);
 673         g_object_set (G_OBJECT (selector), "active-pad", gst_element_get_pad (selector, sinkpad), NULL);
 674         g_object_get (G_OBJECT (selector), "active-pad", &active_pad, NULL);
 675         gchar *name;
 676         name = gst_pad_get_name (active_pad);
 677         eDebug ("switched audio to (%s)", name);
 678         g_free(name);
 679         m_currentAudioStream = i;
 680         return 0;
 681 }
 682
 683 int eServiceMP3::getCurrentChannel()
 684 {
 685         return STEREO;
 686 }
 687
 688 RESULT eServiceMP3::selectChannel(int i)
 689 {
 690         eDebug("eServiceMP3::selectChannel(%i)",i);
 691         return 0;
 692 }
 693
 694 RESULT eServiceMP3::getTrackInfo(struct iAudioTrackInfo &info, unsigned int i)
 695 {
 696 //      eDebug("eServiceMP3::getTrackInfo(&info, %i)",i);
 697         if (i >= m_audioStreams.size())
 698                 return -2;
 699         if (m_audioStreams[i].type == audioStream::atMP2)
 700                 info.m_description = "MP2";
 701         else if (m_audioStreams[i].type == audioStream::atMP3)
 702                 info.m_description = "MP3";
 703         else if (m_audioStreams[i].type == audioStream::atAC3)
 704                 info.m_description = "AC3";
 705         else if (m_audioStreams[i].type == audioStream::atAAC)
 706                 info.m_description = "AAC";
 707         else if (m_audioStreams[i].type == audioStream::atDTS)
 708                 info.m_description = "DTS";
 709         else if (m_audioStreams[i].type == audioStream::atPCM)
 710                 info.m_description = "PCM";
 711         else if (m_audioStreams[i].type == audioStream::atOGG)
 712                 info.m_description = "OGG";
 713         else
 714                 info.m_description = "???";
 715         if (info.m_language.empty())
 716                 info.m_language = m_audioStreams[i].language_code;
 717         return 0;
 718 }
 719
 720 void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
 721 {
 722         if (!msg)
 723                 return;
 724         gchar *sourceName;
 725         GstObject *source;
 726
 727         source = GST_MESSAGE_SRC(msg);
 728         sourceName = gst_object_get_name(source);
 729
 730         if (gst_message_get_structure(msg))
 731         {
 732                 gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
 733                 eDebug("gst_message from %s: %s", sourceName, string);
 734                 g_free(string);
 735         }
 736         else
 737                 eDebug("gst_message from %s: %s (without structure)", sourceName, GST_MESSAGE_TYPE_NAME(msg));
 738
 739         switch (GST_MESSAGE_TYPE (msg))
 740         {
 741         case GST_MESSAGE_EOS:
 742                 m_event((iPlayableService*)this, evEOF);
 743                 break;
 744         case GST_MESSAGE_ERROR:
 745         {
 746                 gchar *debug;
 747                 GError *err;
 748
 749                 gst_message_parse_error (msg, &err, &debug);
 750                 g_free (debug);
 751                 eWarning("Gstreamer error: %s (%i)", err->message, err->code );
 752                 if ( err->domain == GST_STREAM_ERROR && err->code == GST_STREAM_ERROR_DECODE )
 753                 {
 754                         if ( g_strrstr(sourceName, "videosink") )
 755                                 m_event((iPlayableService*)this, evUser+11);
 756                 }
 757                 g_error_free(err);
 758                         /* TODO: signal error condition to user */
 759                 break;
 760         }
 761         case GST_MESSAGE_TAG:
 762         {
 763                 GstTagList *tags, *result;
 764                 gst_message_parse_tag(msg, &tags);
 765
 766                 result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_PREPEND);
 767                 if (result)
 768                 {
 769                         if (m_stream_tags)
 770                                 gst_tag_list_free(m_stream_tags);
 771                         m_stream_tags = result;
 772                 }
 773                 gchar *g_audiocodec;
 774                 if (gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_audiocodec) && m_audioStreams.size())
 775                 {
 776                         std::vector<audioStream>::iterator IterAudioStream = m_audioStreams.begin();
 777                         while ( IterAudioStream->language_code.length() && IterAudioStream != m_audioStreams.end())
 778                                 IterAudioStream++;
 779                         if ( g_strrstr(g_audiocodec, "MPEG-1 layer 2") )
 780                                 IterAudioStream->type = audioStream::atMP2;
 781                         else if ( g_strrstr(g_audiocodec, "MPEG-1 layer 3") )
 782                                 IterAudioStream->type = audioStream::atMP3;
 783                         else if ( g_strrstr(g_audiocodec, "AC-3 audio") )
 784                                 IterAudioStream->type = audioStream::atAC3;
 785                         else if ( g_strrstr(g_audiocodec, "Uncompressed\ 16-bit\ PCM\ audio") )
 786                                 IterAudioStream->type = audioStream::atPCM;
 787                         gchar *g_language;
 788                         if ( gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_language) )
 789                                 IterAudioStream->language_code = std::string(g_language);
 790                         g_free (g_language);
 791                         g_free (g_audiocodec);
 792                 }
 793                 break;
 794         }
 795         default:
 796                 break;
 797         }
 798         g_free (sourceName);
 799 }
 800
 801 GstBusSyncReply eServiceMP3::gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data)
 802 {
 803         eServiceMP3 *_this = (eServiceMP3*)user_data;
 804         _this->m_pump.send(1);
 805                 /* wake */
 806         return GST_BUS_PASS;
 807 }
 808
 809 void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
 810 {
 811         eServiceMP3 *_this = (eServiceMP3*)user_data;
 812         GstBin *pipeline = GST_BIN(_this->m_gst_pipeline);
 813         gchar *name;
 814         name = gst_pad_get_name (pad);
 815         eDebug ("A new pad %s was created", name);
 816         if (g_strrstr(name,"audio")) // mpegdemux, matroskademux, avidemux use video_nn with n=0,1,.., flupsdemux uses stream id
 817         {
 818                 GstElement *selector = gst_bin_get_by_name(pipeline , "switch_audio" );
 819                 audioStream audio;
 820                 audio.pad = pad;
 821                 _this->m_audioStreams.push_back(audio);
 822                 if ( selector )
 823                 {
 824                         GstPadLinkReturn ret = gst_pad_link(pad, gst_element_get_request_pad (selector, "sink%d"));
 825                         if ( _this->m_audioStreams.size() == 1 )
 826                         {
 827                                 _this->selectTrack(0);
 828                                 gst_element_set_state (_this->m_gst_pipeline, GST_STATE_PLAYING);
 829                         }
 830                         else
 831                                 g_object_set (G_OBJECT (selector), "select-all", FALSE, NULL);
 832                 }
 833                 else
 834                         gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_audio"), "sink"));
 835         }
 836         if (g_strrstr(name,"video"))
 837         {
 838                 gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_video"), "sink"));
 839         }
 840         g_free (name);
 841 }
 842
 843 void eServiceMP3::gstCBfilterPadAdded(GstElement *filter, GstPad *pad, gpointer user_data)
 844 {
 845         eServiceMP3 *_this = (eServiceMP3*)user_data;
 846         GstElement *decoder = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"decoder");
 847         gst_pad_link(pad, gst_element_get_static_pad (decoder, "sink"));
 848 }
 849
 850 void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last, gpointer user_data)
 851 {
 852         eServiceMP3 *_this = (eServiceMP3*)user_data;
 853         GstCaps *caps;
 854         GstStructure *str;
 855         GstPad *audiopad;
 856
 857         /* only link once */
 858         GstElement *audio = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"audiobin");
 859         audiopad = gst_element_get_static_pad (audio, "sink");
 860         if ( !audiopad || GST_PAD_IS_LINKED (audiopad)) {
 861                 eDebug("audio already linked!");
 862                 g_object_unref (audiopad);
 863                 return;
 864         }
 865
 866         /* check media type */
 867         caps = gst_pad_get_caps (pad);
 868         str = gst_caps_get_structure (caps, 0);
 869         eDebug("gst new pad! %s", gst_structure_get_name (str));
 870
 871         if (!g_strrstr (gst_structure_get_name (str), "audio")) {
 872                 gst_caps_unref (caps);
 873                 gst_object_unref (audiopad);
 874                 return;
 875         }
 876
 877         gst_caps_unref (caps);
 878         gst_pad_link (pad, audiopad);
 879 }
 880
 881 void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
 882 {
 883         GstStructure *str;
 884
 885         /* check media type */
 886         caps = gst_pad_get_caps (pad);
 887         str = gst_caps_get_structure (caps, 0);
 888         eDebug("unknown type: %s - this can't be decoded.", gst_structure_get_name (str));
 889         gst_caps_unref (caps);
 890 }
 891
 892 void eServiceMP3::gstPoll(const int&)
 893 {
 894                 /* ok, we have a serious problem here. gstBusSyncHandler sends
 895                    us the wakup signal, but likely before it was posted.
 896                    the usleep, an EVIL HACK (DON'T DO THAT!!!) works around this.
 897
 898                    I need to understand the API a bit more to make this work
 899                    proplerly. */
 900         usleep(1);
 901
 902         GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline));
 903         GstMessage *message;
 904         while ((message = gst_bus_pop (bus)))
 905         {
 906                 gstBusCall(bus, message);
 907                 gst_message_unref (message);
 908         }
 909 }
 910
 911 eAutoInitPtr<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
 912 #else
 913 #warning gstreamer not available, not building media player
 914 #endif