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test-onvif example: Add support for AAC backchannel

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Detect upstream backchannel codec and support AAC in MPEG4-GENERIC
encapsulation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8378>
This commit is contained in:
Jan Schmidt 2025-01-29 20:42:55 +11:00 committed by GStreamer Marge Bot
parent d2dcfea052
commit 9adc7bd9f0
1 changed files with 53 additions and 3 deletions

56
subprojects/gst-plugins-good/tests/examples/rtsp/test-onvif.c

@ -35,10 +35,60 @@ setup_backchannel_shoveler (GstElement * rtspsrc, GstCaps * caps)
{
GstElement *appsink;
backpipe = gst_parse_launch ("audiotestsrc is-live=true wave=red-noise ! "
"mulawenc ! rtppcmupay ! appsink name=out", NULL);
if (!backpipe)
gst_println ("Have audio backchannel caps %" GST_PTR_FORMAT "\n", caps);
const GstStructure *s = gst_caps_get_structure (caps, 0);
const gchar *encoding = gst_structure_get_string (s, "encoding-name");
if (encoding == NULL) {
g_error
("Could not setup backchannel pipeline: Missing encoding-name field");
g_main_loop_quit (loop);
}
GError *error = NULL;
if (g_str_equal (encoding, "PCMU")) {
backpipe =
gst_parse_launch
("audiotestsrc is-live=true wave=red-noise ! capsfilter name=ratefilter ! "
"rtppcmupay ! appsink name=out", &error);
} else if (g_str_equal (encoding, "MPEG4-GENERIC")) {
backpipe =
gst_parse_launch
("audiotestsrc is-live=true wave=red-noise ! capsfilter name=ratefilter ! "
"voaacenc ! aacparse ! rtpmp4gpay ! appsink name=out", &error);
} else {
g_error ("Could not setup backchannel pipeline: Unsupported encoding %s",
encoding);
g_main_loop_quit (loop);
}
if (!backpipe) {
g_error ("Could not setup backchannel pipeline");
if (error != NULL) {
g_error ("Error: %s", error->message);
g_clear_error (&error);
}
g_main_loop_quit (loop);
}
gint rate = 32000;
if (!gst_structure_get_int (s, "clock-rate", &rate)) {
g_error ("Could not setup backchannel pipeline: Missing clock-rate field");
g_main_loop_quit (loop);
}
GstElement *rate_filter =
gst_bin_get_by_name (GST_BIN (backpipe), "ratefilter");
g_assert (rate_filter != NULL);
GstCaps *rate_caps =
gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, rate, NULL);
g_object_set (G_OBJECT (rate_filter), "caps", rate_caps, NULL);
gst_caps_unref (rate_caps);
gst_object_unref (GST_OBJECT (rate_filter));
appsink = gst_bin_get_by_name (GST_BIN (backpipe), "out");
g_object_set (G_OBJECT (appsink), "caps", caps, "emit-signals", TRUE, NULL);