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-rw-r--r--gst/rtsp/gstrtpdec.c119
-rw-r--r--gst/rtsp/gstrtpdec.h7
-rw-r--r--gst/rtsp/gstrtspsrc.c203
-rw-r--r--gst/rtsp/gstrtspsrc.h2
4 files changed, 273 insertions, 58 deletions
diff --git a/gst/rtsp/gstrtpdec.c b/gst/rtsp/gstrtpdec.c
index f91b7276..0adfc7aa 100644
--- a/gst/rtsp/gstrtpdec.c
+++ b/gst/rtsp/gstrtpdec.c
@@ -78,6 +78,13 @@ enum
{
SIGNAL_REQUEST_PT_MAP,
SIGNAL_CLEAR_PT_MAP,
+
+ SIGNAL_ON_NEW_SSRC,
+ SIGNAL_ON_SSRC_COLLISION,
+ SIGNAL_ON_SSRC_VALIDATED,
+ SIGNAL_ON_BYE_SSRC,
+ SIGNAL_ON_BYE_TIMEOUT,
+ SIGNAL_ON_TIMEOUT,
LAST_SIGNAL
};
@@ -255,6 +262,37 @@ gst_rtp_dec_marshal_BOXED__UINT_UINT (GClosure * closure,
g_value_take_boxed (return_value, v_return);
}
+void
+gst_rtp_dec_marshal_VOID__UINT_UINT (GClosure * closure,
+ GValue * return_value,
+ guint n_param_values,
+ const GValue * param_values,
+ gpointer invocation_hint, gpointer marshal_data)
+{
+ typedef void (*GMarshalFunc_VOID__UINT_UINT) (gpointer data1,
+ guint arg_1, guint arg_2, gpointer data2);
+ register GMarshalFunc_VOID__UINT_UINT callback;
+ register GCClosure *cc = (GCClosure *) closure;
+ register gpointer data1, data2;
+
+ g_return_if_fail (n_param_values == 3);
+
+ if (G_CCLOSURE_SWAP_DATA (closure)) {
+ data1 = closure->data;
+ data2 = g_value_peek_pointer (param_values + 0);
+ } else {
+ data1 = g_value_peek_pointer (param_values + 0);
+ data2 = closure->data;
+ }
+ callback =
+ (GMarshalFunc_VOID__UINT_UINT) (marshal_data ? marshal_data : cc->
+ callback);
+
+ callback (data1,
+ g_marshal_value_peek_uint (param_values + 1),
+ g_marshal_value_peek_uint (param_values + 2), data2);
+}
+
static void
gst_rtp_dec_class_init (GstRTPDecClass * g_class)
{
@@ -294,6 +332,87 @@ gst_rtp_dec_class_init (GstRTPDecClass * g_class)
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, clear_pt_map),
NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+ /**
+ * GstRTPDec::on-new-ssrc:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of a new SSRC that entered @session.
+ */
+ gst_rtp_dec_signals[SIGNAL_ON_NEW_SSRC] =
+ g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_new_ssrc),
+ NULL, NULL, gst_rtp_dec_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+ /**
+ * GstRTPDec::on-ssrc_collision:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify when we have an SSRC collision
+ */
+ gst_rtp_dec_signals[SIGNAL_ON_SSRC_COLLISION] =
+ g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_ssrc_collision),
+ NULL, NULL, gst_rtp_dec_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+ /**
+ * GstRTPDec::on-ssrc_validated:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of a new SSRC that became validated.
+ */
+ gst_rtp_dec_signals[SIGNAL_ON_SSRC_VALIDATED] =
+ g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_ssrc_validated),
+ NULL, NULL, gst_rtp_dec_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+
+ /**
+ * GstRTPDec::on-bye-ssrc:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of an SSRC that became inactive because of a BYE packet.
+ */
+ gst_rtp_dec_signals[SIGNAL_ON_BYE_SSRC] =
+ g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_bye_ssrc),
+ NULL, NULL, gst_rtp_dec_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+ /**
+ * GstRTPDec::on-bye-timeout:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of an SSRC that has timed out because of BYE
+ */
+ gst_rtp_dec_signals[SIGNAL_ON_BYE_TIMEOUT] =
+ g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_bye_timeout),
+ NULL, NULL, gst_rtp_dec_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+ /**
+ * GstRTPDec::on-timeout:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of an SSRC that has timed out
+ */
+ gst_rtp_dec_signals[SIGNAL_ON_TIMEOUT] =
+ g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_timeout),
+ NULL, NULL, gst_rtp_dec_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+
+
gstelement_class->provide_clock =
GST_DEBUG_FUNCPTR (gst_rtp_dec_provide_clock);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_dec_change_state);
diff --git a/gst/rtsp/gstrtpdec.h b/gst/rtsp/gstrtpdec.h
index 03260b41..08128f48 100644
--- a/gst/rtsp/gstrtpdec.h
+++ b/gst/rtsp/gstrtpdec.h
@@ -72,6 +72,13 @@ struct _GstRTPDecClass {
GstCaps* (*request_pt_map) (GstRTPDec *rtpdec, guint session, guint pt);
void (*clear_pt_map) (GstRTPDec *rtpdec);
+
+ void (*on_new_ssrc) (GstRTPDec *rtpdec, guint session, guint32 ssrc);
+ void (*on_ssrc_collision) (GstRTPDec *rtpdec, guint session, guint32 ssrc);
+ void (*on_ssrc_validated) (GstRTPDec *rtpdec, guint session, guint32 ssrc);
+ void (*on_bye_ssrc) (GstRTPDec *rtpdec, guint session, guint32 ssrc);
+ void (*on_bye_timeout) (GstRTPDec *rtpdec, guint session, guint32 ssrc);
+ void (*on_timeout) (GstRTPDec *rtpdec, guint session, guint32 ssrc);
};
GType gst_rtp_dec_get_type(void);
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index e02114e5..3bfc37d7 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -133,13 +133,14 @@ enum
LAST_SIGNAL
};
-#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
-#define DEFAULT_RETRY 20
-#define DEFAULT_TIMEOUT 5000000
-#define DEFAULT_TCP_TIMEOUT 20000000
-#define DEFAULT_LATENCY_MS 3000
+#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
+#define DEFAULT_RETRY 20
+#define DEFAULT_TIMEOUT 5000000
+#define DEFAULT_TCP_TIMEOUT 20000000
+#define DEFAULT_LATENCY_MS 3000
+#define DEFAULT_CONNECTION_SPEED 0
enum
{
@@ -151,6 +152,7 @@ enum
PROP_TIMEOUT,
PROP_TCP_TIMEOUT,
PROP_LATENCY,
+ PROP_CONNECTION_SPEED
};
#define GST_TYPE_RTSP_LOWER_TRANS (gst_rtsp_lower_trans_get_type())
@@ -201,6 +203,8 @@ static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
static void gst_rtspsrc_loop (GstRTSPSrc * src);
+static void gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
+ GstRTSPStream * stream, GstEvent * event);
static void gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
/* commands we send to out loop to notify it of events */
@@ -292,6 +296,12 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
"Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
+ g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
+ g_param_spec_uint ("connection-speed", "Connection Speed",
+ "Network connection speed in kbps (0 = unknown)",
+ 0, G_MAXINT / 1000, DEFAULT_CONNECTION_SPEED,
+ G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
+
gstelement_class->change_state = gst_rtspsrc_change_state;
gstbin_class->handle_message = gst_rtspsrc_handle_message;
@@ -376,6 +386,9 @@ gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
case PROP_LATENCY:
rtspsrc->latency = g_value_get_uint (value);
break;
+ case PROP_CONNECTION_SPEED:
+ rtspsrc->connection_speed = g_value_get_uint (value);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@@ -418,6 +431,9 @@ gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
case PROP_LATENCY:
g_value_set_uint (value, rtspsrc->latency);
break;
+ case PROP_CONNECTION_SPEED:
+ g_value_set_uint (value, rtspsrc->connection_speed);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@@ -487,6 +503,18 @@ find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
return -1;
}
+GstRTSPStream *
+find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
+{
+ GList *lstream;
+
+ /* find and get stream */
+ if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
+ return (GstRTSPStream *) lstream->data;
+
+ return NULL;
+}
+
static GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
{
@@ -520,8 +548,7 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
/* If we have a dynamic payload type, see if we have a stream with the
* same payload number. If there is one, they are part of the same
* container and we only need to add one pad. */
- if (g_list_find_custom (src->streams, GINT_TO_POINTER (stream->pt),
- (GCompareFunc) find_stream_by_pt)) {
+ if (find_stream (src, GINT_TO_POINTER (stream->pt), find_stream_by_pt)) {
stream->container = TRUE;
}
}
@@ -532,7 +559,7 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
* the RTP-Info header field returned from PLAY. */
control_url = gst_sdp_media_get_attribute_val (media, "control");
- GST_DEBUG_OBJECT (src, "stream %d", stream->id);
+ GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
GST_DEBUG_OBJECT (src, " container: %d", stream->container);
GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
@@ -1382,14 +1409,10 @@ new_session_pad (GstElement * session, GstPad * pad, GstRTSPSrc * src)
GST_DEBUG_OBJECT (src, "stream: %u, SSRC %d, PT %d", id, ssrc, pt);
- lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (id),
- (GCompareFunc) find_stream_by_id);
- if (lstream == NULL)
+ stream = find_stream (src, GINT_TO_POINTER (id), find_stream_by_id);
+ if (stream == NULL)
goto unknown_stream;
- /* get stream */
- stream = (GstRTSPStream *) lstream->data;
-
/* create a new pad we will use to stream to */
template = gst_static_pad_template_get (&rtptemplate);
stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
@@ -1436,18 +1459,15 @@ static GstCaps *
request_pt_map (GstElement * sess, guint session, guint pt, GstRTSPSrc * src)
{
GstRTSPStream *stream;
- GList *lstream;
GstCaps *caps;
GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
GST_RTSP_STATE_LOCK (src);
- lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (session),
- (GCompareFunc) find_stream_by_id);
- if (!lstream)
+ stream = find_stream (src, GINT_TO_POINTER (session), find_stream_by_id);
+ if (!stream)
goto unknown_stream;
- stream = (GstRTSPStream *) lstream->data;
caps = stream->caps;
GST_RTSP_STATE_UNLOCK (src);
@@ -1461,6 +1481,55 @@ unknown_stream:
}
}
+static void
+gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, guint session)
+{
+ GstRTSPStream *stream;
+
+ GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", session);
+
+ /* get stream for session */
+ stream = find_stream (src, GINT_TO_POINTER (session), find_stream_by_id);
+ if (!stream)
+ goto unknown_stream;
+
+ if (stream->eos)
+ goto was_eos;
+
+ stream->eos = TRUE;
+ gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
+ return;
+
+ /* ERRORS */
+unknown_stream:
+ {
+ GST_DEBUG_OBJECT (src, "unknown stream for session %u", session);
+ return;
+ }
+was_eos:
+ {
+ GST_DEBUG_OBJECT (src, "stream for session %u was EOS already %u", session);
+ return;
+ }
+}
+
+static void
+on_bye_ssrc (GstElement * manager, guint session, guint32 ssrc,
+ GstRTSPSrc * src)
+{
+ GST_DEBUG_OBJECT (src, "SSRC %08x in session %u received BYE", ssrc, session);
+
+ gst_rtspsrc_do_stream_eos (src, session);
+}
+
+static void
+on_timeout (GstElement * manager, guint session, guint32 ssrc, GstRTSPSrc * src)
+{
+ GST_DEBUG_OBJECT (src, "SSRC %08x in session %u timed out", ssrc, session);
+
+ gst_rtspsrc_do_stream_eos (src, session);
+}
+
/* try to get and configure a manager */
static gboolean
gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
@@ -1505,13 +1574,20 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
g_object_set (src->session, "latency", src->latency, NULL);
/* connect to signals if we did not already do so */
- GST_DEBUG_OBJECT (src, "connect to signals on session manager");
+ GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
+ stream);
src->session_sig_id =
g_signal_connect (src->session, "pad-added",
(GCallback) new_session_pad, src);
src->session_ptmap_id =
g_signal_connect (src->session, "request-pt-map",
(GCallback) request_pt_map, src);
+ g_signal_connect (src->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
+ src);
+ g_signal_connect (src->session, "on-bye-timeout", (GCallback) on_timeout,
+ src);
+ g_signal_connect (src->session, "on-timeout", (GCallback) on_timeout,
+ src);
}
/* we stream directly to the manager, get some pads. Each RTSP stream goes
@@ -2056,6 +2132,34 @@ done:
}
static void
+gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
+ GstEvent * event)
+{
+ /* only streams that have a connection to the outside world */
+ if (stream->srcpad == NULL)
+ goto done;
+
+ if (stream->channelpad[0]) {
+ gst_event_ref (event);
+ if (GST_PAD_IS_SRC (stream->channelpad[0]))
+ gst_pad_push_event (stream->channelpad[0], event);
+ else
+ gst_pad_send_event (stream->channelpad[0], event);
+ }
+
+ if (stream->channelpad[1]) {
+ gst_event_ref (event);
+ if (GST_PAD_IS_SRC (stream->channelpad[1]))
+ gst_pad_push_event (stream->channelpad[1], event);
+ else
+ gst_pad_send_event (stream->channelpad[1], event);
+ }
+
+done:
+ gst_event_unref (event);
+}
+
+static void
gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
{
GList *streams;
@@ -2063,25 +2167,8 @@ gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
- /* only streams that have a connection to the outside world */
- if (ostream->srcpad == NULL)
- continue;
-
- if (ostream->channelpad[0]) {
- gst_event_ref (event);
- if (GST_PAD_IS_SRC (ostream->channelpad[0]))
- gst_pad_push_event (ostream->channelpad[0], event);
- else
- gst_pad_send_event (ostream->channelpad[0], event);
- }
-
- if (ostream->channelpad[1]) {
- gst_event_ref (event);
- if (GST_PAD_IS_SRC (ostream->channelpad[1]))
- gst_pad_push_event (ostream->channelpad[1], event);
- else
- gst_pad_send_event (ostream->channelpad[1], event);
- }
+ gst_event_ref (event);
+ gst_rtspsrc_stream_push_event (src, ostream, event);
}
gst_event_unref (event);
}
@@ -2168,7 +2255,6 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
GstRTSPMessage message = { 0 };
GstRTSPResult res;
gint channel;
- GList *lstream;
GstRTSPStream *stream;
GstPad *outpad = NULL;
guint8 *data;
@@ -2210,6 +2296,8 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
/* unset flushing so we can do something else */
gst_rtsp_connection_flush (src->connection, FALSE);
goto interrupt;
+ case GST_RTSP_ETIMEOUT:
+ goto timeout;
default:
goto receive_error;
}
@@ -2238,12 +2326,10 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
channel = message.type_data.data.channel;
- lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (channel),
- (GCompareFunc) find_stream_by_channel);
- if (!lstream)
+ stream = find_stream (src, GINT_TO_POINTER (channel), find_stream_by_channel);
+ if (!stream)
goto unknown_stream;
- stream = (GstRTSPStream *) lstream->data;
if (channel == stream->channel[0]) {
outpad = stream->channelpad[0];
is_rtcp = FALSE;
@@ -2319,6 +2405,13 @@ unknown_stream:
gst_rtsp_message_unset (&message);
return;
}
+timeout:
+ {
+ GST_DEBUG_OBJECT (src, "we got a timeout");
+ gst_rtsp_message_unset (&message);
+ ret = GST_FLOW_UNEXPECTED;
+ goto need_pause;
+ }
interrupt:
{
GST_DEBUG_OBJECT (src, "we got interrupted");
@@ -3745,13 +3838,8 @@ gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
/* remove leading whitespace */
fields[j] = g_strchug (fields[j]);
if (g_str_has_prefix (fields[j], "url=")) {
- GList *lstream;
-
/* get the url and the stream */
- lstream = g_list_find_custom (src->streams, (fields[j] + 4),
- (GCompareFunc) find_stream_by_setup);
- if (lstream)
- stream = (GstRTSPStream *) lstream->data;
+ stream = find_stream (src, (fields[j] + 4), find_stream_by_setup);
} else if (g_str_has_prefix (fields[j], "seq=")) {
seqbase = atoi (fields[j] + 4);
} else if (g_str_has_prefix (fields[j], "rtptime=")) {
@@ -3983,7 +4071,6 @@ gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
case GST_MESSAGE_ERROR:
{
GstObject *udpsrc;
- GList *lstream;
GstRTSPStream *stream;
GstFlowReturn ret;
@@ -3992,13 +4079,10 @@ gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
GST_ELEMENT_NAME (udpsrc));
- lstream = g_list_find_custom (rtspsrc->streams, udpsrc,
- (GCompareFunc) find_stream_by_udpsrc);
- if (!lstream)
+ stream = find_stream (rtspsrc, udpsrc, find_stream_by_udpsrc);
+ if (!stream)
goto forward;
- stream = (GstRTSPStream *) lstream->data;
-
/* we ignore the RTCP udpsrc */
if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
goto done;
@@ -4072,6 +4156,9 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
goto done;
switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+ ret = GST_STATE_CHANGE_SUCCESS;
+ break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
index e0b32e9f..81256a9e 100644
--- a/gst/rtsp/gstrtspsrc.h
+++ b/gst/rtsp/gstrtspsrc.h
@@ -90,6 +90,7 @@ struct _GstRTSPStream {
GstFlowReturn last_ret;
gboolean added;
gboolean disabled;
+ gboolean eos;
/* for interleaved mode */
guint8 channel[2];
@@ -152,6 +153,7 @@ struct _GstRTSPSrc {
guint64 udp_timeout;
GTimeVal tcp_timeout;
guint latency;
+ guint connection_speed;
/* state */
GstRTSPState state;
generated by cgit (git 2.34.1) at 2024-09-27 00:24:00 +0000