X-Git-Url: https://git.cweiske.de/enigma2.git/blobdiff_plain/1a0088bffb8804bebd31ac6377656191684e4cae..c6de231f8b5fbd3656d65a4a62b12cdfbd546171:/lib/service/servicemp3.cpp diff --git a/lib/service/servicemp3.cpp b/lib/service/servicemp3.cpp index 80356dc5..3e6de282 100644 --- a/lib/service/servicemp3.cpp +++ b/lib/service/servicemp3.cpp @@ -17,6 +17,12 @@ /* for subtitles */ #include +#ifndef GST_SEEK_FLAG_SKIP +#warning Compiling for legacy gstreamer, things will break +#define GST_SEEK_FLAG_SKIP 0 +#define GST_TAG_HOMEPAGE "" +#endif + // eServiceFactoryMP3 eServiceFactoryMP3::eServiceFactoryMP3() @@ -42,6 +48,7 @@ eServiceFactoryMP3::eServiceFactoryMP3() extensions.push_back("mp4"); extensions.push_back("mov"); extensions.push_back("m4a"); + extensions.push_back("m2ts"); sc->addServiceFactory(eServiceFactoryMP3::id, this, extensions); } @@ -63,7 +70,7 @@ DEFINE_REF(eServiceFactoryMP3) RESULT eServiceFactoryMP3::play(const eServiceReference &ref, ePtr &ptr) { // check resources... - ptr = new eServiceMP3(ref.path.c_str()); + ptr = new eServiceMP3(ref); return 0; } @@ -161,11 +168,16 @@ 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); + if ( ref.name.length() ) + name = ref.name; else - name = ref.path; + { + size_t last = ref.path.rfind('/'); + if (last != std::string::npos) + name = ref.path.substr(last+1); + else + name = ref.path; + } return 0; } @@ -176,27 +188,27 @@ int eStaticServiceMP3Info::getLength(const eServiceReference &ref) // eServiceMP3 -eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1) +eServiceMP3::eServiceMP3(eServiceReference ref) + :m_ref(ref), m_pump(eApp, 1) { m_seekTimeout = eTimer::create(eApp); + m_subtitle_sync_timer = eTimer::create(eApp); m_stream_tags = 0; m_currentAudioStream = 0; m_currentSubtitleStream = 0; m_subtitle_widget = 0; m_currentTrickRatio = 0; + m_subs_to_pull = 0; + m_buffer_size = 1*1024*1024; CONNECT(m_seekTimeout->timeout, eServiceMP3::seekTimeoutCB); + CONNECT(m_subtitle_sync_timer->timeout, eServiceMP3::pushSubtitles); 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. */ + eDebug("eServiceMP3::construct!"); + const char *filename = m_ref.path.c_str(); const char *ext = strrchr(filename, '.'); if (!ext) ext = filename; @@ -243,40 +255,19 @@ eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eAp sourceinfo.containertype = ctVCD; sourceinfo.is_video = TRUE; } - if ( (strncmp(filename, "http://", 7)) == 0 ) + if ( (strncmp(filename, "http://", 7)) == 0 || (strncmp(filename, "udp://", 6)) == 0 || (strncmp(filename, "rtp://", 6)) == 0 || (strncmp(filename, "https://", 8)) == 0 || (strncmp(filename, "mms://", 6)) == 0 || (strncmp(filename, "rtsp://", 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"; + gchar *uri; 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"; + uri = g_strdup_printf ("%s", filename); } 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); - } + int i_track = atoi(filename+18); + uri = g_strdup_printf ("cdda://%i", i_track); } else if ( sourceinfo.containertype == ctVCD ) { @@ -285,314 +276,67 @@ eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eAp 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); + uri = g_strdup_printf ("vcd://"); else - m_error_message = "GStreamer can't open filesrc " + (std::string)filename + "!\n"; + uri = g_strdup_printf ("file://%s", filename); } - if ( sourceinfo.is_video ) - { - /* filesrc -> mpegdemux -> | queue_audio -> dvbaudiosink - | queue_video -> dvbvideosink */ + else - audio = gst_element_factory_make("dvbaudiosink", "audiosink"); - if (!audio) - m_error_message += "failed to create Gstreamer element dvbaudiosink\n"; + uri = g_strdup_printf ("file://%s", filename); - video = gst_element_factory_make("dvbvideosink", "videosink"); - if (!video) - m_error_message += "failed to create Gstreamer element dvbvideosink\n"; + eDebug("eServiceMP3::playbin2 uri=%s", uri); - queue_audio = gst_element_factory_make("queue", "queue_audio"); - queue_video = gst_element_factory_make("queue", "queue_video"); + m_gst_playbin = gst_element_factory_make("playbin2", "playbin"); + if (!m_gst_playbin) + m_error_message = "failed to create GStreamer pipeline!\n"; - 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"; + g_object_set (G_OBJECT (m_gst_playbin), "uri", uri, NULL); - switch_audio = gst_element_factory_make ("input-selector", "switch_audio"); - if (!switch_audio) - m_error_message = "GStreamer plugin input-selector not available!\n"; + int flags = 0x47; // ( == GST_PLAY_FLAG_VIDEO | GST_PLAY_FLAG_AUDIO | GST_PLAY_FLAG_NATIVE_VIDEO | GST_PLAY_FLAG_TEXT ) + g_object_set (G_OBJECT (m_gst_playbin), "flags", flags, NULL); - 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; - } - } + g_free(uri); - } - if (m_gst_pipeline && all_ok) + GstElement *subsink = gst_element_factory_make("appsink", "subtitle_sink"); + if (!subsink) + eDebug("eServiceMP3::sorry, can't play: missing gst-plugin-appsink"); + else { - 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); + g_signal_connect (subsink, "new-buffer", G_CALLBACK (gstCBsubtitleAvail), this); + g_object_set (G_OBJECT (m_gst_playbin), "text-sink", subsink, NULL); + } - } else /* is audio*/ + if ( m_gst_playbin ) + { + gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_playbin)), gstBusSyncHandler, this); + 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) { - 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); - } + std::string suburi = "file://" + (std::string)srt_filename; + eDebug("eServiceMP3::subtitle uri: %s",suburi.c_str()); + g_object_set (G_OBJECT (m_gst_playbin), "suburi", suburi.c_str(), NULL); + subtitleStream subs; + subs.type = stSRT; + subs.language_code = std::string("und"); + m_subtitleStreams.push_back(subs); } } 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; + if (m_gst_playbin) + gst_object_unref(GST_OBJECT(m_gst_playbin)); + + eDebug("eServiceMP3::sorry, can't play: %s",m_error_message.c_str()); + m_gst_playbin = 0; } - gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING); + gst_element_set_state (m_gst_playbin, GST_STATE_PLAYING); + setBufferSize(m_buffer_size); } eServiceMP3::~eServiceMP3() @@ -604,10 +348,10 @@ eServiceMP3::~eServiceMP3() if (m_stream_tags) gst_tag_list_free(m_stream_tags); - if (m_gst_pipeline) + if (m_gst_playbin) { - gst_object_unref (GST_OBJECT (m_gst_pipeline)); - eDebug("SERVICEMP3 destruct!"); + gst_object_unref (GST_OBJECT (m_gst_playbin)); + eDebug("eServiceMP3::destruct!"); } } @@ -624,10 +368,10 @@ RESULT eServiceMP3::start() ASSERT(m_state == stIdle); m_state = stRunning; - if (m_gst_pipeline) + if (m_gst_playbin) { - eDebug("starting pipeline"); - gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING); + eDebug("eServiceMP3::starting pipeline"); + gst_element_set_state (m_gst_playbin, GST_STATE_PLAYING); } m_event(this, evStart); return 0; @@ -638,8 +382,8 @@ 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); + eDebug("eServiceMP3::stop %s", m_ref.path.c_str()); + gst_element_set_state(m_gst_playbin, GST_STATE_NULL); m_state = stStopped; return 0; } @@ -657,18 +401,16 @@ RESULT eServiceMP3::pause(ePtr &ptr) RESULT eServiceMP3::setSlowMotion(int ratio) { - /* we can't do slomo yet */ - return -1; + if (!ratio) + return 0; + eDebug("eServiceMP3::setSlowMotion ratio=%f",1/(float)ratio); + return trickSeek(1/(float)ratio); } RESULT eServiceMP3::setFastForward(int ratio) { - m_currentTrickRatio = ratio; - if (ratio) - m_seekTimeout->start(1000, 0); - else - m_seekTimeout->stop(); - return 0; + eDebug("eServiceMP3::setFastForward ratio=%i",ratio); + return trickSeek(ratio); } void eServiceMP3::seekTimeoutCB() @@ -696,12 +438,9 @@ void eServiceMP3::seekTimeoutCB() // iPausableService RESULT eServiceMP3::pause() { - if (m_state != stRunning) - return; - - if (!m_gst_pipeline) + if (!m_gst_playbin || m_state != stRunning) return -1; - GstStateChangeReturn res = gst_element_set_state(m_gst_pipeline, GST_STATE_PAUSED); + GstStateChangeReturn res = gst_element_set_state(m_gst_playbin, GST_STATE_PAUSED); if (res == GST_STATE_CHANGE_ASYNC) { pts_t ppos; @@ -713,14 +452,12 @@ RESULT eServiceMP3::pause() RESULT eServiceMP3::unpause() { - if (m_state != stRunning) - return; - - if (!m_gst_pipeline) + m_subtitle_pages.clear(); + if (!m_gst_playbin || m_state != stRunning) return -1; GstStateChangeReturn res; - res = gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING); + res = gst_element_set_state(m_gst_playbin, GST_STATE_PLAYING); return 0; } @@ -733,7 +470,7 @@ RESULT eServiceMP3::seek(ePtr &ptr) RESULT eServiceMP3::getLength(pts_t &pts) { - if (!m_gst_pipeline) + if (!m_gst_playbin) return -1; if (m_state != stRunning) return -1; @@ -741,9 +478,8 @@ RESULT eServiceMP3::getLength(pts_t &pts) GstFormat fmt = GST_FORMAT_TIME; gint64 len; - if (!gst_element_query_duration(m_gst_pipeline, &fmt, &len)) + if (!gst_element_query_duration(m_gst_playbin, &fmt, &len)) return -1; - /* len is in nanoseconds. we have 90 000 pts per second. */ pts = len / 11111; @@ -752,24 +488,71 @@ RESULT eServiceMP3::getLength(pts_t &pts) RESULT eServiceMP3::seekTo(pts_t to) { - if (!m_gst_pipeline) + if (!m_gst_playbin) 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, + if (!gst_element_seek (m_gst_playbin, 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"); + eDebug("eServiceMP3::seekTo failed"); + return -1; + } + + m_subtitle_pages.clear(); + eSingleLocker l(m_subs_to_pull_lock); + m_subs_to_pull = 0; + + return 0; +} + +RESULT eServiceMP3::trickSeek(gdouble ratio) +{ + if (!m_gst_playbin) + return -1; + if (!ratio) + return seekRelative(0, 0); + + GstEvent *s_event; + GstSeekFlags flags; + flags = GST_SEEK_FLAG_NONE; + flags |= GstSeekFlags (GST_SEEK_FLAG_FLUSH); +// flags |= GstSeekFlags (GST_SEEK_FLAG_ACCURATE); + flags |= GstSeekFlags (GST_SEEK_FLAG_KEY_UNIT); +// flags |= GstSeekFlags (GST_SEEK_FLAG_SEGMENT); +// flags |= GstSeekFlags (GST_SEEK_FLAG_SKIP); + + GstFormat fmt = GST_FORMAT_TIME; + gint64 pos, len; + gst_element_query_duration(m_gst_playbin, &fmt, &len); + gst_element_query_position(m_gst_playbin, &fmt, &pos); + + if ( ratio >= 0 ) + { + s_event = gst_event_new_seek (ratio, GST_FORMAT_TIME, flags, GST_SEEK_TYPE_SET, pos, GST_SEEK_TYPE_SET, len); + + eDebug("eServiceMP3::trickSeek with rate %lf to %" GST_TIME_FORMAT " ", ratio, GST_TIME_ARGS (pos)); + } + else + { + s_event = gst_event_new_seek (ratio, GST_FORMAT_TIME, GST_SEEK_FLAG_SKIP|GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_NONE, -1, GST_SEEK_TYPE_NONE, -1); + } + + if (!gst_element_send_event ( GST_ELEMENT (m_gst_playbin), s_event)) + { + eDebug("eServiceMP3::trickSeek failed"); return -1; } + return 0; } + RESULT eServiceMP3::seekRelative(int direction, pts_t to) { - if (!m_gst_pipeline) + if (!m_gst_playbin) return -1; pts_t ppos; @@ -784,19 +567,40 @@ RESULT eServiceMP3::seekRelative(int direction, pts_t to) RESULT eServiceMP3::getPlayPosition(pts_t &pts) { - if (!m_gst_pipeline) + GstFormat fmt = GST_FORMAT_TIME; + gint64 pos; + GstElement *sink; + pts = 0; + + if (!m_gst_playbin) 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)) + g_object_get (G_OBJECT (m_gst_playbin), "audio-sink", &sink, NULL); + + if (!sink) + g_object_get (G_OBJECT (m_gst_playbin), "video-sink", &sink, NULL); + + if (!sink) return -1; - /* len is in nanoseconds. we have 90 000 pts per second. */ - pts = len / 11111; + gchar *name = gst_element_get_name(sink); + gboolean use_get_decoder_time = strstr(name, "dvbaudiosink") || strstr(name, "dvbvideosink"); + g_free(name); + + if (use_get_decoder_time) + g_signal_emit_by_name(sink, "get-decoder-time", &pos); + + gst_object_unref(sink); + + if (!use_get_decoder_time && !gst_element_query_position(m_gst_playbin, &fmt, &pos)) { + eDebug("gst_element_query_position failed in getPlayPosition"); + return -1; + } + + /* pos is in nanoseconds. we have 90 000 pts per second. */ + pts = pos / 11111; return 0; } @@ -819,41 +623,106 @@ RESULT eServiceMP3::info(ePtr&i) RESULT eServiceMP3::getName(std::string &name) { - name = m_filename; - size_t n = name.rfind('/'); - if (n != std::string::npos) - name = name.substr(n + 1); + std::string title = m_ref.getName(); + if (title.empty()) + { + name = m_ref.path; + size_t n = name.rfind('/'); + if (n != std::string::npos) + name = name.substr(n + 1); + } + else + name = title; return 0; } + int eServiceMP3::getInfo(int w) { - gchar *tag = 0; + const gchar *tag = 0; switch (w) { + case sServiceref: return m_ref; 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 sTagTitle: + case sTagArtist: + case sTagAlbum: + case sTagTitleSortname: + case sTagArtistSortname: + case sTagAlbumSortname: + case sTagDate: + case sTagComposer: + case sTagGenre: + case sTagComment: + case sTagExtendedComment: + case sTagLocation: + case sTagHomepage: + case sTagDescription: + case sTagVersion: + case sTagISRC: + case sTagOrganization: + case sTagCopyright: + case sTagCopyrightURI: + case sTagContact: + case sTagLicense: + case sTagLicenseURI: + case sTagCodec: + case sTagAudioCodec: + case sTagVideoCodec: + case sTagEncoder: + case sTagLanguageCode: + case sTagKeywords: + case sTagChannelMode: case sUser+12: return resIsString; - case sCurrentTitle: + case sTagTrackGain: + case sTagTrackPeak: + case sTagAlbumGain: + case sTagAlbumPeak: + case sTagReferenceLevel: + case sTagBeatsPerMinute: + case sTagImage: + case sTagPreviewImage: + case sTagAttachment: + return resIsPyObject; + case sTagTrackNumber: tag = GST_TAG_TRACK_NUMBER; break; - case sTotalTitles: + case sTagTrackCount: tag = GST_TAG_TRACK_COUNT; break; + case sTagAlbumVolumeNumber: + tag = GST_TAG_ALBUM_VOLUME_NUMBER; + break; + case sTagAlbumVolumeCount: + tag = GST_TAG_ALBUM_VOLUME_COUNT; + break; + case sTagBitrate: + tag = GST_TAG_BITRATE; + break; + case sTagNominalBitrate: + tag = GST_TAG_NOMINAL_BITRATE; + break; + case sTagMinimumBitrate: + tag = GST_TAG_MINIMUM_BITRATE; + break; + case sTagMaximumBitrate: + tag = GST_TAG_MAXIMUM_BITRATE; + break; + case sTagSerial: + tag = GST_TAG_SERIAL; + break; + case sTagEncoderVersion: + tag = GST_TAG_ENCODER_VERSION; + break; + case sTagCRC: + tag = "has-crc"; + break; default: return resNA; } @@ -864,51 +733,110 @@ int eServiceMP3::getInfo(int w) guint value; if (gst_tag_list_get_uint(m_stream_tags, tag, &value)) return (int) value; - - return 0; + return 0; } std::string eServiceMP3::getInfoString(int w) { - if ( !m_stream_tags ) + if ( !m_stream_tags && w < sUser && w > 26 ) return ""; - gchar *tag = 0; + const gchar *tag = 0; switch (w) { - case sTitle: + case sTagTitle: tag = GST_TAG_TITLE; break; - case sArtist: + case sTagArtist: tag = GST_TAG_ARTIST; break; - case sAlbum: + case sTagAlbum: tag = GST_TAG_ALBUM; break; - case sComment: - tag = GST_TAG_COMMENT; - break; - case sTracknumber: - tag = GST_TAG_TRACK_NUMBER; + case sTagTitleSortname: + tag = GST_TAG_TITLE_SORTNAME; break; - case sGenre: - tag = GST_TAG_GENRE; - break; - case sUser+10: - tag = GST_TAG_AUDIO_CODEC; + case sTagArtistSortname: + tag = GST_TAG_ARTIST_SORTNAME; break; - case sVideoType: - tag = GST_TAG_VIDEO_CODEC; + case sTagAlbumSortname: + tag = GST_TAG_ALBUM_SORTNAME; break; - case sTimeCreate: + case sTagDate: 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); + g_date_strftime (res, sizeof(res), "%Y-%M-%D", date); return (std::string)res; } break; + case sTagComposer: + tag = GST_TAG_COMPOSER; + break; + case sTagGenre: + tag = GST_TAG_GENRE; + break; + case sTagComment: + tag = GST_TAG_COMMENT; + break; + case sTagExtendedComment: + tag = GST_TAG_EXTENDED_COMMENT; + break; + case sTagLocation: + tag = GST_TAG_LOCATION; + break; + case sTagHomepage: + tag = GST_TAG_HOMEPAGE; + break; + case sTagDescription: + tag = GST_TAG_DESCRIPTION; + break; + case sTagVersion: + tag = GST_TAG_VERSION; + break; + case sTagISRC: + tag = GST_TAG_ISRC; + break; + case sTagOrganization: + tag = GST_TAG_ORGANIZATION; + break; + case sTagCopyright: + tag = GST_TAG_COPYRIGHT; + break; + case sTagCopyrightURI: + tag = GST_TAG_COPYRIGHT_URI; + break; + case sTagContact: + tag = GST_TAG_CONTACT; + break; + case sTagLicense: + tag = GST_TAG_LICENSE; + break; + case sTagLicenseURI: + tag = GST_TAG_LICENSE_URI; + break; + case sTagCodec: + tag = GST_TAG_CODEC; + break; + case sTagAudioCodec: + tag = GST_TAG_AUDIO_CODEC; + break; + case sTagVideoCodec: + tag = GST_TAG_VIDEO_CODEC; + break; + case sTagEncoder: + tag = GST_TAG_ENCODER; + break; + case sTagLanguageCode: + tag = GST_TAG_LANGUAGE_CODE; + break; + case sTagKeywords: + tag = GST_TAG_KEYWORDS; + break; + case sTagChannelMode: + tag = "channel-mode"; + break; case sUser+12: return m_error_message; default: @@ -926,6 +854,68 @@ std::string eServiceMP3::getInfoString(int w) return ""; } +PyObject *eServiceMP3::getInfoObject(int w) +{ + const gchar *tag = 0; + bool isBuffer = false; + switch (w) + { + case sTagTrackGain: + tag = GST_TAG_TRACK_GAIN; + break; + case sTagTrackPeak: + tag = GST_TAG_TRACK_PEAK; + break; + case sTagAlbumGain: + tag = GST_TAG_ALBUM_GAIN; + break; + case sTagAlbumPeak: + tag = GST_TAG_ALBUM_PEAK; + break; + case sTagReferenceLevel: + tag = GST_TAG_REFERENCE_LEVEL; + break; + case sTagBeatsPerMinute: + tag = GST_TAG_BEATS_PER_MINUTE; + break; + case sTagImage: + tag = GST_TAG_IMAGE; + isBuffer = true; + break; + case sTagPreviewImage: + tag = GST_TAG_PREVIEW_IMAGE; + isBuffer = true; + break; + case sTagAttachment: + tag = GST_TAG_ATTACHMENT; + isBuffer = true; + break; + default: + break; + } + gdouble value; + if ( !tag || !m_stream_tags ) + value = 0.0; + PyObject *pyValue; + if ( isBuffer ) + { + const GValue *gv_buffer = gst_tag_list_get_value_index(m_stream_tags, tag, 0); + if ( gv_buffer ) + { + GstBuffer *buffer; + buffer = gst_value_get_buffer (gv_buffer); + pyValue = PyBuffer_FromMemory(GST_BUFFER_DATA(buffer), GST_BUFFER_SIZE(buffer)); + } + } + else + { + gst_tag_list_get_double(m_stream_tags, tag, &value); + pyValue = PyFloat_FromDouble(value); + } + + return pyValue; +} + RESULT eServiceMP3::audioChannel(ePtr &ptr) { ptr = this; @@ -967,27 +957,16 @@ RESULT eServiceMP3::selectTrack(unsigned int i) 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 ) + int current_audio; + g_object_set (G_OBJECT (m_gst_playbin), "current-audio", i, NULL); + g_object_get (G_OBJECT (m_gst_playbin), "current-audio", ¤t_audio, NULL); + if ( current_audio == i ) { - eDebug("can't switch audio tracks! gst-plugin-selector needed"); - return -1; + eDebug ("eServiceMP3::switched to audio stream %i", current_audio); + m_currentAudioStream = i; + return 0; } - 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; + return -1; } int eServiceMP3::getCurrentChannel() @@ -1003,10 +982,10 @@ RESULT eServiceMP3::selectChannel(int i) 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 = m_audioStreams[i].codec; +/* if (m_audioStreams[i].type == atMPEG) info.m_description = "MPEG"; else if (m_audioStreams[i].type == atMP3) info.m_description = "MP3"; @@ -1020,8 +999,10 @@ RESULT eServiceMP3::getTrackInfo(struct iAudioTrackInfo &info, unsigned int i) info.m_description = "PCM"; else if (m_audioStreams[i].type == atOGG) info.m_description = "OGG"; + else if (m_audioStreams[i].type == atFLAC) + info.m_description = "FLAC"; else - info.m_description = "???"; + info.m_description = "???";*/ if (info.m_language.empty()) info.m_language = m_audioStreams[i].language_code; return 0; @@ -1036,31 +1017,79 @@ void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg) source = GST_MESSAGE_SRC(msg); sourceName = gst_object_get_name(source); -#if 1 +#if 0 if (gst_message_get_structure(msg)) { gchar *string = gst_structure_to_string(gst_message_get_structure(msg)); - eDebug("gst_message from %s: %s", sourceName, string); + eDebug("eServiceMP3::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)); + eDebug("eServiceMP3::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_STATE_CHANGED: + { + if(GST_MESSAGE_SRC(msg) != GST_OBJECT(m_gst_playbin)) + break; + + GstState old_state, new_state; + gst_message_parse_state_changed(msg, &old_state, &new_state, NULL); + + if(old_state == new_state) + break; + + eDebug("eServiceMP3::state transition %s -> %s", gst_element_state_get_name(old_state), gst_element_state_get_name(new_state)); + + GstStateChange transition = (GstStateChange)GST_STATE_TRANSITION(old_state, new_state); + + switch(transition) + { + case GST_STATE_CHANGE_NULL_TO_READY: + { + } break; + case GST_STATE_CHANGE_READY_TO_PAUSED: + { + GstElement *sink; + g_object_get (G_OBJECT (m_gst_playbin), "text-sink", &sink, NULL); + if (sink) + { + g_object_set (G_OBJECT (sink), "max-buffers", 2, NULL); + g_object_set (G_OBJECT (sink), "sync", FALSE, NULL); + g_object_set (G_OBJECT (sink), "async", FALSE, NULL); + g_object_set (G_OBJECT (sink), "emit-signals", TRUE, NULL); + gst_object_unref(sink); + } + } break; + case GST_STATE_CHANGE_PAUSED_TO_PLAYING: + { + } break; + case GST_STATE_CHANGE_PLAYING_TO_PAUSED: + { + } break; + case GST_STATE_CHANGE_PAUSED_TO_READY: + { + } break; + case GST_STATE_CHANGE_READY_TO_NULL: + { + } break; + } + 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 ) { + eDebug("err->code %d", err->code); if ( err->code == GST_STREAM_ERROR_CODEC_NOT_FOUND ) { if ( g_strrstr(sourceName, "videosink") ) @@ -1068,16 +1097,6 @@ void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg) 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; @@ -1102,7 +1121,7 @@ void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg) GstTagList *tags, *result; gst_message_parse_tag(msg, &tags); - result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_PREPEND); + result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_REPLACE); if (result) { if (m_stream_tags) @@ -1110,65 +1129,94 @@ void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg) 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)); + int ret = write(fd, GST_BUFFER_DATA(buf_image), GST_BUFFER_SIZE(buf_image)); close(fd); + eDebug("eServiceMP3::/tmp/.id3coverart %d bytes written ", ret); m_event((iPlayableService*)this, evUser+13); } - gst_tag_list_free(tags); m_event((iPlayableService*)this, evUpdatedInfo); break; } case GST_MESSAGE_ASYNC_DONE: { + if(GST_MESSAGE_SRC(msg) != GST_OBJECT(m_gst_playbin)) + break; + GstTagList *tags; - for (std::vector::iterator IterAudioStream(m_audioStreams.begin()); IterAudioStream != m_audioStreams.end(); ++IterAudioStream) + gint i, active_idx, n_video = 0, n_audio = 0, n_text = 0; + + g_object_get (m_gst_playbin, "n-video", &n_video, NULL); + g_object_get (m_gst_playbin, "n-audio", &n_audio, NULL); + g_object_get (m_gst_playbin, "n-text", &n_text, NULL); + + eDebug("eServiceMP3::async-done - %d video, %d audio, %d subtitle", n_video, n_audio, n_text); + + active_idx = 0; + + m_audioStreams.clear(); + m_subtitleStreams.clear(); + + for (i = 0; i < n_audio; i++) { - if ( IterAudioStream->pad ) + audioStream audio; + gchar *g_codec, *g_lang; + GstPad* pad = 0; + g_signal_emit_by_name (m_gst_playbin, "get-audio-pad", i, &pad); + GstCaps* caps = gst_pad_get_negotiated_caps(pad); + if (!caps) + continue; + GstStructure* str = gst_caps_get_structure(caps, 0); + gchar *g_type; + g_type = gst_structure_get_name(str); + eDebug("AUDIO STRUCT=%s", g_type); + audio.type = gstCheckAudioPad(str); + g_codec = g_strdup(g_type); + g_lang = g_strdup_printf ("und"); + g_signal_emit_by_name (m_gst_playbin, "get-audio-tags", i, &tags); + if ( tags && gst_is_tag_list(tags) ) { - 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); - } + gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_codec); + gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_lang); + gst_tag_list_free(tags); } + audio.language_code = std::string(g_lang); + audio.codec = std::string(g_codec); + eDebug("eServiceMP3::audio stream=%i codec=%s language=%s", i, g_codec, g_lang); + m_audioStreams.push_back(audio); + g_free (g_lang); + g_free (g_codec); + gst_caps_unref(caps); } - 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); - } - } + + for (i = 0; i < n_text; i++) + { + gchar *g_lang; +// gchar *g_type; +// GstPad* pad = 0; +// g_signal_emit_by_name (m_gst_playbin, "get-text-pad", i, &pad); +// GstCaps* caps = gst_pad_get_negotiated_caps(pad); +// GstStructure* str = gst_caps_get_structure(caps, 0); +// g_type = gst_structure_get_name(str); +// g_signal_emit_by_name (m_gst_playbin, "get-text-tags", i, &tags); + subtitleStream subs; + subs.type = stPlainText; + g_lang = g_strdup_printf ("und"); + if ( tags && gst_is_tag_list(tags) ) + gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_lang); + subs.language_code = std::string(g_lang); + eDebug("eServiceMP3::subtitle stream=%i language=%s"/* type=%s*/, i, g_lang/*, g_type*/); + m_subtitleStreams.push_back(subs); + g_free (g_lang); +// g_free (g_type); } + m_event((iPlayableService*)this, evUpdatedEventInfo); } case GST_MESSAGE_ELEMENT: { @@ -1209,6 +1257,14 @@ void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg) } } } + break; + } + case GST_MESSAGE_BUFFERING: + { + GstBufferingMode mode; + gst_message_parse_buffering(msg, &(m_bufferInfo.bufferPercent)); + gst_message_parse_buffering_stats(msg, &mode, &(m_bufferInfo.avgInRate), &(m_bufferInfo.avgOutRate), &(m_bufferInfo.bufferingLeft)); + m_event((iPlayableService*)this, evBuffering); } default: break; @@ -1226,261 +1282,170 @@ GstBusSyncReply eServiceMP3::gstBusSyncHandler(GstBus *bus, GstMessage *message, 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 (!structure) + return atUnknown; + + if ( gst_structure_has_name (structure, "audio/mpeg")) + { + gint mpegversion, layer = -1; + if (!gst_structure_get_int (structure, "mpegversion", &mpegversion)) + return atUnknown; + + switch (mpegversion) { + case 1: { + gst_structure_get_int (structure, "layer", &layer); if ( layer == 3 ) return atMP3; else return atMPEG; + break; } - case 2: - return atMPEG; - case 4: - return atAAC; - default: - return atUnknown; - } + case 2: + return atAAC; + 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; } + + else if ( gst_structure_has_name (structure, "audio/x-ac3") || gst_structure_has_name (structure, "audio/ac3") ) + return atAC3; + else if ( gst_structure_has_name (structure, "audio/x-dts") || gst_structure_has_name (structure, "audio/dts") ) + return atDTS; + else if ( gst_structure_has_name (structure, "audio/x-raw-int") ) + return atPCM; + return atUnknown; } -void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data) +void eServiceMP3::gstPoll(const int &msg) { - 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")) + /* 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. */ + if (msg == 1) { - audioStream audio; - audio.type = _this->gstCheckAudioPad(str); - GstElement *switch_audio = gst_bin_get_by_name(pipeline , "switch_audio"); - if ( switch_audio ) + GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (m_gst_playbin)); + GstMessage *message; + usleep(1); + while ((message = gst_bus_pop (bus))) { - 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")); + gstBusCall(bus, message); + gst_message_unref (message); } } - 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"); + else + pullSubtitle(); } -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")); -} +eAutoInitPtr init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3"); -void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last, gpointer user_data) +void eServiceMP3::gstCBsubtitleAvail(GstElement *appsink, 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); + eSingleLocker l(_this->m_subs_to_pull_lock); + ++_this->m_subs_to_pull; + _this->m_pump.send(2); } -void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data) +void eServiceMP3::pullSubtitle() { - 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))) + GstElement *sink; + g_object_get (G_OBJECT (m_gst_playbin), "text-sink", &sink, NULL); + if (sink) { - gstBusCall(bus, message); - gst_message_unref (message); + while (m_subs_to_pull && m_subtitle_pages.size() < 2) + { + GstBuffer *buffer; + { + eSingleLocker l(m_subs_to_pull_lock); + --m_subs_to_pull; + } + g_signal_emit_by_name (sink, "pull-buffer", &buffer); + if (buffer) + { + gint64 buf_pos = GST_BUFFER_TIMESTAMP(buffer); + gint64 duration_ns = GST_BUFFER_DURATION(buffer); + size_t len = GST_BUFFER_SIZE(buffer); + unsigned char line[len+1]; + memcpy(line, GST_BUFFER_DATA(buffer), len); + line[len] = 0; + eDebug("got new subtitle @ buf_pos = %lld ns (in pts=%lld): '%s' ", buf_pos, buf_pos/11111, line); + ePangoSubtitlePage page; + gRGB rgbcol(0xD0,0xD0,0xD0); + page.m_elements.push_back(ePangoSubtitlePageElement(rgbcol, (const char*)line)); + page.show_pts = buf_pos / 11111L; + page.m_timeout = duration_ns / 1000000; + m_subtitle_pages.push_back(page); + pushSubtitles(); + gst_buffer_unref(buffer); + } + } + gst_object_unref(sink); } + else + eDebug("no subtitle sink!"); } -eAutoInitPtr init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3"); - -void eServiceMP3::gstCBsubtitleAvail(GstElement *element, GstBuffer *buffer, GstPad *pad, gpointer user_data) +void eServiceMP3::pushSubtitles() { - 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; + pts_t running_pts; + while ( !m_subtitle_pages.empty() ) { - 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); + getPlayPosition(running_pts); + page = m_subtitle_pages.front(); + gint64 diff_ms = ( page.show_pts - running_pts ) / 90; + eDebug("eServiceMP3::pushSubtitles show_pts = %lld running_pts = %lld diff = %lld", page.show_pts, running_pts, diff_ms); + if (diff_ms < -100) + { + GstFormat fmt = GST_FORMAT_TIME; + gint64 now; + if (gst_element_query_position(m_gst_playbin, &fmt, &now) != -1) + { + now /= 11111; + diff_ms = abs((now - running_pts) / 90); + eDebug("diff < -100ms check decoder/pipeline diff: decoder: %lld, pipeline: %lld, diff: %lld", running_pts, now, diff_ms); + if (diff_ms > 100000) + { + eDebug("high decoder/pipeline difference.. assume decoder has now started yet.. check again in 1sec"); + m_subtitle_sync_timer->start(1000, true); + break; + } + } + else + eDebug("query position for decoder/pipeline check failed!"); + eDebug("subtitle to late... drop"); + m_subtitle_pages.pop_front(); + } + else if ( diff_ms > 20 ) + { +// eDebug("start recheck timer"); + m_subtitle_sync_timer->start(diff_ms > 1000 ? 1000 : diff_ms, true); + break; + } + else // immediate show + { + if (m_subtitle_widget) + m_subtitle_widget->setPage(page); + m_subtitle_pages.pop_front(); + } } + if (m_subtitle_pages.empty()) + pullSubtitle(); } 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"); + int pid, type; + gint text_pid = 0; if (!PyTuple_Check(tuple)) goto error_out; @@ -1495,46 +1460,28 @@ RESULT eServiceMP3::enableSubtitles(eWidget *parent, ePyObject tuple) goto error_out; type = PyInt_AsLong(entry); - switch ((subtype_t)type) + if (m_currentSubtitleStream != pid) { - 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; + g_object_set (G_OBJECT (m_gst_playbin), "current-text", pid, NULL); + m_currentSubtitleStream = pid; + eSingleLocker l(m_subs_to_pull_lock); + m_subs_to_pull = 0; + m_subtitle_pages.clear(); } + m_subtitle_widget = 0; 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; + g_object_get (G_OBJECT (m_gst_playbin), "current-text", &text_pid, NULL); + + eDebug ("eServiceMP3::switched to subtitle stream %i", text_pid); - 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" + eDebug("eServiceMP3::enableSubtitles needs a tuple as 2nd argument!\n" "for gst subtitles (2, subtitle_stream_count, subtitle_type)"); return -1; } @@ -1542,6 +1489,7 @@ error_out: RESULT eServiceMP3::disableSubtitles(eWidget *parent) { eDebug("eServiceMP3::disableSubtitles"); + m_subtitle_pages.clear(); delete m_subtitle_widget; m_subtitle_widget = 0; return 0; @@ -1549,7 +1497,7 @@ RESULT eServiceMP3::disableSubtitles(eWidget *parent) PyObject *eServiceMP3::getCachedSubtitle() { - eDebug("eServiceMP3::getCachedSubtitle"); +// eDebug("eServiceMP3::getCachedSubtitle"); Py_RETURN_NONE; } @@ -1578,6 +1526,31 @@ PyObject *eServiceMP3::getSubtitleList() return l; } +RESULT eServiceMP3::streamed(ePtr &ptr) +{ + ptr = this; + return 0; +} + +PyObject *eServiceMP3::getBufferCharge() +{ + ePyObject tuple = PyTuple_New(5); + PyTuple_SET_ITEM(tuple, 0, PyInt_FromLong(m_bufferInfo.bufferPercent)); + PyTuple_SET_ITEM(tuple, 1, PyInt_FromLong(m_bufferInfo.avgInRate)); + PyTuple_SET_ITEM(tuple, 2, PyInt_FromLong(m_bufferInfo.avgOutRate)); + PyTuple_SET_ITEM(tuple, 3, PyInt_FromLong(m_bufferInfo.bufferingLeft)); + PyTuple_SET_ITEM(tuple, 4, PyInt_FromLong(m_buffer_size)); + return tuple; +} + +int eServiceMP3::setBufferSize(int size) +{ + m_buffer_size = size; + g_object_set (G_OBJECT (m_gst_playbin), "buffer-size", m_buffer_size, NULL); + return 0; +} + + #else #warning gstreamer not available, not building media player #endif