X-Git-Url: https://git.cweiske.de/enigma2.git/blobdiff_plain/fe813cde98c0c550137b47dd7a75ec2d4d9e6f34..a9939e31b86e340d93aad3d6eb450352ca8323d7:/lib/service/servicemp3.cpp

diff --git a/lib/service/servicemp3.cpp b/lib/service/servicemp3.cpp
index f550afc0..f74cc8d0 100644
--- a/lib/service/servicemp3.cpp
+++ b/lib/service/servicemp3.cpp
@@ -1,3 +1,7 @@
+#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>
@@ -6,6 +10,7 @@
 #include <lib/service/service.h>
 #include <lib/base/init_num.h>
 #include <lib/base/init.h>
+#include <gst/gst.h>
 
 // eServiceFactoryMP3
 
@@ -13,9 +18,18 @@ 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("mp3");
+		extensions.push_back("ogg");
+		extensions.push_back("mpg");
+		extensions.push_back("vob");
+		extensions.push_back("wav");
+		extensions.push_back("wave");
+		sc->addServiceFactory(eServiceFactoryMP3::id, this, extensions);
+	}
 
 	m_service_info = new eStaticServiceMP3Info();
 }
@@ -24,7 +38,7 @@ eServiceFactoryMP3::~eServiceFactoryMP3()
 {
 	ePtr<eServiceCenter> sc;
 	
-	eServiceCenter::getInstance(sc);
+	eServiceCenter::getPrivInstance(sc);
 	if (sc)
 		sc->removeServiceFactory(eServiceFactoryMP3::id);
 }
@@ -57,6 +71,13 @@ RESULT eServiceFactoryMP3::info(const eServiceReference &ref, ePtr<iStaticServic
 	return 0;
 }
 
+RESULT eServiceFactoryMP3::offlineOperations(const eServiceReference &, ePtr<iServiceOfflineOperations> &ptr)
+{
+	ptr = 0;
+	return -1;
+}
+
+
 // eStaticServiceMP3Info
 
 
@@ -75,7 +96,11 @@ eStaticServiceMP3Info::eStaticServiceMP3Info()
 
 RESULT eStaticServiceMP3Info::getName(const eServiceReference &ref, std::string &name)
 {
-	name = "MP3 file: " + ref.path;
+	size_t last = ref.path.rfind('/');
+	if (last != std::string::npos)
+		name = ref.path.substr(last+1);
+	else
+		name = ref.path;
 	return 0;
 }
 
@@ -86,23 +111,227 @@ int eStaticServiceMP3Info::getLength(const eServiceReference &ref)
 
 // eServiceMP3
 
-void eServiceMP3::test_end()
-{
-	eDebug("end of mp3!");
-	stop();
-}
-
-eServiceMP3::eServiceMP3(const char *filename): filename(filename), test(eApp)
+eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1)
 {
+	m_stream_tags = 0;
+	CONNECT(m_pump.recv_msg, eServiceMP3::gstPoll);
+	GstElement *source = 0;
+	
+	GstElement *filter = 0, *decoder = 0, *conv = 0, *flt = 0, *sink = 0; /* for audio */
+	
+	GstElement *audio = 0, *queue_audio = 0, *video = 0, *queue_video = 0, *mpegdemux = 0;
+	
 	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;
+
+	int is_mpeg_ps = !(strcasecmp(ext, ".mpeg") && strcasecmp(ext, ".mpg") && strcasecmp(ext, ".vob") && strcasecmp(ext, ".bin"));
+	int is_mpeg_ts = !strcasecmp(ext, ".ts");
+	int is_mp3 = !strcasecmp(ext, ".mp3"); /* force mp3 instead of decodebin */
+	int is_video = is_mpeg_ps || is_mpeg_ts;
+	int is_streaming = !strncmp(filename, "http://", 7);
+	
+	eDebug("filename: %s, is_mpeg_ps: %d, is_mpeg_ts: %d, is_video: %d, is_streaming: %d, is_mp3: %d", filename, is_mpeg_ps, is_mpeg_ts, is_video, is_streaming, is_mp3);
+	
+	int is_audio = !is_video;
+	
+	int all_ok = 0;
+
+	m_gst_pipeline = gst_pipeline_new ("audio-player");
+	if (!m_gst_pipeline)
+		eWarning("failed to create pipeline");
+
+	if (!is_streaming)
+		source = gst_element_factory_make ("filesrc", "file-source");
+	else
+	{
+		source = gst_element_factory_make ("neonhttpsrc", "http-source");
+		if (source)
+			g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
+	}
+
+	if (!source)
+		eWarning("failed to create %s", is_streaming ? "neonhttpsrc" : "filesrc");
+	else
+				/* configure source */
+		g_object_set (G_OBJECT (source), "location", filename, NULL);
+
+	if (is_audio)
+	{
+			/* filesrc -> decodebin -> audioconvert -> capsfilter -> alsasink */
+		const char *decodertype = is_mp3 ? "mad" : "decodebin";
+
+		decoder = gst_element_factory_make (decodertype, "decoder");
+		if (!decoder)
+			eWarning("failed to create %s decoder", decodertype);
+
+			/* mp3 decoding needs id3demux to extract ID3 data. 'decodebin' would do that internally. */
+		if (is_mp3)
+		{
+			filter = gst_element_factory_make ("id3demux", "filter");
+			if (!filter)
+				eWarning("failed to create id3demux");
+		}
+
+		conv = gst_element_factory_make ("audioconvert", "converter");
+		if (!conv)
+			eWarning("failed to create audioconvert");
+
+		flt = gst_element_factory_make ("capsfilter", "flt");
+		if (!flt)
+			eWarning("failed to create capsfilter");
+
+			/* 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, (char*)0);
+			g_object_set (G_OBJECT (flt), "caps", caps, (char*)0);
+			gst_caps_unref(caps);
+		}
+
+		sink = gst_element_factory_make ("alsasink", "alsa-output");
+		if (!sink)
+			eWarning("failed to create osssink");
+
+		if (source && decoder && conv && sink)
+			all_ok = 1;
+	} else /* is_video */
+	{
+			/* filesrc -> mpegdemux -> | queue_audio -> dvbaudiosink
+			                           | queue_video -> dvbvideosink */
+
+		audio = gst_element_factory_make("dvbaudiosink", "audio");
+		queue_audio = gst_element_factory_make("queue", "queue_audio");
+		
+		video = gst_element_factory_make("dvbvideosink", "video");
+		queue_video = gst_element_factory_make("queue", "queue_video");
+		
+		if (is_mpeg_ps)
+			mpegdemux = gst_element_factory_make("flupsdemux", "mpegdemux");
+		else
+			mpegdemux = gst_element_factory_make("flutsdemux", "mpegdemux");
+			
+		if (!mpegdemux)
+		{
+			eDebug("fluendo mpegdemux not available, falling back to mpegdemux\n");
+			mpegdemux = gst_element_factory_make("mpegdemux", "mpegdemux");
+		}
+		
+		eDebug("audio: %p, queue_audio %p, video %p, queue_video %p, mpegdemux %p", audio, queue_audio, video, queue_video, mpegdemux);
+		if (audio && queue_audio && video && queue_video && mpegdemux)
+		{
+			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-time", (guint64)0, NULL);
+			all_ok = 1;
+		}
+	}
+	
+	if (m_gst_pipeline && all_ok)
+	{
+		gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline)), gstBusSyncHandler, this);
+
+		if (is_audio)
+		{
+			if (!is_mp3)
+			{
+					/* decodebin has dynamic pads. When they get created, we connect them to the audio bin */
+				g_signal_connect (decoder, "new-decoded-pad", G_CALLBACK(gstCBnewPad), this);
+				g_signal_connect (decoder, "unknown-type", G_CALLBACK(gstCBunknownType), this);
+			}
+
+				/* gst_bin will take the 'floating references' */
+			gst_bin_add_many (GST_BIN (m_gst_pipeline),
+						source, decoder, NULL);
+
+			if (filter)
+			{
+					/* id3demux also has dynamic pads, which need to be connected to the decoder (this is done in the 'gstCBfilterPadAdded' CB) */
+				gst_bin_add(GST_BIN(m_gst_pipeline), filter);
+				gst_element_link(source, filter);
+				m_decoder = decoder;
+				g_signal_connect (filter, "pad-added", G_CALLBACK(gstCBfilterPadAdded), this);
+			} else
+					/* 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(source, decoder);
+
+				/* create audio bin with the audioconverter, the capsfilter and the audiosink */
+			m_gst_audio = gst_bin_new ("audiobin");
+
+			GstPad *audiopad = gst_element_get_pad (conv, "sink");
+			gst_bin_add_many(GST_BIN(m_gst_audio), conv, flt, sink, (char*)0);
+			gst_element_link_many(conv, flt, sink, (char*)0);
+			gst_element_add_pad(m_gst_audio, gst_ghost_pad_new ("sink", audiopad));
+			gst_object_unref(audiopad);
+			gst_bin_add (GST_BIN(m_gst_pipeline), m_gst_audio);
+
+				/* in mad's case, we can directly connect the decoder to the audiobin. otherwise, we do this in gstCBnewPad */
+			if (is_mp3)
+				gst_element_link(decoder, m_gst_audio);
+		} else
+		{
+			gst_bin_add_many(GST_BIN(m_gst_pipeline), source, mpegdemux, audio, queue_audio, video, queue_video, NULL);
+			gst_element_link(source, mpegdemux);
+			gst_element_link(queue_audio, audio);
+			gst_element_link(queue_video, video);
+			
+			m_gst_audioqueue = queue_audio;
+			m_gst_videoqueue = queue_video;
+			
+			g_signal_connect(mpegdemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
+		}
+	} else
+	{
+		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 (mpegdemux)
+			gst_object_unref(GST_OBJECT(mpegdemux));
+
+		eDebug("sorry, can't play.");
+		m_gst_pipeline = 0;
+	}
+	
+	gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
 }
 
 eServiceMP3::~eServiceMP3()
 {
-	eDebug("SERVICEMP3 destruct!");
 	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);	
@@ -118,11 +347,11 @@ RESULT eServiceMP3::start()
 	assert(m_state == stIdle);
 	
 	m_state = stRunning;
-
-	printf("mp3 starts\n");
-	printf("MP3: %s start\n", filename.c_str());
-	test.start(1000, 1);
-	CONNECT(test.timeout, eServiceMP3::test_end);
+	if (m_gst_pipeline)
+	{
+		eDebug("starting pipeline");
+		gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
+	}
 	m_event(this, evStart);
 	return 0;
 }
@@ -132,25 +361,361 @@ RESULT eServiceMP3::stop()
 	assert(m_state != stIdle);
 	if (m_state == stStopped)
 		return -1;
-	test.stop();
-	printf("MP3: %s stop\n", filename.c_str());
+	printf("MP3: %s stop\n", m_filename.c_str());
+	gst_element_set_state(m_gst_pipeline, GST_STATE_NULL);
 	m_state = stStopped;
-	m_event(this, evEnd);
 	return 0;
 }
 
-RESULT eServiceMP3::pause(ePtr<iPauseableService> &ptr) { ptr=this; return 0; }
+RESULT eServiceMP3::setTarget(int target)
+{
+	return -1;
+}
+
+RESULT eServiceMP3::pause(ePtr<iPauseableService> &ptr)
+{
+	ptr=this;
+	return 0;
+}
+
+RESULT eServiceMP3::setSlowMotion(int ratio)
+{
+	return -1;
+}
 
+RESULT eServiceMP3::setFastForward(int ratio)
+{
+	return -1;
+}
+  
 		// 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;
+	gst_element_set_state(m_gst_pipeline, GST_STATE_PAUSED);
+	return 0;
+}
 
-RESULT eServiceMP3::info(ePtr<iServiceInformation>&i) { i = this; return 0; }
+RESULT eServiceMP3::unpause()
+{
+	if (!m_gst_pipeline)
+		return -1;
+	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;
+
+	pause();
+
+	pts_t ppos;
+	getPlayPosition(ppos);
+	ppos += to * direction;
+	if (ppos < 0)
+		ppos = 0;
+	seekTo(ppos);
+	
+	unpause();
+
+	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 currently doesn't make any sense for us. */
+	return -1;
+}
+
+RESULT eServiceMP3::isCurrentlySeekable()
+{
+	return 1;
+}
+
+RESULT eServiceMP3::info(ePtr<iServiceInformation>&i)
+{
+	i = this;
+	return 0;
+}
 
 RESULT eServiceMP3::getName(std::string &name)
 {
-	name = "MP3 File: " + filename;
+	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)
+{
+	switch (w)
+	{
+	case sTitle:
+	case sArtist:
+	case sAlbum:
+	case sComment:
+	case sTracknumber:
+	case sGenre:
+		return resIsString;
+
+	default:
+		return resNA;
+	}
+}
+
+std::string eServiceMP3::getInfoString(int w)
+{
+	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;
+	default:
+		return "";
+	}
+	
+	if (!m_stream_tags || !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 "";
+}
+
+
+		void foreach(const GstTagList *list, const gchar *tag, gpointer user_data)
+		{
+			if (tag)
+				eDebug("Tag: %c%c%c%c", tag[0], tag[1], tag[2], tag[3]);
+			
+		}
+
+void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
+{
+	if (msg)
+	{
+		gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
+		eDebug("gst_message: %s", string);
+		g_free(string);
+	}
+	
+	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", err->message);
+		g_error_free(err);
+			/* TODO: signal error condition to user */
+		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;
+		}
+		gst_tag_list_free(tags);
+		
+		m_event((iPlayableService*)this, evUpdatedInfo);
+		break;
+	}
+	default:
+		break;
+	}
+}
+
+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;
+}
+
+void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
+{
+	eServiceMP3 *_this = (eServiceMP3*)user_data;
+	
+	gchar *name;
+	name = gst_pad_get_name (pad);
+	g_print ("A new pad %s was created\n", name);
+	if (!strncmp(name, "audio_", 6)) // mpegdemux uses video_nn with n=0,1,.., flupsdemux uses stream id
+		gst_pad_link(pad, gst_element_get_pad (_this->m_gst_audioqueue, "sink"));
+	if (!strncmp(name, "video_", 6))
+		gst_pad_link(pad, gst_element_get_pad (_this->m_gst_videoqueue, "sink"));
+	g_free (name);
+	
+}
+
+void eServiceMP3::gstCBfilterPadAdded(GstElement *filter, GstPad *pad, gpointer user_data)
+{
+	eServiceMP3 *_this = (eServiceMP3*)user_data;
+	gst_pad_link(pad, gst_element_get_pad (_this->m_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 */
+	audiopad = gst_element_get_pad (_this->m_gst_audio, "sink");
+	if (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)
+{
+	eServiceMP3 *_this = (eServiceMP3*)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");
+#else
+#warning gstreamer not available, not building media player
+#endif