#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 #include #include #include #include #include #include #include #include #include #include #include /* for subtitles */ #include // eServiceFactoryMP3 eServiceFactoryMP3::eServiceFactoryMP3() { ePtr sc; eServiceCenter::getPrivInstance(sc); if (sc) { std::list 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("mov"); extensions.push_back("m4a"); sc->addServiceFactory(eServiceFactoryMP3::id, this, extensions); } m_service_info = new eStaticServiceMP3Info(); } eServiceFactoryMP3::~eServiceFactoryMP3() { ePtr sc; eServiceCenter::getPrivInstance(sc); if (sc) sc->removeServiceFactory(eServiceFactoryMP3::id); } DEFINE_REF(eServiceFactoryMP3) // iServiceHandler RESULT eServiceFactoryMP3::play(const eServiceReference &ref, ePtr &ptr) { // check resources... ptr = new eServiceMP3(ref.path.c_str()); return 0; } RESULT eServiceFactoryMP3::record(const eServiceReference &ref, ePtr &ptr) { ptr=0; return -1; } RESULT eServiceFactoryMP3::list(const eServiceReference &, ePtr &ptr) { ptr=0; return -1; } RESULT eServiceFactoryMP3::info(const eServiceReference &ref, ePtr &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 &); }; 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 res; if (getListOfFilenames(res)) return -1; eBackgroundFileEraser *eraser = eBackgroundFileEraser::getInstance(); if (!eraser) eDebug("FATAL !! can't get background file eraser"); for (std::list::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 &res) { res.clear(); res.push_back(m_ref.path); return 0; } RESULT eServiceFactoryMP3::offlineOperations(const eServiceReference &ref, ePtr &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): m_filename(filename), m_pump(eApp, 1) { 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_aspect = m_width = m_height = m_framerate = m_progressive = -1; 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 || strcasecmp(ext, ".mov") == 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); eDebug("servicemp3: this is a 'REAL' video cd... we use vcdsrc !"); } } } 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 = "mpegtsdemux"; break; case ctMPEGPS: case ctVCD: demux_type = "mpegpsdemux"; 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("servicemp3: 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() { 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 &event, ePtr &connection) { connection = new eConnection((iPlayableService*)this, m_event.connect(event)); return 0; } RESULT eServiceMP3::start() { ASSERT(m_state == stIdle); 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 &ptr) { ptr=this; return 0; } RESULT eServiceMP3::setSlowMotion(int ratio) { /* 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; } 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() { if (m_state != stRunning) return; 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_state != stRunning) return; 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 &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&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 sVideoHeight: return m_height; case sVideoWidth: return m_width; case sFrameRate: return m_framerate; case sProgressive: return m_progressive; case sAspect: return m_aspect; case sTitle: case sArtist: case sAlbum: case sComment: case sTracknumber: case sGenre: case sVideoType: case sTimeCreate: case sUser+10: 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 sUser+10: tag = GST_TAG_AUDIO_CODEC; 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 &ptr) { ptr = this; return 0; } RESULT eServiceMP3::audioTracks(ePtr &ptr) { ptr = this; return 0; } RESULT eServiceMP3::subtitle(ePtr &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 1 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) from %s", err->message, err->code, sourceName ); if ( err->domain == GST_STREAM_ERROR ) { if ( err->code == GST_STREAM_ERROR_CODEC_NOT_FOUND ) { if ( g_strrstr(sourceName, "videosink") ) m_event((iPlayableService*)this, evUser+11); else if ( g_strrstr(sourceName, "audiosink") ) m_event((iPlayableService*)this, evUser+10); } else if ( err->code == GST_STREAM_ERROR_FAILED && g_strrstr(sourceName, "file-source") ) { eWarning("error in tag parsing, linking mp3parse directly to file-sink, bypassing id3demux..."); GstElement *source = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"file-source"); GstElement *parser = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"audiosink"); gst_element_set_state(m_gst_pipeline, GST_STATE_NULL); gst_element_unlink(source, gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"id3demux")); gst_element_link(source, parser); gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING); } } 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::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::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 = gst_structure_get_name(msgstruct); if ( eventname ) { if (!strcmp(eventname, "eventSizeChanged") || !strcmp(eventname, "eventSizeAvail")) { gst_structure_get_int (msgstruct, "aspect_ratio", &m_aspect); gst_structure_get_int (msgstruct, "width", &m_width); gst_structure_get_int (msgstruct, "height", &m_height); if (strstr(eventname, "Changed")) m_event((iPlayableService*)this, evVideoSizeChanged); } else if (!strcmp(eventname, "eventFrameRateChanged") || !strcmp(eventname, "eventFrameRateAvail")) { gst_structure_get_int (msgstruct, "frame_rate", &m_framerate); if (strstr(eventname, "Changed")) m_event((iPlayableService*)this, evVideoFramerateChanged); } else if (!strcmp(eventname, "eventProgressiveChanged") || !strcmp(eventname, "eventProgressiveAvail")) { gst_structure_get_int (msgstruct, "progressive", &m_progressive); if (strstr(eventname, "Changed")) m_event((iPlayableService*)this, evVideoProgressiveChanged); } } } } 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 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); size_t len = GST_BUFFER_SIZE(buffer); unsigned char tmp[len+1]; memcpy(tmp, GST_BUFFER_DATA(buffer), len); tmp[len] = 0; eDebug("gstCBsubtitleAvail: %s", tmp); 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*)tmp)); 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::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; } #else #warning gstreamer not available, not building media player #endif