rtspsrc: Implement ONVIF backchannel support
Set backchannel=onvif to enable, and use the 'push-backchannel-sample'
action signal with the correct stream id.
diff --git a/configure.ac b/configure.ac
index 90e8f1a..6736c4b 100644
--- a/configure.ac
+++ b/configure.ac
@@ -1294,6 +1294,7 @@
tests/examples/jack/Makefile
tests/examples/level/Makefile
tests/examples/rtp/Makefile
+tests/examples/rtsp/Makefile
tests/examples/shapewipe/Makefile
tests/examples/spectrum/Makefile
tests/examples/v4l2/Makefile
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index d636073..e424f47 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -126,6 +126,7 @@
SIGNAL_REQUEST_RTCP_KEY,
SIGNAL_ACCEPT_CERTIFICATE,
SIGNAL_BEFORE_SEND,
+ SIGNAL_PUSH_BACKCHANNEL_BUFFER,
LAST_SIGNAL
};
@@ -200,6 +201,32 @@
return ntp_time_source_type;
}
+enum _GstRtspBackchannel
+{
+ BACKCHANNEL_NONE,
+ BACKCHANNEL_ONVIF
+};
+
+#define GST_TYPE_RTSP_BACKCHANNEL (gst_rtsp_backchannel_get_type())
+static GType
+gst_rtsp_backchannel_get_type (void)
+{
+ static GType backchannel_type = 0;
+ static const GEnumValue backchannel_values[] = {
+ {BACKCHANNEL_NONE, "No backchannel", "none"},
+ {BACKCHANNEL_ONVIF, "ONVIF audio backchannel", "onvif"},
+ {0, NULL, NULL},
+ };
+
+ if (G_UNLIKELY (backchannel_type == 0)) {
+ backchannel_type =
+ g_enum_register_static ("GstRTSPBackchannel", backchannel_values);
+ }
+ return backchannel_type;
+}
+
+#define BACKCHANNEL_ONVIF_HDR_REQUIRE_VAL "www.onvif.org/ver20/backchannel"
+
#define DEFAULT_LOCATION NULL
#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
#define DEFAULT_DEBUG FALSE
@@ -236,6 +263,7 @@
#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT G_GUINT64_CONSTANT(0)
#define DEFAULT_MAX_TS_OFFSET G_GINT64_CONSTANT(3000000000)
#define DEFAULT_VERSION GST_RTSP_VERSION_1_0
+#define DEFAULT_BACKCHANNEL GST_RTSP_BACKCHANNEL_NONE
enum
{
@@ -279,6 +307,7 @@
PROP_MAX_TS_OFFSET_ADJUSTMENT,
PROP_MAX_TS_OFFSET,
PROP_DEFAULT_VERSION,
+ PROP_BACKCHANNEL,
};
#define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
@@ -359,6 +388,9 @@
static void
gst_rtspsrc_print_sdp_message (GstRTSPSrc * src, const GstSDPMessage * msg);
+static GstFlowReturn gst_rtspsrc_push_backchannel_buffer (GstRTSPSrc * src,
+ guint id, GstSample * sample);
+
typedef struct
{
guint8 pt;
@@ -830,6 +862,20 @@
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
+ * GstRtpSrc:backchannel
+ *
+ * Select a type of backchannel to setup with the RTSP server.
+ * Default value is "none". Allowed values are "none" and "onvif".
+ *
+ * Since: 1.14
+ */
+ g_object_class_install_property (gobject_class, PROP_BACKCHANNEL,
+ g_param_spec_enum ("backchannel", "Backchannel type",
+ "The type of backchannel to setup. Default is 'none'.",
+ GST_TYPE_RTSP_BACKCHANNEL, BACKCHANNEL_NONE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
* GstRTSPSrc::handle-request:
* @rtspsrc: a #GstRTSPSrc
* @request: a #GstRTSPMessage
@@ -965,6 +1011,19 @@
g_cclosure_marshal_generic, G_TYPE_BOOLEAN,
1, GST_TYPE_RTSP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
+ /**
+ * GstRTSPSrc::push-backchannel-buffer:
+ * @rtspsrc: a #GstRTSPSrc
+ * @buffer: RTP buffer to send back
+ *
+ *
+ */
+ gst_rtspsrc_signals[SIGNAL_PUSH_BACKCHANNEL_BUFFER] =
+ g_signal_new ("push-backchannel-buffer", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRTSPSrcClass,
+ push_backchannel_buffer), NULL, NULL, NULL, GST_TYPE_FLOW_RETURN, 2,
+ G_TYPE_UINT, GST_TYPE_BUFFER);
+
gstelement_class->send_event = gst_rtspsrc_send_event;
gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
gstelement_class->change_state = gst_rtspsrc_change_state;
@@ -980,6 +1039,8 @@
gstbin_class->handle_message = gst_rtspsrc_handle_message;
+ klass->push_backchannel_buffer = gst_rtspsrc_push_backchannel_buffer;
+
gst_rtsp_ext_list_init ();
}
@@ -1335,6 +1396,9 @@
case PROP_DEFAULT_VERSION:
rtspsrc->default_version = g_value_get_enum (value);
break;
+ case PROP_BACKCHANNEL:
+ rtspsrc->backchannel = g_value_get_enum (value);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@@ -1705,7 +1769,9 @@
else
goto unknown_proto;
- if (gst_sdp_media_get_attribute_val (media, "recvonly") != NULL)
+ if (gst_sdp_media_get_attribute_val (media, "recvonly") != NULL &&
+ /* We want to setup caps for streams configured as backchannel */
+ !stream->is_backchannel)
goto recvonly_media;
/* Parse global SDP attributes once */
@@ -1779,7 +1845,7 @@
}
recvonly_media:
{
- GST_DEBUG_OBJECT (src, "recvonly media ignored");
+ GST_WARNING_OBJECT (src, "recvonly media ignored, no backchannel");
return;
}
}
@@ -1839,10 +1905,16 @@
stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
stream->mikey = NULL;
stream->stream_id = NULL;
+ stream->is_backchannel = FALSE;
g_mutex_init (&stream->conninfo.send_lock);
g_mutex_init (&stream->conninfo.recv_lock);
g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
+ /* stream is recvonly and onvif backchannel is requested */
+ if (gst_sdp_media_get_attribute_val (media, "recvonly") != NULL &&
+ src->backchannel != BACKCHANNEL_NONE)
+ stream->is_backchannel = TRUE;
+
/* collect bandwidth information for this steam. FIXME, configure in the RTP
* session manager to scale RTCP. */
gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
@@ -1940,10 +2012,10 @@
gst_object_unref (stream->udpsink[i]);
}
}
- if (stream->fakesrc) {
- gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
- gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
- gst_object_unref (stream->fakesrc);
+ if (stream->rtpsrc) {
+ gst_element_set_state (stream->rtpsrc, GST_STATE_NULL);
+ gst_bin_remove (GST_BIN_CAST (src), stream->rtpsrc);
+ gst_object_unref (stream->rtpsrc);
}
if (stream->srcpad) {
gst_pad_set_active (stream->srcpad, FALSE);
@@ -2763,6 +2835,32 @@
return res;
}
+static GstFlowReturn
+gst_rtspsrc_push_backchannel_buffer (GstRTSPSrc * src, guint id,
+ GstSample * sample)
+{
+ GstFlowReturn res = GST_FLOW_OK;
+ GstRTSPStream *stream;
+
+ if (!src->conninfo.connected || src->state != GST_RTSP_STATE_PLAYING)
+ goto out;
+
+ stream = find_stream (src, &id, (gpointer) find_stream_by_id);
+ if (stream == NULL) {
+ GST_ERROR_OBJECT (src, "no stream with id %u", id);
+ goto out;
+ }
+
+ g_signal_emit_by_name (stream->rtpsrc, "push-sample", sample, &res);
+ GST_DEBUG_OBJECT (src, "sent backchannel RTP sample %p: %s", sample,
+ gst_flow_get_name (res));
+
+out:
+ gst_sample_unref (sample);
+
+ return res;
+}
+
static GstPadProbeReturn
pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
{
@@ -2801,6 +2899,35 @@
return TRUE;
}
+static gboolean
+add_backchannel_fakesink (GstRTSPSrc * src, GstRTSPStream * stream,
+ GstPad * srcpad)
+{
+ GstPad *sinkpad;
+ GstElement *fakesink;
+
+ fakesink = gst_element_factory_make ("fakesink", NULL);
+ if (fakesink == NULL) {
+ GST_ERROR_OBJECT (src, "no fakesink");
+ return FALSE;
+ }
+
+ sinkpad = gst_element_get_static_pad (fakesink, "sink");
+
+ GST_DEBUG_OBJECT (src, "backchannel stream %p, hooking fakesink", stream);
+
+ gst_bin_add (GST_BIN_CAST (src), fakesink);
+ if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
+ GST_WARNING_OBJECT (src, "could not link to fakesink");
+ return FALSE;
+ }
+
+ gst_object_unref (sinkpad);
+
+ gst_element_sync_state_with_parent (fakesink);
+ return TRUE;
+}
+
/* this callback is called when the session manager generated a new src pad with
* payloaded RTP packets. We simply ghost the pad here. */
static void
@@ -2868,7 +2995,12 @@
gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
gst_pad_set_active (stream->srcpad, TRUE);
gst_pad_sticky_events_foreach (pad, copy_sticky_events, stream->srcpad);
- gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
+
+ /* don't add the srcpad if this is a recvonly stream */
+ if (stream->is_backchannel)
+ add_backchannel_fakesink (src, stream, stream->srcpad);
+ else
+ gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
if (all_added) {
GST_DEBUG_OBJECT (src, "We added all streams");
@@ -3898,7 +4030,7 @@
goto no_destination;
/* try to construct the fakesrc to the RTP port of the server to open up any
- * NAT firewalls */
+ * NAT firewalls or, if backchannel, construct an appsrc */
if (do_rtp) {
GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
rtp_port);
@@ -3932,25 +4064,36 @@
g_object_unref (socket);
}
- /* the source for the dummy packets to open up NAT */
- stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
- if (stream->fakesrc == NULL)
- goto no_fakesrc_element;
+ if (stream->is_backchannel) {
+ /* appsrc is for the app to shovel data using push-backchannel-buffer */
+ stream->rtpsrc = gst_element_factory_make ("appsrc", NULL);
+ if (stream->rtpsrc == NULL)
+ goto no_appsrc_element;
- /* random data in 5 buffers, a size of 200 bytes should be fine */
- g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
- "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
+ /* interal use only, don't emit signals */
+ g_object_set (G_OBJECT (stream->rtpsrc), "emit-signals", TRUE,
+ "is-live", TRUE, NULL);
+ } else {
+ /* the source for the dummy packets to open up NAT */
+ stream->rtpsrc = gst_element_factory_make ("fakesrc", NULL);
+ if (stream->rtpsrc == NULL)
+ goto no_fakesrc_element;
+
+ /* random data in 5 buffers, a size of 200 bytes should be fine */
+ g_object_set (G_OBJECT (stream->rtpsrc), "filltype", 3, "num-buffers", 5,
+ "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
+ }
/* keep everything locked */
gst_element_set_locked_state (stream->udpsink[0], TRUE);
- gst_element_set_locked_state (stream->fakesrc, TRUE);
+ gst_element_set_locked_state (stream->rtpsrc, TRUE);
gst_object_ref (stream->udpsink[0]);
gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
- gst_object_ref (stream->fakesrc);
- gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
+ gst_object_ref (stream->rtpsrc);
+ gst_bin_add (GST_BIN_CAST (src), stream->rtpsrc);
- gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
+ gst_element_link_pads_full (stream->rtpsrc, "src", stream->udpsink[0],
"sink", GST_PAD_LINK_CHECK_NOTHING);
}
if (do_rtcp) {
@@ -4021,6 +4164,11 @@
GST_ERROR_OBJECT (src, "no UDP sink element found");
return FALSE;
}
+no_appsrc_element:
+ {
+ GST_ERROR_OBJECT (src, "no appsrc element found");
+ return FALSE;
+ }
no_fakesrc_element:
{
GST_ERROR_OBJECT (src, "no fakesrc element found");
@@ -4094,8 +4242,8 @@
case GST_RTSP_LOWER_TRANS_UDP:
if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
goto transport_failed;
- /* configure udpsinks back to the server for RTCP messages and for the
- * dummy RTP messages to open NAT. */
+ /* configure udpsinks back to the server for RTCP messages, for the
+ * dummy RTP messages to open NAT, and for the backchannel */
if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
goto transport_failed;
break;
@@ -4103,8 +4251,12 @@
goto unknown_transport;
}
- if (outpad) {
- GST_DEBUG_OBJECT (src, "creating ghostpad");
+ /* using backchannel and no manager, hence no srcpad for this stream */
+ if (outpad && stream->is_backchannel) {
+ add_backchannel_fakesink (src, stream, outpad);
+ gst_object_unref (outpad);
+ } else if (outpad) {
+ GST_DEBUG_OBJECT (src, "creating ghostpad for stream %p", stream);
gst_pad_use_fixed_caps (outpad);
@@ -4128,17 +4280,17 @@
/* ERRORS */
transport_failed:
{
- GST_DEBUG_OBJECT (src, "failed to configure transport");
+ GST_WARNING_OBJECT (src, "failed to configure transport");
return FALSE;
}
unknown_transport:
{
- GST_DEBUG_OBJECT (src, "unknown transport");
+ GST_WARNING_OBJECT (src, "unknown transport");
return FALSE;
}
no_manager:
{
- GST_DEBUG_OBJECT (src, "cannot get a session manager");
+ GST_WARNING_OBJECT (src, "cannot get a session manager");
return FALSE;
}
}
@@ -4157,13 +4309,18 @@
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
- if (stream->fakesrc && stream->udpsink[0]) {
+ if (!stream->rtpsrc || !stream->udpsink[0])
+ continue;
+
+ if (stream->is_backchannel)
+ GST_DEBUG_OBJECT (src, "starting backchannel stream %p", stream);
+ else
GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
- gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
- gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
- gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
- gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
- }
+
+ gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
+ gst_element_set_state (stream->rtpsrc, GST_STATE_NULL);
+ gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
+ gst_element_set_state (stream->rtpsrc, GST_STATE_PLAYING);
}
return TRUE;
}
@@ -4205,7 +4362,10 @@
/* add the pad */
if (!stream->added) {
GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
- gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
+ if (stream->is_backchannel)
+ add_backchannel_fakesink (src, stream, stream->srcpad);
+ else
+ gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
stream->added = TRUE;
}
}
@@ -6529,7 +6689,7 @@
caps = stream_get_caps_for_pt (stream, stream->default_pt);
if (caps == NULL) {
- GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
+ GST_WARNING_OBJECT (src, "skipping stream %p, no caps", stream);
continue;
}
@@ -6574,13 +6734,14 @@
/* skip setup if we have no URL for it */
if (stream->conninfo.location == NULL) {
- GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
+ GST_WARNING_OBJECT (src, "skipping stream %p, no setup", stream);
continue;
}
if (src->conninfo.connection == NULL) {
if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
- GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
+ GST_WARNING_OBJECT (src, "skipping stream %p, failed to connect",
+ stream);
continue;
}
conninfo = &stream->conninfo;
@@ -6653,6 +6814,10 @@
/* select transport */
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
+ if (stream->is_backchannel && src->backchannel == BACKCHANNEL_ONVIF)
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE,
+ BACKCHANNEL_ONVIF_HDR_REQUIRE_VAL);
+
/* set up keys */
if (stream->profile == GST_RTSP_PROFILE_SAVP ||
stream->profile == GST_RTSP_PROFILE_SAVPF) {
@@ -7168,6 +7333,11 @@
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
"application/sdp");
+ if (src->backchannel == BACKCHANNEL_ONVIF)
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE,
+ BACKCHANNEL_ONVIF_HDR_REQUIRE_VAL);
+ /* TODO: Handle the case when backchannel is unsupported and goto restart */
+
/* send DESCRIBE */
GST_DEBUG_OBJECT (src, "send describe...");
@@ -7395,6 +7565,10 @@
if (res < 0)
goto create_request_failed;
+ if (stream->is_backchannel && src->backchannel == BACKCHANNEL_ONVIF)
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE,
+ BACKCHANNEL_ONVIF_HDR_REQUIRE_VAL);
+
if (async)
GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
@@ -7736,6 +7910,13 @@
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SEEK_STYLE,
seek_style);
+ /* when we have an ONVIF audio backchannel, the PLAY request must have the
+ * Require: header when doing either aggregate or non-aggregate control */
+ if (src->backchannel == BACKCHANNEL_ONVIF &&
+ (control || stream->is_backchannel))
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE,
+ BACKCHANNEL_ONVIF_HDR_REQUIRE_VAL);
+
if (async)
GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
@@ -7856,17 +8037,17 @@
/* ERRORS */
open_failed:
{
- GST_DEBUG_OBJECT (src, "failed to open stream");
+ GST_WARNING_OBJECT (src, "failed to open stream");
goto done;
}
not_supported:
{
- GST_DEBUG_OBJECT (src, "PLAY is not supported");
+ GST_WARNING_OBJECT (src, "PLAY is not supported");
goto done;
}
was_playing:
{
- GST_DEBUG_OBJECT (src, "we were already PLAYING");
+ GST_WARNING_OBJECT (src, "we were already PLAYING");
goto done;
}
create_request_failed:
@@ -7950,6 +8131,13 @@
setup_url)) < 0)
goto create_request_failed;
+ /* when we have an ONVIF audio backchannel, the PAUSE request must have the
+ * Require: header when doing either aggregate or non-aggregate control */
+ if (src->backchannel == BACKCHANNEL_ONVIF &&
+ (control || stream->is_backchannel))
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE,
+ BACKCHANNEL_ONVIF_HDR_REQUIRE_VAL);
+
if ((res =
gst_rtspsrc_send (src, conninfo, &request, &response, NULL,
NULL)) < 0)
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
index 57921d2..4e5adef 100644
--- a/gst/rtsp/gstrtspsrc.h
+++ b/gst/rtsp/gstrtspsrc.h
@@ -123,8 +123,8 @@
GstElement *udpsink[2];
GstPad *rtcppad;
- /* fakesrc for sending dummy data */
- GstElement *fakesrc;
+ /* fakesrc for sending dummy data or appsrc for sending backchannel data */
+ GstElement *rtpsrc;
/* state */
guint port;
@@ -161,6 +161,7 @@
gchar *destination;
gboolean is_multicast;
guint ttl;
+ gboolean is_backchannel;
/* A unique and stable id we will use for the stream start event */
gchar *stream_id;
@@ -254,6 +255,7 @@
guint64 max_ts_offset_adjustment;
gint64 max_ts_offset;
gboolean max_ts_offset_is_set;
+ gint backchannel;
/* state */
GstRTSPState state;
@@ -298,6 +300,8 @@
struct _GstRTSPSrcClass {
GstBinClass parent_class;
+
+ GstFlowReturn (*push_backchannel_buffer) (GstRTSPSrc *src, guint id, GstSample *sample);
};
GType gst_rtspsrc_get_type(void);
diff --git a/tests/examples/Makefile.am b/tests/examples/Makefile.am
index 3f9cee6..bc1ada4 100644
--- a/tests/examples/Makefile.am
+++ b/tests/examples/Makefile.am
@@ -17,9 +17,9 @@
endif
SUBDIRS = audiofx equalizer $(GTK_DIR) $(JACK_DIR) level \
- rtp shapewipe spectrum v4l2 $(CAIRO_DIR)
+ rtp rtsp shapewipe spectrum v4l2 $(CAIRO_DIR)
DIST_SUBDIRS = audiofx equalizer gtk jack level \
- rtp shapewipe spectrum v4l2 cairo
+ rtp rtsp shapewipe spectrum v4l2 cairo
include $(top_srcdir)/common/parallel-subdirs.mak
diff --git a/tests/examples/meson.build b/tests/examples/meson.build
index aa871a4..19f9a0d 100644
--- a/tests/examples/meson.build
+++ b/tests/examples/meson.build
@@ -4,6 +4,7 @@
subdir('level')
#FIXME: subdir('qt')
subdir('rtp')
+subdir('rtsp')
subdir('shapewipe')
subdir('v4l2')
diff --git a/tests/examples/rtsp/Makefile.am b/tests/examples/rtsp/Makefile.am
new file mode 100644
index 0000000..126ce8c
--- /dev/null
+++ b/tests/examples/rtsp/Makefile.am
@@ -0,0 +1,3 @@
+noinst_PROGRAMS = test-onvif
+test_onvif_CFLAGS = $(GST_CFLAGS)
+test_onvif_LDADD = $(GST_LIBS)
diff --git a/tests/examples/rtsp/meson.build b/tests/examples/rtsp/meson.build
new file mode 100644
index 0000000..7434c8a
--- /dev/null
+++ b/tests/examples/rtsp/meson.build
@@ -0,0 +1,5 @@
+executable('onvif-test', 'onvif-test.c',
+ dependencies: [gst_dep],
+ c_args : gst_plugins_good_args,
+ include_directories : [configinc],
+ install: false)
diff --git a/tests/examples/rtsp/test-onvif.c b/tests/examples/rtsp/test-onvif.c
new file mode 100644
index 0000000..671c20f
--- /dev/null
+++ b/tests/examples/rtsp/test-onvif.c
@@ -0,0 +1,107 @@
+#include <gst/gst.h>
+
+static GMainLoop *loop = NULL;
+static GstElement *backpipe = NULL;
+static gint stream_id = -1;
+
+#define PCMU_CAPS "application/x-rtp, media=audio, payload=0, clock-rate=8000, encoding-name=PCMU"
+
+static GstFlowReturn
+new_sample (GstElement * appsink, GstElement * rtspsrc)
+{
+ GstSample *sample;
+ GstFlowReturn ret = GST_FLOW_OK;
+
+ g_assert (stream_id != -1);
+
+ g_signal_emit_by_name (appsink, "pull-sample", &sample);
+
+ if (!sample)
+ goto out;
+
+ g_signal_emit_by_name (rtspsrc, "push-backchannel-buffer", stream_id, sample,
+ &ret);
+
+out:
+ return ret;
+}
+
+static void
+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)
+ g_error ("Could not setup backchannel pipeline");
+
+ appsink = gst_bin_get_by_name (GST_BIN (backpipe), "out");
+ g_object_set (G_OBJECT (appsink), "caps", caps, "emit-signals", TRUE, NULL);
+
+ g_signal_connect (appsink, "new-sample", G_CALLBACK (new_sample), rtspsrc);
+
+ g_print ("Playing backchannel shoveler\n");
+ gst_element_set_state (backpipe, GST_STATE_PLAYING);
+}
+
+static gboolean
+remove_extra_fields (GQuark field_id, GValue * value G_GNUC_UNUSED,
+ gpointer user_data G_GNUC_UNUSED)
+{
+ return !g_str_has_prefix (g_quark_to_string (field_id), "a-");
+}
+
+static gboolean
+find_backchannel (GstElement * rtspsrc, guint idx, GstCaps * caps,
+ gpointer user_data G_GNUC_UNUSED)
+{
+ GstStructure *s;
+ gchar *caps_str = gst_caps_to_string (caps);
+ g_print ("Selecting stream idx %u, caps %s\n", idx, caps_str);
+ g_free (caps_str);
+
+ s = gst_caps_get_structure (caps, 0);
+ if (gst_structure_has_field (s, "a-recvonly")) {
+ stream_id = idx;
+ caps = gst_caps_new_empty ();
+ s = gst_structure_copy (s);
+ gst_structure_set_name (s, "application/x-rtp");
+ gst_structure_filter_and_map_in_place (s, remove_extra_fields, NULL);
+ gst_caps_append_structure (caps, s);
+ setup_backchannel_shoveler (rtspsrc, caps);
+ }
+
+ return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+ GstElement *pipeline, *rtspsrc;
+ const gchar *location;
+
+ gst_init (&argc, &argv);
+
+ if (argc >= 2)
+ location = argv[1];
+ else
+ location = "rtsp://127.0.0.1:8554/test";
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ pipeline = gst_parse_launch ("rtspsrc backchannel=onvif debug=true name=r "
+ "r. ! queue ! decodebin ! queue ! xvimagesink async=false "
+ "r. ! queue ! decodebin ! queue ! pulsesink async=false ", NULL);
+ if (!pipeline)
+ g_error ("Failed to parse pipeline");
+
+ rtspsrc = gst_bin_get_by_name (GST_BIN (pipeline), "r");
+ g_object_set (G_OBJECT (rtspsrc), "location", location, NULL);
+ g_signal_connect (rtspsrc, "select-stream", G_CALLBACK (find_backchannel),
+ NULL);
+
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+ g_main_loop_run (loop);
+}
