mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-01-10 17:35:59 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtspsrc: Implement ONVIF backchannel support

Browse source 
Set backchannel=onvif to enable, and use the 'push-backchannel-sample'
action signal with the correct stream id.
This commit is contained in:
Nirbheek Chauhan 2017-10-13 18:05:54 +03:00 committed by Sebastian Dröge
parent d61066e6b5
commit befa41cdf6
8 changed files with 352 additions and 43 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

1
configure.ac
View file

@ -1294,6 +1294,7 @@ tests/examples/gtk/Makefile
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

266
gst/rtsp/gstrtspsrc.c
View file

@ -126,6 +126,7 @@ enum
SIGNAL_REQUEST_RTCP_KEY,
SIGNAL_ACCEPT_CERTIFICATE,
SIGNAL_BEFORE_SEND,
SIGNAL_PUSH_BACKCHANNEL_BUFFER,
LAST_SIGNAL
};
@ -200,6 +201,32 @@ gst_rtsp_src_ntp_time_source_get_type (void)
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 @@ gst_rtsp_src_ntp_time_source_get_type (void)
#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 @@ enum
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 @@ gst_rtspsrc_print_rtsp_message (GstRTSPSrc * src, const GstRTSPMessage * msg);
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;
@ -829,6 +861,20 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
"changed to 0 (no limit)", 0, G_MAXINT64, DEFAULT_MAX_TS_OFFSET,
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
@ -965,6 +1011,19 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
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 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
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 @@ gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
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 @@ gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
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 @@ unknown_proto:
}
recvonly_media:
{
GST_DEBUG_OBJECT (src, "recvonly media ignored");
GST_WARNING_OBJECT (src, "recvonly media ignored, no backchannel");
return;
}
}
@ -1839,10 +1905,16 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx,
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_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
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 @@ gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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 @@ copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
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 @@ new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
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 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
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 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
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 @@ no_sink_element:
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 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
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 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
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 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
/* 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 @@ gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
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 @@ gst_rtspsrc_activate_streams (GstRTSPSrc * src)
/* 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 @@ gst_rtspsrc_setup_streams_start (GstRTSPSrc * src, gboolean async)
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 @@ gst_rtspsrc_setup_streams_start (GstRTSPSrc * src, gboolean async)
/* 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 @@ gst_rtspsrc_setup_streams_start (GstRTSPSrc * src, gboolean async)
/* 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 @@ restart:
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 @@ gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
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 @@ restart:
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 @@ done:
/* 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 @@ gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
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)

8
gst/rtsp/gstrtspsrc.h
View file

@ -123,8 +123,8 @@ struct _GstRTSPStream {
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 @@ struct _GstRTSPStream {
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 @@ struct _GstRTSPSrc {
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 _GstRTSPSrc {
struct _GstRTSPSrcClass {
GstBinClass parent_class;
GstFlowReturn (*push_backchannel_buffer) (GstRTSPSrc *src, guint id, GstSample *sample);
};
GType gst_rtspsrc_get_type(void);

4
tests/examples/Makefile.am
View file

@ -17,9 +17,9 @@ CAIRO_DIR=
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

1
tests/examples/meson.build
View file

@ -4,6 +4,7 @@ subdir('cairo')
subdir('level')
#FIXME: subdir('qt')
subdir('rtp')
subdir('rtsp')
subdir('shapewipe')
subdir('v4l2')

3
tests/examples/rtsp/Makefile.am Normal file
View file

@ -0,0 +1,3 @@
noinst_PROGRAMS = test-onvif
test_onvif_CFLAGS = $(GST_CFLAGS)
test_onvif_LDADD = $(GST_LIBS)

5
tests/examples/rtsp/meson.build Normal file
View file

@ -0,0 +1,5 @@
executable('onvif-test', 'onvif-test.c',
dependencies: [gst_dep],
c_args : gst_plugins_good_args,
include_directories : [configinc],
install: false)

107
tests/examples/rtsp/test-onvif.c Normal file
View file

@ -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);
}