static void _check_message_received (GstBus * bus, GstMessage * message, gpointer user_data) { GMainLoop *main_loop = (GMainLoop *) user_data; GST_DEBUG_OBJECT (GST_MESSAGE_SRC (message), "%s: %" GST_PTR_FORMAT, GST_MESSAGE_TYPE_NAME (message), message); switch (GST_MESSAGE_TYPE (message)) { case GST_MESSAGE_STATE_CHANGED: if (GST_IS_PIPELINE (GST_MESSAGE_SRC (message))) { GstState oldstate, newstate, pending; gst_message_parse_state_changed (message, &oldstate, &newstate, &pending); switch (GST_STATE_TRANSITION (oldstate, newstate)) { case GST_STATE_CHANGE_PAUSED_TO_PLAYING: goto done; break; default: break; } } break; case GST_MESSAGE_ERROR: GST_WARNING_OBJECT (GST_MESSAGE_SRC (message), "error: %" GST_PTR_FORMAT, message); _check_run_main_loop_error = TRUE; goto done; case GST_MESSAGE_SEGMENT_DONE: case GST_MESSAGE_EOS: if (GST_IS_PIPELINE (GST_MESSAGE_SRC (message))) { GST_INFO_OBJECT (GST_MESSAGE_SRC (message), "%s: %" GST_PTR_FORMAT, GST_MESSAGE_TYPE_NAME (message), message); goto done; } break; default: GST_WARNING_OBJECT (GST_MESSAGE_SRC (message), "unexpected messges: %s", GST_MESSAGE_TYPE_NAME (message)); } return; done: if (g_main_loop_is_running (main_loop)) g_main_loop_quit (main_loop); }
static gboolean gst_switchui_handle_message (GstBus * bus, GstMessage * message, gpointer data) { GstSwitchUI *switchui = (GstSwitchUI *) data; switch (GST_MESSAGE_TYPE (message)) { case GST_MESSAGE_EOS: gst_switchui_handle_eos (switchui); break; case GST_MESSAGE_ERROR: { GError *error = NULL; gchar *debug; gst_message_parse_error (message, &error, &debug); gst_switchui_handle_error (switchui, error, debug); } break; case GST_MESSAGE_WARNING: { GError *error = NULL; gchar *debug; gst_message_parse_warning (message, &error, &debug); gst_switchui_handle_warning (switchui, error, debug); } break; case GST_MESSAGE_INFO: { GError *error = NULL; gchar *debug; gst_message_parse_info (message, &error, &debug); gst_switchui_handle_info (switchui, error, debug); } break; case GST_MESSAGE_TAG: { GstTagList *tag_list; gst_message_parse_tag (message, &tag_list); if (verbose) g_print ("tag\n"); } break; case GST_MESSAGE_STATE_CHANGED: { GstState oldstate, newstate, pending; gst_message_parse_state_changed (message, &oldstate, &newstate, &pending); if (GST_ELEMENT (message->src) == switchui->pipeline) { if (verbose) g_print ("state change from %s to %s\n", gst_element_state_get_name (oldstate), gst_element_state_get_name (newstate)); switch (GST_STATE_TRANSITION (oldstate, newstate)) { case GST_STATE_CHANGE_NULL_TO_READY: gst_switchui_handle_null_to_ready (switchui); break; case GST_STATE_CHANGE_READY_TO_PAUSED: gst_switchui_handle_ready_to_paused (switchui); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: gst_switchui_handle_paused_to_playing (switchui); break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: gst_switchui_handle_playing_to_paused (switchui); break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_switchui_handle_paused_to_ready (switchui); break; case GST_STATE_CHANGE_READY_TO_NULL: gst_switchui_handle_ready_to_null (switchui); break; default: if (verbose) g_print ("unknown state change from %s to %s\n", gst_element_state_get_name (oldstate), gst_element_state_get_name (newstate)); } } } break; case GST_MESSAGE_BUFFERING: { int percent; gst_message_parse_buffering (message, &percent); //g_print("buffering %d\n", percent); if (!switchui->paused_for_buffering && percent < 100) { g_print ("pausing for buffing\n"); switchui->paused_for_buffering = TRUE; gst_element_set_state (switchui->pipeline, GST_STATE_PAUSED); } else if (switchui->paused_for_buffering && percent == 100) { g_print ("unpausing for buffing\n"); switchui->paused_for_buffering = FALSE; gst_element_set_state (switchui->pipeline, GST_STATE_PLAYING); } } break; case GST_MESSAGE_STATE_DIRTY: case GST_MESSAGE_CLOCK_PROVIDE: case GST_MESSAGE_CLOCK_LOST: case GST_MESSAGE_NEW_CLOCK: case GST_MESSAGE_STRUCTURE_CHANGE: case GST_MESSAGE_STREAM_STATUS: break; case GST_MESSAGE_STEP_DONE: case GST_MESSAGE_APPLICATION: case GST_MESSAGE_ELEMENT: case GST_MESSAGE_SEGMENT_START: case GST_MESSAGE_SEGMENT_DONE: //case GST_MESSAGE_DURATION: case GST_MESSAGE_LATENCY: case GST_MESSAGE_ASYNC_START: case GST_MESSAGE_ASYNC_DONE: case GST_MESSAGE_REQUEST_STATE: case GST_MESSAGE_STEP_START: case GST_MESSAGE_QOS: default: if (verbose) { g_print ("message: %s\n", GST_MESSAGE_TYPE_NAME (message)); } break; } return TRUE; }
void Pipeline::handleBusMessage(GstMessage *message) { switch (GST_MESSAGE_TYPE(message)) { case GST_MESSAGE_ELEMENT: { // The only message we are handling here is the prepare-xwindow-id one if (gst_structure_has_name (message->structure, "prepare-xwindow-id")) { gst_x_overlay_set_xwindow_id(GST_X_OVERLAY(viewfinder), windowId); } break; } case GST_MESSAGE_ERROR: { GError *gerror = 0; gchar *debug = 0; gst_message_parse_error(message, &gerror, &debug); qCritical() << "Debug" << debug << " Error " << gerror->message; g_free(debug); g_error_free(gerror); break; } case GST_MESSAGE_WARNING: { GError *gerror = 0; gchar *debug = 0; gst_message_parse_warning(message, &gerror, &debug); qWarning() << "Debug" << debug << " Warning " << gerror->message; g_free(debug); g_error_free(gerror); break; } case GST_MESSAGE_INFO: { GError *gerror = 0; gchar *debug = 0; gst_message_parse_info(message, &gerror, &debug); qDebug() << "Debug" << debug << " Info " << gerror->message; g_free(debug); g_error_free(gerror); break; } case GST_MESSAGE_STATE_CHANGED: { if (GST_ELEMENT(GST_MESSAGE_SRC(message)) == camerabin) { GstState oldstate, newstate, pending; gst_message_parse_state_changed(message, &oldstate, &newstate, &pending); qDebug() << Q_FUNC_INFO << gst_element_state_get_name(oldstate) << "->" << gst_element_state_get_name(newstate) << "=>" << gst_element_state_get_name(pending); GstStateChange stateTransition = GST_STATE_TRANSITION(oldstate, newstate); switch (stateTransition) { case GST_STATE_CHANGE_PAUSED_TO_PLAYING: QMetaObject::invokeMethod(this, "pipelinePlaying", Qt::QueuedConnection); break; default: break; } } break; } default: break; } }
void eServiceMP3Record::gstBusCall(GstMessage *msg) { if (!msg) return; ePtr<iRecordableService> ptr = this; gchar *sourceName; GstObject *source; source = GST_MESSAGE_SRC(msg); if (!GST_IS_OBJECT(source)) return; sourceName = gst_object_get_name(source); switch (GST_MESSAGE_TYPE (msg)) { case GST_MESSAGE_EOS: eDebug("[eMP3ServiceRecord] gstBusCall eos event"); // Stream end -> stop recording m_event((iRecordableService*)this, evGstRecordEnded); break; case GST_MESSAGE_STATE_CHANGED: { if(GST_MESSAGE_SRC(msg) != GST_OBJECT(m_recording_pipeline)) break; GstState old_state, new_state; gst_message_parse_state_changed(msg, &old_state, &new_state, NULL); if(old_state == new_state) break; GstStateChange transition = (GstStateChange)GST_STATE_TRANSITION(old_state, new_state); eDebug("[eMP3ServiceRecord] gstBusCall state transition %s -> %s", gst_element_state_get_name(old_state), gst_element_state_get_name(new_state)); switch(transition) { case GST_STATE_CHANGE_PAUSED_TO_PLAYING: { if (m_streamingsrc_timeout) m_streamingsrc_timeout->stop(); break; } default: break; } break; } case GST_MESSAGE_ERROR: { gchar *debug; GError *err; gst_message_parse_error(msg, &err, &debug); g_free(debug); if (err->code != GST_STREAM_ERROR_CODEC_NOT_FOUND) eWarning("[eServiceMP3Record] gstBusCall Gstreamer error: %s (%i) from %s", err->message, err->code, sourceName); g_error_free(err); break; } case GST_MESSAGE_ELEMENT: { const GstStructure *msgstruct = gst_message_get_structure(msg); if (msgstruct) { if (gst_is_missing_plugin_message(msg)) { GstCaps *caps = NULL; gst_structure_get (msgstruct, "detail", GST_TYPE_CAPS, &caps, NULL); if (caps) { std::string codec = (const char*) gst_caps_to_string(caps); eDebug("[eServiceMP3Record] gstBusCall cannot record because of incompatible codecs %s", codec.c_str()); gst_caps_unref(caps); } } else { const gchar *eventname = gst_structure_get_name(msgstruct); if (eventname) { if (!strcmp(eventname, "redirect")) { const char *uri = gst_structure_get_string(msgstruct, "new-location"); eDebug("[eServiceMP3Record] gstBusCall redirect to %s", uri); gst_element_set_state (m_recording_pipeline, GST_STATE_NULL); g_object_set(G_OBJECT (m_source), "uri", uri, NULL); gst_element_set_state (m_recording_pipeline, GST_STATE_PLAYING); } } } } break; } case GST_MESSAGE_STREAM_STATUS: { GstStreamStatusType type; GstElement *owner; gst_message_parse_stream_status (msg, &type, &owner); if (type == GST_STREAM_STATUS_TYPE_CREATE) { if (GST_IS_PAD(source)) owner = gst_pad_get_parent_element(GST_PAD(source)); else if (GST_IS_ELEMENT(source)) owner = GST_ELEMENT(source); else owner = 0; if (owner) { GstState state; gst_element_get_state(m_recording_pipeline, &state, NULL, 0LL); GstElementFactory *factory = gst_element_get_factory(GST_ELEMENT(owner)); const gchar *name = gst_plugin_feature_get_name(GST_PLUGIN_FEATURE(factory)); if (!strcmp(name, "souphttpsrc") && (state == GST_STATE_READY) && !m_streamingsrc_timeout->isActive()) { m_streamingsrc_timeout->start(HTTP_TIMEOUT*1000, true); g_object_set (G_OBJECT (owner), "timeout", HTTP_TIMEOUT, NULL); eDebug("[eServiceMP3Record] gstBusCall setting timeout on %s to %is", name, HTTP_TIMEOUT); } } if (GST_IS_PAD(source)) gst_object_unref(owner); } break; } default: break; } g_free(sourceName); }
static gboolean bus_message (GstBus * bus, GstMessage * message, App * app) { gchar *sourceName; GstObject *source; gchar *string; GstState current_state; if (!message) return FALSE; source = GST_MESSAGE_SRC (message); if (!GST_IS_OBJECT (source)) return FALSE; sourceName = gst_object_get_name (source); if (gst_message_get_structure (message)) string = gst_structure_to_string (gst_message_get_structure (message)); else string = g_strdup (GST_MESSAGE_TYPE_NAME (message)); GST_DEBUG("gst_message from %s: %s", sourceName, string); g_free (string); switch (GST_MESSAGE_TYPE (message)) { case GST_MESSAGE_ERROR: { GError *gerror; gchar *debug; gst_message_parse_error (message, &gerror, &debug); gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug); g_error_free (gerror); g_free (debug); g_main_loop_quit (app->loop); break; } case GST_MESSAGE_WARNING: { GError *gerror; gchar *debug; gst_message_parse_warning (message, &gerror, &debug); gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug); g_error_free (gerror); g_free (debug); // g_main_loop_quit (app->loop); break; } case GST_MESSAGE_EOS: g_message ("received EOS"); g_main_loop_quit (app->loop); break; case GST_MESSAGE_ASYNC_DONE: break; case GST_MESSAGE_ELEMENT: { const GstStructure *msgstruct = gst_message_get_structure (message); if (msgstruct) { const gchar *eventname = gst_structure_get_name (msgstruct); if (!strcmp (eventname, "seekable")) app->is_seekable = TRUE; } break; } case GST_MESSAGE_STATE_CHANGED: { GstState old_state, new_state; GstStateChange transition; if (GST_MESSAGE_SRC (message) != GST_OBJECT (app->tsdemux)) break; gst_message_parse_state_changed (message, &old_state, &new_state, NULL); 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: 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_SEGMENT_DONE: { GST_DEBUG ("GST_MESSAGE_SEGMENT_DONE!!!"); do_seek (app); } default: break; } gst_element_get_state (app->pipeline, ¤t_state, NULL, 0); if (app->current_segment == 0 && app->segment_count /*&& app->is_seekable*/ && current_state == GST_STATE_PLAYING) do_seek (app); GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(app->pipeline),GST_DEBUG_GRAPH_SHOW_ALL,"bdremux_pipelinegraph_message"); return TRUE; }
void eServiceMP3::gstBusCall(GstMessage *msg) { if (!msg) return; gchar *sourceName; GstObject *source; source = GST_MESSAGE_SRC(msg); if (!GST_IS_OBJECT(source)) return; sourceName = gst_object_get_name(source); #if 0 gchar *string; if (gst_message_get_structure(msg)) string = gst_structure_to_string(gst_message_get_structure(msg)); else string = g_strdup(GST_MESSAGE_TYPE_NAME(msg)); eDebug("eTsRemoteSource::gst_message from %s: %s", sourceName, string); g_free(string); #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 *subsink = gst_bin_get_by_name(GST_BIN(m_gst_playbin), "subtitle_sink"); if (subsink) { #ifdef GSTREAMER_SUBTITLE_SYNC_MODE_BUG /* * HACK: disable sync mode for now, gstreamer suffers from a bug causing sparse streams to loose sync, after pause/resume / skip * see: https://bugzilla.gnome.org/show_bug.cgi?id=619434 * Sideeffect of using sync=false is that we receive subtitle buffers (far) ahead of their * display time. * Not too far ahead for subtitles contained in the media container. * But for external srt files, we could receive all subtitles at once. * And not just once, but after each pause/resume / skip. * So as soon as gstreamer has been fixed to keep sync in sparse streams, sync needs to be re-enabled. */ g_object_set (G_OBJECT (subsink), "sync", FALSE, NULL); #endif #if 0 /* we should not use ts-offset to sync with the decoder time, we have to do our own decoder timekeeping */ g_object_set (G_OBJECT (subsink), "ts-offset", -2L * GST_SECOND, NULL); /* late buffers probably will not occur very often */ g_object_set (G_OBJECT (subsink), "max-lateness", 0L, NULL); /* avoid prerolling (it might not be a good idea to preroll a sparse stream) */ g_object_set (G_OBJECT (subsink), "async", TRUE, NULL); #endif eDebug("eServiceMP3::subsink properties set!"); gst_object_unref(subsink); } setAC3Delay(ac3_delay); setPCMDelay(pcm_delay); } break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: { if ( m_sourceinfo.is_streaming && m_streamingsrc_timeout ) m_streamingsrc_timeout->stop(); } 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 ) { 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); } } 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_REPLACE); if (result) { if (m_stream_tags) gst_tag_list_free(m_stream_tags); m_stream_tags = result; } 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); 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; 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); if ( n_video + n_audio <= 0 ) stop(); active_idx = 0; m_audioStreams.clear(); m_subtitleStreams.clear(); for (i = 0; i < n_audio; i++) { 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); const gchar *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) ) { 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 (i = 0; i < n_text; i++) { gchar *g_codec = NULL, *g_lang = NULL; g_signal_emit_by_name (m_gst_playbin, "get-text-tags", i, &tags); subtitleStream subs; 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); gst_tag_list_get_string(tags, GST_TAG_SUBTITLE_CODEC, &g_codec); gst_tag_list_free(tags); } subs.language_code = std::string(g_lang); eDebug("eServiceMP3::subtitle stream=%i language=%s codec=%s", i, g_lang, g_codec); GstPad* pad = 0; g_signal_emit_by_name (m_gst_playbin, "get-text-pad", i, &pad); if ( pad ) g_signal_connect (G_OBJECT (pad), "notify::caps", G_CALLBACK (gstTextpadHasCAPS), this); subs.type = getSubtitleType(pad, g_codec); m_subtitleStreams.push_back(subs); g_free (g_lang); } m_event((iPlayableService*)this, evUpdatedInfo); if ( m_errorInfo.missing_codec != "" ) { if ( m_errorInfo.missing_codec.find("video/") == 0 || ( m_errorInfo.missing_codec.find("audio/") == 0 && getNumberOfTracks() == 0 ) ) m_event((iPlayableService*)this, evUser+12); } break; } case GST_MESSAGE_ELEMENT: { if (const GstStructure *msgstruct = gst_message_get_structure(msg)) { if ( gst_is_missing_plugin_message(msg) ) { GstCaps *caps; gst_structure_get (msgstruct, "detail", GST_TYPE_CAPS, &caps, NULL); std::string codec = (const char*) gst_caps_to_string(caps); gchar *description = gst_missing_plugin_message_get_description(msg); if ( description ) { eDebug("eServiceMP3::m_errorInfo.missing_codec = %s", codec.c_str()); m_errorInfo.error_message = "GStreamer plugin " + (std::string)description + " not available!\n"; m_errorInfo.missing_codec = codec.substr(0,(codec.find_first_of(','))); g_free(description); } gst_caps_unref(caps); } else { 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); } } } } 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); break; } case GST_MESSAGE_STREAM_STATUS: { GstStreamStatusType type; GstElement *owner; gst_message_parse_stream_status (msg, &type, &owner); if ( type == GST_STREAM_STATUS_TYPE_CREATE && m_sourceinfo.is_streaming ) { if ( GST_IS_PAD(source) ) owner = gst_pad_get_parent_element(GST_PAD(source)); else if ( GST_IS_ELEMENT(source) ) owner = GST_ELEMENT(source); else owner = 0; if ( owner ) { GstElementFactory *factory = gst_element_get_factory(GST_ELEMENT(owner)); const gchar *name = gst_plugin_feature_get_name(GST_PLUGIN_FEATURE(factory)); if (!strcmp(name, "souphttpsrc")) { m_streamingsrc_timeout->start(HTTP_TIMEOUT*1000, true); g_object_set (G_OBJECT (owner), "timeout", HTTP_TIMEOUT, NULL); eDebug("eServiceMP3::GST_STREAM_STATUS_TYPE_CREATE -> setting timeout on %s to %is", name, HTTP_TIMEOUT); } } if ( GST_IS_PAD(source) ) gst_object_unref(owner); } break; } default: break; } g_free (sourceName); }
GstBusSyncReply Gst_bus_call(GstBus * bus, GstMessage *msg, gpointer user_data) { gchar * sourceName; // source GstObject * source; source = GST_MESSAGE_SRC(msg); if (!GST_IS_OBJECT(source)) return GST_BUS_DROP; sourceName = gst_object_get_name(source); switch (GST_MESSAGE_TYPE(msg)) { case GST_MESSAGE_EOS: { g_message("End-of-stream"); end_eof = 1; break; } case GST_MESSAGE_ERROR: { gchar * debug; GError *err; gst_message_parse_error(msg, &err, &debug); g_free (debug); lt_info_c( "%s:%s - GST_MESSAGE_ERROR: %s (%i) from %s\n", FILENAME, __FUNCTION__, 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") ) lt_info_c( "%s:%s - GST_MESSAGE_ERROR: videosink\n", FILENAME, __FUNCTION__ ); //FIXME: how shall playback handle this event??? else if ( g_strrstr(sourceName, "audiosink") ) lt_info_c( "%s:%s - GST_MESSAGE_ERROR: audioSink\n", FILENAME, __FUNCTION__ ); //FIXME: how shall playback handle this event??? } } g_error_free(err); end_eof = 1; // NOTE: just to exit 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") ) lt_info_c( "%s:%s - GST_MESSAGE_INFO: videosink\n", FILENAME, __FUNCTION__ ); //FIXME: how shall playback handle this event??? } 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_REPLACE); if (result) { if (m_stream_tags) gst_tag_list_free(m_stream_tags); m_stream_tags = result; } 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); if(fd >= 0) { int ret = write(fd, GST_BUFFER_DATA(buf_image), GST_BUFFER_SIZE(buf_image)); close(fd); lt_info_c( "%s:%s - GST_MESSAGE_INFO: cPlayback::state /tmp/.id3coverart %d bytes written\n", FILENAME, __FUNCTION__ , ret); } //FIXME: how shall playback handle this event??? } gst_tag_list_free(tags); lt_info_c( "%s:%s - GST_MESSAGE_INFO: update info tags\n", FILENAME, __FUNCTION__); //FIXME: how shall playback handle this event??? 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; lt_info_c( "%s:%s - GST_MESSAGE_STATE_CHANGED: state transition %s -> %s\n", FILENAME, __FUNCTION__, 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: { GstIterator *children; if (audioSink) { gst_object_unref(GST_OBJECT(audioSink)); audioSink = NULL; } if (videoSink) { gst_object_unref(GST_OBJECT(videoSink)); videoSink = NULL; } children = gst_bin_iterate_recurse(GST_BIN(m_gst_playbin)); audioSink = GST_ELEMENT_CAST(gst_iterator_find_custom(children, (GCompareFunc)match_sinktype, (gpointer)"GstDVBAudioSink")); videoSink = GST_ELEMENT_CAST(gst_iterator_find_custom(children, (GCompareFunc)match_sinktype, (gpointer)"GstDVBVideoSink")); gst_iterator_free(children); } break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: { } break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: { } break; case GST_STATE_CHANGE_PAUSED_TO_READY: { if (audioSink) { gst_object_unref(GST_OBJECT(audioSink)); audioSink = NULL; } if (videoSink) { gst_object_unref(GST_OBJECT(videoSink)); videoSink = NULL; } } break; case GST_STATE_CHANGE_READY_TO_NULL: { } break; } break; } #if 0 case GST_MESSAGE_ELEMENT: { if(gst_structure_has_name(gst_message_get_structure(msg), "prepare-xwindow-id")) { // set window id gst_x_overlay_set_xwindow_id(GST_X_OVERLAY(GST_MESSAGE_SRC (msg)), glfb->getWindowID()); // reshape window gst_x_overlay_set_render_rectangle(GST_X_OVERLAY(GST_MESSAGE_SRC (msg)), 0, 0, glfb->getOSDWidth(), glfb->getOSDHeight()); // sync frames gst_x_overlay_expose(GST_X_OVERLAY(GST_MESSAGE_SRC (msg))); } } #endif break; default: break; } return GST_BUS_DROP; }
static gboolean gst_PlayRegion_handle_message (GstBus * bus, GstMessage * message, gpointer data) { GstPlayRegion *PlayRegion = (GstPlayRegion *) data; switch (GST_MESSAGE_TYPE (message)) { case GST_MESSAGE_APPLICATION: if (gst_message_has_name (message, "ExPrerolled")) { /* it's our message */ g_print ("we are all prerolled, do seek\n"); gst_element_seek (PlayRegion->pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE, GST_SEEK_TYPE_SET, 2 * GST_SECOND, GST_SEEK_TYPE_SET, 5 * GST_SECOND); gst_element_set_state (PlayRegion->pipeline, GST_STATE_PLAYING); } break; case GST_MESSAGE_EOS: gst_PlayRegion_handle_eos (PlayRegion); break; case GST_MESSAGE_ERROR: { GError *error = NULL; gchar *debug; gst_message_parse_error (message, &error, &debug); gst_PlayRegion_handle_error (PlayRegion, error, debug); } break; case GST_MESSAGE_WARNING: { GError *error = NULL; gchar *debug; gst_message_parse_warning (message, &error, &debug); gst_PlayRegion_handle_warning (PlayRegion, error, debug); } break; case GST_MESSAGE_INFO: { GError *error = NULL; gchar *debug; gst_message_parse_info (message, &error, &debug); gst_PlayRegion_handle_info (PlayRegion, error, debug); } break; case GST_MESSAGE_TAG: { GstTagList *tag_list; gst_message_parse_tag (message, &tag_list); if (verbose) g_print ("tag\n"); } break; case GST_MESSAGE_STATE_CHANGED: { GstState oldstate, newstate, pending; gst_message_parse_state_changed (message, &oldstate, &newstate, &pending); if (GST_ELEMENT (message->src) == PlayRegion->pipeline) { if (verbose) g_print ("state change from %s to %s\n", gst_element_state_get_name (oldstate), gst_element_state_get_name (newstate)); switch (GST_STATE_TRANSITION (oldstate, newstate)) { case GST_STATE_CHANGE_NULL_TO_READY: gst_PlayRegion_handle_null_to_ready (PlayRegion); break; case GST_STATE_CHANGE_READY_TO_PAUSED: gst_PlayRegion_handle_ready_to_paused (PlayRegion); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: gst_PlayRegion_handle_paused_to_playing (PlayRegion); break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: gst_PlayRegion_handle_playing_to_paused (PlayRegion); break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_PlayRegion_handle_paused_to_ready (PlayRegion); break; case GST_STATE_CHANGE_READY_TO_NULL: gst_PlayRegion_handle_ready_to_null (PlayRegion); break; default: if (verbose) g_print ("unknown state change from %s to %s\n", gst_element_state_get_name (oldstate), gst_element_state_get_name (newstate)); } } } break; case GST_MESSAGE_BUFFERING: { int percent; gst_message_parse_buffering (message, &percent); //g_print("buffering %d\n", percent); if (!PlayRegion->paused_for_buffering && percent < 100) { g_print ("pausing for buffing\n"); PlayRegion->paused_for_buffering = TRUE; gst_element_set_state (PlayRegion->pipeline, GST_STATE_PAUSED); } else if (PlayRegion->paused_for_buffering && percent == 100) { g_print ("unpausing for buffing\n"); PlayRegion->paused_for_buffering = FALSE; gst_element_set_state (PlayRegion->pipeline, GST_STATE_PLAYING); } } break; case GST_MESSAGE_STATE_DIRTY: case GST_MESSAGE_CLOCK_PROVIDE: case GST_MESSAGE_CLOCK_LOST: case GST_MESSAGE_NEW_CLOCK: case GST_MESSAGE_STRUCTURE_CHANGE: case GST_MESSAGE_STREAM_STATUS: break; case GST_MESSAGE_STEP_DONE: case GST_MESSAGE_ELEMENT: case GST_MESSAGE_SEGMENT_START: case GST_MESSAGE_SEGMENT_DONE: case GST_MESSAGE_DURATION: case GST_MESSAGE_LATENCY: case GST_MESSAGE_ASYNC_START: case GST_MESSAGE_ASYNC_DONE: case GST_MESSAGE_REQUEST_STATE: case GST_MESSAGE_STEP_START: case GST_MESSAGE_QOS: default: if (verbose) { g_print ("message: %s\n", GST_MESSAGE_TYPE_NAME (message)); } break; } return TRUE; }