<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"> <head> <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> <meta name="Generator" content="Microsoft Word 15 (filtered medium)"> <style></style> </head> <body lang="NO-BOK" link="#0563C1" vlink="#954F72" style="word-wrap:break-word"> <div class="WordSection1"> Hello. I have a GStreamer pipeline that sets up a live stream from a camera which is split into several branches via a tee. One of the branches is recording video and one branch should be served via RTSP to a client which may connect to it at a later point in time. I have some problems in getting this to work.<o:p></o:p> <o:p> </o:p> To debug this, I looked at the example <a href="https://github.com/GStreamer/gst-rtsp-server/blob/master/examples/test-appsrc2.c"> https://github.com/GStreamer/gst-rtsp-server/blob/master/examples/test-appsrc2.c</a>. In this example, a pipeline is set up that ends in an appsink. The RTSP server sets up a pipeline that starts with an appsrc that pulls data from the appsink. The pipeline that ends in the appsink is started in the media_configure function.<o:p></o:p> <o:p> </o:p> This does not work for me, as I need the pipeline to start running to feed the other branches (e.g. the one that records the video).<o:p></o:p> <o:p> </o:p> I tried to modify the example to set the pipeline to PLAYING state when it starts up. If I do this, I am no longer able to connect to it (see below).<o:p></o:p> <o:p> </o:p> D:\gstreamer\1.0\msvc_x86_64\bin>gst-launch-1.0 -v playbin uri=rtsp://localhost:8554/test<o:p></o:p> Setting pipeline to PAUSED ...<o:p></o:p> Pipeline is live and does not need PREROLL ...<o:p></o:p> /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: ring-buffer-max-size = 0<o:p></o:p> /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-size = -1<o:p></o:p> /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-duration = -1<o:p></o:p> /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: force-sw-decoders = false<o:p></o:p> /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: use-buffering = false<o:p></o:p> /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: download = false<o:p></o:p> /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: uri = rtsp://localhost:8554/test<o:p></o:p> /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: connection-speed = 0<o:p></o:p> /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: source = "\(GstRTSPSrc\)\ source"<o:p></o:p> Progress: (open) Opening Stream<o:p></o:p> Pipeline is PREROLLED ...<o:p></o:p> Prerolled, waiting for progress to finish...<o:p></o:p> Progress: (connect) Connecting to rtsp://localhost:8554/test<o:p></o:p> Progress: (open) Retrieving server options<o:p></o:p> Progress: (open) Retrieving media info<o:p></o:p> ERROR: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source: Could not read from resource.<o:p></o:p> Additional debug info:<o:p></o:p> ../gst/rtsp/gstrtspsrc.c(6408): gst_rtsp_src_receive_response (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source:<o:p></o:p> Could not receive message. (Timeout while waiting for server response)<o:p></o:p> ERROR: pipeline doesn't want to preroll.<o:p></o:p> Setting pipeline to NULL ...<o:p></o:p> ERROR: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source: Could not read from resource.<o:p></o:p> Additional debug info:<o:p></o:p> ../gst/rtsp/gstrtspsrc.c(6506): gst_rtspsrc_try_send (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source:<o:p></o:p> Could not receive message. (Timeout while waiting for server response)<o:p></o:p> ERROR: pipeline doesn't want to preroll.<o:p></o:p> Freeing pipeline ...<o:p></o:p> <o:p> </o:p> What is the reason for this? How do you recommend to set up my pipeline if I would like my other branches of the pipeline to play and allow the RTSP client to connect at a later point in time?<o:p></o:p> <o:p> </o:p> See my code below<o:p></o:p> <o:p> </o:p> ---------- code below------------<o:p></o:p> <o:p> </o:p> #include <gst/app/app.h><o:p></o:p> #include <gst/gst.h><o:p></o:p> #include <gst/rtsp-server/rtsp-server.h><o:p></o:p> #include <stdio.h><o:p></o:p> <o:p> </o:p> typedef struct {<o:p></o:p> GstElement *vid_appsink;<o:p></o:p> GstElement *vid_appsrc;<o:p></o:p> } MyContext;<o:p></o:p> <o:p> </o:p> // Main pipeline<o:p></o:p> GstElement *pipeline;<o:p></o:p> <o:p> </o:p> /* called when we need to give data to an appsrc */<o:p></o:p> static void need_data(GstElement *appsrc, guint unused, MyContext *ctx) {<o:p></o:p> GstSample *sample;<o:p></o:p> GstFlowReturn ret;<o:p></o:p> <o:p> </o:p> sample = gst_app_sink_pull_sample(GST_APP_SINK(ctx->vid_appsink));<o:p></o:p> <o:p> </o:p> if (sample) {<o:p></o:p> GstBuffer *buffer = gst_sample_get_buffer(sample);<o:p></o:p> GstSegment *seg = gst_sample_get_segment(sample);<o:p></o:p> GstClockTime pts, dts;<o:p></o:p> <o:p> </o:p> /* Convert the PTS/DTS to running time so they start from 0 */<o:p></o:p> pts = GST_BUFFER_PTS(buffer);<o:p></o:p> if (GST_CLOCK_TIME_IS_VALID(pts))<o:p></o:p> pts = gst_segment_to_running_time(seg, GST_FORMAT_TIME, pts);<o:p></o:p> <o:p> </o:p> dts = GST_BUFFER_DTS(buffer);<o:p></o:p> if (GST_CLOCK_TIME_IS_VALID(dts))<o:p></o:p> dts = gst_segment_to_running_time(seg, GST_FORMAT_TIME, dts);<o:p></o:p> <o:p> </o:p> if (buffer) {<o:p></o:p> /* Make writable so we can adjust the timestamps */<o:p></o:p> buffer = gst_buffer_copy(buffer);<o:p></o:p> GST_BUFFER_PTS(buffer) = pts;<o:p></o:p> GST_BUFFER_DTS(buffer) = dts;<o:p></o:p> g_signal_emit_by_name(appsrc, "push-buffer", buffer, &ret);<o:p></o:p> }<o:p></o:p> <o:p> </o:p> /* we don't need the appsink sample anymore */<o:p></o:p> gst_sample_unref(sample);<o:p></o:p> }<o:p></o:p> }<o:p></o:p> <o:p> </o:p> static void ctx_free(MyContext *ctx) {<o:p></o:p> gst_object_unref(ctx->vid_appsrc);<o:p></o:p> g_free(ctx);<o:p></o:p> }<o:p></o:p> <o:p> </o:p> /* called when a new media pipeline is constructed. We can query the<o:p></o:p> * pipeline and configure our appsrc */<o:p></o:p> static void media_configure(GstRTSPMediaFactory *factory, GstRTSPMedia *media,<o:p></o:p> gpointer user_data) {<o:p></o:p> GstElement *element, *appsrc, *appsink;<o:p></o:p> GstCaps *caps;<o:p></o:p> MyContext *ctx;<o:p></o:p> <o:p> </o:p> ctx = g_new0(MyContext, 1);<o:p></o:p> <o:p> </o:p> g_object_set_data_full(G_OBJECT(media), "rtsp-extra-data", ctx,<o:p></o:p> (GDestroyNotify)ctx_free);<o:p></o:p> <o:p> </o:p> element = gst_rtsp_media_get_element(media);<o:p></o:p> <o:p> </o:p> caps = gst_caps_new_simple("video/x-h264", "stream-format", G_TYPE_STRING,<o:p></o:p> "byte-stream", "alignment", G_TYPE_STRING, "au",<o:p></o:p> "width", G_TYPE_INT, 384, "height", G_TYPE_INT,<o:p></o:p> 288, "framerate", GST_TYPE_FRACTION, 15, 1, NULL);<o:p></o:p> ctx->vid_appsrc = appsrc =<o:p></o:p> gst_bin_get_by_name_recurse_up(GST_BIN(element), "videosrc");<o:p></o:p> ctx->vid_appsink = appsink = gst_bin_get_by_name(GST_BIN(pipeline), "vid");<o:p></o:p> <o:p> </o:p> gst_util_set_object_arg(G_OBJECT(appsrc), "format", "time");<o:p></o:p> g_object_set(G_OBJECT(appsrc), "caps", caps, NULL);<o:p></o:p> g_object_set(G_OBJECT(appsink), "caps", caps, NULL);<o:p></o:p> <o:p> </o:p> /* install the callback that will be called when a buffer is needed */<o:p></o:p> g_signal_connect(appsrc, "need-data", (GCallback)need_data, ctx);<o:p></o:p> gst_caps_unref(caps);<o:p></o:p> <o:p> </o:p> gst_object_unref(element);<o:p></o:p> }<o:p></o:p> <o:p> </o:p> int main(int argc, char *argv[]) {<o:p></o:p> GMainLoop *loop;<o:p></o:p> GstRTSPServer *server;<o:p></o:p> GstRTSPMountPoints *mounts;<o:p></o:p> GstRTSPMediaFactory *factory;<o:p></o:p> <o:p> </o:p> gst_init(&argc, &argv);<o:p></o:p> <o:p> </o:p> loop = g_main_loop_new(NULL, FALSE);<o:p></o:p> <o:p> </o:p> server = gst_rtsp_server_new();<o:p></o:p> mounts = gst_rtsp_server_get_mount_points(server);<o:p></o:p> <o:p> </o:p> factory = gst_rtsp_media_factory_new();<o:p></o:p> gst_rtsp_media_factory_set_launch(<o:p></o:p> factory,<o:p></o:p> "( appsrc name=videosrc ! h264parse ! rtph264pay name=pay0 pt=96 )");<o:p></o:p> <o:p> </o:p> g_signal_connect(factory, "media-configure", (GCallback)media_configure,<o:p></o:p> NULL);<o:p></o:p> <o:p> </o:p> gst_rtsp_mount_points_add_factory(mounts, "/test", factory);<o:p></o:p> <o:p> </o:p> g_object_unref(mounts);<o:p></o:p> gst_rtsp_server_attach(server, NULL);<o:p></o:p> <o:p> </o:p> pipeline = gst_parse_launch(<o:p></o:p> "videotestsrc is-live=true ! x264enc speed-preset=superfast "<o:p></o:p> "tune=zerolatency ! h264parse ! appsink name=vid max-buffers=1 "<o:p></o:p> "drop=true",<o:p></o:p> NULL);<o:p></o:p> <o:p> </o:p> /* Start playing */<o:p></o:p> gst_element_set_state(pipeline, GST_STATE_PLAYING);<o:p></o:p> <o:p> </o:p> /* start serving */<o:p></o:p> g_print("stream ready at rtsp://127.0.0.1:8554/test\n");<o:p></o:p> g_main_loop_run(loop);<o:p></o:p> <o:p> </o:p> return 0;<o:p></o:p> }<o:p></o:p> <o:p> </o:p> Best regards,<o:p></o:p> -�ystein<o:p></o:p> <o:p> </o:p> </div> <img src="https://t.sidekickopen14.com/s3t/o/5/f18dQhb0S7n28cN5byW5lN9tK2zGCvGW40Fv_62SXv86W6W0__H3LyCNMN4WJ49z1DJ1Kf3fTmtg03?si=8000000002644914&pi=3d0d794c-ab0f-44d4-b714-f428444cfc89" width="1" height="1" style="display:none!important;"> </body> </html>