question: how to properly set up pipeline rtspsrc->decodebin->playsink

[maksim]

Hi, I have set up pipeline to play rtsp stream, utilizing rtspsrc, two
decodebin (audio and video) and playsink for output, so the pipeline can
play either audio or video stream but i can not making working both. Can
somebody give me clue what is wrong in my pipeline?

here is snippet code:
    /* Create our own GLib Main Context and make it the default one */
    context = g_main_context_new ();
    g_main_context_push_thread_default(context);
 
    /* Create a GLib Main Loop */
    GST_DEBUG ("Create main loop...");
    main_loop = g_main_loop_new (context, FALSE);

    /* Build pipeline */
    pipeline = gst_pipeline_new("pipeline");
   
    /* Create source element */
    source = gst_element_factory_make("rtspsrc", "source");
    if (!source) {
      GST_ERROR("Could not create rtspsrc source");
      return;
    }

    /* Set video Source */
    g_object_set(G_OBJECT(source), "do-rtcp", TRUE, NULL);
    g_object_set(G_OBJECT(source), "latency", 0, NULL);
    g_object_set(G_OBJECT(source), "tls-validation-flags",
G_TLS_CERTIFICATE_VALIDATE_ALL, NULL);

    NSString *resources = [[NSBundle mainBundle] resourcePath];
    const gchar *resources_dir = [resources UTF8String];
    gchar *ca_certificates = g_build_filename (resources_dir, "ssl",
"certs", "ca-certificates.crt", NULL);
    g_setenv ("CA_CERTIFICATES", ca_certificates, TRUE);

    GTlsDatabase *db = NULL;
    if (ca_certificates) {
        GTlsBackend *backend = g_tls_backend_get_default();
        if (backend) {
        db = g_tls_file_database_new(ca_certificates, NULL);
        if (db)
            g_tls_backend_set_default_database(backend, db);
        }
    }

    if (!db) {
        GST_ERROR("failed to parse CA CERT: %s\n", error->message);
        return;
    }

    g_object_set(G_OBJECT(source), "tls-database", db, NULL);

    /* create decode sinc */
    decodeAudioSink = gst_element_factory_make("decodebin", "adecoder");
    decodeVideoSink = gst_element_factory_make("decodebin", "vdecoder");

    /* create audio/video output */
    playSink = gst_element_factory_make ("playsink", "sink");
    gst_util_set_object_arg(G_OBJECT(playSink), "flags",
"soft-colorbalance+soft-volume+vis+text+audio+video");

    /* Instruct the bus to emit signals for each received message, and
connect to the interesting signals */
    bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
    gst_bus_add_watch(bus, my_bus_callback, main_loop);
    gst_object_unref(bus);

    /* link elements */
    gst_bin_add_many(GST_BIN(pipeline), source, decodeAudioSink,
decodeVideoSink, playSink, NULL);
    gst_element_link(source, decodeAudioSink);
    gst_element_link(source, decodeVideoSink);
    gst_element_link(decodeVideoSink, playSink);
    gst_element_link(decodeAudioSink, playSink);

    g_signal_connect(source, "pad-added", G_CALLBACK(cb_pad_added),
(__bridge void *)self);
    g_signal_connect(decodeAudioSink, "pad-added", G_CALLBACK(cb_pad_added),
(__bridge void *)self);
    g_signal_connect(decodeVideoSink, "pad-added", G_CALLBACK(cb_pad_added),
(__bridge void *)self);

    /* Set the pipeline to READY, so it can already accept a window handle
*/
    gst_element_set_state(pipeline, GST_STATE_READY);

    videoSink = gst_bin_get_by_interface(GST_BIN(pipeline),
GST_TYPE_VIDEO_OVERLAY);
    if (!videoSink) {
        GST_ERROR("Could not retrieve video sink");
        return;
    }
       
    gst_video_overlay_set_window_handle(GST_VIDEO_OVERLAY(videoSink),
(guintptr)(id)videoView);

*and cb_pad_added callback
*

static void cb_pad_added(GstElement *dec, GstPad *pad, GStreamerBackend
*self) {
    GstPad *sinkpad = NULL;
    GstPadLinkReturn ret;
    GstCaps *caps = NULL;
    GstStructure *str = NULL;
    const gchar *name = NULL;

    /* check media type */
    caps = gst_pad_query_caps(pad, NULL);
    str = gst_caps_get_structure(caps, 0);
    name = gst_structure_get_name(str);
    g_print("*** Linking to %s ***\n", name);

    /* We can now link this pad with the rtsp-decoder sink pad */
    if (g_str_has_prefix(name, "audio")) {
        GstElementClass *klass = GST_ELEMENT_GET_CLASS(self->playSink);
        GstPadTemplate *templ = gst_element_class_get_pad_template(klass,
"audio_sink");
        sinkpad = gst_element_request_pad(self->playSink, templ, NULL,
NULL);
    } else if (g_str_has_prefix (name, "video")) {
        GstElementClass *klass = GST_ELEMENT_GET_CLASS(self->playSink);
        GstPadTemplate *templ = gst_element_class_get_pad_template(klass,
"video_sink");
        sinkpad = gst_element_request_pad(self->playSink, templ, NULL,
NULL);
    } else if (g_str_has_prefix (name, "text")) {
        GstElementClass *klass = GST_ELEMENT_GET_CLASS(self->playSink);
        GstPadTemplate *templ = gst_element_class_get_pad_template(klass,
"text_sink");
        sinkpad = gst_element_request_pad(self->playSink, templ, NULL,
NULL);
    } else {
        const gchar *media = gst_structure_get_string(str, "media");
        if (g_strrstr(media, "audio")) {
            sinkpad = gst_element_get_static_pad(self->decodeAudioSink,
"sink");
        } else if (g_strrstr(media, "video")) {
            sinkpad = gst_element_get_static_pad(self->decodeVideoSink,
"sink");
        }
    }

    gst_caps_unref(caps);
   
    /* If our converter is already linked, we have nothing to do here */
    if (gst_pad_is_linked(sinkpad)) {
        g_print("*** We are already linked ***\n");
        gst_object_unref(sinkpad);
        return;
    } else {
        g_print("proceeding to linking ...\n");
    }
    ret = gst_pad_link(pad, sinkpad);

    if (GST_PAD_LINK_FAILED(ret)) {
        //failed
        g_print("failed to link dynamically\n");
    } else {
        //pass
        g_print("dynamically link successful\n");
    }

    gst_object_unref (sinkpad);
}





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[Mariano Koremblum]

I did not read all the code, but did you set the "location" parameter? the
one that tells where to actually read from.



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[Patrick Cusack]

I am sending a decklink signal (audio and video) to an RTMP server. I can send the video fine, but the audio does not come through. I am confident that the audio is properly routed. Would anything in my script explain why the audio isn’t muxed into my stream?

RTMP_DEST=rtmp://www.myaddress.com:1935/live/stream1

gst-launch-1.0  \
        decklinkvideosrc device-number=0 mode=1080p2398  connection=sdi \
        ! videoconvert \
        ! videorate \
        ! x264enc speed-preset=veryfast tune=zerolatency bitrate=2000 \
        ! video/x-h264, profile=baseline \
        ! queue \
        ! flvmux name=mux \
        ! rtmpsink location=$RTMP_DEST
        decklinkaudiosrc device-number=0 \
        ! audioconvert \
        ! audioresample \
        ! audio/x-raw,channels=2,rate=48000 \
        ! voaacenc bitrate=128000 \
        ! audio/mpeg \
        ! aacparse \
        ! audio/mpeg, mpegversion=4 \
        ! mux.

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