【发布时间】：2015-06-17 16:37:00
【问题描述】：
-实际上我从链接下载了 gstreamer 的示例教程，
http://cgit.freedesktop.org/~slomo/gst-sdk-tutorials/
git://people.freedesktop.org/~slomo/gst-sdk-tutorials
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现在我在教程 3 中修改了以下代码，
-(void) app_function { GstBus *bus; GSource *bus_source; GError *error = NULL; GST_DEBUG ("Creating pipeline"); pipeline = gst_pipeline_new ("e-pipeline"); /* Create our own GLib Main Context and make it the default one */ context = g_main_context_new (); g_main_context_push_thread_default(context); /* Build pipeline */ // pipeline = gst_parse_launch("videotestsrc ! warptv ! videoconvert ! autovideosink", &error); source = gst_element_factory_make("udpsrc", "source"); g_object_set( G_OBJECT ( source), "port", 8001, NULL ); GstCaps *caps; caps = gst_caps_new_simple ("application/x-rtp", "encoding-name", G_TYPE_STRING, "H264", "payload", G_TYPE_INT, 96, "clock-rate", G_TYPE_INT, 90000, NULL); g_object_set (source, "caps", caps, NULL); rtp264depay = gst_element_factory_make ("rtph264depay", "rtph264depay"); h264parse = gst_element_factory_make ("h264parse", "h264parse"); vtdec = gst_element_factory_make ("vtdec", "vtdec"); glimagesink = gst_element_factory_make ("glimagesink", "glimagesink"); gst_bin_add_many (GST_BIN(pipeline), source, rtp264depay, h264parse, vtdec, glimagesink, NULL); if (error) { gchar *message = g_strdup_printf("Unable to build pipeline: %s", error->message); g_clear_error (&error); [self setUIMessage:message]; g_free (message); return; } /* Set the pipeline to READY, so it can already accept a window handle */ gst_element_set_state(pipeline, GST_STATE_READY); video_sink = gst_bin_get_by_interface(GST_BIN(pipeline), GST_TYPE_VIDEO_OVERLAY); if (!video_sink) { GST_ERROR ("Could not retrieve video sink"); return; } gst_video_overlay_set_window_handle(GST_VIDEO_OVERLAY(video_sink), (guintptr) (id) ui_video_view); /* Instruct the bus to emit signals for each received message, and connect to the interesting signals */ bus = gst_element_get_bus (pipeline); bus_source = gst_bus_create_watch (bus); g_source_set_callback (bus_source, (GSourceFunc) gst_bus_async_signal_func, NULL, NULL); g_source_attach (bus_source, context); g_source_unref (bus_source); g_signal_connect (G_OBJECT (bus), "message::error", (GCallback)error_cb, (__bridge void *)self); g_signal_connect (G_OBJECT (bus), "message::state-changed", (GCallback)state_changed_cb, (__bridge void *)self); gst_object_unref (bus); /* Create a GLib Main Loop and set it to run */ GST_DEBUG ("Entering main loop..."); main_loop = g_main_loop_new (context, FALSE); [self check_initialization_complete]; g_main_loop_run (main_loop); GST_DEBUG ("Exited main loop"); g_main_loop_unref (main_loop); main_loop = NULL; /* Free resources */ g_main_context_pop_thread_default(context); g_main_context_unref (context); gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); return;}
-现在我在 ipad 上运行应用程序，应用程序开始播放。
- 现在正在进入后台并返回前台 Gstreamer 流式更新在 UI 中不可见，但在 xcode 的网络使用情况下，我可以看到数据包正在接收....:(
提前致谢....iOS GEEKS....
【问题讨论】：
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我在 iPhone 的 Gstreamer 中也面临同样的问题。如果您找到解决方案，请发布。 :(
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pipeline = gst_parse_launch("udpsrc host=192.168.7.45 port=5000 !application/x-rtp, payload=96 !rtpjitterbuffer !rtph264depay !avdec_h264 !autovideosink sync=false text-overlay=false", ＆错误）;错误管道错误无法构建管道：没有元素“udpsrc”
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请解释您使用了哪些额外的变量以及类型，例如，来源，rtp264depay 等
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嗨，我是 Gstremer 的新手，我正在运行一个演示 // pipeline = gst_parse_launch("videotestsrc !warptv !videoconvert !autovideosink", &error);上面的行在我这边没有注释，我得到了错误 - “没有为 warptv 找到元素”？你有什么解决办法请帮帮我谢谢
标签： ios gstreamer ios8-extension