<html> <head> <meta content="text/html; charset=utf-8" http-equiv="Content-Type"> </head> <body text="#000000" bgcolor="#FFFFFF"> Thanks for your suggestion, you are right so what I have done is writing the pipeline, making, adding and linking elements one by one. What happens to me now is that, when dynamically adding audio and video pads from tsdemux to their own queues before the parsers, they are refused and there is no playout. The function for dynamically adding pads is (I know the conditional instructions are not implemented in an efficient way as I wrote them just to check where it failed): <meta http-equiv="content-type" content="text/html; charset=utf-8"> <blockquote> <pre style="background-color:#ffffff;color:#000000;font-family:'DejaVu Sans Mono';font-size:9,0pt;">static void dynamic_addpad(GstElement *src, GstPad *new_pad, CustomData *data) { char* pad_name = gst_pad_get_name(new_pad); g_print(" In dynamic ADDING PAD %s\n", pad_name); if (g_str_has_prefix(pad_name,"audio")) { GstElement *q_audio; q_audio = gst_bin_get_by_name (GST_BIN(data->pipeline), "queue_audio"); GstPad *audiodemuxsink = gst_element_get_static_pad(q_audio,"sink"); if(gst_pad_link(new_pad,audiodemuxsink) == GST_PAD_LINK_OK) g_print("Audio successfully linked!\n"); else if(gst_pad_link(new_pad,audiodemuxsink) == GST_PAD_LINK_WRONG_HIERARCHY) g_print("Audio link wrong hierarchy!\n"); else if(gst_pad_link(new_pad,audiodemuxsink) == GST_PAD_LINK_WAS_LINKED) g_print("Audio already linked!\n"); else if(gst_pad_link(new_pad,audiodemuxsink) == GST_PAD_LINK_WRONG_DIRECTION) g_print("Audio link wrong direction!\n"); else if(gst_pad_link(new_pad,audiodemuxsink) == GST_PAD_LINK_NOFORMAT) g_print("Audio link noformat!\n"); else if(gst_pad_link(new_pad,audiodemuxsink) == GST_PAD_LINK_NOSCHED) g_print("Audio link nosched!\n"); else if(gst_pad_link(new_pad,audiodemuxsink) == GST_PAD_LINK_REFUSED) g_print("Audio link refused!\n"); g_print ("Sink pad link: '%s'\n", pad_name); gst_object_unref(q_audio); } else if (g_str_has_prefix(pad_name,"video")) { GstElement *q_video; q_video = gst_bin_get_by_name(GST_BIN(data->pipeline), "queue_video"); GstPad *videodemuxsink = gst_element_get_static_pad(q_video,"sink"); if(gst_pad_link(new_pad,videodemuxsink) == GST_PAD_LINK_OK) g_print("Video successfully linked!\n"); else if(gst_pad_link(new_pad,videodemuxsink) == GST_PAD_LINK_WRONG_HIERARCHY) g_print("Video link wrong hierarchy!\n"); else if(gst_pad_link(new_pad,videodemuxsink) == GST_PAD_LINK_WAS_LINKED) g_print("Video link already linked!\n"); else if(gst_pad_link(new_pad,videodemuxsink) == GST_PAD_LINK_WRONG_DIRECTION) g_print("Video link wrong direction!\n"); else if(gst_pad_link(new_pad,videodemuxsink) == GST_PAD_LINK_NOFORMAT) g_print("Video link noformat!\n"); else if(gst_pad_link(new_pad,videodemuxsink) == GST_PAD_LINK_NOSCHED) g_print("Video link nosched!\n"); else if(gst_pad_link(new_pad,videodemuxsink) == GST_PAD_LINK_REFUSED) g_print("Video link refused!\n"); g_print ("Sink pad link: '%s'\n", pad_name); gst_object_unref(q_video); } }</pre> </blockquote> So the log prints the "Video link refused!" sentence and after that one: <blockquote>05-23 12:38:58.675 8461-8526/com.sample.app W/GStreamer+mpegtspacketizer: 0:00:01.503897667 0x9e11a260 mpegtspacketizer.c:2509:mpegts_packetizer_pts_to_ts Not enough information to calculate proper timestamp </blockquote> Which is understandable as there is no linking and therefore it has no info. There the log stops and the app does not crash but keeps a black video surface. Anyway I know where the problem is (dynamic linking) but I would appreciate any suggestions to solve it, I don't know why there is a link refusement for both audio and video streams, with the gst-launch way it was working correctly at this point. And the pc-equivalent code works fine exactly as shown on this post. The modifications implemented for the main method in android c source are as seen here: <meta http-equiv="content-type" content="text/html; charset=utf-8"> <blockquote> <pre style="background-color:#ffffff;color:#000000;font-family:'DejaVu Sans Mono';font-size:9,0pt;">static void *app_function (void *userdata) { [...] GST_DEBUG ("Creating pipeline in CustomData at %p", data); /* Create our own GLib Main Context and make it the default one */ data->context = g_main_context_new (); g_main_context_push_thread_default(data->context); /* Build pipeline */ data->pipeline = gst_pipeline_new ("pipeline"); data->httpsrc = gst_element_factory_make ("souphttpsrc", "http_src"); data->tsdemux = gst_element_factory_make ("tsdemux", "demux"); data->queue_video = gst_element_factory_make("queue2", "queue_video"); data->h264parse = gst_element_factory_make ("h264parse", "h264_parse"); data->h264dec = gst_element_factory_make ("avdec_h264", "h264dec"); data->video_convert = gst_element_factory_make("videoconvert","video_convert"); data->videosink = gst_element_factory_make ("glimagesink", "video_sink"); data->queue_audio = gst_element_factory_make ("queue2", "queue_audio"); data->aacparse = gst_element_factory_make ("aacparse", "aacparse"); data->faad = gst_element_factory_make ("faad", "faad"); data->audio_convert = gst_element_factory_make("audioconvert", "audio_convert"); data->audiosink = gst_element_factory_make ("autoaudiosink", "audio_sink"); g_signal_connect (data->tsdemux, "pad-added", G_CALLBACK (dynamic_addpad), &data); gst_bin_add_many(GST_BIN(data->pipeline), data->httpsrc, data->tsdemux, data->queue_video, data->h264parse, data->h264dec, data->video_convert, data->videosink, data->queue_audio, data->aacparse, data->faad, data->audio_convert, data->audiosink, NULL); gst_element_link(data->httpsrc, data->tsdemux); gst_element_link_many(data->queue_video, data->h264parse, data->h264dec, data->video_convert, data->videosink, NULL); gst_element_link_many(data->queue_audio, data->aacparse, data->faad, data->audio_convert, data->audiosink, NULL); if (error) { gchar *message = g_strdup_printf("Unable to build pipeline: %s", error->message); g_clear_error (&error); set_ui_message(message, data); g_free (message); return NULL; } [...] }</pre> </blockquote> Thanks for your time, Dani <div class="moz-cite-prefix">El 18/05/16 a las 14:31, Sebastian Dröge escribió: </div> <blockquote cite="mid:1463574707.2897.23.camel@centricular.com" type="cite"> <pre wrap="">On Di, 2016-05-17 at 11:12 +0200, Dani wrote: </pre> <blockquote type="cite"> <pre wrap=""> So as this function is called, when it used to be a playbin, just by adding the GST_STATE_READY and the g_object_set() for the uri worked fine. Now I have a working pipeline with a checked .ts and it initially works, but not when this function is called. The debug logs for the location property gets changed but what happens is that the app gets paused (no crash at all) with the outdated video. The initially working pipeline -using gst_parse_launch()- is: souphttpsrc name=http_src location=<a class="moz-txt-link-freetext" href="http://192.168.0.32/videos/sintel">http://192.168.0.32/videos/sintel</a>. ts ! queue ! tsdemux name=demux demux. ! queue ! h264parse ! avdec_h264 ! videoconvert ! glimagesink name=video_sink demux. ! queue ! aacparse ! faad ! audioconvert ! autoaudiosink name=audio_sink The related log is: [...] </pre> </blockquote> <pre wrap=""> Can you get a full log? But in general, re-using gst_parse_launch() pipelines (that is, going back to READY/NULL and starting them again) is generally not working well. It's better to recreate the whole pipeline. If you want to re-use pipelines, you better build them manually instead and handle the linking of pads in your code. </pre> <fieldset class="mimeAttachmentHeader"></fieldset> <pre wrap="">_______________________________________________ gstreamer-devel mailing list <a class="moz-txt-link-abbreviated" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> <a class="moz-txt-link-freetext" href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </pre> </blockquote> </body> </html>