331f7a246507b96a1d080e58091061eee869ce2b
[enigma2.git] / 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
313                                 /* in mad's case, we can directly connect the decoder to the audiobin. otherwise, we do this in gstCBnewPad */
314                         if (is_mp3)
315                                 gst_element_link(decoder, audio);
316                 } else /* is_video */
317                 {
318                         gst_bin_add_many(GST_BIN(m_gst_pipeline), source, videodemux, audio, queue_audio, video, queue_video, NULL);
319                         switch_audio = gst_element_factory_make ("input-selector", "switch_audio");
320                         if (switch_audio)
321                         {
322                                 g_object_set (G_OBJECT (switch_audio), "select-all", TRUE, NULL);
323                                 gst_bin_add(GST_BIN(m_gst_pipeline), switch_audio);
324                                 gst_element_link(switch_audio, queue_audio);
325                         }
326                         gst_element_link(source, videodemux);
327                         gst_element_link(queue_audio, audio);
328                         gst_element_link(queue_video, video);
329                         g_signal_connect(videodemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
330                 }
331         } else
332         {
333                 if (m_gst_pipeline)
334                         gst_object_unref(GST_OBJECT(m_gst_pipeline));
335                 if (source)
336                         gst_object_unref(GST_OBJECT(source));
337                 if (decoder)
338                         gst_object_unref(GST_OBJECT(decoder));
339                 if (conv)
340                         gst_object_unref(GST_OBJECT(conv));
341                 if (sink)
342                         gst_object_unref(GST_OBJECT(sink));
343
344                 if (audio)
345                         gst_object_unref(GST_OBJECT(audio));
346                 if (queue_audio)
347                         gst_object_unref(GST_OBJECT(queue_audio));
348                 if (video)
349                         gst_object_unref(GST_OBJECT(video));
350                 if (queue_video)
351                         gst_object_unref(GST_OBJECT(queue_video));
352                 if (videodemux)
353                         gst_object_unref(GST_OBJECT(videodemux));
354
355                 eDebug("sorry, can't play.");
356                 m_gst_pipeline = 0;
357         }
358         
359         gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
360 }
361
362 eServiceMP3::~eServiceMP3()
363 {
364         if (m_state == stRunning)
365                 stop();
366         
367         if (m_stream_tags)
368                 gst_tag_list_free(m_stream_tags);
369         
370         if (m_gst_pipeline)
371         {
372                 gst_object_unref (GST_OBJECT (m_gst_pipeline));
373                 eDebug("SERVICEMP3 destruct!");
374         }
375 }
376
377 DEFINE_REF(eServiceMP3);        
378
379 RESULT eServiceMP3::connectEvent(const Slot2<void,iPlayableService*,int> &event, ePtr<eConnection> &connection)
380 {
381         connection = new eConnection((iPlayableService*)this, m_event.connect(event));
382         return 0;
383 }
384
385 RESULT eServiceMP3::start()
386 {
387         assert(m_state == stIdle);
388         
389         m_state = stRunning;
390         if (m_gst_pipeline)
391         {
392                 eDebug("starting pipeline");
393                 gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
394         }
395         m_event(this, evStart);
396         return 0;
397 }
398
399 RESULT eServiceMP3::stop()
400 {
401         assert(m_state != stIdle);
402         if (m_state == stStopped)
403                 return -1;
404         eDebug("MP3: %s stop\n", m_filename.c_str());
405         gst_element_set_state(m_gst_pipeline, GST_STATE_NULL);
406         m_state = stStopped;
407         return 0;
408 }
409
410 RESULT eServiceMP3::setTarget(int target)
411 {
412         return -1;
413 }
414
415 RESULT eServiceMP3::pause(ePtr<iPauseableService> &ptr)
416 {
417         ptr=this;
418         return 0;
419 }
420
421 RESULT eServiceMP3::setSlowMotion(int ratio)
422 {
423         return -1;
424 }
425
426 RESULT eServiceMP3::setFastForward(int ratio)
427 {
428         return -1;
429 }
430   
431                 // iPausableService
432 RESULT eServiceMP3::pause()
433 {
434         if (!m_gst_pipeline)
435                 return -1;
436         gst_element_set_state(m_gst_pipeline, GST_STATE_PAUSED);
437         return 0;
438 }
439
440 RESULT eServiceMP3::unpause()
441 {
442         if (!m_gst_pipeline)
443                 return -1;
444         gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING);
445         return 0;
446 }
447
448         /* iSeekableService */
449 RESULT eServiceMP3::seek(ePtr<iSeekableService> &ptr)
450 {
451         ptr = this;
452         return 0;
453 }
454
455 RESULT eServiceMP3::getLength(pts_t &pts)
456 {
457         if (!m_gst_pipeline)
458                 return -1;
459         if (m_state != stRunning)
460                 return -1;
461         
462         GstFormat fmt = GST_FORMAT_TIME;
463         gint64 len;
464         
465         if (!gst_element_query_duration(m_gst_pipeline, &fmt, &len))
466                 return -1;
467         
468                 /* len is in nanoseconds. we have 90 000 pts per second. */
469         
470         pts = len / 11111;
471         return 0;
472 }
473
474 RESULT eServiceMP3::seekTo(pts_t to)
475 {
476         if (!m_gst_pipeline)
477                 return -1;
478
479                 /* convert pts to nanoseconds */
480         gint64 time_nanoseconds = to * 11111LL;
481         if (!gst_element_seek (m_gst_pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH,
482                 GST_SEEK_TYPE_SET, time_nanoseconds,
483                 GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE))
484         {
485                 eDebug("SEEK failed");
486                 return -1;
487         }
488         return 0;
489 }
490
491 RESULT eServiceMP3::seekRelative(int direction, pts_t to)
492 {
493         if (!m_gst_pipeline)
494                 return -1;
495
496         pause();
497
498         pts_t ppos;
499         getPlayPosition(ppos);
500         ppos += to * direction;
501         if (ppos < 0)
502                 ppos = 0;
503         seekTo(ppos);
504         
505         unpause();
506
507         return 0;
508 }
509
510 RESULT eServiceMP3::getPlayPosition(pts_t &pts)
511 {
512         if (!m_gst_pipeline)
513                 return -1;
514         if (m_state != stRunning)
515                 return -1;
516         
517         GstFormat fmt = GST_FORMAT_TIME;
518         gint64 len;
519         
520         if (!gst_element_query_position(m_gst_pipeline, &fmt, &len))
521                 return -1;
522         
523                 /* len is in nanoseconds. we have 90 000 pts per second. */
524         pts = len / 11111;
525         return 0;
526 }
527
528 RESULT eServiceMP3::setTrickmode(int trick)
529 {
530                 /* trickmode currently doesn't make any sense for us. */
531         return -1;
532 }
533
534 RESULT eServiceMP3::isCurrentlySeekable()
535 {
536         return 1;
537 }
538
539 RESULT eServiceMP3::info(ePtr<iServiceInformation>&i)
540 {
541         i = this;
542         return 0;
543 }
544
545 RESULT eServiceMP3::getName(std::string &name)
546 {
547         name = m_filename;
548         size_t n = name.rfind('/');
549         if (n != std::string::npos)
550                 name = name.substr(n + 1);
551         return 0;
552 }
553
554 int eServiceMP3::getInfo(int w)
555 {
556         gchar *tag = 0;
557
558         switch (w)
559         {
560         case sTitle:
561         case sArtist:
562         case sAlbum:
563         case sComment:
564         case sTracknumber:
565         case sGenre:
566         case sVideoType:
567                 return resIsString;
568         case sCurrentTitle:
569                 tag = GST_TAG_TRACK_NUMBER;
570                 break;
571         case sTotalTitles:
572                 tag = GST_TAG_TRACK_COUNT;
573                 break;
574         default:
575                 return resNA;
576         }
577
578         if (!m_stream_tags || !tag)
579                 return 0;
580         
581         guint value;
582         if (gst_tag_list_get_uint(m_stream_tags, tag, &value))
583                 return (int) value;
584         
585         return 0;
586
587 }
588
589 std::string eServiceMP3::getInfoString(int w)
590 {
591         gchar *tag = 0;
592         switch (w)
593         {
594         case sTitle:
595                 tag = GST_TAG_TITLE;
596                 break;
597         case sArtist:
598                 tag = GST_TAG_ARTIST;
599                 break;
600         case sAlbum:
601                 tag = GST_TAG_ALBUM;
602                 break;
603         case sComment:
604                 tag = GST_TAG_COMMENT;
605                 break;
606         case sTracknumber:
607                 tag = GST_TAG_TRACK_NUMBER;
608                 break;
609         case sGenre:
610                 tag = GST_TAG_GENRE;
611                 break;
612         case sVideoType:
613                 tag = GST_TAG_VIDEO_CODEC;
614                 break;
615         default:
616                 return "";
617         }
618         
619         if (!m_stream_tags || !tag)
620                 return "";
621         
622         gchar *value;
623         
624         if (gst_tag_list_get_string(m_stream_tags, tag, &value))
625         {
626                 std::string res = value;
627                 g_free(value);
628                 return res;
629         }
630         
631         return "";
632 }
633
634 RESULT eServiceMP3::audioChannel(ePtr<iAudioChannelSelection> &ptr)
635 {
636         ptr = this;
637         return 0;
638 }
639
640 RESULT eServiceMP3::audioTracks(ePtr<iAudioTrackSelection> &ptr)
641 {
642         ptr = this;
643         return 0;
644 }
645
646 int eServiceMP3::getNumberOfTracks()
647 {
648         return m_audioStreams.size();
649 }
650
651 int eServiceMP3::getCurrentTrack()
652 {
653         return m_currentAudioStream;
654 }
655
656 RESULT eServiceMP3::selectTrack(unsigned int i)
657 {
658         gint nb_sources;
659         GstPad *active_pad;
660         GstElement *selector = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_audio");
661         if ( !selector)
662         {
663                 eDebug("can't switch audio tracks! gst-plugin-selector needed");
664                 return -1;
665         }
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         if ( i >= m_audioStreams.size() || i >= nb_sources || m_currentAudioStream >= m_audioStreams.size() )
669                 return -2;
670         char sinkpad[8];
671         sprintf(sinkpad, "sink%d", i);
672         g_object_set (G_OBJECT (selector), "active-pad", gst_element_get_pad (selector, sinkpad), NULL);
673         g_object_get (G_OBJECT (selector), "active-pad", &active_pad, NULL);
674         gchar *name;
675         name = gst_pad_get_name (active_pad);
676         eDebug ("switched audio to (%s)", name);
677         g_free(name);
678         m_currentAudioStream = i;
679         return 0;
680 }
681
682 int eServiceMP3::getCurrentChannel()
683 {
684         return STEREO;
685 }
686
687 RESULT eServiceMP3::selectChannel(int i)
688 {
689         eDebug("eServiceMP3::selectChannel(%i)",i);
690         return 0;
691 }
692
693 RESULT eServiceMP3::getTrackInfo(struct iAudioTrackInfo &info, unsigned int i)
694 {
695 //      eDebug("eServiceMP3::getTrackInfo(&info, %i)",i);
696         if (i >= m_audioStreams.size())
697                 return -2;
698         if (m_audioStreams[i].type == audioStream::atMP2)
699                 info.m_description = "MP2";
700         else if (m_audioStreams[i].type == audioStream::atMP3)
701                 info.m_description = "MP3";
702         else if (m_audioStreams[i].type == audioStream::atAC3)
703                 info.m_description = "AC3";
704         else if (m_audioStreams[i].type == audioStream::atAAC)
705                 info.m_description = "AAC";
706         else  if (m_audioStreams[i].type == audioStream::atDTS)
707                 info.m_description = "DTS";
708         else
709                 info.m_description = "???";
710         if (info.m_language.empty())
711                 info.m_language = m_audioStreams[i].language_code;
712         return 0;
713 }
714
715 void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
716 {
717         if (!msg)
718                 return;
719         gchar *sourceName;
720         GstObject *source;
721
722         source = GST_MESSAGE_SRC(msg);
723         sourceName = gst_object_get_name(source);       
724
725         if (gst_message_get_structure(msg))
726         {
727                 gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
728                 eDebug("gst_message from %s: %s", sourceName, string);
729                 g_free(string);
730         }
731         else
732                 eDebug("gst_message from %s: %s (without structure)", sourceName, GST_MESSAGE_TYPE_NAME(msg));
733
734         switch (GST_MESSAGE_TYPE (msg))
735         {
736         case GST_MESSAGE_EOS:
737                 m_event((iPlayableService*)this, evEOF);
738                 break;
739         case GST_MESSAGE_ERROR:
740         {
741                 gchar *debug;
742                 GError *err;
743
744                 gst_message_parse_error (msg, &err, &debug);
745                 g_free (debug);
746                 eWarning("Gstreamer error: %s (%i)", err->message, err->code );
747                 if ( err->domain == GST_STREAM_ERROR && err->code == GST_STREAM_ERROR_DECODE )
748                 {
749                         if ( g_strrstr(sourceName, "videosink") )
750                                 m_event((iPlayableService*)this, evUser+11);
751                 }
752                 g_error_free(err);
753                         /* TODO: signal error condition to user */
754                 break;
755         }
756         case GST_MESSAGE_TAG:
757         {
758                 GstTagList *tags, *result;
759                 gst_message_parse_tag(msg, &tags);
760
761                 result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_PREPEND);
762                 if (result)
763                 {
764                         if (m_stream_tags)
765                                 gst_tag_list_free(m_stream_tags);
766                         m_stream_tags = result;
767                 }
768                 gchar *g_audiocodec;
769                 if (gst_tag_list_get_string(m_stream_tags, GST_TAG_AUDIO_CODEC, &g_audiocodec))
770                 {
771                         std::vector<audioStream>::iterator IterAudioStream = m_audioStreams.begin();
772                         while ( IterAudioStream->language_code.length() && IterAudioStream != m_audioStreams.end())
773                                 IterAudioStream++;
774                         if ( g_strrstr(g_audiocodec, "MPEG-1 layer 2") )
775                                 IterAudioStream->type = audioStream::atMP2;
776                         else if ( g_strrstr(g_audiocodec, "AC-3 audio") )
777                                 IterAudioStream->type = audioStream::atAC3;
778                         gchar *g_language;
779                         if ( gst_tag_list_get_string(m_stream_tags, GST_TAG_LANGUAGE_CODE, &g_language) )
780                                 IterAudioStream->language_code = std::string(g_language);
781                         g_free (g_language);
782                         g_free (g_audiocodec);
783                 }
784                 break;
785         }
786         default:
787                 break;
788         }
789         g_free (sourceName);
790 }
791
792 GstBusSyncReply eServiceMP3::gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data)
793 {
794         eServiceMP3 *_this = (eServiceMP3*)user_data;
795         _this->m_pump.send(1);
796                 /* wake */
797         return GST_BUS_PASS;
798 }
799
800 void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
801 {
802         eServiceMP3 *_this = (eServiceMP3*)user_data;
803         GstBin *pipeline = GST_BIN(_this->m_gst_pipeline);
804         gchar *name;
805         name = gst_pad_get_name (pad);
806         eDebug ("A new pad %s was created", name);
807         if (g_strrstr(name,"audio")) // mpegdemux, matroskademux, avidemux use video_nn with n=0,1,.., flupsdemux uses stream id
808         {
809                 GstElement *selector = gst_bin_get_by_name(pipeline , "switch_audio" );
810                 audioStream audio;
811                 audio.pad = pad;
812                 _this->m_audioStreams.push_back(audio);
813                 if ( selector )
814                 {
815                         GstPadLinkReturn ret = gst_pad_link(pad, gst_element_get_request_pad (selector, "sink%d"));
816                         if ( _this->m_audioStreams.size() == 1 )
817                         {
818                                 _this->selectTrack(0);
819                                 gst_element_set_state (_this->m_gst_pipeline, GST_STATE_PLAYING);
820                         }
821                         else
822                                 g_object_set (G_OBJECT (selector), "select-all", FALSE, NULL);
823                 }
824                 else
825                         gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_audio"), "sink"));
826         }
827         if (g_strrstr(name,"video"))
828         {
829                 gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_video"), "sink"));
830         }
831         g_free (name);
832 }
833
834 void eServiceMP3::gstCBfilterPadAdded(GstElement *filter, GstPad *pad, gpointer user_data)
835 {
836         eServiceMP3 *_this = (eServiceMP3*)user_data;
837         GstElement *decoder = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"decoder");
838         gst_pad_link(pad, gst_element_get_static_pad (decoder, "sink"));
839 }
840
841 void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last, gpointer user_data)
842 {
843         eServiceMP3 *_this = (eServiceMP3*)user_data;
844         GstCaps *caps;
845         GstStructure *str;
846         GstPad *audiopad;
847
848         /* only link once */
849         GstElement *audio = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"audiobin");
850         audiopad = gst_element_get_static_pad (audio, "sink");
851         if ( !audiopad || GST_PAD_IS_LINKED (audiopad)) {
852                 eDebug("audio already linked!");
853                 g_object_unref (audiopad);
854                 return;
855         }
856
857         /* check media type */
858         caps = gst_pad_get_caps (pad);
859         str = gst_caps_get_structure (caps, 0);
860         eDebug("gst new pad! %s", gst_structure_get_name (str));
861         
862         if (!g_strrstr (gst_structure_get_name (str), "audio")) {
863                 gst_caps_unref (caps);
864                 gst_object_unref (audiopad);
865                 return;
866         }
867         
868         gst_caps_unref (caps);
869         gst_pad_link (pad, audiopad);
870 }
871
872 void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
873 {
874         GstStructure *str;
875
876         /* check media type */
877         caps = gst_pad_get_caps (pad);
878         str = gst_caps_get_structure (caps, 0);
879         eDebug("unknown type: %s - this can't be decoded.", gst_structure_get_name (str));
880         gst_caps_unref (caps);
881 }
882
883 void eServiceMP3::gstPoll(const int&)
884 {
885                 /* ok, we have a serious problem here. gstBusSyncHandler sends 
886                    us the wakup signal, but likely before it was posted.
887                    the usleep, an EVIL HACK (DON'T DO THAT!!!) works around this.
888                    
889                    I need to understand the API a bit more to make this work 
890                    proplerly. */
891         usleep(1);
892         
893         GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline));
894         GstMessage *message;
895         while ((message = gst_bus_pop (bus)))
896         {
897                 gstBusCall(bus, message);
898                 gst_message_unref (message);
899         }
900 }
901
902 eAutoInitPtr<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
903 #else
904 #warning gstreamer not available, not building media player
905 #endif