Hi When chaning the code yo improve my debug output I mis-takenly removed my uridecodebin. Here is the code now, but it still does not link correctly and I get no audio out. virtual bool init() { if (!GStreamerPlayerBase::init()) { return false; } pipeline = gst_pipeline_new(""); // Audio bin audiobin = gst_bin_new("audiobin"); qDebug() << "adding audiobin to pipeline"; gst_bin_add(GST_BIN(pipeline), audiobin); qDebug() << "adding elements"; _sourceQ = gst_element_factory_make("queue2", NULL); quint64 maxBufferTime = 0; // disable guint maxBufferBytes = Settings::getValue<int>(Settings::SECTION_RECEIVE + Settings::RECEIVE_MAX_BUFFER_SIZE); guint highPercent = Settings::getValue<int>(Settings::SECTION_RECEIVE + Settings::RECEIVE_BUFFER_HIGHPERC); guint lowPercent = Settings::getValue<int>(Settings::SECTION_RECEIVE + Settings::RECEIVE_BUFFER_LOWPERC); bool useBuffering = Settings::getValue<int>(Settings::SECTION_RECEIVE + Settings::RECEIVE_USE_BUFFER); //this queue is used to force buffering of more data, the intention //being to help with internet radio drop out g_object_set(G_OBJECT(_sourceQ), "max-size-buffers", 0, "max-size-time", maxBufferTime, "max-size-bytes", maxBufferBytes, "use-buffering", useBuffering, "high-percent", highPercent, "low-percent", lowPercent, NULL); qDebug() << "add source to bin elements"; gst_bin_add(GST_BIN(audiobin), _sourceQ); qDebug() << "cret sourceQ sink pad"; GstPad* pad = gst_element_get_pad(_sourceQ, "sink"); gst_element_add_pad(audiobin, gst_ghost_pad_new("sink", pad)); gst_object_unref(pad); uriDecodebin = gst_element_factory_make("uridecodebin", NULL); gst_bin_add(GST_BIN(pipeline), uriDecodebin); g_object_set(G_OBJECT(uriDecodebin), "buffer-size", 150000, NULL); g_object_set(G_OBJECT(uriDecodebin), "download", false, NULL); g_object_set(G_OBJECT(uriDecodebin), "use-buffering", false, NULL); g_signal_connect(G_OBJECT(uriDecodebin), "drained", G_CALLBACK(sourceDrainedCallback), this); /* connect uridecodebin to _sourceQ when it creates its output pad */ qDebug() << "adding uriDeecode callback"; g_signal_connect(G_OBJECT(uriDecodebin), "pad-added", G_CALLBACK(callbackPadAdded), this); qDebug() << "init output stage"; initOutputStage(GST_BIN(pipeline)); qDebug() << "ref elements"; gst_object_ref(uriDecodebin); gst_object_ref(_sourceQ); <div class="gmail_quote"><div> return true; } Cheers, Stuart </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"> Message: 5 Date: Wed, 5 Oct 2011 16:15:50 +0800 From: Stuart Gray <<a href="mailto:stuart68@gmail.com">stuart68@gmail.com</a>> Subject: Re: Name queue20 is not unique in bin , not adding - when connecting uridecodebin to queue2 To: <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> Message-ID: <<a href="mailto:CAG2RAVYtZWYgTumW1-Socf92MAjnytt4oFPTDaY_yRBZ_GLKuw@mail.gmail.com">CAG2RAVYtZWYgTumW1-Socf92MAjnytt4oFPTDaY_yRBZ_GLKuw@mail.gmail.com</a>> Content-Type: text/plain; charset="iso-8859-1" Hi Tim, I had already begun to think of an alternative way to do this. I was thinking about using BINs. As I am running within an embedded system I need to queue2 to put the second part of the pipeline into a different thread and also to buffer data in a way that I can control. I have re-written the code as following: virtual bool init() { if (!GStreamerPlayerBase::init()) { return false; } pipeline = gst_pipeline_new(""); // Audio bin audiobin = gst_bin_new("audiobin"); qDebug() << "adding audiobin to pipeline"; gst_bin_add(GST_BIN(pipeline), audiobin); qDebug() << "adding elements"; _sourceQ = gst_element_factory_make("queue2", NULL); quint64 maxBufferTime = 0; // disable guint maxBufferBytes = Settings::getValue<int>(Settings::SECTION_RECEIVE + Settings::RECEIVE_MAX_BUFFER_SIZE); guint highPercent = Settings::getValue<int>(Settings::SECTION_RECEIVE + Settings::RECEIVE_BUFFER_HIGHPERC); guint lowPercent = Settings::getValue<int>(Settings::SECTION_RECEIVE + Settings::RECEIVE_BUFFER_LOWPERC); bool useBuffering = Settings::getValue<int>(Settings::SECTION_RECEIVE + Settings::RECEIVE_USE_BUFFER); //this queue is used to force buffering of more data, the intention //being to help with internet radio drop out g_object_set(G_OBJECT(_sourceQ), "max-size-buffers", 0, "max-size-time", maxBufferTime, "max-size-bytes", maxBufferBytes, "use-buffering", useBuffering, "high-percent", highPercent, "low-percent", lowPercent, NULL); qDebug() << "add source to bin elements"; gst_bin_add(GST_BIN(audiobin), _sourceQ); qDebug() << "create sourceQ sink pad"; GstPad* pad = gst_element_get_pad(_sourceQ, "sink"); gst_element_add_pad(audiobin, gst_ghost_pad_new("sink", pad)); gst_object_unref(pad); qDebug() << "init output stage"; initOutputStage(GST_BIN(pipeline)); qDebug() << "ref elements"; gst_object_ref(uriDecodebin); gst_object_ref(_sourceQ); return true; } my callbackPadAdded is: static void callbackPadAdded(GstElement *uriDecodebin, GstPad *pad, gpointer self) { GStreamerDecodebinPrivate* obj = reinterpret_cast<GStreamerDecodebinPrivate*>(self); GstPad* const audiopad = gst_element_get_pad(obj->audiobin, "sink"); qDebug() << "uridecodebin pad add started"; Q_UNUSED(uriDecodebin); if (GST_PAD_IS_LINKED(audiopad)) { qDebug() << "audiopad is already linked. Unlinking old pad."; gst_pad_unlink(audiopad, GST_PAD_PEER(audiopad)); } qDebug() << "uridecodebin pad added"; gst_pad_link(pad, audiopad); gst_object_unref(audiopad); } For the above I would like to add setting of caps to the uridecodebin. Setting of the output stage is: void initOutputStage(GstBin* outputBin) { QString dev = Settings::getValue<QString>( Settings::SECTION_DEVICES + Settings::AUDIO_OUTPUT ); _pcmSink = gst_element_factory_make("alsasink", NULL); g_object_set(G_OBJECT(_pcmSink), "device", dev.toLatin1().data(), NULL); GstBaseAudioSinkSlaveMethod slaveMethod = GST_BASE_AUDIO_SINK_SLAVE_NONE; g_object_set(G_OBJECT(_pcmSink), "sync", 0, "async", 0, "slave-method", slaveMethod, NULL); gst_bin_add(GST_BIN(outputBin), _pcmSink); // link the static parts together gst_element_link(_pcmSink, NULL); bool usePlayBin = Settings::getValue<bool>(Settings::SECTION_AUDIO + Settings::USE_PLAYBIN); } I get no audio from the code when I try to run this. I do not see how to link the BUN together and get it into the playing state. I have the following: gst_element_link_many(uriDecodebin, _sourceQ, _pcmSink, NULL); During initialization I get the following warning: ref elements (<unknown>:2847): GStreamer-CRITICAL **: gst_object_ref: assertion `object != NULL' failed When I try to link it all together I get: link many in streaming (<unknown>:2847): GStreamer-CRITICAL **: gst_element_link_many: assertion `GST_IS_ELEMENT (element_1)' failed Code: virtual void linkStreaming() { qDebug() << "linking in streaming"; gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_NULL); if (passThroughLinked) { gst_element_set_state(GST_ELEMENT(recPipeline), GST_STATE_NULL); // gst_bin_remove_many(GST_BIN(pipeline), passThroughFakeSource, passThroughCapsFilter, _audioResample, NULL); } qDebug() << "link many in streaming"; gst_element_link_many(uriDecodebin, _sourceQ, _pcmSink, NULL); passThroughLinked = false; } I think if I use the BIN to put _sourceQ into I can get the callBackPadAdded to work correctly. But with BINs I am unsure how to link them together correctly and I suspect this is where my error is. My goal is to have the very simple pipeline: uridecodebin ! queue2 ! alsasink Thanks, Stuart On 5 October 2011 03:00, <<a href="mailto:gstreamer-devel-request@lists.freedesktop.org">gstreamer-devel-request@lists.freedesktop.org</a>>wrote: > Send gstreamer-devel mailing list submissions to > <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> > > To subscribe or unsubscribe via the World Wide Web, visit > <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> > or, via email, send a message with subject or body 'help' to > <a href="mailto:gstreamer-devel-request@lists.freedesktop.org">gstreamer-devel-request@lists.freedesktop.org</a> > > You can reach the person managing the list at > <a href="mailto:gstreamer-devel-owner@lists.freedesktop.org">gstreamer-devel-owner@lists.freedesktop.org</a> > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of gstreamer-devel digest..." > > > Today's Topics: > > 1. Re: Name queue20 is not unique in bin , not adding - when > connecting uridecodebin to queue2 (Tim-Philipp M?ller) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 04 Oct 2011 09:25:52 +0100 > From: Tim-Philipp M?ller <<a href="mailto:t.i.m@zen.co.uk">t.i.m@zen.co.uk</a>> > Subject: Re: Name queue20 is not unique in bin , not adding - when > connecting uridecodebin to queue2 > To: <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> > Message-ID: <1317716752.4063.3.camel@zingle> > Content-Type: text/plain; charset="UTF-8" > > On Tue, 2011-10-04 at 10:43 +0800, Stuart Gray wrote: > > > In trying to connect uridecodebin to queue2 I get the following output > > from gstreamer: > > > > (<unknown>:1363): GStreamer-WARNING **: Name queue20 is not unique in > > bin , not adding > > > > My code is as follows: > > > > pipeline = gst_pipeline_new(""); > > > > uriDecodebin = gst_element_factory_make("uridecodebin", NULL); > > > > g_object_set(G_OBJECT(uriDecodebin), "buffer-size", 150000, > > NULL); > > g_object_set(G_OBJECT(uriDecodebin), "download", false, NULL); > > g_object_set(G_OBJECT(uriDecodebin), "use-buffering", false, > > NULL); > > g_signal_connect(G_OBJECT(uriDecodebin), "drained", > > G_CALLBACK(sourceDrainedCallback), this); > > /* connect uridecodebin to _sourceQ when it creates its output > > pad */ > > g_signal_connect(G_OBJECT(uriDecodebin), "pad-added", > > G_CALLBACK(callbackPadAdded), this); > > > > outputBin = gst_bin_new("output-bin"); > > ... > > _sourceQ = gst_element_factory_make("queue2", NULL); > > gst_bin_add_many(GST_BIN(outputBin), _sourceQ, _pcmSink, > > NULL); > > ... > > What does your callbackPadAdded look like? > > I'm guessing you are creating "queue" elements in there. It's basically > a bug in the way GStreamer creates default names for elements if you > don't specify one. The first "queue2" instance will get named "queue20" > and the twentieth "queue" instance will get named "queue20" as well, > which may cause problems if you are using both at the same time. It's > hard to change now though, because it would break code that relies on > the the naming scheme. > > In this case, you could just use a normal queue instead of queue2 before > the sink (or just not use a queue at all?). > > Cheers > -Tim > > > > > ------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> > <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> > > > End of gstreamer-devel Digest, Vol 9, Issue 8 > ********************************************* > -------------- next part -------------- An HTML attachment was scrubbed... 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