X-Git-Url: https://git.cweiske.de/enigma2.git/blobdiff_plain/02da1f586ef7945fef385e6d4743ef53441bc2fa..8c811bc11d54848b4a35bba44e8ef1b8d6948806:/lib/service/servicemp3.cpp diff --git a/lib/service/servicemp3.cpp b/lib/service/servicemp3.cpp index a6d19287..f3f44177 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 #include #include @@ -6,23 +10,37 @@ #include #include #include +#include // eServiceFactoryMP3 -eServiceFactoryMP3::eServiceFactoryMP3(): ref(0) +eServiceFactoryMP3::eServiceFactoryMP3() { ePtr sc; - eServiceCenter::getInstance(sc); + eServiceCenter::getPrivInstance(sc); if (sc) - sc->addServiceFactory(eServiceFactoryMP3::id, this); + { + std::list 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"); + extensions.push_back("mkv"); + extensions.push_back("avi"); + sc->addServiceFactory(eServiceFactoryMP3::id, this, extensions); + } + + m_service_info = new eStaticServiceMP3Info(); } eServiceFactoryMP3::~eServiceFactoryMP3() { ePtr sc; - eServiceCenter::getInstance(sc); + eServiceCenter::getPrivInstance(sc); if (sc) sc->removeServiceFactory(eServiceFactoryMP3::id); } @@ -49,33 +67,318 @@ RESULT eServiceFactoryMP3::list(const eServiceReference &, ePtr &ptr) +{ + ptr = m_service_info; + return 0; +} + +RESULT eServiceFactoryMP3::offlineOperations(const eServiceReference &, ePtr &ptr) +{ + ptr = 0; + return -1; +} + + +// 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. -void eServiceMP3::test_end() +DEFINE_REF(eStaticServiceMP3Info) + +eStaticServiceMP3Info::eStaticServiceMP3Info() { - eDebug("end of mp3!"); - stop(); } -eServiceMP3::eServiceMP3(const char *filename): ref(0), filename(filename), test(eApp) +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_stream_tags = 0; + m_audioStreams.clear(); + 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 */ + + GstElement *audio = 0, *switch_audio = 0, *queue_audio = 0, *video = 0, *queue_video = 0, *videodemux = 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_matroska = !strcasecmp(ext, ".mkv"); + int is_avi = !strcasecmp(ext, ".avi"); + int is_mp3 = !strcasecmp(ext, ".mp3"); /* force mp3 instead of decodebin */ + int is_video = is_mpeg_ps || is_mpeg_ts || is_matroska || is_avi; + int is_streaming = !strncmp(filename, "http://", 7); + int is_AudioCD = !(strncmp(filename, "/autofs/", 8) || strncmp(filename+strlen(filename)-13, "/track-", 7) || strcasecmp(ext, ".wav")); + + eDebug("filename: %s, is_mpeg_ps: %d, is_mpeg_ts: %d, is_video: %d, is_streaming: %d, is_mp3: %d, is_matroska: %d, is_avi: %d, is_AudioCD: %d", filename, is_mpeg_ps, is_mpeg_ts, is_video, is_streaming, is_mp3, is_matroska, is_avi, is_AudioCD); + + int is_audio = !is_video; + + int all_ok = 0; + + m_gst_pipeline = gst_pipeline_new ("mediaplayer"); + if (!m_gst_pipeline) + eWarning("failed to create pipeline"); + + if (is_AudioCD) + { + source = gst_element_factory_make ("cdiocddasrc", "cda-source"); + if (source) + g_object_set (G_OBJECT (source), "device", "/dev/cdroms/cdrom0", NULL); + else + is_AudioCD = 0; + } + if ( !is_streaming && !is_AudioCD ) + source = gst_element_factory_make ("filesrc", "file-source"); + else if ( is_streaming ) + { + 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"); + /* configure source */ + else if (!is_AudioCD) + g_object_set (G_OBJECT (source), "location", filename, NULL); + else + { + int track = atoi(filename+18); + eDebug("play audio CD track #%i",track); + if (track > 0) + g_object_set (G_OBJECT (source), "track", track, 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", "audiosink"); + queue_audio = gst_element_factory_make("queue", "queue_audio"); + + video = gst_element_factory_make("dvbvideosink", "videosink"); + queue_video = gst_element_factory_make("queue", "queue_video"); + + if (is_mpeg_ps) + videodemux = gst_element_factory_make("flupsdemux", "videodemux"); + else if (is_mpeg_ts) + videodemux = gst_element_factory_make("flutsdemux", "videodemux"); + else if (is_matroska) + videodemux = gst_element_factory_make("matroskademux", "videodemux"); + else if (is_avi) + videodemux = gst_element_factory_make("avidemux", "videodemux"); + + if (!videodemux) + { + eDebug("fluendo mpegdemux not available, falling back to mpegdemux\n"); + videodemux = gst_element_factory_make("mpegdemux", "videodemux"); + } + + 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-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); + 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_AudioCD) + { + 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 (is_audio) + { + queue_audio = gst_element_factory_make("queue", "queue_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); + 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); + + 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); + 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 */ + audio = gst_bin_new ("audiobin"); + + GstPad *audiopad = gst_element_get_static_pad (conv, "sink"); + gst_bin_add_many(GST_BIN(audio), conv, flt, sink, (char*)0); + gst_element_link_many(conv, flt, sink, (char*)0); + gst_element_add_pad(audio, gst_ghost_pad_new ("sink", audiopad)); + gst_object_unref(audiopad); + 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) + gst_element_link(decoder, audio); + } else /* is_video */ + { + gst_bin_add_many(GST_BIN(m_gst_pipeline), source, videodemux, audio, queue_audio, video, queue_video, NULL); + switch_audio = gst_element_factory_make ("input-selector", "switch_audio"); + if (switch_audio) + { + g_object_set (G_OBJECT (switch_audio), "select-all", TRUE, NULL); + gst_bin_add(GST_BIN(m_gst_pipeline), switch_audio); + gst_element_link(switch_audio, queue_audio); + } + gst_element_link(source, videodemux); + gst_element_link(queue_audio, audio); + gst_element_link(queue_video, video); + g_signal_connect(videodemux, "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 (videodemux) + gst_object_unref(GST_OBJECT(videodemux)); + + 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); RESULT eServiceMP3::connectEvent(const Slot2 &event, ePtr &connection) { - connection = new eConnection(this, m_event.connect(event)); + connection = new eConnection((iPlayableService*)this, m_event.connect(event)); return 0; } @@ -84,11 +387,11 @@ 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; } @@ -98,25 +401,505 @@ RESULT eServiceMP3::stop() assert(m_state != stIdle); if (m_state == stStopped) return -1; - test.stop(); - printf("MP3: %s stop\n", 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; - m_event(this, evEnd); return 0; } -RESULT eServiceMP3::getIPausableService(ePtr &ptr) { ptr=this; return 0; } +RESULT eServiceMP3::setTarget(int target) +{ + return -1; +} +RESULT eServiceMP3::pause(ePtr &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::unpause() +{ + if (!m_gst_pipeline) + return -1; + 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::getIServiceInformation(ePtr&i) { i = this; 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&i) +{ + i = this; + return 0; +} -RESULT eServiceMP3::getName(eString &name) +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) +{ + gchar *tag = 0; + + switch (w) + { + case sTitle: + case sArtist: + case sAlbum: + case sComment: + case sTracknumber: + case sGenre: + case sVideoType: + 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) +{ + 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; + 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 ""; +} + +RESULT eServiceMP3::audioChannel(ePtr &ptr) +{ + ptr = this; + return 0; +} + +RESULT eServiceMP3::audioTracks(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) +{ + gint nb_sources; + GstPad *active_pad; + GstElement *selector = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_audio"); + if ( !selector) + { + eDebug("can't switch audio tracks! gst-plugin-selector needed"); + return -1; + } + 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); + 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 == 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) +{ + if (!msg) + return; + gchar *sourceName; + GstObject *source; + + source = GST_MESSAGE_SRC(msg); + sourceName = gst_object_get_name(source); + + 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)); + + 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_DECODE ) + { + if ( g_strrstr(sourceName, "videosink") ) + m_event((iPlayableService*)this, evUser+11); + } + 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; + } + gchar *g_audiocodec; + if (gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_audiocodec)) + { + std::vector::iterator IterAudioStream = m_audioStreams.begin(); + while ( IterAudioStream->language_code.length() && IterAudioStream != m_audioStreams.end()) + IterAudioStream++; + if ( g_strrstr(g_audiocodec, "MPEG-1 layer 2") ) + IterAudioStream->type = audioStream::atMP2; + else if ( g_strrstr(g_audiocodec, "AC-3 audio") ) + IterAudioStream->type = audioStream::atAC3; + gchar *g_language; + if ( gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_language) ) + IterAudioStream->language_code = std::string(g_language); + g_free (g_language); + g_free (g_audiocodec); + } + break; + } + 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; +} + +void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data) +{ + eServiceMP3 *_this = (eServiceMP3*)user_data; + GstBin *pipeline = GST_BIN(_this->m_gst_pipeline); + gchar *name; + name = gst_pad_get_name (pad); + 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 + { + GstElement *selector = gst_bin_get_by_name(pipeline , "switch_audio" ); + audioStream audio; + audio.pad = pad; + _this->m_audioStreams.push_back(audio); + if ( selector ) + { + GstPadLinkReturn ret = gst_pad_link(pad, gst_element_get_request_pad (selector, "sink%d")); + if ( _this->m_audioStreams.size() == 1 ) + { + _this->selectTrack(0); + gst_element_set_state (_this->m_gst_pipeline, GST_STATE_PLAYING); + } + else + g_object_set (G_OBJECT (selector), "select-all", FALSE, NULL); + } + else + gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_audio"), "sink")); + } + if (g_strrstr(name,"video")) + { + gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_video"), "sink")); + } + g_free (name); +} + +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 *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); + 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"); +#else +#warning gstreamer not available, not building media player +#endif