X-Git-Url: https://git.cweiske.de/enigma2.git/blobdiff_plain/d63d2c3c6cbbd574dda4f8b00ebe6c661735edd5..f14fe40f1fc2d6ba0c993bc0dabbfe250428aaa1:/lib/service/servicemp3.cpp

diff --git a/lib/service/servicemp3.cpp b/lib/service/servicemp3.cpp
index fb5993e4..86b7696f 100644
--- a/lib/service/servicemp3.cpp
+++ b/lib/service/servicemp3.cpp
@@ -1,27 +1,57 @@
+#ifdef HAVE_GSTREAMER
+
+	/* note: this requires gstreamer 0.10.x and a big list of plugins. */
+	/* it's currently hardcoded to use a big-endian alsasink as sink. */
 #include <lib/base/eerror.h>
 #include <lib/base/object.h>
+#include <lib/base/ebase.h>
 #include <string>
 #include <lib/service/servicemp3.h>
 #include <lib/service/service.h>
+#include <lib/components/file_eraser.h>
 #include <lib/base/init_num.h>
 #include <lib/base/init.h>
+#include <gst/gst.h>
+#include <gst/pbutils/missing-plugins.h>
+#include <sys/stat.h>
+/* for subtitles */
+#include <lib/gui/esubtitle.h>
 
 // eServiceFactoryMP3
 
-eServiceFactoryMP3::eServiceFactoryMP3(): ref(0)
+eServiceFactoryMP3::eServiceFactoryMP3()
 {
 	ePtr<eServiceCenter> sc;
 	
-	eServiceCenter::getInstance(sc);
+	eServiceCenter::getPrivInstance(sc);
 	if (sc)
-		sc->addServiceFactory(eServiceFactoryMP3::id, this);
+	{
+		std::list<std::string> extensions;
+		extensions.push_back("mp2");
+		extensions.push_back("mp3");
+		extensions.push_back("ogg");
+		extensions.push_back("mpg");
+		extensions.push_back("vob");
+		extensions.push_back("wav");
+		extensions.push_back("wave");
+		extensions.push_back("mkv");
+		extensions.push_back("avi");
+		extensions.push_back("divx");
+		extensions.push_back("dat");
+		extensions.push_back("flac");
+		extensions.push_back("mp4");
+		extensions.push_back("m4a");
+		sc->addServiceFactory(eServiceFactoryMP3::id, this, extensions);
+	}
+
+	m_service_info = new eStaticServiceMP3Info();
 }
 
 eServiceFactoryMP3::~eServiceFactoryMP3()
 {
 	ePtr<eServiceCenter> sc;
 	
-	eServiceCenter::getInstance(sc);
+	eServiceCenter::getPrivInstance(sc);
 	if (sc)
 		sc->removeServiceFactory(eServiceFactoryMP3::id);
 }
@@ -31,15 +61,8 @@ DEFINE_REF(eServiceFactoryMP3)
 	// iServiceHandler
 RESULT eServiceFactoryMP3::play(const eServiceReference &ref, ePtr<iPlayableService> &ptr)
 {
-	RESULT res;
 		// check resources...
 	ptr = new eServiceMP3(ref.path.c_str());
-	res = ptr->start();
-	if (res)
-	{
-		ptr = 0;
-		return res;
-	}
 	return 0;
 }
 
@@ -55,35 +78,1466 @@ RESULT eServiceFactoryMP3::list(const eServiceReference &, ePtr<iListableService
 	return -1;
 }
 
+RESULT eServiceFactoryMP3::info(const eServiceReference &ref, ePtr<iStaticServiceInformation> &ptr)
+{
+	ptr = m_service_info;
+	return 0;
+}
+
+class eMP3ServiceOfflineOperations: public iServiceOfflineOperations
+{
+	DECLARE_REF(eMP3ServiceOfflineOperations);
+	eServiceReference m_ref;
+public:
+	eMP3ServiceOfflineOperations(const eServiceReference &ref);
+	
+	RESULT deleteFromDisk(int simulate);
+	RESULT getListOfFilenames(std::list<std::string> &);
+};
+
+DEFINE_REF(eMP3ServiceOfflineOperations);
+
+eMP3ServiceOfflineOperations::eMP3ServiceOfflineOperations(const eServiceReference &ref): m_ref((const eServiceReference&)ref)
+{
+}
+
+RESULT eMP3ServiceOfflineOperations::deleteFromDisk(int simulate)
+{
+	if (simulate)
+		return 0;
+	else
+	{
+		std::list<std::string> res;
+		if (getListOfFilenames(res))
+			return -1;
+		
+		eBackgroundFileEraser *eraser = eBackgroundFileEraser::getInstance();
+		if (!eraser)
+			eDebug("FATAL !! can't get background file eraser");
+		
+		for (std::list<std::string>::iterator i(res.begin()); i != res.end(); ++i)
+		{
+			eDebug("Removing %s...", i->c_str());
+			if (eraser)
+				eraser->erase(i->c_str());
+			else
+				::unlink(i->c_str());
+		}
+		
+		return 0;
+	}
+}
+
+RESULT eMP3ServiceOfflineOperations::getListOfFilenames(std::list<std::string> &res)
+{
+	res.clear();
+	res.push_back(m_ref.path);
+	return 0;
+}
+
+
+RESULT eServiceFactoryMP3::offlineOperations(const eServiceReference &ref, ePtr<iServiceOfflineOperations> &ptr)
+{
+	ptr = new eMP3ServiceOfflineOperations(ref);
+	return 0;
+}
+
+// eStaticServiceMP3Info
+
+
+// eStaticServiceMP3Info is seperated from eServiceMP3 to give information
+// about unopened files.
+
+// probably eServiceMP3 should use this class as well, and eStaticServiceMP3Info
+// should have a database backend where ID3-files etc. are cached.
+// this would allow listing the mp3 database based on certain filters.
+
+DEFINE_REF(eStaticServiceMP3Info)
+
+eStaticServiceMP3Info::eStaticServiceMP3Info()
+{
+}
+
+RESULT eStaticServiceMP3Info::getName(const eServiceReference &ref, std::string &name)
+{
+	size_t last = ref.path.rfind('/');
+	if (last != std::string::npos)
+		name = ref.path.substr(last+1);
+	else
+		name = ref.path;
+	return 0;
+}
+
+int eStaticServiceMP3Info::getLength(const eServiceReference &ref)
+{
+	return -1;
+}
+
 // eServiceMP3
 
-eServiceMP3::eServiceMP3(const char *filename): filename(filename), ref(0)
+eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1)
 {
-	printf("MP3: %s start\n", filename);
+	m_seekTimeout = eTimer::create(eApp);
+	m_stream_tags = 0;
+	m_currentAudioStream = 0;
+	m_currentSubtitleStream = 0;
+	m_subtitle_widget = 0;
+	m_currentTrickRatio = 0;
+	CONNECT(m_seekTimeout->timeout, eServiceMP3::seekTimeoutCB);
+	CONNECT(m_pump.recv_msg, eServiceMP3::gstPoll);
+	GstElement *source = 0;
+	GstElement *decoder = 0, *conv = 0, *flt = 0, *parser = 0, *sink = 0; /* for audio */
+	GstElement *audio = 0, *switch_audio = 0, *queue_audio = 0, *video = 0, *queue_video = 0, *videodemux = 0, *audiodemux = 0, *id3demux;
+	
+	m_state = stIdle;
+	eDebug("SERVICEMP3 construct!");
+	
+		/* FIXME: currently, decodebin isn't possible for 
+		   video streams. in that case, make a manual pipeline. */
+
+	const char *ext = strrchr(filename, '.');
+	if (!ext)
+		ext = filename;
+
+	sourceStream sourceinfo;
+	sourceinfo.is_video = FALSE;
+	sourceinfo.audiotype = atUnknown;
+	if ( (strcasecmp(ext, ".mpeg") && strcasecmp(ext, ".mpg") && strcasecmp(ext, ".vob") && strcasecmp(ext, ".bin") && strcasecmp(ext, ".dat") ) == 0 )
+	{
+		sourceinfo.containertype = ctMPEGPS;
+		sourceinfo.is_video = TRUE;
+	}
+	else if ( strcasecmp(ext, ".ts") == 0 )
+	{
+		sourceinfo.containertype = ctMPEGTS;
+		sourceinfo.is_video = TRUE;
+	}
+	else if ( strcasecmp(ext, ".mkv") == 0 )
+	{
+		sourceinfo.containertype = ctMKV;
+		sourceinfo.is_video = TRUE;
+	}
+	else if ( strcasecmp(ext, ".avi") == 0 || strcasecmp(ext, ".divx") == 0)
+	{
+		sourceinfo.containertype = ctAVI;
+		sourceinfo.is_video = TRUE;
+	}
+	else if ( strcasecmp(ext, ".mp4") == 0 )
+	{
+		sourceinfo.containertype = ctMP4;
+		sourceinfo.is_video = TRUE;
+	}
+	else if ( strcasecmp(ext, ".m4a") == 0 )
+	{
+		sourceinfo.containertype = ctMP4;
+		sourceinfo.audiotype = atAAC;
+	}
+	else if ( strcasecmp(ext, ".mp3") == 0 )
+		sourceinfo.audiotype = atMP3;
+	else if ( (strncmp(filename, "/autofs/", 8) || strncmp(filename+strlen(filename)-13, "/track-", 7) || strcasecmp(ext, ".wav")) == 0 )
+		sourceinfo.containertype = ctCDA;
+	if ( strcasecmp(ext, ".dat") == 0 )
+	{
+		sourceinfo.containertype = ctVCD;
+		sourceinfo.is_video = TRUE;
+	}
+	if ( (strncmp(filename, "http://", 7)) == 0 )
+		sourceinfo.is_streaming = TRUE;
+
+	eDebug("filename=%s, containertype=%d, is_video=%d, is_streaming=%d", filename, sourceinfo.containertype, sourceinfo.is_video, sourceinfo.is_streaming);
+
+	int all_ok = 0;
+
+	m_gst_pipeline = gst_pipeline_new ("mediaplayer");
+	if (!m_gst_pipeline)
+		m_error_message = "failed to create GStreamer pipeline!\n";
+
+	if ( sourceinfo.is_streaming )
+	{
+		eDebug("play webradio!");
+		source = gst_element_factory_make ("neonhttpsrc", "http-source");
+		if (source)
+		{
+			g_object_set (G_OBJECT (source), "location", filename, NULL);
+			g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
+		}
+		else
+			m_error_message = "GStreamer plugin neonhttpsrc not available!\n";
+	}
+	else if ( sourceinfo.containertype == ctCDA )
+	{
+		source = gst_element_factory_make ("cdiocddasrc", "cda-source");
+		if (source)
+		{
+			g_object_set (G_OBJECT (source), "device", "/dev/cdroms/cdrom0", NULL);
+			int track = atoi(filename+18);
+			eDebug("play audio CD track #%i",track);
+			if (track > 0)
+				g_object_set (G_OBJECT (source), "track", track, NULL);
+		}
+	}
+	else if ( sourceinfo.containertype == ctVCD )
+	{
+		int fd = open(filename,O_RDONLY);
+		char tmp[128*1024];
+		int ret = read(fd, tmp, 128*1024);
+		close(fd);
+		if ( ret == -1 ) // this is a "REAL" VCD
+			source = gst_element_factory_make ("vcdsrc", "vcd-source");
+			if (source)
+				g_object_set (G_OBJECT (source), "device", "/dev/cdroms/cdrom0", NULL);
+	}
+	if ( !source && !sourceinfo.is_streaming )
+	{
+		source = gst_element_factory_make ("filesrc", "file-source");
+		if (source)
+			g_object_set (G_OBJECT (source), "location", filename, NULL);
+		else
+			m_error_message = "GStreamer can't open filesrc " + (std::string)filename + "!\n";
+	}
+	if ( sourceinfo.is_video )
+	{
+			/* filesrc -> mpegdemux -> | queue_audio -> dvbaudiosink
+			                           | queue_video -> dvbvideosink */
+
+		audio = gst_element_factory_make("dvbaudiosink", "audiosink");
+		if (!audio)
+			m_error_message += "failed to create Gstreamer element dvbaudiosink\n";
+
+		video = gst_element_factory_make("dvbvideosink", "videosink");
+		if (!video)
+			m_error_message += "failed to create Gstreamer element dvbvideosink\n";
+
+		queue_audio = gst_element_factory_make("queue", "queue_audio");
+		queue_video = gst_element_factory_make("queue", "queue_video");
+
+		std::string demux_type;
+		switch (sourceinfo.containertype)
+		{
+			case ctMPEGTS:
+				demux_type = "flutsdemux";
+				break;
+			case ctMPEGPS:
+			case ctVCD:
+				demux_type = "flupsdemux";
+				break;
+			case ctMKV:
+				demux_type = "matroskademux";
+				break;
+			case ctAVI:
+				demux_type = "avidemux";
+				break;
+			case ctMP4:
+				demux_type = "qtdemux";
+				break;
+			default:
+				break;
+		}
+		videodemux = gst_element_factory_make(demux_type.c_str(), "videodemux");
+		if (!videodemux)
+			m_error_message = "GStreamer plugin " + demux_type + " not available!\n";
+
+		switch_audio = gst_element_factory_make ("input-selector", "switch_audio");
+		if (!switch_audio)
+			m_error_message = "GStreamer plugin input-selector not available!\n";
+
+		if (audio && queue_audio && video && queue_video && videodemux && switch_audio)
+		{
+			g_object_set (G_OBJECT (queue_audio), "max-size-bytes", 256*1024, NULL);
+			g_object_set (G_OBJECT (queue_audio), "max-size-buffers", 0, NULL);
+			g_object_set (G_OBJECT (queue_audio), "max-size-time", (guint64)0, NULL);
+			g_object_set (G_OBJECT (queue_video), "max-size-buffers", 0, NULL);
+			g_object_set (G_OBJECT (queue_video), "max-size-bytes", 2*1024*1024, NULL);
+			g_object_set (G_OBJECT (queue_video), "max-size-time", (guint64)0, NULL);
+			g_object_set (G_OBJECT (switch_audio), "select-all", TRUE, NULL);
+			all_ok = 1;
+		}
+	} else /* is audio */
+	{
+		std::string demux_type;
+		switch ( sourceinfo.containertype )
+		{
+			case ctMP4:
+				demux_type = "qtdemux";
+				break;
+			default:
+				break;
+		}
+		if ( demux_type.length() )
+		{
+			audiodemux = gst_element_factory_make(demux_type.c_str(), "audiodemux");
+			if (!audiodemux)
+				m_error_message = "GStreamer plugin " + demux_type + " not available!\n";
+		}
+		switch ( sourceinfo.audiotype )
+		{
+			case atMP3:
+			{
+				id3demux = gst_element_factory_make("id3demux", "id3demux");
+				if ( !id3demux )
+				{
+					m_error_message += "failed to create Gstreamer element id3demux\n";
+					break;
+				}
+				parser = gst_element_factory_make("mp3parse", "audiosink");
+				if ( !parser )
+				{
+					m_error_message += "failed to create Gstreamer element mp3parse\n";
+					break;
+				}
+				sink = gst_element_factory_make("dvbaudiosink", "audiosink2");
+				if ( !sink )
+					m_error_message += "failed to create Gstreamer element dvbaudiosink\n";
+				else
+					all_ok = 1;
+				break;
+			}
+			case atAAC:
+			{
+				if ( !audiodemux )
+				{
+					m_error_message += "cannot parse raw AAC audio\n";
+					break;
+				}
+				sink = gst_element_factory_make("dvbaudiosink", "audiosink");
+				if (!sink)
+					m_error_message += "failed to create Gstreamer element dvbaudiosink\n";
+				else
+					all_ok = 1;
+				break;
+			}
+			case atAC3:
+			{
+				if ( !audiodemux )
+				{
+					m_error_message += "cannot parse raw AC3 audio\n";
+					break;
+				}
+				sink = gst_element_factory_make("dvbaudiosink", "audiosink");
+				if ( !sink )
+					m_error_message += "failed to create Gstreamer element dvbaudiosink\n";
+				else
+					all_ok = 1;
+				break;
+			}
+			default:
+			{	/* filesrc -> decodebin -> audioconvert -> capsfilter -> alsasink */
+				decoder = gst_element_factory_make ("decodebin", "decoder");
+				if (!decoder)
+					m_error_message += "failed to create Gstreamer element decodebin\n";
+		
+				conv = gst_element_factory_make ("audioconvert", "converter");
+				if (!conv)
+					m_error_message += "failed to create Gstreamer element audioconvert\n";
+		
+				flt = gst_element_factory_make ("capsfilter", "flt");
+				if (!flt)
+					m_error_message += "failed to create Gstreamer element capsfilter\n";
+		
+					/* for some reasons, we need to set the sample format to depth/width=16, because auto negotiation doesn't work. */
+					/* endianness, however, is not required to be set anymore. */
+				if (flt)
+				{
+					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, */NULL);
+					g_object_set (G_OBJECT (flt), "caps", caps, NULL);
+					gst_caps_unref(caps);
+				}
+		
+				sink = gst_element_factory_make ("alsasink", "alsa-output");
+				if (!sink)
+					m_error_message += "failed to create Gstreamer element alsasink\n";
+		
+				if (source && decoder && conv && sink)
+					all_ok = 1;
+				break;
+			}
+		}
+
+	}
+	if (m_gst_pipeline && all_ok)
+	{
+		gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline)), gstBusSyncHandler, this);
+
+		if ( sourceinfo.containertype == ctCDA )
+		{
+			queue_audio = gst_element_factory_make("queue", "queue_audio");
+			g_object_set (G_OBJECT (sink), "preroll-queue-len", 80, NULL);
+			gst_bin_add_many (GST_BIN (m_gst_pipeline), source, queue_audio, conv, sink, NULL);
+			gst_element_link_many(source, queue_audio, conv, sink, NULL);
+		}
+		else if ( sourceinfo.is_video )
+		{
+			char srt_filename[strlen(filename)+1];
+			strncpy(srt_filename,filename,strlen(filename)-3);
+			srt_filename[strlen(filename)-3]='\0';
+			strcat(srt_filename, "srt");
+			struct stat buffer;
+			if (stat(srt_filename, &buffer) == 0)
+			{
+				eDebug("subtitle file found: %s",srt_filename);
+				GstElement *subsource = gst_element_factory_make ("filesrc", "srt_source");
+				g_object_set (G_OBJECT (subsource), "location", srt_filename, NULL);
+				gst_bin_add(GST_BIN (m_gst_pipeline), subsource);
+				GstPad *switchpad = gstCreateSubtitleSink(this, stSRT);
+				gst_pad_link(gst_element_get_pad (subsource, "src"), switchpad);
+				subtitleStream subs;
+				subs.pad = switchpad;
+				subs.type = stSRT;
+				subs.language_code = std::string("und");
+				m_subtitleStreams.push_back(subs);
+			}
+			gst_bin_add_many(GST_BIN(m_gst_pipeline), source, videodemux, audio, queue_audio, video, queue_video, switch_audio, NULL);
+
+			if ( sourceinfo.containertype == ctVCD && gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"file-source") )
+			{
+				eDebug("this is a fake video cd... we use filesrc ! cdxaparse !");
+				GstElement *cdxaparse = gst_element_factory_make("cdxaparse", "cdxaparse");
+				gst_bin_add(GST_BIN(m_gst_pipeline), cdxaparse);
+				gst_element_link(source, cdxaparse);
+				gst_element_link(cdxaparse, videodemux);
+			}
+			else
+				gst_element_link(source, videodemux);
+
+			gst_element_link(switch_audio, queue_audio);
+			gst_element_link(queue_audio, audio);
+			gst_element_link(queue_video, video);
+			g_signal_connect(videodemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
+
+		} else /* is audio*/
+		{
+			if ( decoder )
+			{
+				queue_audio = gst_element_factory_make("queue", "queue_audio");
+	
+				g_signal_connect (decoder, "new-decoded-pad", G_CALLBACK(gstCBnewPad), this);
+				g_signal_connect (decoder, "unknown-type", G_CALLBACK(gstCBunknownType), this);
+	
+				g_object_set (G_OBJECT (sink), "preroll-queue-len", 80, NULL);
+	
+					/* gst_bin will take the 'floating references' */
+				gst_bin_add_many (GST_BIN (m_gst_pipeline),
+							source, queue_audio, decoder, NULL);
+	
+					/* in decodebin's case we can just connect the source with the decodebin, and decodebin will take care about id3demux (or whatever is required) */
+				gst_element_link_many(source, queue_audio, decoder, NULL);
+	
+					/* create audio bin with the audioconverter, the capsfilter and the audiosink */
+				audio = gst_bin_new ("audiobin");
+	
+				GstPad *audiopad = gst_element_get_static_pad (conv, "sink");
+				gst_bin_add_many(GST_BIN(audio), conv, flt, sink, NULL);
+				gst_element_link_many(conv, flt, sink, NULL);
+				gst_element_add_pad(audio, gst_ghost_pad_new ("sink", audiopad));
+				gst_object_unref(audiopad);
+				gst_bin_add (GST_BIN(m_gst_pipeline), audio);
+			}
+			else
+			{
+				gst_bin_add_many (GST_BIN (m_gst_pipeline), source, sink, NULL);
+				if ( parser && id3demux )
+				{
+					gst_bin_add_many (GST_BIN (m_gst_pipeline), parser, id3demux, NULL);
+					gst_element_link(source, id3demux);
+					g_signal_connect(id3demux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
+					gst_element_link(parser, sink);
+				}
+				if ( audiodemux )
+				{
+					gst_bin_add (GST_BIN (m_gst_pipeline), audiodemux);
+					g_signal_connect(audiodemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
+					gst_element_link(source, audiodemux);
+				}
+				audioStream audio;
+				audio.type = sourceinfo.audiotype;
+				m_audioStreams.push_back(audio);
+			}
+		}
+	} else
+	{
+		m_event((iPlayableService*)this, evUser+12);
+
+		if (m_gst_pipeline)
+			gst_object_unref(GST_OBJECT(m_gst_pipeline));
+		if (source)
+			gst_object_unref(GST_OBJECT(source));
+		if (decoder)
+			gst_object_unref(GST_OBJECT(decoder));
+		if (conv)
+			gst_object_unref(GST_OBJECT(conv));
+		if (sink)
+			gst_object_unref(GST_OBJECT(sink));
+
+		if (audio)
+			gst_object_unref(GST_OBJECT(audio));
+		if (queue_audio)
+			gst_object_unref(GST_OBJECT(queue_audio));
+		if (video)
+			gst_object_unref(GST_OBJECT(video));
+		if (queue_video)
+			gst_object_unref(GST_OBJECT(queue_video));
+		if (videodemux)
+			gst_object_unref(GST_OBJECT(videodemux));
+		if (switch_audio)
+			gst_object_unref(GST_OBJECT(switch_audio));
+
+		eDebug("sorry, can't play: %s",m_error_message.c_str());
+		m_gst_pipeline = 0;
+	}
+
+	gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
 }
 
 eServiceMP3::~eServiceMP3()
 {
-	printf("MP3: %s stop\n", filename.c_str());
+	delete m_subtitle_widget;
+	if (m_state == stRunning)
+		stop();
+	
+	if (m_stream_tags)
+		gst_tag_list_free(m_stream_tags);
+	
+	if (m_gst_pipeline)
+	{
+		gst_object_unref (GST_OBJECT (m_gst_pipeline));
+		eDebug("SERVICEMP3 destruct!");
+	}
 }
+
+DEFINE_REF(eServiceMP3);	
+
+RESULT eServiceMP3::connectEvent(const Slot2<void,iPlayableService*,int> &event, ePtr<eConnection> &connection)
+{
+	connection = new eConnection((iPlayableService*)this, m_event.connect(event));
+	return 0;
+}
+
+RESULT eServiceMP3::start()
+{
+	assert(m_state == stIdle);
 	
-void eServiceMP3::AddRef()
+	m_state = stRunning;
+	if (m_gst_pipeline)
+	{
+		eDebug("starting pipeline");
+		gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
+	}
+	m_event(this, evStart);
+	return 0;
+}
+
+RESULT eServiceMP3::stop()
+{
+	assert(m_state != stIdle);
+	if (m_state == stStopped)
+		return -1;
+	eDebug("MP3: %s stop\n", m_filename.c_str());
+	gst_element_set_state(m_gst_pipeline, GST_STATE_NULL);
+	m_state = stStopped;
+	return 0;
+}
+
+RESULT eServiceMP3::setTarget(int target)
+{
+	return -1;
+}
+
+RESULT eServiceMP3::pause(ePtr<iPauseableService> &ptr)
 {
-	++ref;
+	ptr=this;
+	return 0;
 }
 
-void eServiceMP3::Release()
+RESULT eServiceMP3::setSlowMotion(int ratio)
 {
-	if (!--ref)
-		delete this;
+	/* we can't do slomo yet */
+	return -1;
+}
+
+RESULT eServiceMP3::setFastForward(int ratio)
+{
+	m_currentTrickRatio = ratio;
+	if (ratio)
+		m_seekTimeout->start(1000, 0);
+	else
+		m_seekTimeout->stop();
+	return 0;
 }
 
-RESULT eServiceMP3::start() { printf("mp3 starts\n"); return 0; }
-RESULT eServiceMP3::getIPausableService(ePtr<iPauseableService> &ptr) { ptr=this; return 0; }
+void eServiceMP3::seekTimeoutCB()
+{
+	pts_t ppos, len;
+	getPlayPosition(ppos);
+	getLength(len);
+	ppos += 90000*m_currentTrickRatio;
+	
+	if (ppos < 0)
+	{
+		ppos = 0;
+		m_seekTimeout->stop();
+	}
+	if (ppos > len)
+	{
+		ppos = 0;
+		stop();
+		m_seekTimeout->stop();
+		return;
+	}
+	seekTo(ppos);
+}
 
 		// iPausableService
-RESULT eServiceMP3::pause() { printf("mp3 pauses!\n"); return 0; }
-RESULT eServiceMP3::unpause() { printf("mp3 unpauses!\n"); return 0; }
+RESULT eServiceMP3::pause()
+{
+	if (!m_gst_pipeline)
+		return -1;
+	GstStateChangeReturn res = gst_element_set_state(m_gst_pipeline, GST_STATE_PAUSED);
+	if (res == GST_STATE_CHANGE_ASYNC)
+	{
+		pts_t ppos;
+		getPlayPosition(ppos);
+		seekTo(ppos);
+	}
+	return 0;
+}
+
+RESULT eServiceMP3::unpause()
+{
+	if (!m_gst_pipeline)
+		return -1;
+
+	GstStateChangeReturn res;
+	res = gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING);
+	return 0;
+}
+
+	/* iSeekableService */
+RESULT eServiceMP3::seek(ePtr<iSeekableService> &ptr)
+{
+	ptr = this;
+	return 0;
+}
+
+RESULT eServiceMP3::getLength(pts_t &pts)
+{
+	if (!m_gst_pipeline)
+		return -1;
+	if (m_state != stRunning)
+		return -1;
+	
+	GstFormat fmt = GST_FORMAT_TIME;
+	gint64 len;
+	
+	if (!gst_element_query_duration(m_gst_pipeline, &fmt, &len))
+		return -1;
+	
+		/* len is in nanoseconds. we have 90 000 pts per second. */
+	
+	pts = len / 11111;
+	return 0;
+}
+
+RESULT eServiceMP3::seekTo(pts_t to)
+{
+	if (!m_gst_pipeline)
+		return -1;
+
+		/* convert pts to nanoseconds */
+	gint64 time_nanoseconds = to * 11111LL;
+	if (!gst_element_seek (m_gst_pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH,
+		GST_SEEK_TYPE_SET, time_nanoseconds,
+		GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE))
+	{
+		eDebug("SEEK failed");
+		return -1;
+	}
+	return 0;
+}
+
+RESULT eServiceMP3::seekRelative(int direction, pts_t to)
+{
+	if (!m_gst_pipeline)
+		return -1;
+
+	pts_t ppos;
+	getPlayPosition(ppos);
+	ppos += to * direction;
+	if (ppos < 0)
+		ppos = 0;
+	seekTo(ppos);
+	
+	return 0;
+}
+
+RESULT eServiceMP3::getPlayPosition(pts_t &pts)
+{
+	if (!m_gst_pipeline)
+		return -1;
+	if (m_state != stRunning)
+		return -1;
+	
+	GstFormat fmt = GST_FORMAT_TIME;
+	gint64 len;
+	
+	if (!gst_element_query_position(m_gst_pipeline, &fmt, &len))
+		return -1;
+	
+		/* len is in nanoseconds. we have 90 000 pts per second. */
+	pts = len / 11111;
+	return 0;
+}
+
+RESULT eServiceMP3::setTrickmode(int trick)
+{
+		/* trickmode is not yet supported by our dvbmediasinks. */
+	return -1;
+}
+
+RESULT eServiceMP3::isCurrentlySeekable()
+{
+	return 1;
+}
+
+RESULT eServiceMP3::info(ePtr<iServiceInformation>&i)
+{
+	i = this;
+	return 0;
+}
+
+RESULT eServiceMP3::getName(std::string &name)
+{
+	name = m_filename;
+	size_t n = name.rfind('/');
+	if (n != std::string::npos)
+		name = name.substr(n + 1);
+	return 0;
+}
+
+int eServiceMP3::getInfo(int w)
+{
+	gchar *tag = 0;
+
+	switch (w)
+	{
+	case sTitle:
+	case sArtist:
+	case sAlbum:
+	case sComment:
+	case sTracknumber:
+	case sGenre:
+	case sVideoType:
+	case sTimeCreate:
+	case sUser+12:
+		return resIsString;
+	case sCurrentTitle:
+		tag = GST_TAG_TRACK_NUMBER;
+		break;
+	case sTotalTitles:
+		tag = GST_TAG_TRACK_COUNT;
+		break;
+	default:
+		return resNA;
+	}
+
+	if (!m_stream_tags || !tag)
+		return 0;
+	
+	guint value;
+	if (gst_tag_list_get_uint(m_stream_tags, tag, &value))
+		return (int) value;
+	
+	return 0;
+
+}
+
+std::string eServiceMP3::getInfoString(int w)
+{
+	if ( !m_stream_tags )
+		return "";
+	gchar *tag = 0;
+	switch (w)
+	{
+	case sTitle:
+		tag = GST_TAG_TITLE;
+		break;
+	case sArtist:
+		tag = GST_TAG_ARTIST;
+		break;
+	case sAlbum:
+		tag = GST_TAG_ALBUM;
+		break;
+	case sComment:
+		tag = GST_TAG_COMMENT;
+		break;
+	case sTracknumber:
+		tag = GST_TAG_TRACK_NUMBER;
+		break;
+	case sGenre:
+		tag = GST_TAG_GENRE;
+		break;
+	case sVideoType:
+		tag = GST_TAG_VIDEO_CODEC;
+		break;
+	case sTimeCreate:
+		GDate *date;
+		if (gst_tag_list_get_date(m_stream_tags, GST_TAG_DATE, &date))
+		{
+			gchar res[5];
+ 			g_date_strftime (res, sizeof(res), "%Y", date); 
+			return (std::string)res;
+		}
+		break;
+	case sUser+12:
+		return m_error_message;
+	default:
+		return "";
+	}
+	if ( !tag )
+		return "";
+	gchar *value;
+	if (gst_tag_list_get_string(m_stream_tags, tag, &value))
+	{
+		std::string res = value;
+		g_free(value);
+		return res;
+	}
+	return "";
+}
+
+RESULT eServiceMP3::audioChannel(ePtr<iAudioChannelSelection> &ptr)
+{
+	ptr = this;
+	return 0;
+}
+
+RESULT eServiceMP3::audioTracks(ePtr<iAudioTrackSelection> &ptr)
+{
+	ptr = this;
+	return 0;
+}
+
+RESULT eServiceMP3::subtitle(ePtr<iSubtitleOutput> &ptr)
+{
+	ptr = this;
+	return 0;
+}
+
+int eServiceMP3::getNumberOfTracks()
+{
+ 	return m_audioStreams.size();
+}
+
+int eServiceMP3::getCurrentTrack()
+{
+	return m_currentAudioStream;
+}
+
+RESULT eServiceMP3::selectTrack(unsigned int i)
+{
+	int ret = selectAudioStream(i);
+	/* flush */
+	pts_t ppos;
+	getPlayPosition(ppos);
+	seekTo(ppos);
+
+	return ret;
+}
+
+int eServiceMP3::selectAudioStream(int i)
+{
+	gint nb_sources;
+	GstPad *active_pad;
+	GstElement *switch_audio = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_audio");
+	if ( !switch_audio )
+	{
+		eDebug("can't switch audio tracks! gst-plugin-selector needed");
+		return -1;
+	}
+	g_object_get (G_OBJECT (switch_audio), "n-pads", &nb_sources, NULL);
+ 	if ( (unsigned int)i >= m_audioStreams.size() || i >= nb_sources || (unsigned int)m_currentAudioStream >= m_audioStreams.size() )
+		return -2;
+	char sinkpad[8];
+	sprintf(sinkpad, "sink%d", i);
+	g_object_set (G_OBJECT (switch_audio), "active-pad", gst_element_get_pad (switch_audio, sinkpad), NULL);
+	g_object_get (G_OBJECT (switch_audio), "active-pad", &active_pad, NULL);
+	gchar *name;
+	name = gst_pad_get_name (active_pad);
+	eDebug ("switched audio to (%s)", name);
+	g_free(name);
+	m_currentAudioStream = i;
+	return 0;
+}
+
+int eServiceMP3::getCurrentChannel()
+{
+	return STEREO;
+}
+
+RESULT eServiceMP3::selectChannel(int i)
+{
+	eDebug("eServiceMP3::selectChannel(%i)",i);
+	return 0;
+}
+
+RESULT eServiceMP3::getTrackInfo(struct iAudioTrackInfo &info, unsigned int i)
+{
+// 	eDebug("eServiceMP3::getTrackInfo(&info, %i)",i);
+ 	if (i >= m_audioStreams.size())
+		return -2;
+	if (m_audioStreams[i].type == atMPEG)
+		info.m_description = "MPEG";
+	else if (m_audioStreams[i].type == atMP3)
+		info.m_description = "MP3";
+	else if (m_audioStreams[i].type == atAC3)
+		info.m_description = "AC3";
+	else if (m_audioStreams[i].type == atAAC)
+		info.m_description = "AAC";
+	else if (m_audioStreams[i].type == atDTS)
+		info.m_description = "DTS";
+	else if (m_audioStreams[i].type == atPCM)
+		info.m_description = "PCM";
+	else if (m_audioStreams[i].type == atOGG)
+		info.m_description = "OGG";
+	else
+		info.m_description = "???";
+	if (info.m_language.empty())
+		info.m_language = m_audioStreams[i].language_code;
+	return 0;
+}
+
+void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
+{
+	if (!msg)
+		return;
+	gchar *sourceName;
+	GstObject *source;
+
+	source = GST_MESSAGE_SRC(msg);
+	sourceName = gst_object_get_name(source);
+#if 0
+	if (gst_message_get_structure(msg))
+	{
+		gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
+		eDebug("gst_message from %s: %s", sourceName, string);
+		g_free(string);
+	}
+	else
+		eDebug("gst_message from %s: %s (without structure)", sourceName, GST_MESSAGE_TYPE_NAME(msg));
+#endif
+	switch (GST_MESSAGE_TYPE (msg))
+	{
+		case GST_MESSAGE_EOS:
+			m_event((iPlayableService*)this, evEOF);
+			break;
+		case GST_MESSAGE_ERROR:
+		{
+			gchar *debug;
+			GError *err;
+	
+			gst_message_parse_error (msg, &err, &debug);
+			g_free (debug);
+			eWarning("Gstreamer error: %s (%i)", err->message, err->code );
+			if ( err->domain == GST_STREAM_ERROR && err->code == GST_STREAM_ERROR_CODEC_NOT_FOUND )
+			{
+				if ( g_strrstr(sourceName, "videosink") )
+					m_event((iPlayableService*)this, evUser+11);
+			}
+			g_error_free(err);
+			break;
+		}
+		case GST_MESSAGE_INFO:
+		{
+			gchar *debug;
+			GError *inf;
+	
+			gst_message_parse_info (msg, &inf, &debug);
+			g_free (debug);
+			if ( inf->domain == GST_STREAM_ERROR && inf->code == GST_STREAM_ERROR_DECODE )
+			{
+				if ( g_strrstr(sourceName, "videosink") )
+					m_event((iPlayableService*)this, evUser+14);
+			}
+			g_error_free(inf);
+			break;
+		}
+		case GST_MESSAGE_TAG:
+		{
+			GstTagList *tags, *result;
+			gst_message_parse_tag(msg, &tags);
+	
+			result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_PREPEND);
+			if (result)
+			{
+				if (m_stream_tags)
+					gst_tag_list_free(m_stream_tags);
+				m_stream_tags = result;
+			}
+	
+			gchar *g_audiocodec;
+			if ( gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_audiocodec) && m_audioStreams.size() == 0 )
+			{
+				GstPad* pad = gst_element_get_pad (GST_ELEMENT(source), "src");
+				GstCaps* caps = gst_pad_get_caps(pad);
+				GstStructure* str = gst_caps_get_structure(caps, 0);
+				if ( !str )
+					break;
+				audioStream audio;
+				audio.type = gstCheckAudioPad(str);
+				m_audioStreams.push_back(audio);
+			}
+	
+			const GValue *gv_image = gst_tag_list_get_value_index(tags, GST_TAG_IMAGE, 0);
+			if ( gv_image )
+			{
+				GstBuffer *buf_image;
+				buf_image = gst_value_get_buffer (gv_image);
+				int fd = open("/tmp/.id3coverart", O_CREAT|O_WRONLY|O_TRUNC, 0644);
+				write(fd, GST_BUFFER_DATA(buf_image), GST_BUFFER_SIZE(buf_image));
+				close(fd);
+				m_event((iPlayableService*)this, evUser+13);
+			}
+	
+			gst_tag_list_free(tags);
+			m_event((iPlayableService*)this, evUpdatedInfo);
+			break;
+		}
+		case GST_MESSAGE_ASYNC_DONE:
+		{
+			GstTagList *tags;
+			for (std::vector<audioStream>::iterator IterAudioStream(m_audioStreams.begin()); IterAudioStream != m_audioStreams.end(); ++IterAudioStream)
+			{
+				if ( IterAudioStream->pad )
+				{
+					g_object_get(IterAudioStream->pad, "tags", &tags, NULL);
+					gchar *g_language;
+					if ( tags && gst_is_tag_list(tags) && gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_language) )
+					{
+						eDebug("found audio language %s",g_language);
+						IterAudioStream->language_code = std::string(g_language);
+						g_free (g_language);
+					}
+				}
+			}
+			for (std::vector<subtitleStream>::iterator IterSubtitleStream(m_subtitleStreams.begin()); IterSubtitleStream != m_subtitleStreams.end(); ++IterSubtitleStream)
+			{
+				if ( IterSubtitleStream->pad )
+				{
+					g_object_get(IterSubtitleStream->pad, "tags", &tags, NULL);
+					gchar *g_language;
+					if ( tags && gst_is_tag_list(tags) && gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_language) )
+					{
+						eDebug("found subtitle language %s",g_language);
+						IterSubtitleStream->language_code = std::string(g_language);
+						g_free (g_language);
+					}
+				}
+			}
+		}
+		case GST_MESSAGE_ELEMENT:
+		{
+			if ( gst_is_missing_plugin_message(msg) )
+			{
+				gchar *description = gst_missing_plugin_message_get_description(msg);
+				if ( description )
+				{
+					m_error_message = "GStreamer plugin " + (std::string)description + " not available!\n";
+					g_free(description);
+					m_event((iPlayableService*)this, evUser+12);
+				}
+			}
+			else if (const GstStructure *msgstruct = gst_message_get_structure(msg))
+			{
+				const gchar *eventname;
+				if ( eventname = gst_structure_get_name(msgstruct) )
+				{
+					if (!strcmp(eventname, "eventSizeChanged"))
+					{
+						gint aspect_ratio, width, height = 0;
+						gst_structure_get_int (msgstruct, "aspect_ratio", &aspect_ratio);
+						gst_structure_get_int (msgstruct, "width", &width);
+						gst_structure_get_int (msgstruct, "height", &height);
+						eDebug("****** decoder threw eventSizeChanged! aspect_ratio=%i, width=%i, height=%i", aspect_ratio, width, height);
+					}
+					if (!strcmp(eventname, "eventFrameRateChanged"))
+					{
+						gint frame_rate = 0;
+						gst_structure_get_int (msgstruct, "frame_rate", &frame_rate);
+						eDebug("****** decoder threw eventFrameRateChanged! frame_rate=%i", frame_rate);
+					}
+					if (!strcmp(eventname, "eventProgressiveChanged"))
+					{
+						gint progressive = 0;
+						gst_structure_get_int (msgstruct, "progressive", &progressive);
+						eDebug("****** decoder threw eventProgressiveChanged! progressive=%i", progressive);
+					}
+				}
+			}
+		}
+		default:
+			break;
+	}
+	g_free (sourceName);
+}
+
+GstBusSyncReply eServiceMP3::gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data)
+{
+	eServiceMP3 *_this = (eServiceMP3*)user_data;
+	_this->m_pump.send(1);
+		/* wake */
+	return GST_BUS_PASS;
+}
+
+audiotype_t eServiceMP3::gstCheckAudioPad(GstStructure* structure)
+{
+	const gchar* type;
+	type = gst_structure_get_name(structure);
+
+	if (!strcmp(type, "audio/mpeg")) {
+			gint mpegversion, layer = 0;
+			gst_structure_get_int (structure, "mpegversion", &mpegversion);
+			gst_structure_get_int (structure, "layer", &layer);
+			eDebug("mime audio/mpeg version %d layer %d", mpegversion, layer);
+			switch (mpegversion) {
+				case 1:
+				{
+					if ( layer == 3 )
+						return atMP3;
+					else
+						return atMPEG;
+				}
+				case 2:
+					return atMPEG;
+				case 4:
+					return atAAC;
+				default:
+					return atUnknown;
+			}
+		}
+	else
+	{
+		eDebug("mime %s", type);
+		if (!strcmp(type, "audio/x-ac3") || !strcmp(type, "audio/ac3"))
+			return atAC3;
+		else if (!strcmp(type, "audio/x-dts") || !strcmp(type, "audio/dts"))
+			return atDTS;
+		else if (!strcmp(type, "audio/x-raw-int"))
+			return atPCM;
+	}
+	return atUnknown;
+}
+
+void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
+{
+	const gchar* type;
+	GstCaps* caps;
+	GstStructure* str;
+	caps = gst_pad_get_caps(pad);
+	str = gst_caps_get_structure(caps, 0);
+	type = gst_structure_get_name(str);
+
+	eDebug("A new pad %s:%s was created", GST_OBJECT_NAME (decodebin), GST_OBJECT_NAME (pad));
+
+	eServiceMP3 *_this = (eServiceMP3*)user_data;
+	GstBin *pipeline = GST_BIN(_this->m_gst_pipeline);
+	if (g_strrstr(type,"audio"))
+	{
+		audioStream audio;
+		audio.type = _this->gstCheckAudioPad(str);
+		GstElement *switch_audio = gst_bin_get_by_name(pipeline , "switch_audio");
+		if ( switch_audio )
+		{
+			GstPad *sinkpad = gst_element_get_request_pad (switch_audio, "sink%d");
+			gst_pad_link(pad, sinkpad);
+			audio.pad = sinkpad;
+			_this->m_audioStreams.push_back(audio);
+		
+			if ( _this->m_audioStreams.size() == 1 )
+			{
+				_this->selectAudioStream(0);
+				gst_element_set_state (_this->m_gst_pipeline, GST_STATE_PLAYING);
+			}
+			else
+				g_object_set (G_OBJECT (switch_audio), "select-all", FALSE, NULL);
+		}
+		else
+		{
+			GstElement *queue_audio = gst_bin_get_by_name(pipeline , "queue_audio");
+			if ( queue_audio )
+			{
+				gst_pad_link(pad, gst_element_get_static_pad(queue_audio, "sink"));
+				_this->m_audioStreams.push_back(audio);
+			}
+			else
+				gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline , "audiosink"), "sink"));
+		}
+	}
+	if (g_strrstr(type,"video"))
+	{
+		gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_video"), "sink"));
+	}
+	if (g_strrstr(type,"application/x-ssa") || g_strrstr(type,"application/x-ass"))
+	{
+		GstPad *switchpad = _this->gstCreateSubtitleSink(_this, stSSA);
+		gst_pad_link(pad, switchpad);
+		subtitleStream subs;
+		subs.pad = switchpad;
+		subs.type = stSSA;
+		_this->m_subtitleStreams.push_back(subs);
+	}
+	if (g_strrstr(type,"text/plain"))
+	{
+		GstPad *switchpad = _this->gstCreateSubtitleSink(_this, stPlainText);
+		gst_pad_link(pad, switchpad);
+		subtitleStream subs;
+		subs.pad = switchpad;
+		subs.type = stPlainText;
+		_this->m_subtitleStreams.push_back(subs);
+	}
+}
+
+GstPad* eServiceMP3::gstCreateSubtitleSink(eServiceMP3* _this, subtype_t type)
+{
+	GstBin *pipeline = GST_BIN(_this->m_gst_pipeline);
+	GstElement *switch_subparse = gst_bin_get_by_name(pipeline,"switch_subparse");
+	if ( !switch_subparse )
+	{
+		switch_subparse = gst_element_factory_make ("input-selector", "switch_subparse");
+		GstElement *sink = gst_element_factory_make("fakesink", "sink_subtitles");
+		gst_bin_add_many(pipeline, switch_subparse, sink, NULL);
+		gst_element_link(switch_subparse, sink);
+		g_object_set (G_OBJECT(sink), "signal-handoffs", TRUE, NULL);
+		g_object_set (G_OBJECT(sink), "sync", TRUE, NULL);
+		g_object_set (G_OBJECT(sink), "async", FALSE, NULL);
+		g_signal_connect(sink, "handoff", G_CALLBACK(_this->gstCBsubtitleAvail), _this);
+	
+		// order is essential since requested sink pad names can't be explicitely chosen
+		GstElement *switch_substream_plain = gst_element_factory_make ("input-selector", "switch_substream_plain");
+		gst_bin_add(pipeline, switch_substream_plain);
+		GstPad *sinkpad_plain = gst_element_get_request_pad (switch_subparse, "sink%d");
+		gst_pad_link(gst_element_get_pad (switch_substream_plain, "src"), sinkpad_plain);
+	
+		GstElement *switch_substream_ssa = gst_element_factory_make ("input-selector", "switch_substream_ssa");
+		GstElement *ssaparse = gst_element_factory_make("ssaparse", "ssaparse");
+		gst_bin_add_many(pipeline, switch_substream_ssa, ssaparse, NULL);
+		GstPad *sinkpad_ssa = gst_element_get_request_pad (switch_subparse, "sink%d");
+		gst_element_link(switch_substream_ssa, ssaparse);
+		gst_pad_link(gst_element_get_pad (ssaparse, "src"), sinkpad_ssa);
+	
+		GstElement *switch_substream_srt = gst_element_factory_make ("input-selector", "switch_substream_srt");
+		GstElement *srtparse = gst_element_factory_make("subparse", "srtparse");
+		gst_bin_add_many(pipeline, switch_substream_srt, srtparse, NULL);
+		GstPad *sinkpad_srt = gst_element_get_request_pad (switch_subparse, "sink%d");
+		gst_element_link(switch_substream_srt, srtparse);
+		gst_pad_link(gst_element_get_pad (srtparse, "src"), sinkpad_srt);
+		g_object_set (G_OBJECT(srtparse), "subtitle-encoding", "ISO-8859-15", NULL);
+	}
+
+	switch (type)
+	{
+		case stSSA:
+			return gst_element_get_request_pad (gst_bin_get_by_name(pipeline,"switch_substream_ssa"), "sink%d");
+		case stSRT:
+			return gst_element_get_request_pad (gst_bin_get_by_name(pipeline,"switch_substream_srt"), "sink%d");
+		case stPlainText:
+		default:
+			break;
+	}
+	return gst_element_get_request_pad (gst_bin_get_by_name(pipeline,"switch_substream_plain"), "sink%d");
+}
+
+void eServiceMP3::gstCBfilterPadAdded(GstElement *filter, GstPad *pad, gpointer user_data)
+{
+	eServiceMP3 *_this = (eServiceMP3*)user_data;
+	GstElement *decoder = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"decoder");
+	gst_pad_link(pad, gst_element_get_static_pad (decoder, "sink"));
+}
+
+void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last, gpointer user_data)
+{
+	eServiceMP3 *_this = (eServiceMP3*)user_data;
+	GstCaps *caps;
+	GstStructure *str;
+	GstPad *audiopad;
+
+	/* only link once */
+	GstElement *audiobin = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"audiobin");
+	audiopad = gst_element_get_static_pad (audiobin, "sink");
+	if ( !audiopad || GST_PAD_IS_LINKED (audiopad)) {
+		eDebug("audio already linked!");
+		g_object_unref (audiopad);
+		return;
+	}
+
+	/* check media type */
+	caps = gst_pad_get_caps (pad);
+	str = gst_caps_get_structure (caps, 0);
+	eDebug("gst new pad! %s", gst_structure_get_name (str));
+
+	if (!g_strrstr (gst_structure_get_name (str), "audio")) {
+		gst_caps_unref (caps);
+		gst_object_unref (audiopad);
+		return;
+	}
+	
+	gst_caps_unref (caps);
+	gst_pad_link (pad, audiopad);
+}
+
+void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
+{
+	GstStructure *str;
+
+	/* check media type */
+	caps = gst_pad_get_caps (pad);
+	str = gst_caps_get_structure (caps, 0);
+	eDebug("unknown type: %s - this can't be decoded.", gst_structure_get_name (str));
+	gst_caps_unref (caps);
+}
+
+void eServiceMP3::gstPoll(const int&)
+{
+		/* ok, we have a serious problem here. gstBusSyncHandler sends 
+		   us the wakup signal, but likely before it was posted.
+		   the usleep, an EVIL HACK (DON'T DO THAT!!!) works around this.
+		   
+		   I need to understand the API a bit more to make this work 
+		   proplerly. */
+	usleep(1);
+	
+	GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline));
+	GstMessage *message;
+	while ((message = gst_bus_pop (bus)))
+	{
+		gstBusCall(bus, message);
+		gst_message_unref (message);
+	}
+}
+
+eAutoInitPtr<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
+
+void eServiceMP3::gstCBsubtitleAvail(GstElement *element, GstBuffer *buffer, GstPad *pad, gpointer user_data)
+{
+	gint64 duration_ns = GST_BUFFER_DURATION(buffer);
+	const unsigned char *text = (unsigned char *)GST_BUFFER_DATA(buffer);
+	eDebug("gstCBsubtitleAvail: %s",text);
+	eServiceMP3 *_this = (eServiceMP3*)user_data;
+	if ( _this->m_subtitle_widget )
+	{
+		ePangoSubtitlePage page;
+		gRGB rgbcol(0xD0,0xD0,0xD0);
+		page.m_elements.push_back(ePangoSubtitlePageElement(rgbcol, (const char*)text));
+		page.m_timeout = duration_ns / 1000000;
+		(_this->m_subtitle_widget)->setPage(page);
+	}
+}
+
+RESULT eServiceMP3::enableSubtitles(eWidget *parent, ePyObject tuple)
+{
+	ePyObject entry;
+	int tuplesize = PyTuple_Size(tuple);
+	int pid;
+	int type;
+	gint nb_sources;
+	GstPad *active_pad;
+	GstElement *switch_substream = NULL;
+	GstElement *switch_subparse = gst_bin_get_by_name (GST_BIN(m_gst_pipeline), "switch_subparse");
+
+	if (!PyTuple_Check(tuple))
+		goto error_out;
+	if (tuplesize < 1)
+		goto error_out;
+	entry = PyTuple_GET_ITEM(tuple, 1);
+	if (!PyInt_Check(entry))
+		goto error_out;
+	pid = PyInt_AsLong(entry);
+	entry = PyTuple_GET_ITEM(tuple, 2);
+	if (!PyInt_Check(entry))
+		goto error_out;
+	type = PyInt_AsLong(entry);
+
+	switch ((subtype_t)type)
+	{
+		case stPlainText:
+			switch_substream = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_substream_plain");
+			break;
+		case stSSA:
+			switch_substream = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_substream_ssa");
+			break;
+		case stSRT:
+			switch_substream = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_substream_srt");
+			break;
+		default:
+			goto error_out;
+	}
 
+	m_subtitle_widget = new eSubtitleWidget(parent);
+	m_subtitle_widget->resize(parent->size()); /* full size */
+
+	if ( !switch_substream )
+	{
+		eDebug("can't switch subtitle tracks! gst-plugin-selector needed");
+		return -2;
+	}
+	g_object_get (G_OBJECT (switch_substream), "n-pads", &nb_sources, NULL);
+ 	if ( (unsigned int)pid >= m_subtitleStreams.size() || pid >= nb_sources || (unsigned int)m_currentSubtitleStream >= m_subtitleStreams.size() )
+		return -2;
+	g_object_get (G_OBJECT (switch_subparse), "n-pads", &nb_sources, NULL);
+	if ( type < 0 || type >= nb_sources )
+		return -2;
+
+	char sinkpad[6];
+	sprintf(sinkpad, "sink%d", type);
+	g_object_set (G_OBJECT (switch_subparse), "active-pad", gst_element_get_pad (switch_subparse, sinkpad), NULL);
+	sprintf(sinkpad, "sink%d", pid);
+	g_object_set (G_OBJECT (switch_substream), "active-pad", gst_element_get_pad (switch_substream, sinkpad), NULL);
+	m_currentSubtitleStream = pid;
+
+	return 0;
+error_out:
+	eDebug("enableSubtitles needs a tuple as 2nd argument!\n"
+		"for gst subtitles (2, subtitle_stream_count, subtitle_type)");
+	return -1;
+}
+
+RESULT eServiceMP3::disableSubtitles(eWidget *parent)
+{
+	eDebug("eServiceMP3::disableSubtitles");
+	delete m_subtitle_widget;
+	m_subtitle_widget = 0;
+	return 0;
+}
+
+PyObject *eServiceMP3::getCachedSubtitle()
+{
+	eDebug("eServiceMP3::getCachedSubtitle");
+	Py_RETURN_NONE;
+}
+
+PyObject *eServiceMP3::getSubtitleList()
+{
+	eDebug("eServiceMP3::getSubtitleList");
+
+	ePyObject l = PyList_New(0);
+	int stream_count[sizeof(subtype_t)];
+	for ( unsigned int i = 0; i < sizeof(subtype_t); i++ )
+		stream_count[i] = 0;
+
+	for (std::vector<subtitleStream>::iterator IterSubtitleStream(m_subtitleStreams.begin()); IterSubtitleStream != m_subtitleStreams.end(); ++IterSubtitleStream)
+	{
+		subtype_t type = IterSubtitleStream->type;
+		ePyObject tuple = PyTuple_New(5);
+		PyTuple_SET_ITEM(tuple, 0, PyInt_FromLong(2));
+		PyTuple_SET_ITEM(tuple, 1, PyInt_FromLong(stream_count[type]));
+		PyTuple_SET_ITEM(tuple, 2, PyInt_FromLong(int(type)));
+		PyTuple_SET_ITEM(tuple, 3, PyInt_FromLong(0));
+		PyTuple_SET_ITEM(tuple, 4, PyString_FromString((IterSubtitleStream->language_code).c_str()));
+		PyList_Append(l, tuple);
+		Py_DECREF(tuple);
+		stream_count[type]++;
+	}
+	return l;
+}
 
-eAutoInitP0<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
+#else
+#warning gstreamer not available, not building media player
+#endif