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>
9 #include <lib/service/servicemp3.h>
10 #include <lib/service/service.h>
11 #include <lib/components/file_eraser.h>
12 #include <lib/base/init_num.h>
13 #include <lib/base/init.h>
17 #include <lib/gui/esubtitle.h>
21 eServiceFactoryMP3::eServiceFactoryMP3()
23 ePtr<eServiceCenter> sc;
25 eServiceCenter::getPrivInstance(sc);
28 std::list<std::string> extensions;
29 extensions.push_back("mp3");
30 extensions.push_back("ogg");
31 extensions.push_back("mpg");
32 extensions.push_back("vob");
33 extensions.push_back("wav");
34 extensions.push_back("wave");
35 extensions.push_back("mkv");
36 extensions.push_back("avi");
37 sc->addServiceFactory(eServiceFactoryMP3::id, this, extensions);
40 m_service_info = new eStaticServiceMP3Info();
43 eServiceFactoryMP3::~eServiceFactoryMP3()
45 ePtr<eServiceCenter> sc;
47 eServiceCenter::getPrivInstance(sc);
49 sc->removeServiceFactory(eServiceFactoryMP3::id);
52 DEFINE_REF(eServiceFactoryMP3)
55 RESULT eServiceFactoryMP3::play(const eServiceReference &ref, ePtr<iPlayableService> &ptr)
58 ptr = new eServiceMP3(ref.path.c_str());
62 RESULT eServiceFactoryMP3::record(const eServiceReference &ref, ePtr<iRecordableService> &ptr)
68 RESULT eServiceFactoryMP3::list(const eServiceReference &, ePtr<iListableService> &ptr)
74 RESULT eServiceFactoryMP3::info(const eServiceReference &ref, ePtr<iStaticServiceInformation> &ptr)
80 class eMP3ServiceOfflineOperations: public iServiceOfflineOperations
82 DECLARE_REF(eMP3ServiceOfflineOperations);
83 eServiceReference m_ref;
85 eMP3ServiceOfflineOperations(const eServiceReference &ref);
87 RESULT deleteFromDisk(int simulate);
88 RESULT getListOfFilenames(std::list<std::string> &);
91 DEFINE_REF(eMP3ServiceOfflineOperations);
93 eMP3ServiceOfflineOperations::eMP3ServiceOfflineOperations(const eServiceReference &ref): m_ref((const eServiceReference&)ref)
97 RESULT eMP3ServiceOfflineOperations::deleteFromDisk(int simulate)
103 std::list<std::string> res;
104 if (getListOfFilenames(res))
107 eBackgroundFileEraser *eraser = eBackgroundFileEraser::getInstance();
109 eDebug("FATAL !! can't get background file eraser");
111 for (std::list<std::string>::iterator i(res.begin()); i != res.end(); ++i)
113 eDebug("Removing %s...", i->c_str());
115 eraser->erase(i->c_str());
117 ::unlink(i->c_str());
124 RESULT eMP3ServiceOfflineOperations::getListOfFilenames(std::list<std::string> &res)
127 res.push_back(m_ref.path);
132 RESULT eServiceFactoryMP3::offlineOperations(const eServiceReference &ref, ePtr<iServiceOfflineOperations> &ptr)
134 ptr = new eMP3ServiceOfflineOperations(ref);
138 // eStaticServiceMP3Info
141 // eStaticServiceMP3Info is seperated from eServiceMP3 to give information
142 // about unopened files.
144 // probably eServiceMP3 should use this class as well, and eStaticServiceMP3Info
145 // should have a database backend where ID3-files etc. are cached.
146 // this would allow listing the mp3 database based on certain filters.
148 DEFINE_REF(eStaticServiceMP3Info)
150 eStaticServiceMP3Info::eStaticServiceMP3Info()
154 RESULT eStaticServiceMP3Info::getName(const eServiceReference &ref, std::string &name)
156 size_t last = ref.path.rfind('/');
157 if (last != std::string::npos)
158 name = ref.path.substr(last+1);
164 int eStaticServiceMP3Info::getLength(const eServiceReference &ref)
171 eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1)
174 m_audioStreams.clear();
175 m_subtitleStreams.clear();
176 m_currentAudioStream = 0;
177 m_currentSubtitleStream = 0;
178 m_subtitle_widget = 0;
179 m_currentTrickRatio = 0;
180 CONNECT(m_seekTimeout.timeout, eServiceMP3::seekTimeoutCB);
181 CONNECT(m_pump.recv_msg, eServiceMP3::gstPoll);
182 GstElement *source = 0;
184 GstElement *filter = 0, *decoder = 0, *conv = 0, *flt = 0, *sink = 0; /* for audio */
186 GstElement *audio = 0, *switch_audio = 0, *queue_audio = 0, *video = 0, *queue_video = 0, *videodemux = 0;
189 eDebug("SERVICEMP3 construct!");
191 /* FIXME: currently, decodebin isn't possible for
192 video streams. in that case, make a manual pipeline. */
194 const char *ext = strrchr(filename, '.');
198 int is_mpeg_ps = !(strcasecmp(ext, ".mpeg") && strcasecmp(ext, ".mpg") && strcasecmp(ext, ".vob") && strcasecmp(ext, ".bin") && strcasecmp(ext, ".dat"));
199 int is_mpeg_ts = !strcasecmp(ext, ".ts");
200 int is_matroska = !strcasecmp(ext, ".mkv");
201 int is_avi = !strcasecmp(ext, ".avi");
202 int is_mp3 = !strcasecmp(ext, ".mp3"); /* force mp3 instead of decodebin */
203 int is_video = is_mpeg_ps || is_mpeg_ts || is_matroska || is_avi;
204 int is_streaming = !strncmp(filename, "http://", 7);
205 int is_AudioCD = !(strncmp(filename, "/autofs/", 8) || strncmp(filename+strlen(filename)-13, "/track-", 7) || strcasecmp(ext, ".wav"));
207 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);
209 int is_audio = !is_video;
213 m_gst_pipeline = gst_pipeline_new ("mediaplayer");
215 eWarning("failed to create pipeline");
219 source = gst_element_factory_make ("cdiocddasrc", "cda-source");
221 g_object_set (G_OBJECT (source), "device", "/dev/cdroms/cdrom0", NULL);
225 if ( !is_streaming && !is_AudioCD )
226 source = gst_element_factory_make ("filesrc", "file-source");
227 else if ( is_streaming )
229 source = gst_element_factory_make ("neonhttpsrc", "http-source");
231 g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
235 eWarning("failed to create %s", is_streaming ? "neonhttpsrc" : "filesrc");
236 /* configure source */
237 else if (!is_AudioCD)
238 g_object_set (G_OBJECT (source), "location", filename, NULL);
241 int track = atoi(filename+18);
242 eDebug("play audio CD track #%i",track);
244 g_object_set (G_OBJECT (source), "track", track, NULL);
249 /* filesrc -> decodebin -> audioconvert -> capsfilter -> alsasink */
250 const char *decodertype = is_mp3 ? "mad" : "decodebin";
252 decoder = gst_element_factory_make (decodertype, "decoder");
254 eWarning("failed to create %s decoder", decodertype);
256 /* mp3 decoding needs id3demux to extract ID3 data. 'decodebin' would do that internally. */
259 filter = gst_element_factory_make ("id3demux", "filter");
261 eWarning("failed to create id3demux");
264 conv = gst_element_factory_make ("audioconvert", "converter");
266 eWarning("failed to create audioconvert");
268 flt = gst_element_factory_make ("capsfilter", "flt");
270 eWarning("failed to create capsfilter");
272 /* for some reasons, we need to set the sample format to depth/width=16, because auto negotiation doesn't work. */
273 /* endianness, however, is not required to be set anymore. */
276 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);
277 g_object_set (G_OBJECT (flt), "caps", caps, (char*)0);
278 gst_caps_unref(caps);
281 sink = gst_element_factory_make ("alsasink", "alsa-output");
283 eWarning("failed to create osssink");
285 if (source && decoder && conv && sink)
287 } else /* is_video */
289 /* filesrc -> mpegdemux -> | queue_audio -> dvbaudiosink
290 | queue_video -> dvbvideosink */
292 audio = gst_element_factory_make("dvbaudiosink", "audiosink");
293 queue_audio = gst_element_factory_make("queue", "queue_audio");
295 video = gst_element_factory_make("dvbvideosink", "videosink");
296 queue_video = gst_element_factory_make("queue", "queue_video");
299 videodemux = gst_element_factory_make("flupsdemux", "videodemux");
301 videodemux = gst_element_factory_make("flutsdemux", "videodemux");
302 else if (is_matroska)
303 videodemux = gst_element_factory_make("matroskademux", "videodemux");
305 videodemux = gst_element_factory_make("avidemux", "videodemux");
309 eDebug("fluendo mpegdemux not available, falling back to mpegdemux\n");
310 videodemux = gst_element_factory_make("mpegdemux", "videodemux");
313 eDebug("audio: %p, queue_audio %p, video %p, queue_video %p, videodemux %p", audio, queue_audio, video, queue_video, videodemux);
314 if (audio && queue_audio && video && queue_video && videodemux)
316 g_object_set (G_OBJECT (queue_audio), "max-size-bytes", 256*1024, NULL);
317 g_object_set (G_OBJECT (queue_audio), "max-size-buffers", 0, NULL);
318 g_object_set (G_OBJECT (queue_audio), "max-size-time", (guint64)0, NULL);
319 g_object_set (G_OBJECT (queue_video), "max-size-buffers", 0, NULL);
320 g_object_set (G_OBJECT (queue_video), "max-size-bytes", 2*1024*1024, NULL);
321 g_object_set (G_OBJECT (queue_video), "max-size-time", (guint64)0, NULL);
326 if (m_gst_pipeline && all_ok)
328 gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline)), gstBusSyncHandler, this);
332 queue_audio = gst_element_factory_make("queue", "queue_audio");
333 g_object_set (G_OBJECT (sink), "preroll-queue-len", 80, NULL);
334 gst_bin_add_many (GST_BIN (m_gst_pipeline), source, queue_audio, conv, sink, NULL);
335 gst_element_link_many(source, queue_audio, conv, sink, NULL);
339 queue_audio = gst_element_factory_make("queue", "queue_audio");
343 /* decodebin has dynamic pads. When they get created, we connect them to the audio bin */
344 g_signal_connect (decoder, "new-decoded-pad", G_CALLBACK(gstCBnewPad), this);
345 g_signal_connect (decoder, "unknown-type", G_CALLBACK(gstCBunknownType), this);
346 g_object_set (G_OBJECT (sink), "preroll-queue-len", 80, NULL);
349 /* gst_bin will take the 'floating references' */
350 gst_bin_add_many (GST_BIN (m_gst_pipeline),
351 source, queue_audio, decoder, NULL);
355 /* id3demux also has dynamic pads, which need to be connected to the decoder (this is done in the 'gstCBfilterPadAdded' CB) */
356 gst_bin_add(GST_BIN(m_gst_pipeline), filter);
357 gst_element_link(source, filter);
358 g_signal_connect (filter, "pad-added", G_CALLBACK(gstCBfilterPadAdded), this);
360 /* in decodebin's case we can just connect the source with the decodebin, and decodebin will take care about id3demux (or whatever is required) */
361 gst_element_link_many(source, queue_audio, decoder, NULL);
363 /* create audio bin with the audioconverter, the capsfilter and the audiosink */
364 audio = gst_bin_new ("audiobin");
366 GstPad *audiopad = gst_element_get_static_pad (conv, "sink");
367 gst_bin_add_many(GST_BIN(audio), conv, flt, sink, (char*)0);
368 gst_element_link_many(conv, flt, sink, (char*)0);
369 gst_element_add_pad(audio, gst_ghost_pad_new ("sink", audiopad));
370 gst_object_unref(audiopad);
371 gst_bin_add (GST_BIN(m_gst_pipeline), audio);
372 /* in mad's case, we can directly connect the decoder to the audiobin. otherwise, we do this in gstCBnewPad */
374 gst_element_link(decoder, audio);
375 audioStream audioStreamElem;
376 m_audioStreams.push_back(audioStreamElem);
377 } else /* is_video */
379 char srt_filename[strlen(filename)+1];
380 strncpy(srt_filename,filename,strlen(filename)-3);
381 srt_filename[strlen(filename)-3]='\0';
382 strcat(srt_filename, "srt");
384 if (stat(srt_filename, &buffer) == 0)
386 eDebug("subtitle file found: %s",srt_filename);
387 GstElement *subsource;
388 subsource = gst_element_factory_make ("filesrc", "srt_source");
389 g_object_set (G_OBJECT (subsource), "location", filename, NULL);
390 GstElement *parser = gst_element_factory_make("subparse", "srt_parse");
391 eDebug ("subparse = %p", parser);
392 GstElement *sink = gst_element_factory_make("fakesink", "srt_sink");
393 eDebug ("fakesink = %p", sink);
394 g_object_set (G_OBJECT(sink), "signal-handoffs", TRUE, NULL);
395 gst_bin_add_many(GST_BIN (m_gst_pipeline), subsource, parser, sink, NULL);
396 gboolean res = gst_element_link(subsource, parser);
397 eDebug ("parser link = %d", res);
398 res = gst_element_link(parser, sink);
399 eDebug ("sink link = %d", res);
400 g_signal_connect(sink, "handoff", G_CALLBACK(gstCBsubtitleAvail), this);
403 m_subtitleStreams.push_back(subs);
406 eDebug("subtitle file not found: %s",srt_filename);
408 gst_bin_add_many(GST_BIN(m_gst_pipeline), source, videodemux, audio, queue_audio, video, queue_video, NULL);
409 switch_audio = gst_element_factory_make ("input-selector", "switch_audio");
412 g_object_set (G_OBJECT (switch_audio), "select-all", TRUE, NULL);
413 gst_bin_add(GST_BIN(m_gst_pipeline), switch_audio);
414 gst_element_link(switch_audio, queue_audio);
416 gst_element_link(source, videodemux);
417 gst_element_link(queue_audio, audio);
418 gst_element_link(queue_video, video);
419 g_signal_connect(videodemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
424 gst_object_unref(GST_OBJECT(m_gst_pipeline));
426 gst_object_unref(GST_OBJECT(source));
428 gst_object_unref(GST_OBJECT(decoder));
430 gst_object_unref(GST_OBJECT(conv));
432 gst_object_unref(GST_OBJECT(sink));
435 gst_object_unref(GST_OBJECT(audio));
437 gst_object_unref(GST_OBJECT(queue_audio));
439 gst_object_unref(GST_OBJECT(video));
441 gst_object_unref(GST_OBJECT(queue_video));
443 gst_object_unref(GST_OBJECT(videodemux));
445 gst_object_unref(GST_OBJECT(switch_audio));
447 eDebug("sorry, can't play.");
451 gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
454 eServiceMP3::~eServiceMP3()
456 delete m_subtitle_widget;
457 if (m_state == stRunning)
461 gst_tag_list_free(m_stream_tags);
465 gst_object_unref (GST_OBJECT (m_gst_pipeline));
466 eDebug("SERVICEMP3 destruct!");
470 DEFINE_REF(eServiceMP3);
472 RESULT eServiceMP3::connectEvent(const Slot2<void,iPlayableService*,int> &event, ePtr<eConnection> &connection)
474 connection = new eConnection((iPlayableService*)this, m_event.connect(event));
478 RESULT eServiceMP3::start()
480 assert(m_state == stIdle);
485 eDebug("starting pipeline");
486 gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
488 m_event(this, evStart);
492 RESULT eServiceMP3::stop()
494 assert(m_state != stIdle);
495 if (m_state == stStopped)
497 eDebug("MP3: %s stop\n", m_filename.c_str());
498 gst_element_set_state(m_gst_pipeline, GST_STATE_NULL);
503 RESULT eServiceMP3::setTarget(int target)
508 RESULT eServiceMP3::pause(ePtr<iPauseableService> &ptr)
514 RESULT eServiceMP3::setSlowMotion(int ratio)
516 /* we can't do slomo yet */
520 RESULT eServiceMP3::setFastForward(int ratio)
522 m_currentTrickRatio = ratio;
524 m_seekTimeout.start(1000, 0);
526 m_seekTimeout.stop();
530 void eServiceMP3::seekTimeoutCB()
533 getPlayPosition(ppos);
535 ppos += 90000*m_currentTrickRatio;
540 m_seekTimeout.stop();
546 m_seekTimeout.stop();
553 RESULT eServiceMP3::pause()
557 GstStateChangeReturn res = gst_element_set_state(m_gst_pipeline, GST_STATE_PAUSED);
558 if (res == GST_STATE_CHANGE_ASYNC)
561 getPlayPosition(ppos);
567 RESULT eServiceMP3::unpause()
572 GstStateChangeReturn res;
573 res = gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING);
577 /* iSeekableService */
578 RESULT eServiceMP3::seek(ePtr<iSeekableService> &ptr)
584 RESULT eServiceMP3::getLength(pts_t &pts)
588 if (m_state != stRunning)
591 GstFormat fmt = GST_FORMAT_TIME;
594 if (!gst_element_query_duration(m_gst_pipeline, &fmt, &len))
597 /* len is in nanoseconds. we have 90 000 pts per second. */
603 RESULT eServiceMP3::seekTo(pts_t to)
608 /* convert pts to nanoseconds */
609 gint64 time_nanoseconds = to * 11111LL;
610 if (!gst_element_seek (m_gst_pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH,
611 GST_SEEK_TYPE_SET, time_nanoseconds,
612 GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE))
614 eDebug("SEEK failed");
620 RESULT eServiceMP3::seekRelative(int direction, pts_t to)
626 getPlayPosition(ppos);
627 ppos += to * direction;
635 RESULT eServiceMP3::getPlayPosition(pts_t &pts)
639 if (m_state != stRunning)
642 GstFormat fmt = GST_FORMAT_TIME;
645 if (!gst_element_query_position(m_gst_pipeline, &fmt, &len))
648 /* len is in nanoseconds. we have 90 000 pts per second. */
653 RESULT eServiceMP3::setTrickmode(int trick)
655 /* trickmode is not yet supported by our dvbmediasinks. */
659 RESULT eServiceMP3::isCurrentlySeekable()
664 RESULT eServiceMP3::info(ePtr<iServiceInformation>&i)
670 RESULT eServiceMP3::getName(std::string &name)
673 size_t n = name.rfind('/');
674 if (n != std::string::npos)
675 name = name.substr(n + 1);
679 int eServiceMP3::getInfo(int w)
694 tag = GST_TAG_TRACK_NUMBER;
697 tag = GST_TAG_TRACK_COUNT;
703 if (!m_stream_tags || !tag)
707 if (gst_tag_list_get_uint(m_stream_tags, tag, &value))
714 std::string eServiceMP3::getInfoString(int w)
723 tag = GST_TAG_ARTIST;
729 tag = GST_TAG_COMMENT;
732 tag = GST_TAG_TRACK_NUMBER;
738 tag = GST_TAG_VIDEO_CODEC;
744 if (!m_stream_tags || !tag)
749 if (gst_tag_list_get_string(m_stream_tags, tag, &value))
751 std::string res = value;
759 RESULT eServiceMP3::audioChannel(ePtr<iAudioChannelSelection> &ptr)
765 RESULT eServiceMP3::audioTracks(ePtr<iAudioTrackSelection> &ptr)
771 RESULT eServiceMP3::subtitle(ePtr<iSubtitleOutput> &ptr)
777 int eServiceMP3::getNumberOfTracks()
779 return m_audioStreams.size();
782 int eServiceMP3::getCurrentTrack()
784 return m_currentAudioStream;
787 RESULT eServiceMP3::selectTrack(unsigned int i)
789 int ret = selectAudioStream(i);
792 getPlayPosition(ppos);
798 int eServiceMP3::selectAudioStream(int i)
802 GstElement *selector = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_audio");
805 eDebug("can't switch audio tracks! gst-plugin-selector needed");
808 g_object_get (G_OBJECT (selector), "n-pads", &nb_sources, NULL);
809 if ( (unsigned int)i >= m_audioStreams.size() || i >= nb_sources || (unsigned int)m_currentAudioStream >= m_audioStreams.size() )
812 sprintf(sinkpad, "sink%d", i);
813 g_object_set (G_OBJECT (selector), "active-pad", gst_element_get_pad (selector, sinkpad), NULL);
814 g_object_get (G_OBJECT (selector), "active-pad", &active_pad, NULL);
816 name = gst_pad_get_name (active_pad);
817 eDebug ("switched audio to (%s)", name);
819 m_currentAudioStream = i;
823 int eServiceMP3::getCurrentChannel()
828 RESULT eServiceMP3::selectChannel(int i)
830 eDebug("eServiceMP3::selectChannel(%i)",i);
834 RESULT eServiceMP3::getTrackInfo(struct iAudioTrackInfo &info, unsigned int i)
836 // eDebug("eServiceMP3::getTrackInfo(&info, %i)",i);
837 if (i >= m_audioStreams.size())
839 if (m_audioStreams[i].type == audioStream::atMP2)
840 info.m_description = "MP2";
841 else if (m_audioStreams[i].type == audioStream::atMP3)
842 info.m_description = "MP3";
843 else if (m_audioStreams[i].type == audioStream::atAC3)
844 info.m_description = "AC3";
845 else if (m_audioStreams[i].type == audioStream::atAAC)
846 info.m_description = "AAC";
847 else if (m_audioStreams[i].type == audioStream::atDTS)
848 info.m_description = "DTS";
849 else if (m_audioStreams[i].type == audioStream::atPCM)
850 info.m_description = "PCM";
851 else if (m_audioStreams[i].type == audioStream::atOGG)
852 info.m_description = "OGG";
854 info.m_description = "???";
855 if (info.m_language.empty())
856 info.m_language = m_audioStreams[i].language_code;
860 void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
867 source = GST_MESSAGE_SRC(msg);
868 sourceName = gst_object_get_name(source);
870 if (gst_message_get_structure(msg))
872 gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
873 eDebug("gst_message from %s: %s", sourceName, string);
877 eDebug("gst_message from %s: %s (without structure)", sourceName, GST_MESSAGE_TYPE_NAME(msg));
879 switch (GST_MESSAGE_TYPE (msg))
881 case GST_MESSAGE_EOS:
882 m_event((iPlayableService*)this, evEOF);
884 case GST_MESSAGE_ERROR:
889 gst_message_parse_error (msg, &err, &debug);
891 eWarning("Gstreamer error: %s (%i)", err->message, err->code );
892 if ( err->domain == GST_STREAM_ERROR && err->code == GST_STREAM_ERROR_DECODE )
894 if ( g_strrstr(sourceName, "videosink") )
895 m_event((iPlayableService*)this, evUser+11);
898 /* TODO: signal error condition to user */
901 case GST_MESSAGE_TAG:
903 GstTagList *tags, *result;
904 gst_message_parse_tag(msg, &tags);
906 result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_PREPEND);
910 gst_tag_list_free(m_stream_tags);
911 m_stream_tags = result;
914 if (gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_audiocodec) && m_audioStreams.size())
916 std::vector<audioStream>::iterator IterAudioStream = m_audioStreams.begin();
917 while ( IterAudioStream != m_audioStreams.end() && (!IterAudioStream->language_code.empty() || IterAudioStream->type != audioStream::atUnknown))
919 if ( g_strrstr(g_audiocodec, "MPEG-1 layer 2") )
920 IterAudioStream->type = audioStream::atMP2;
921 else if ( g_strrstr(g_audiocodec, "MPEG-1 layer 3") )
922 IterAudioStream->type = audioStream::atMP3;
923 else if ( g_strrstr(g_audiocodec, "AAC audio") ) // dont checked if correct
924 IterAudioStream->type = audioStream::atAAC;
925 else if ( g_strrstr(g_audiocodec, "DTS audio") )
926 IterAudioStream->type = audioStream::atDTS;
927 else if ( g_strrstr(g_audiocodec, "AC-3 audio") )
928 IterAudioStream->type = audioStream::atAC3;
929 else if ( g_strrstr(g_audiocodec, "Uncompressed 16-bit PCM audio") )
930 IterAudioStream->type = audioStream::atPCM;
932 eDebug("unknown audiocodec '%s'!", g_audiocodec);
934 if ( gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_language) )
935 IterAudioStream->language_code = std::string(g_language);
937 g_free (g_audiocodec);
947 GstBusSyncReply eServiceMP3::gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data)
949 eServiceMP3 *_this = (eServiceMP3*)user_data;
950 _this->m_pump.send(1);
955 void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
957 eServiceMP3 *_this = (eServiceMP3*)user_data;
958 GstBin *pipeline = GST_BIN(_this->m_gst_pipeline);
960 name = gst_pad_get_name (pad);
961 eDebug ("A new pad %s was created", name);
962 if (g_strrstr(name,"audio")) // mpegdemux, matroskademux, avidemux use video_nn with n=0,1,.., flupsdemux uses stream id
964 GstElement *selector = gst_bin_get_by_name(pipeline , "switch_audio" );
967 _this->m_audioStreams.push_back(audio);
970 gst_pad_link(pad, gst_element_get_request_pad (selector, "sink%d"));
971 if ( _this->m_audioStreams.size() == 1 )
973 _this->selectAudioStream(0);
974 gst_element_set_state (_this->m_gst_pipeline, GST_STATE_PLAYING);
977 g_object_set (G_OBJECT (selector), "select-all", FALSE, NULL);
980 gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_audio"), "sink"));
982 if (g_strrstr(name,"video"))
984 gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_video"), "sink"));
986 if (g_strrstr(name,"subtitle"))
989 // const GstStructure *structure;
990 // caps = gst_pad_get_caps(name);
991 // structure = gst_caps_get_structure(caps, 0);
993 sprintf(elemname, "%s_pars", name);
994 GstElement *parser = gst_element_factory_make("ssaparse", elemname);
995 eDebug ("ssaparse %s = %p", elemname, parser);
996 sprintf(elemname, "%s_sink", name);
997 GstElement *sink = gst_element_factory_make("fakesink", elemname);
998 eDebug ("fakesink %s = %p", elemname, sink);
999 g_object_set (G_OBJECT(sink), "signal-handoffs", TRUE, NULL);
1000 gst_bin_add_many(pipeline, parser, sink, NULL);
1001 gboolean res = gst_pad_link(pad, gst_element_get_static_pad(parser, "sink"));
1002 eDebug ("parser link = %d", res);
1003 res = gst_element_link(parser, sink);
1004 eDebug ("sink link = %d", res);
1005 g_signal_connect(sink, "handoff", G_CALLBACK(gstCBsubtitleAvail), _this);
1006 subtitleStream subs;
1007 subs.element = sink;
1008 _this->m_subtitleStreams.push_back(subs);
1013 void eServiceMP3::gstCBfilterPadAdded(GstElement *filter, GstPad *pad, gpointer user_data)
1015 eServiceMP3 *_this = (eServiceMP3*)user_data;
1016 GstElement *decoder = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"decoder");
1017 gst_pad_link(pad, gst_element_get_static_pad (decoder, "sink"));
1020 void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last, gpointer user_data)
1022 eServiceMP3 *_this = (eServiceMP3*)user_data;
1027 /* only link once */
1028 GstElement *audio = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"audiobin");
1029 audiopad = gst_element_get_static_pad (audio, "sink");
1030 if ( !audiopad || GST_PAD_IS_LINKED (audiopad)) {
1031 eDebug("audio already linked!");
1032 g_object_unref (audiopad);
1036 /* check media type */
1037 caps = gst_pad_get_caps (pad);
1038 str = gst_caps_get_structure (caps, 0);
1039 eDebug("gst new pad! %s", gst_structure_get_name (str));
1041 if (!g_strrstr (gst_structure_get_name (str), "audio")) {
1042 gst_caps_unref (caps);
1043 gst_object_unref (audiopad);
1047 gst_caps_unref (caps);
1048 gst_pad_link (pad, audiopad);
1051 void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
1055 /* check media type */
1056 caps = gst_pad_get_caps (pad);
1057 str = gst_caps_get_structure (caps, 0);
1058 eDebug("unknown type: %s - this can't be decoded.", gst_structure_get_name (str));
1059 gst_caps_unref (caps);
1062 void eServiceMP3::gstPoll(const int&)
1064 /* ok, we have a serious problem here. gstBusSyncHandler sends
1065 us the wakup signal, but likely before it was posted.
1066 the usleep, an EVIL HACK (DON'T DO THAT!!!) works around this.
1068 I need to understand the API a bit more to make this work
1072 GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline));
1073 GstMessage *message;
1074 while ((message = gst_bus_pop (bus)))
1076 gstBusCall(bus, message);
1077 gst_message_unref (message);
1081 eAutoInitPtr<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
1083 #warning gstreamer not available, not building media player
1086 void eServiceMP3::gstCBsubtitleAvail(GstElement *element, GstBuffer *buffer, GstPad *pad, gpointer user_data)
1088 const unsigned char *text = (unsigned char *)GST_BUFFER_DATA(buffer);
1089 eServiceMP3 *_this = (eServiceMP3*)user_data;
1091 sourceName = gst_object_get_name(GST_OBJECT(element));
1092 if ( _this->m_subtitle_widget && _this->m_subtitleStreams.at(_this->m_currentSubtitleStream).element == element)
1094 eDVBTeletextSubtitlePage page;
1095 gRGB rgbcol(0xD0,0xD0,0xD0);
1096 page.m_elements.push_back(eDVBTeletextSubtitlePageElement(rgbcol, (const char*)text));
1097 (_this->m_subtitle_widget)->setPage(page);
1100 eDebug("on inactive element: %s (%p) saw subtitle: %s",sourceName, element, (const char*)text);
1103 RESULT eServiceMP3::enableSubtitles(eWidget *parent, ePyObject tuple)
1105 eDebug("eServiceMP3::enableSubtitles");
1108 int tuplesize = PyTuple_Size(tuple);
1112 if (!PyTuple_Check(tuple))
1118 entry = PyTuple_GET_ITEM(tuple, 0);
1120 if (!PyInt_Check(entry))
1123 type = PyInt_AsLong(entry);
1125 entry = PyTuple_GET_ITEM(tuple, 1);
1126 if (!PyInt_Check(entry))
1128 pid = PyInt_AsLong(entry);
1130 m_subtitle_widget = new eSubtitleWidget(parent);
1131 m_subtitle_widget->resize(parent->size()); /* full size */
1132 m_currentSubtitleStream = pid;
1136 eDebug("enableSubtitles needs a tuple as 2nd argument!\n"
1137 "for gst subtitles (2, subtitle_stream_count)");
1141 RESULT eServiceMP3::disableSubtitles(eWidget *parent)
1143 eDebug("eServiceMP3::disableSubtitles");
1144 delete m_subtitle_widget;
1145 m_subtitle_widget = 0;
1149 PyObject *eServiceMP3::getCachedSubtitle()
1151 eDebug("eServiceMP3::eDVBServicePlay");
1155 PyObject *eServiceMP3::getSubtitleList()
1157 eDebug("eServiceMP3::getSubtitleList");
1159 ePyObject l = PyList_New(0);
1161 int stream_count = 0;
1163 for (std::vector<subtitleStream>::iterator IterSubtitleStream(m_subtitleStreams.begin()); IterSubtitleStream != m_subtitleStreams.end(); ++IterSubtitleStream)
1165 ePyObject tuple = PyTuple_New(5);
1166 PyTuple_SET_ITEM(tuple, 0, PyInt_FromLong(2));
1167 PyTuple_SET_ITEM(tuple, 1, PyInt_FromLong(stream_count));
1168 PyTuple_SET_ITEM(tuple, 2, PyInt_FromLong(0));
1169 PyTuple_SET_ITEM(tuple, 3, PyInt_FromLong(0));
1170 sourceName = gst_object_get_name(GST_OBJECT (IterSubtitleStream->element));
1171 PyTuple_SET_ITEM(tuple, 4, PyString_FromString(sourceName));
1172 PyList_Append(l, tuple);