Add basic AudioSelection support for video containers

diff --git a/lib/service/servicemp3.cpp b/lib/service/servicemp3.cpp
index fb515e9b8e73f76b13aa189854c022267fa64c30..1e53a40fd351fdc01b43028dfde1abbba8134d58 100644 (file)
--- a/lib/service/servicemp3.cpp
+++ b/lib/service/servicemp3.cpp
@@ -116,12 +116,13 @@ int eStaticServiceMP3Info::getLength(const eServiceReference &ref)
 eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1)
 {
        m_stream_tags = 0;
 eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1)
 {
        m_stream_tags = 0;

+       m_currentAudioStream = 0;

        CONNECT(m_pump.recv_msg, eServiceMP3::gstPoll);
        GstElement *source = 0;
        
        GstElement *filter = 0, *decoder = 0, *conv = 0, *flt = 0, *sink = 0; /* for audio */
        
        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;
+       GstElement *audio = 0, *switch_audio = 0, *queue_audio = 0, *video = 0, *queue_video = 0, *videodemux = 0;

        
        m_state = stIdle;
        eDebug("SERVICEMP3 construct!");
        
        m_state = stIdle;
        eDebug("SERVICEMP3 construct!");


@@ -148,6 +149,10 @@ eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eAp
        
        int all_ok = 0;
 
        
        int all_ok = 0;
 

+//     GError *blubb = NULL;
+//     GstElement* m_gst_pipeline = gst_parse_launch("filesrc location=/media/hdd/movie/artehd_2lang.mkv ! matroskademux name=demux  demux.audio_00 ! input-selector name=a demux.audio_01 ! a.   a. ! queue ! dvbaudiosink", &blubb);
+//     return;
+

        m_gst_pipeline = gst_pipeline_new ("mediaplayer");
        if (!m_gst_pipeline)
                eWarning("failed to create pipeline");
        m_gst_pipeline = gst_pipeline_new ("mediaplayer");
        if (!m_gst_pipeline)
                eWarning("failed to create pipeline");


@@ -234,22 +239,22 @@ eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eAp
                queue_video = gst_element_factory_make("queue", "queue_video");
                
                if (is_mpeg_ps)
                queue_video = gst_element_factory_make("queue", "queue_video");
                
                if (is_mpeg_ps)

-                       mpegdemux = gst_element_factory_make("flupsdemux", "mpegdemux");
+                       videodemux = gst_element_factory_make("flupsdemux", "videodemux");

                else if (is_mpeg_ts)
                else if (is_mpeg_ts)

-                       mpegdemux = gst_element_factory_make("flutsdemux", "mpegdemux");
+                       videodemux = gst_element_factory_make("flutsdemux", "videodemux");

                else if (is_matroska)
                else if (is_matroska)

-                       mpegdemux = gst_element_factory_make("matroskademux", "mpegdemux");
+                       videodemux = gst_element_factory_make("matroskademux", "videodemux");

                else if (is_avi)
                else if (is_avi)

-                       mpegdemux = gst_element_factory_make("avidemux", "mpegdemux");
+                       videodemux = gst_element_factory_make("avidemux", "videodemux");

 
 

-               if (!mpegdemux)
+               if (!videodemux)

                {
                        eDebug("fluendo mpegdemux not available, falling back to mpegdemux\n");
                {
                        eDebug("fluendo mpegdemux not available, falling back to mpegdemux\n");

-                       mpegdemux = gst_element_factory_make("mpegdemux", "mpegdemux");
+                       videodemux = gst_element_factory_make("mpegdemux", "videodemux");

                }
                }

-               
-               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)
+
+               eDebug("audio: %p, queue_audio %p, video %p, queue_video %p, videodemux %p", audio, queue_audio, video, queue_video, videodemux);
+               if (audio && queue_audio && video && queue_video && videodemux)

                {
                        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-bytes", 256*1024, NULL);
                        g_object_set (G_OBJECT (queue_audio), "max-size-buffers", 0, NULL);


@@ -293,36 +298,34 @@ eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eAp
                                        /* 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);
                                        /* 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_many(source, queue_audio, decoder, NULL);
 
                                /* create audio bin with the audioconverter, the capsfilter and the audiosink */
                                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_many(source, queue_audio, decoder, NULL);
 
                                /* create audio bin with the audioconverter, the capsfilter and the audiosink */

-                       m_gst_audio = gst_bin_new ("audiobin");
+                       audio = gst_bin_new ("audiobin");

 
                        GstPad *audiopad = gst_element_get_static_pad (conv, "sink");
 
                        GstPad *audiopad = gst_element_get_static_pad (conv, "sink");

-                       gst_bin_add_many(GST_BIN(m_gst_audio), conv, flt, sink, (char*)0);
+                       gst_bin_add_many(GST_BIN(audio), conv, flt, sink, (char*)0);

                        gst_element_link_many(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_element_add_pad(audio, gst_ghost_pad_new ("sink", audiopad));

                        gst_object_unref(audiopad);
                        gst_object_unref(audiopad);

-                       gst_bin_add (GST_BIN(m_gst_pipeline), m_gst_audio);
+                       gst_bin_add (GST_BIN(m_gst_pipeline), audio);

 
                                /* in mad's case, we can directly connect the decoder to the audiobin. otherwise, we do this in gstCBnewPad */
                        if (is_mp3)
 
                                /* 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);
+                               gst_element_link(decoder, audio);

                } else /* is_video */
                {
                } else /* is_video */
                {

-                       gst_bin_add_many(GST_BIN(m_gst_pipeline), source, mpegdemux, audio, queue_audio, video, queue_video, NULL);
-                       gst_element_link(source, mpegdemux);
+                       switch_audio = gst_element_factory_make ("input-selector", "switch_audio");
+                       g_object_set (G_OBJECT (switch_audio), "select-all", TRUE, NULL);
+                       gst_bin_add_many(GST_BIN(m_gst_pipeline), source, videodemux, switch_audio, audio, queue_audio, video, queue_video, NULL);
+                       gst_element_link(source, videodemux);
+                       gst_element_link(switch_audio, queue_audio);

                        gst_element_link(queue_audio, audio);
                        gst_element_link(queue_video, video);
                        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);
+                       g_signal_connect(videodemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);

                }
        } else
        {
                }
        } else
        {


@@ -345,8 +348,8 @@ eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eAp
                        gst_object_unref(GST_OBJECT(video));
                if (queue_video)
                        gst_object_unref(GST_OBJECT(queue_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));
+               if (videodemux)
+                       gst_object_unref(GST_OBJECT(videodemux));

 
                eDebug("sorry, can't play.");
                m_gst_pipeline = 0;
 
                eDebug("sorry, can't play.");
                m_gst_pipeline = 0;


@@ -397,7 +400,7 @@ RESULT eServiceMP3::stop()
        assert(m_state != stIdle);
        if (m_state == stStopped)
                return -1;
        assert(m_state != stIdle);
        if (m_state == stStopped)
                return -1;

-       printf("MP3: %s stop\n", m_filename.c_str());
+       eDebug("MP3: %s stop\n", m_filename.c_str());

        gst_element_set_state(m_gst_pipeline, GST_STATE_NULL);
        m_state = stStopped;
        return 0;
        gst_element_set_state(m_gst_pipeline, GST_STATE_NULL);
        m_state = stStopped;
        return 0;


@@ -627,28 +630,111 @@ std::string eServiceMP3::getInfoString(int w)
        return "";
 }
 
        return "";
 }
 

+RESULT eServiceMP3::audioChannel(ePtr<iAudioChannelSelection> &ptr)
+{
+       ptr = this;
+       return 0;
+}

 
 

-               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]);
-                       
-               }
+RESULT eServiceMP3::audioTracks(ePtr<iAudioTrackSelection> &ptr)
+{
+       ptr = this;
+       return 0;
+}
+
+int eServiceMP3::getNumberOfTracks()
+{
+       eDebug("eServiceMP3::getNumberOfTracks()=%i",m_audioStreams.size());
+       return m_audioStreams.size();
+}
+
+int eServiceMP3::getCurrentTrack()
+{
+       eDebug("eServiceMP3::getCurrentTrack()=%i",m_currentAudioStream);
+       return m_currentAudioStream;
+}
+
+RESULT eServiceMP3::selectTrack(unsigned int i)
+{
+       eDebug("eServiceMP3::selectTrack(%i)",i);
+
+       GstPadLinkReturn ret;
+       gint nb_sources;
+       GstPad *active_pad;
+       
+       GstElement *selector = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_audio");
+       g_object_get (G_OBJECT (selector), "n-pads", &nb_sources, NULL);
+       g_object_get (G_OBJECT (selector), "active-pad", &active_pad, NULL);
+       
+       if ( i >= m_audioStreams.size() || i >= nb_sources || m_currentAudioStream >= m_audioStreams.size() )
+               return -2;
+
+       char sinkpad[8];
+       sprintf(sinkpad, "sink%d", i);
+
+       g_object_set (G_OBJECT (selector), "active-pad", gst_element_get_pad (selector, sinkpad), NULL);
+       g_object_get (G_OBJECT (selector), "active-pad", &active_pad, NULL);
+
+       gchar *name;
+       name = gst_pad_get_name (active_pad);
+       eDebug ("switched audio to (%s)", 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 == audioStream::atMP2)
+               info.m_description = "MP2";
+       else if (m_audioStreams[i].type == audioStream::atMP3)
+               info.m_description = "MP3";
+       else if (m_audioStreams[i].type == audioStream::atAC3)
+               info.m_description = "AC3";
+       else if (m_audioStreams[i].type == audioStream::atAAC)
+               info.m_description = "AAC";
+       else  if (m_audioStreams[i].type == audioStream::atDTS)
+               info.m_description = "DTS";
+       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)
 {
 
 void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
 {

-       if (msg)
+       if (!msg)
+               return;
+       gchar *sourceName;
+       GstObject *source;
+
+       source = GST_MESSAGE_SRC(msg);
+       sourceName = gst_object_get_name(source);       
+
+       if (gst_message_get_structure(msg))

        {
        {

-               if (gst_message_get_structure(msg))
-               {
-                       gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
-                       eDebug("gst_message: %s", string);
-                       g_free(string);
-               }
-               else
-                       eDebug("gst_message: %s (without structure)", GST_MESSAGE_TYPE_NAME(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));
+

        switch (GST_MESSAGE_TYPE (msg))
        {
        case GST_MESSAGE_EOS:
        switch (GST_MESSAGE_TYPE (msg))
        {
        case GST_MESSAGE_EOS:


@@ -656,16 +742,12 @@ void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
                break;
        case GST_MESSAGE_ERROR:
        {
                break;
        case GST_MESSAGE_ERROR:
        {

-               gchar *debug, *sourceName;
+               gchar *debug;

                GError *err;
                GError *err;

-               GstObject *source;
-
-               source = GST_MESSAGE_SRC(msg);
-               sourceName = gst_object_get_name(source);

 
                gst_message_parse_error (msg, &err, &debug);
                g_free (debug);
 
                gst_message_parse_error (msg, &err, &debug);
                g_free (debug);

-               eWarning("Gstreamer error: %s (%i) from %s", err->message, err->code, sourceName );
+               eWarning("Gstreamer error: %s (%i)", err->message, err->code );

                if ( err->domain == GST_STREAM_ERROR && err->code == GST_STREAM_ERROR_DECODE )
                {
                        if ( g_strrstr(sourceName, "videosink") )
                if ( err->domain == GST_STREAM_ERROR && err->code == GST_STREAM_ERROR_DECODE )
                {
                        if ( g_strrstr(sourceName, "videosink") )


@@ -687,14 +769,30 @@ void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
                                gst_tag_list_free(m_stream_tags);
                        m_stream_tags = result;
                }
                                gst_tag_list_free(m_stream_tags);
                        m_stream_tags = result;
                }

-               gst_tag_list_free(tags);
-               
-               m_event((iPlayableService*)this, evUpdatedInfo);
+
+               gchar *g_audiocodec;
+               if (gst_tag_list_get_string(m_stream_tags, GST_TAG_AUDIO_CODEC, &g_audiocodec))
+               {
+                       static int a_str_cnt = 0;
+                       if ( g_strrstr(g_audiocodec, "MPEG-1 layer 2") )
+                               m_audioStreams[a_str_cnt].type = audioStream::atMP2;
+                       else if ( g_strrstr(g_audiocodec, "AC-3 audio") )
+                               m_audioStreams[a_str_cnt].type = audioStream::atAC3;
+                       gchar *g_language;
+                       if ( gst_tag_list_get_string(m_stream_tags, GST_TAG_LANGUAGE_CODE, &g_language) )
+                       {
+                               m_audioStreams[a_str_cnt].language_code = std::string(g_language);
+                               g_free (g_language);
+                       }
+                       g_free (g_audiocodec);
+                       a_str_cnt++;
+               }

                break;
        }
        default:
                break;
        }
                break;
        }
        default:
                break;
        }

+       g_free (sourceName);

 }
 
 GstBusSyncReply eServiceMP3::gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data)
 }
 
 GstBusSyncReply eServiceMP3::gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data)


@@ -710,20 +808,35 @@ void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer use
        eServiceMP3 *_this = (eServiceMP3*)user_data;
        gchar *name;
        name = gst_pad_get_name (pad);
        eServiceMP3 *_this = (eServiceMP3*)user_data;
        gchar *name;
        name = gst_pad_get_name (pad);

-       g_print ("A new pad %s was created\n", name);
-       GstPad *sinkpad;
-
-       if (g_strrstr(name,"audio")) // mpegdemux uses video_nn with n=0,1,.., flupsdemux uses stream id
-               gst_pad_link(pad, gst_element_get_static_pad (_this->m_gst_audioqueue, "sink"));
+       eDebug ("A new pad %s was created", name);
+       if (g_strrstr(name,"audio")) // mpegdemux, matroskademux, avidemux use video_nn with n=0,1,.., flupsdemux uses stream id
+       {
+               audioStream audio;
+               audio.pad = pad;
+               _this->m_audioStreams.push_back(audio);
+               GstElement *selector = gst_bin_get_by_name( GST_BIN(_this->m_gst_pipeline), "switch_audio" );
+               GstPadLinkReturn ret = gst_pad_link(pad, gst_element_get_request_pad (selector, "sink%d"));
+               if ( _this->m_audioStreams.size() == 1 )
+               {
+                       _this->selectTrack(_this->m_audioStreams.size()-1);
+                       gst_element_set_state (_this->m_gst_pipeline, GST_STATE_PLAYING);
+               }
+               else
+                       g_object_set (G_OBJECT (selector), "select-all", FALSE, NULL);
+       }

        if (g_strrstr(name,"video"))
        if (g_strrstr(name,"video"))

-               gst_pad_link(pad, gst_element_get_static_pad (_this->m_gst_videoqueue, "sink"));
+       {
+               GstElement *video = gst_bin_get_by_name(GST_BIN (_this->m_gst_pipeline),"queue_video");
+               gst_pad_link(pad, gst_element_get_static_pad (video, "sink"));
+       }

        g_free (name);
 }
 
 void eServiceMP3::gstCBfilterPadAdded(GstElement *filter, GstPad *pad, gpointer user_data)
 {
        eServiceMP3 *_this = (eServiceMP3*)user_data;
        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_static_pad (_this->m_decoder, "sink"));
+       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)
 }
 
 void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last, gpointer user_data)


@@ -734,7 +847,8 @@ void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last,
        GstPad *audiopad;
 
        /* only link once */
        GstPad *audiopad;
 
        /* only link once */

-       audiopad = gst_element_get_static_pad (_this->m_gst_audio, "sink");
+       GstElement *audio = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"audiobin");
+       audiopad = gst_element_get_static_pad (audio, "sink");

        if ( !audiopad || GST_PAD_IS_LINKED (audiopad)) {
                eDebug("audio already linked!");
                g_object_unref (audiopad);
        if ( !audiopad || GST_PAD_IS_LINKED (audiopad)) {
                eDebug("audio already linked!");
                g_object_unref (audiopad);


@@ -758,7 +872,6 @@ void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last,
 
 void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
 {
 
 void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
 {

-       eServiceMP3 *_this = (eServiceMP3*)user_data;

        GstStructure *str;
 
        /* check media type */
        GstStructure *str;
 
        /* check media type */

diff --git a/lib/service/servicemp3.h b/lib/service/servicemp3.h
index 08b712c334f12b656d3eea922813f4a5bbec456c..bf29efcf69a79fbc66928462d748f692a2e29910 100644 (file)
--- a/lib/service/servicemp3.h
+++ b/lib/service/servicemp3.h
@@ -4,6 +4,7 @@
 #ifdef HAVE_GSTREAMER
 #include <lib/base/message.h>
 #include <lib/service/iservice.h>
 #ifdef HAVE_GSTREAMER
 #include <lib/base/message.h>
 #include <lib/service/iservice.h>

+#include <lib/dvb/pmt.h>

 #include <gst/gst.h>
 
 class eStaticServiceMP3Info;
 #include <gst/gst.h>
 
 class eStaticServiceMP3Info;


@@ -39,7 +40,7 @@ public:
 typedef struct _GstElement GstElement;
 
 class eServiceMP3: public iPlayableService, public iPauseableService, 
 typedef struct _GstElement GstElement;
 
 class eServiceMP3: public iPlayableService, public iPauseableService, 

-       public iServiceInformation, public iSeekableService, public Object
+       public iServiceInformation, public iSeekableService, public iAudioTrackSelection, public iAudioChannelSelection, public Object

 {
        DECLARE_REF(eServiceMP3);
 public:
 {
        DECLARE_REF(eServiceMP3);
 public:


@@ -56,10 +57,10 @@ public:
        RESULT setFastForward(int ratio);
 
        RESULT seek(ePtr<iSeekableService> &ptr);
        RESULT setFastForward(int ratio);
 
        RESULT seek(ePtr<iSeekableService> &ptr);

+       RESULT audioTracks(ePtr<iAudioTrackSelection> &ptr);
+       RESULT audioChannel(ePtr<iAudioChannelSelection> &ptr);

 
                // not implemented (yet)
 
                // not implemented (yet)

-       RESULT audioChannel(ePtr<iAudioChannelSelection> &ptr) { ptr = 0; return -1; }
-       RESULT audioTracks(ePtr<iAudioTrackSelection> &ptr) { ptr = 0; return -1; }

        RESULT frontendInfo(ePtr<iFrontendInformation> &ptr) { ptr = 0; return -1; }
        RESULT subServices(ePtr<iSubserviceList> &ptr) { ptr = 0; return -1; }
        RESULT timeshift(ePtr<iTimeshiftService> &ptr) { ptr = 0; return -1; }
        RESULT frontendInfo(ePtr<iFrontendInformation> &ptr) { ptr = 0; return -1; }
        RESULT subServices(ePtr<iSubserviceList> &ptr) { ptr = 0; return -1; }
        RESULT timeshift(ePtr<iTimeshiftService> &ptr) { ptr = 0; return -1; }


@@ -88,20 +89,40 @@ public:
        RESULT getName(std::string &name);
        int getInfo(int w);
        std::string getInfoString(int w);
        RESULT getName(std::string &name);
        int getInfo(int w);
        std::string getInfoString(int w);

+
+               // iAudioTrackSelection 
+       int getNumberOfTracks();
+       RESULT selectTrack(unsigned int i);
+       RESULT getTrackInfo(struct iAudioTrackInfo &, unsigned int n);
+       int getCurrentTrack();
+
+               // iAudioChannelSelection       
+       int getCurrentChannel();
+       RESULT selectChannel(int i);
+
+       struct audioStream
+       {
+               GstPad* pad;
+               enum { atMP2, atMP3, atAC3, atDTS, atAAC };
+               int type; // mpeg2, ac3, dts, ...
+               std::string language_code; /* iso-639, if available. */
+       };

 private:
 private:

+       int m_currentAudioStream;
+       std::vector<audioStream> m_audioStreams;

        friend class eServiceFactoryMP3;
        std::string m_filename;
        friend class eServiceFactoryMP3;
        std::string m_filename;

-       eServiceMP3(const char *filename);      
+       eServiceMP3(const char *filename);

        Signal2<void,iPlayableService*,int> m_event;
        enum
        {
                stIdle, stRunning, stStopped,
        };
        int m_state;
        Signal2<void,iPlayableService*,int> m_event;
        enum
        {
                stIdle, stRunning, stStopped,
        };
        int m_state;

-       GstElement *m_gst_pipeline, *m_gst_audio, *m_gst_videoqueue, *m_gst_audioqueue, *m_decoder;
+       GstElement *m_gst_pipeline;

        GstTagList *m_stream_tags;
        eFixedMessagePump<int> m_pump;
        GstTagList *m_stream_tags;
        eFixedMessagePump<int> m_pump;

-       
+

        void gstBusCall(GstBus *bus, GstMessage *msg);
        static GstBusSyncReply gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data);
        static void gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer data); /* for mpegdemux */
        void gstBusCall(GstBus *bus, GstMessage *msg);
        static GstBusSyncReply gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data);
        static void gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer data); /* for mpegdemux */