*/
/**
- * SECTION:element-gstrtpsession
- * @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux
+ * SECTION:element-rtpsession
+ * @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux
*
- * The RTP session manager models one participant with a unique SSRC in an RTP
+ * The RTP session manager models participants with unique SSRC in an RTP
* session. This session can be used to send and receive RTP and RTCP packets.
* Based on what REQUEST pads are requested from the session manager, specific
* functionality can be activated.
* <listitem>
* <para>Scheduling of RR/SR RTCP packets.</para>
* </listitem>
+ * <listitem>
+ * <para>Support for multiple sender SSRC.</para>
+ * </listitem>
* </itemizedlist>
*
- * The gstrtpsession will not demux packets based on SSRC or payload type, nor will
+ * The rtpsession will not demux packets based on SSRC or payload type, nor will
* it correct for packet reordering and jitter. Use #GstRtpsSrcDemux,
* #GstRtpPtDemux and GstRtpJitterBuffer in addition to #GstRtpSession to
* perform these tasks. It is usually a good idea to use #GstRtpBin, which
* that should be sent to all participants in the session.
*
* To use #GstRtpSession as a sender, request a send_rtp_sink pad, which will
- * automatically create a send_rtp_src pad. The session manager will modify the
- * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
- * send_rtp_src pad after updating its internal state.
+ * automatically create a send_rtp_src pad. The session manager will
+ * forward the packets on the send_rtp_src pad after updating its internal state.
*
* The session manager needs the clock-rate of the payload types it is handling
* and will signal the #GstRtpSession::request-pt-map signal when it needs such a
* <refsect2>
* <title>Example pipelines</title>
* |[
- * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
+ * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
* ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
* decoder and display. Note that the application/x-rtp caps on udpsrc should be
* configured based on some negotiation process such as RTSP for this pipeline
* to work correctly.
* |[
- * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession name=session \
+ * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession name=session \
* .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \
* udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink
* ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
* configured based on some negotiation process such as RTSP for this pipeline
* to work correctly.
* |[
- * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession .send_rtp_src ! udpsink port=5000
+ * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession .send_rtp_src ! udpsink port=5000
* ]| Send theora RTP packets through the session manager and out on UDP port
* 5000.
* |[
- * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession name=session .send_rtp_src \
+ * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession name=session .send_rtp_src \
* ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001
* ]| Send theora RTP packets through the session manager and out on UDP port
* 5000. Send RTCP packets on port 5001. Note that this pipeline will not preroll
* packets are sent in the PAUSED state). Applications should manually set and
* keep (see gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state.
* </refsect2>
- *
- * Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
#include <gst/glib-compat-private.h>
-#include "gstrtpbin-marshal.h"
#include "gstrtpsession.h"
#include "rtpsession.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug);
#define GST_CAT_DEFAULT gst_rtp_session_debug
+GType
+gst_rtp_ntp_time_source_get_type (void)
+{
+ static GType type = 0;
+ static const GEnumValue values[] = {
+ {GST_RTP_NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"},
+ {GST_RTP_NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"},
+ {GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME,
+ "Running time based on pipeline clock",
+ "running-time"},
+ {GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"},
+ {0, NULL, NULL},
+ };
+
+ if (!type) {
+ type = g_enum_register_static ("GstRtpNtpTimeSource", values);
+ }
+ return type;
+}
+
/* sink pads */
static GstStaticPadTemplate rtpsession_recv_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink",
SIGNAL_ON_BYE_TIMEOUT,
SIGNAL_ON_TIMEOUT,
SIGNAL_ON_SENDER_TIMEOUT,
+ SIGNAL_ON_NEW_SENDER_SSRC,
+ SIGNAL_ON_SENDER_SSRC_ACTIVE,
LAST_SIGNAL
};
-#define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
-#define DEFAULT_RTCP_FRACTION (RTP_STATS_BANDWIDTH * RTP_STATS_RTCP_FRACTION)
+#define DEFAULT_BANDWIDTH 0
+#define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_FRACTION
#define DEFAULT_RTCP_RR_BANDWIDTH -1
#define DEFAULT_RTCP_RS_BANDWIDTH -1
#define DEFAULT_SDES NULL
#define DEFAULT_USE_PIPELINE_CLOCK FALSE
#define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
#define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
+#define DEFAULT_MAX_DROPOUT_TIME 60000
+#define DEFAULT_MAX_MISORDER_TIME 2000
+#define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP
+#define DEFAULT_NTP_TIME_SOURCE GST_RTP_NTP_TIME_SOURCE_NTP
+#define DEFAULT_RTCP_SYNC_SEND_TIME TRUE
enum
{
PROP_USE_PIPELINE_CLOCK,
PROP_RTCP_MIN_INTERVAL,
PROP_PROBATION,
- PROP_LAST
+ PROP_MAX_DROPOUT_TIME,
+ PROP_MAX_MISORDER_TIME,
+ PROP_STATS,
+ PROP_RTP_PROFILE,
+ PROP_NTP_TIME_SOURCE,
+ PROP_RTCP_SYNC_SEND_TIME
};
-#define GST_RTP_SESSION_GET_PRIVATE(obj) \
- (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_SESSION, GstRtpSessionPrivate))
-
#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->priv->lock)
#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->priv->lock)
GstClockTime send_latency;
gboolean use_pipeline_clock;
+ GstRtpNtpTimeSource ntp_time_source;
+ gboolean rtcp_sync_send_time;
+
+ guint recv_rtx_req_count;
+ guint sent_rtx_req_count;
};
/* callbacks to handle actions from the session manager */
static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
gpointer user_data);
static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data);
-static void gst_rtp_session_request_key_unit (RTPSession * sess,
+static void gst_rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
gboolean all_headers, gpointer user_data);
static GstClockTime gst_rtp_session_request_time (RTPSession * session,
gpointer user_data);
+static void gst_rtp_session_notify_nack (RTPSession * sess,
+ guint16 seqnum, guint16 blp, guint32 ssrc, gpointer user_data);
+static void gst_rtp_session_reconfigure (RTPSession * sess, gpointer user_data);
+static void gst_rtp_session_notify_early_rtcp (RTPSession * sess,
+ gpointer user_data);
+static GstFlowReturn gst_rtp_session_chain_recv_rtp (GstPad * pad,
+ GstObject * parent, GstBuffer * buffer);
+static GstFlowReturn gst_rtp_session_chain_recv_rtcp (GstPad * pad,
+ GstObject * parent, GstBuffer * buffer);
+static GstFlowReturn gst_rtp_session_chain_send_rtp (GstPad * pad,
+ GstObject * parent, GstBuffer * buffer);
+static GstFlowReturn gst_rtp_session_chain_send_rtp_list (GstPad * pad,
+ GstObject * parent, GstBufferList * list);
static RTPSessionCallbacks callbacks = {
gst_rtp_session_process_rtp,
gst_rtp_session_clock_rate,
gst_rtp_session_reconsider,
gst_rtp_session_request_key_unit,
- gst_rtp_session_request_time
+ gst_rtp_session_request_time,
+ gst_rtp_session_notify_nack,
+ gst_rtp_session_reconfigure,
+ gst_rtp_session_notify_early_rtcp
};
/* GObject vmethods */
static void gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession);
+static GstStructure *gst_rtp_session_create_stats (GstRtpSession * rtpsession);
+
static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 };
static void
static void
on_ssrc_collision (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
+ GstPad *send_rtp_sink;
+
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
src->ssrc);
+
+ GST_RTP_SESSION_LOCK (sess);
+ if ((send_rtp_sink = sess->send_rtp_sink))
+ gst_object_ref (send_rtp_sink);
+ GST_RTP_SESSION_UNLOCK (sess);
+
+ if (send_rtp_sink) {
+ GstStructure *structure;
+ GstEvent *event;
+ RTPSource *internal_src;
+ guint32 suggested_ssrc;
+
+ structure = gst_structure_new ("GstRTPCollision", "ssrc", G_TYPE_UINT,
+ (guint) src->ssrc, NULL);
+
+ /* if there is no source using the suggested ssrc, most probably because
+ * this ssrc has just collided, suggest upstream to use it */
+ suggested_ssrc = rtp_session_suggest_ssrc (session, NULL);
+ internal_src = rtp_session_get_source_by_ssrc (session, suggested_ssrc);
+ if (!internal_src)
+ gst_structure_set (structure, "suggested-ssrc", G_TYPE_UINT,
+ (guint) suggested_ssrc, NULL);
+ else
+ g_object_unref (internal_src);
+
+ event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
+ gst_pad_push_event (send_rtp_sink, event);
+ gst_object_unref (send_rtp_sink);
+ }
}
static void
src->ssrc);
}
+static void
+on_new_sender_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
+{
+ g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
+ src->ssrc);
+}
+
+static void
+on_sender_ssrc_active (RTPSession * session, RTPSource * src,
+ GstRtpSession * sess)
+{
+ g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE], 0,
+ src->ssrc);
+}
+
+static void
+on_notify_stats (RTPSession * session, GParamSpec * spec,
+ GstRtpSession * rtpsession)
+{
+ g_object_notify (G_OBJECT (rtpsession), "stats");
+}
+
#define gst_rtp_session_parent_class parent_class
-G_DEFINE_TYPE (GstRtpSession, gst_rtp_session, GST_TYPE_ELEMENT);
+G_DEFINE_TYPE_WITH_PRIVATE (GstRtpSession, gst_rtp_session, GST_TYPE_ELEMENT);
static void
gst_rtp_session_class_init (GstRtpSessionClass * klass)
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
- g_type_class_add_private (klass, sizeof (GstRtpSessionPrivate));
-
gobject_class->finalize = gst_rtp_session_finalize;
gobject_class->set_property = gst_rtp_session_set_property;
gobject_class->get_property = gst_rtp_session_get_property;
gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP] =
g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, request_pt_map),
- NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT, GST_TYPE_CAPS, 1,
- G_TYPE_UINT);
+ NULL, NULL, g_cclosure_marshal_generic, GST_TYPE_CAPS, 1, G_TYPE_UINT);
/**
* GstRtpSession::clear-pt-map:
* @sess: the object which received the signal
*/
gst_rtp_session_signals[SIGNAL_CLEAR_PT_MAP] =
g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
- G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpSessionClass, clear_pt_map),
- NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+ G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
+ G_STRUCT_OFFSET (GstRtpSessionClass, clear_pt_map),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstRtpSession::on-new-ssrc:
on_ssrc_validated), NULL, NULL, g_cclosure_marshal_VOID__UINT,
G_TYPE_NONE, 1, G_TYPE_UINT);
/**
- * GstRtpSession::on-ssrc_active:
+ * GstRtpSession::on-ssrc-active:
* @sess: the object which received the signal
* @ssrc: the SSRC
*
on_sender_timeout), NULL, NULL, g_cclosure_marshal_VOID__UINT,
G_TYPE_NONE, 1, G_TYPE_UINT);
+ /**
+ * GstRtpSession::on-new-sender-ssrc:
+ * @sess: the object which received the signal
+ * @ssrc: the sender SSRC
+ *
+ * Notify of a new sender SSRC that entered @session.
+ *
+ * Since: 1.8
+ */
+ gst_rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC] =
+ g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_new_ssrc),
+ NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpSession::on-sender-ssrc-active:
+ * @sess: the object which received the signal
+ * @ssrc: the sender SSRC
+ *
+ * Notify of a sender SSRC that is active, i.e., sending RTCP.
+ *
+ * Since: 1.8
+ */
+ gst_rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] =
+ g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
+ on_ssrc_active), NULL, NULL, g_cclosure_marshal_VOID__UINT,
+ G_TYPE_NONE, 1, G_TYPE_UINT);
+
g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
g_param_spec_double ("bandwidth", "Bandwidth",
"The bandwidth of the session in bytes per second (0 for auto-discover)",
g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
- "Use the pipeline running-time to set the NTP time in the RTCP SR messages",
+ "Use the pipeline running-time to set the NTP time in the RTCP SR messages "
+ "(DEPRECATED: Use ntp-time-source property)",
DEFAULT_USE_PIPELINE_CLOCK,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
0, G_MAXUINT, DEFAULT_PROBATION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
+ g_param_spec_uint ("max-dropout-time", "Max dropout time",
+ "The maximum time (milliseconds) of missing packets tolerated.",
+ 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
+ g_param_spec_uint ("max-misorder-time", "Max misorder time",
+ "The maximum time (milliseconds) of misordered packets tolerated.",
+ 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpSession::stats:
+ *
+ * Various session statistics. This property returns a GstStructure
+ * with name application/x-rtp-session-stats with the following fields:
+ *
+ * "recv-rtx-req-count G_TYPE_UINT The number of retransmission event
+ * received from downstream (in receiver mode) (Since 1.16)
+ * "sent-rtx-req-count" G_TYPE_UINT The number of retransmission event
+ * sent downstream (in sender mode) (Since 1.16)
+ * "rtx-count" G_TYPE_UINT DEPRECATED Since 1.16, same as
+ * "recv-rtx-req-count".
+ * "rtx-drop-count" G_TYPE_UINT The number of retransmission events
+ * dropped (due to bandwidth constraints)
+ * "sent-nack-count" G_TYPE_UINT Number of NACKs sent
+ * "recv-nack-count" G_TYPE_UINT Number of NACKs received
+ * "source-stats" G_TYPE_BOXED GValueArray of #RTPSource::stats for all
+ * RTP sources (Since 1.8)
+ *
+ * Since: 1.4
+ */
+ g_object_class_install_property (gobject_class, PROP_STATS,
+ g_param_spec_boxed ("stats", "Statistics",
+ "Various statistics", GST_TYPE_STRUCTURE,
+ G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
+ g_param_spec_enum ("rtp-profile", "RTP Profile",
+ "RTP profile to use", GST_TYPE_RTP_PROFILE, DEFAULT_RTP_PROFILE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
+ g_param_spec_enum ("ntp-time-source", "NTP Time Source",
+ "NTP time source for RTCP packets",
+ gst_rtp_ntp_time_source_get_type (), DEFAULT_NTP_TIME_SOURCE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_SEND_TIME,
+ g_param_spec_boolean ("rtcp-sync-send-time", "RTCP Sync Send Time",
+ "Use send time or capture time for RTCP sync "
+ "(TRUE = send time, FALSE = capture time)",
+ DEFAULT_RTCP_SYNC_SEND_TIME,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_session_change_state);
gstelement_class->request_new_pad =
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_session_clear_pt_map);
/* sink pads */
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&rtpsession_recv_rtp_sink_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&rtpsession_recv_rtcp_sink_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&rtpsession_send_rtp_sink_template));
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &rtpsession_recv_rtp_sink_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &rtpsession_recv_rtcp_sink_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &rtpsession_send_rtp_sink_template);
/* src pads */
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&rtpsession_recv_rtp_src_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&rtpsession_sync_src_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&rtpsession_send_rtp_src_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&rtpsession_send_rtcp_src_template));
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &rtpsession_recv_rtp_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &rtpsession_sync_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &rtpsession_send_rtp_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &rtpsession_send_rtcp_src_template);
gst_element_class_set_static_metadata (gstelement_class, "RTP Session",
"Filter/Network/RTP",
GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug,
"rtpsession", 0, "RTP Session");
+
+ GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_recv_rtp);
+ GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_recv_rtcp);
+ GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_send_rtp);
+ GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_send_rtp_list);
+
}
static void
gst_rtp_session_init (GstRtpSession * rtpsession)
{
- rtpsession->priv = GST_RTP_SESSION_GET_PRIVATE (rtpsession);
+ rtpsession->priv = gst_rtp_session_get_instance_private (rtpsession);
g_mutex_init (&rtpsession->priv->lock);
g_cond_init (&rtpsession->priv->cond);
rtpsession->priv->sysclock = gst_system_clock_obtain ();
rtpsession->priv->session = rtp_session_new ();
rtpsession->priv->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
+ rtpsession->priv->rtcp_sync_send_time = DEFAULT_RTCP_SYNC_SEND_TIME;
/* configure callbacks */
rtp_session_set_callbacks (rtpsession->priv->session, &callbacks, rtpsession);
(GCallback) on_timeout, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-sender-timeout",
(GCallback) on_sender_timeout, rtpsession);
+ g_signal_connect (rtpsession->priv->session, "on-new-sender-ssrc",
+ (GCallback) on_new_sender_ssrc, rtpsession);
+ g_signal_connect (rtpsession->priv->session, "on-sender-ssrc-active",
+ (GCallback) on_sender_ssrc_active, rtpsession);
+ g_signal_connect (rtpsession->priv->session, "notify::stats",
+ (GCallback) on_notify_stats, rtpsession);
rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
(GDestroyNotify) gst_caps_unref);
+ rtpsession->recv_rtcp_segment_seqnum = GST_SEQNUM_INVALID;
+
gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
rtpsession->priv->thread_stopped = TRUE;
+
+ rtpsession->priv->recv_rtx_req_count = 0;
+ rtpsession->priv->sent_rtx_req_count = 0;
+
+ rtpsession->priv->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
}
static void
case PROP_PROBATION:
g_object_set_property (G_OBJECT (priv->session), "probation", value);
break;
+ case PROP_MAX_DROPOUT_TIME:
+ g_object_set_property (G_OBJECT (priv->session), "max-dropout-time",
+ value);
+ break;
+ case PROP_MAX_MISORDER_TIME:
+ g_object_set_property (G_OBJECT (priv->session), "max-misorder-time",
+ value);
+ break;
+ case PROP_RTP_PROFILE:
+ g_object_set_property (G_OBJECT (priv->session), "rtp-profile", value);
+ break;
+ case PROP_NTP_TIME_SOURCE:
+ priv->ntp_time_source = g_value_get_enum (value);
+ break;
+ case PROP_RTCP_SYNC_SEND_TIME:
+ priv->rtcp_sync_send_time = g_value_get_boolean (value);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
case PROP_PROBATION:
g_object_get_property (G_OBJECT (priv->session), "probation", value);
break;
+ case PROP_MAX_DROPOUT_TIME:
+ g_object_get_property (G_OBJECT (priv->session), "max-dropout-time",
+ value);
+ break;
+ case PROP_MAX_MISORDER_TIME:
+ g_object_get_property (G_OBJECT (priv->session), "max-misorder-time",
+ value);
+ break;
+ case PROP_STATS:
+ g_value_take_boxed (value, gst_rtp_session_create_stats (rtpsession));
+ break;
+ case PROP_RTP_PROFILE:
+ g_object_get_property (G_OBJECT (priv->session), "rtp-profile", value);
+ break;
+ case PROP_NTP_TIME_SOURCE:
+ g_value_set_enum (value, priv->ntp_time_source);
+ break;
+ case PROP_RTCP_SYNC_SEND_TIME:
+ g_value_set_boolean (value, priv->rtcp_sync_send_time);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
+static GstStructure *
+gst_rtp_session_create_stats (GstRtpSession * rtpsession)
+{
+ GstStructure *s;
+
+ g_object_get (rtpsession->priv->session, "stats", &s, NULL);
+ gst_structure_set (s, "rtx-count", G_TYPE_UINT,
+ rtpsession->priv->recv_rtx_req_count, "recv-rtx-req-count", G_TYPE_UINT,
+ rtpsession->priv->recv_rtx_req_count, "sent-rtx-req-count", G_TYPE_UINT,
+ rtpsession->priv->sent_rtx_req_count, NULL);
+
+ return s;
+}
+
static void
get_current_times (GstRtpSession * rtpsession, GstClockTime * running_time,
guint64 * ntpnstime)
{
- guint64 ntpns;
+ guint64 ntpns = -1;
GstClock *clock;
GstClockTime base_time, rt, clock_time;
if (rtpsession->priv->use_pipeline_clock) {
ntpns = rt;
+ /* add constant to convert from 1970 based time to 1900 based time */
+ ntpns += (2208988800LL * GST_SECOND);
} else {
- GTimeVal current;
-
- /* get current NTP time */
- g_get_current_time (¤t);
- ntpns = GST_TIMEVAL_TO_TIME (current);
+ switch (rtpsession->priv->ntp_time_source) {
+ case GST_RTP_NTP_TIME_SOURCE_NTP:
+ case GST_RTP_NTP_TIME_SOURCE_UNIX:{
+ GTimeVal current;
+
+ /* get current NTP time */
+ g_get_current_time (¤t);
+ ntpns = GST_TIMEVAL_TO_TIME (current);
+
+ /* add constant to convert from 1970 based time to 1900 based time */
+ if (rtpsession->priv->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP)
+ ntpns += (2208988800LL * GST_SECOND);
+ break;
+ }
+ case GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME:
+ ntpns = rt;
+ break;
+ case GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME:
+ ntpns = clock_time;
+ break;
+ default:
+ ntpns = -1;
+ g_assert_not_reached ();
+ break;
+ }
}
- /* add constant to convert from 1970 based time to 1900 based time */
- ntpns += (2208988800LL * GST_SECOND);
-
gst_object_unref (clock);
} else {
GST_OBJECT_UNLOCK (rtpsession);
*ntpnstime = ntpns;
}
+/* must be called with GST_RTP_SESSION_LOCK */
+static void
+signal_waiting_rtcp_thread_unlocked (GstRtpSession * rtpsession)
+{
+ if (rtpsession->priv->wait_send) {
+ GST_LOG_OBJECT (rtpsession, "signal RTCP thread");
+ rtpsession->priv->wait_send = FALSE;
+ GST_RTP_SESSION_SIGNAL (rtpsession);
+ }
+}
+
static void
rtcp_thread (GstRtpSession * rtpsession)
{
GST_RTP_SESSION_LOCK (rtpsession);
while (rtpsession->priv->wait_send) {
- GST_LOG_OBJECT (rtpsession, "waiting for RTP thread");
+ GST_LOG_OBJECT (rtpsession, "waiting for getting started");
GST_RTP_SESSION_WAIT (rtpsession);
GST_LOG_OBJECT (rtpsession, "signaled...");
}
GST_RTP_SESSION_LOCK (rtpsession);
rtpsession->priv->stop_thread = TRUE;
- rtpsession->priv->wait_send = FALSE;
- GST_RTP_SESSION_SIGNAL (rtpsession);
+ signal_waiting_rtcp_thread_unlocked (rtpsession);
if (rtpsession->priv->id)
gst_clock_id_unschedule (rtpsession->priv->id);
GST_RTP_SESSION_UNLOCK (rtpsession);
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_RTP_SESSION_LOCK (rtpsession);
- if (rtpsession->send_rtp_src)
- rtpsession->priv->wait_send = TRUE;
+ rtpsession->priv->wait_send = TRUE;
GST_RTP_SESSION_UNLOCK (rtpsession);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* downstream is now releasing the dataflow and we can join. */
join_rtcp_thread (rtpsession);
+ rtp_session_reset (rtpsession->priv->session);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
static void
gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession)
{
+ GST_RTP_SESSION_LOCK (rtpsession);
g_hash_table_foreach_remove (rtpsession->priv->ptmap, return_true, NULL);
+ GST_RTP_SESSION_UNLOCK (rtpsession);
}
/* called when the session manager has an RTP packet or a list of packets
GST_RTP_SESSION_LOCK (rtpsession);
if ((rtp_src = rtpsession->send_rtp_src))
gst_object_ref (rtp_src);
- if (rtpsession->priv->wait_send) {
- GST_LOG_OBJECT (rtpsession, "signal RTCP thread");
- rtpsession->priv->wait_send = FALSE;
- GST_RTP_SESSION_SIGNAL (rtpsession);
- }
+ signal_waiting_rtcp_thread_unlocked (rtpsession);
GST_RTP_SESSION_UNLOCK (rtpsession);
if (rtp_src) {
GST_RTP_SESSION_UNLOCK (rtpsession);
event = gst_event_new_stream_start (stream_id);
+ rtpsession->recv_rtcp_segment_seqnum = gst_event_get_seqnum (event);
+ gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
if (have_group_id)
gst_event_set_group_id (event, group_id);
gst_pad_push_event (srcpad, event);
gst_segment_init (&seg, GST_FORMAT_TIME);
event = gst_event_new_segment (&seg);
+ gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
gst_pad_push_event (srcpad, event);
}
* well. */
static GstFlowReturn
gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
- GstBuffer * buffer, gboolean eos, gpointer user_data)
+ GstBuffer * buffer, gboolean all_sources_bye, gpointer user_data)
{
GstFlowReturn result;
GstRtpSession *rtpsession;
GST_LOG_OBJECT (rtpsession, "sending RTCP");
result = gst_pad_push (rtcp_src, buffer);
- /* we have to send EOS after this packet */
- if (eos) {
+ /* Forward send an EOS on the RTCP sink if we received an EOS on the
+ * send_rtp_sink. We don't need to check the recv_rtp_sink since in this
+ * case the EOS event would already have been sent */
+ if (all_sources_bye && rtpsession->send_rtp_sink &&
+ GST_PAD_IS_EOS (rtpsession->send_rtp_sink)) {
+ GstEvent *event;
+
GST_LOG_OBJECT (rtpsession, "sending EOS");
- gst_pad_push_event (rtcp_src, gst_event_new_eos ());
+
+ event = gst_event_new_eos ();
+ gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
+ gst_pad_push_event (rtcp_src, event);
}
gst_object_unref (rtcp_src);
} else {
gst_object_ref (sync_src);
GST_RTP_SESSION_UNLOCK (rtpsession);
+ /* set rtcp caps on output pad, this happens
+ * when we receive RTCP muxed with RTP according
+ * to RFC5761. Otherwise we would have forwarded
+ * the events from the recv_rtcp_sink pad already
+ */
+ if (!gst_pad_has_current_caps (sync_src))
+ do_rtcp_events (rtpsession, sync_src);
+
GST_LOG_OBJECT (rtpsession, "sending Sync RTCP");
result = gst_pad_push (sync_src, buffer);
gst_object_unref (sync_src);
}
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
+ rtpsession->recv_rtcp_segment_seqnum = GST_SEQNUM_INVALID;
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
break;
case GST_EVENT_SEGMENT:
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
break;
}
+ case GST_EVENT_EOS:
+ {
+ GstPad *rtcp_src;
+
+ ret =
+ gst_pad_push_event (rtpsession->recv_rtp_src, gst_event_ref (event));
+
+ GST_RTP_SESSION_LOCK (rtpsession);
+ if ((rtcp_src = rtpsession->send_rtcp_src))
+ gst_object_ref (rtcp_src);
+ GST_RTP_SESSION_UNLOCK (rtpsession);
+
+ gst_event_unref (event);
+
+ if (rtcp_src) {
+ event = gst_event_new_eos ();
+ if (rtpsession->recv_rtcp_segment_seqnum != GST_SEQNUM_INVALID)
+ gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
+ ret = gst_pad_push_event (rtcp_src, event);
+ gst_object_unref (rtcp_src);
+ } else {
+ ret = TRUE;
+ }
+ break;
+ }
default:
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
break;
if (pli || fir)
return rtp_session_request_key_unit (rtpsession->priv->session, ssrc,
- gst_clock_get_time (rtpsession->priv->sysclock), fir, count);
+ fir, count);
}
return FALSE;
if (gst_rtp_session_request_remote_key_unit (rtpsession, ssrc, pt,
all_headers, count))
forward = FALSE;
+ } else if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) {
+ guint seqnum, delay, deadline, max_delay, avg_rtt;
+
+ GST_RTP_SESSION_LOCK (rtpsession);
+ rtpsession->priv->recv_rtx_req_count++;
+ GST_RTP_SESSION_UNLOCK (rtpsession);
+
+ if (!gst_structure_get_uint (s, "ssrc", &ssrc))
+ ssrc = -1;
+ if (!gst_structure_get_uint (s, "seqnum", &seqnum))
+ seqnum = -1;
+ if (!gst_structure_get_uint (s, "delay", &delay))
+ delay = 0;
+ if (!gst_structure_get_uint (s, "deadline", &deadline))
+ deadline = 100;
+ if (!gst_structure_get_uint (s, "avg-rtt", &avg_rtt))
+ avg_rtt = 40;
+
+ /* remaining time to receive the packet */
+ max_delay = deadline;
+ if (max_delay > delay)
+ max_delay -= delay;
+ /* estimated RTT */
+ if (max_delay > avg_rtt)
+ max_delay -= avg_rtt;
+ else
+ max_delay = 0;
+
+ if (rtp_session_request_nack (rtpsession->priv->session, ssrc, seqnum,
+ max_delay * GST_MSECOND))
+ forward = FALSE;
}
break;
default:
it = gst_iterator_new_single (GST_TYPE_PAD, &val);
g_value_unset (&val);
gst_object_unref (otherpad);
+ } else {
+ it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
}
return it;
GstFlowReturn ret;
GstClockTime current_time, running_time;
GstClockTime timestamp;
+ guint64 ntpnstime;
rtpsession = GST_RTP_SESSION (parent);
priv = rtpsession->priv;
GST_LOG_OBJECT (rtpsession, "received RTP packet");
+ GST_RTP_SESSION_LOCK (rtpsession);
+ signal_waiting_rtcp_thread_unlocked (rtpsession);
+ GST_RTP_SESSION_UNLOCK (rtpsession);
+
/* get NTP time when this packet was captured, this depends on the timestamp. */
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ timestamp = GST_BUFFER_PTS (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* convert to running time using the segment values */
running_time =
gst_segment_to_running_time (&rtpsession->recv_rtp_seg, GST_FORMAT_TIME,
timestamp);
+ ntpnstime = GST_CLOCK_TIME_NONE;
} else {
- get_current_times (rtpsession, &running_time, NULL);
+ get_current_times (rtpsession, &running_time, &ntpnstime);
}
current_time = gst_clock_get_time (priv->sysclock);
ret = rtp_session_process_rtp (priv->session, buffer, current_time,
- running_time);
+ running_time, ntpnstime);
if (ret != GST_FLOW_OK)
goto push_error;
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEGMENT:
+ /* Make sure that the sync_src pad has caps before the segment event.
+ * Otherwise we might get a segment event before caps from the receive
+ * RTCP pad, and then later when receiving RTCP packets will set caps.
+ * This will results in a sticky event misordering warning
+ */
+ if (!gst_pad_has_current_caps (rtpsession->sync_src)) {
+ GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtcp");
+ gst_pad_set_caps (rtpsession->sync_src, caps);
+ gst_caps_unref (caps);
+ }
+ /* fall through */
default:
ret = gst_pad_push_event (rtpsession->sync_src, event);
break;
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GstClockTime current_time;
+ GstClockTime running_time;
guint64 ntpnstime;
rtpsession = GST_RTP_SESSION (parent);
GST_LOG_OBJECT (rtpsession, "received RTCP packet");
+ GST_RTP_SESSION_LOCK (rtpsession);
+ signal_waiting_rtcp_thread_unlocked (rtpsession);
+ GST_RTP_SESSION_UNLOCK (rtpsession);
+
current_time = gst_clock_get_time (priv->sysclock);
- get_current_times (rtpsession, NULL, &ntpnstime);
+ get_current_times (rtpsession, &running_time, &ntpnstime);
- rtp_session_process_rtcp (priv->session, buffer, current_time, ntpnstime);
+ rtp_session_process_rtcp (priv->session, buffer, current_time, running_time,
+ ntpnstime);
return GST_FLOW_OK; /* always return OK */
}
* because we stop sending. */
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
current_time = gst_clock_get_time (rtpsession->priv->sysclock);
+
GST_DEBUG_OBJECT (rtpsession, "scheduling BYE message");
rtp_session_mark_all_bye (rtpsession->priv->session, "End Of Stream");
rtp_session_schedule_bye (rtpsession->priv->session, current_time);
GstCaps *result;
GstStructure *s1, *s2;
guint ssrc;
+ gboolean is_random;
priv = rtpsession->priv;
- ssrc = rtp_session_get_internal_ssrc (priv->session);
+ ssrc = rtp_session_suggest_ssrc (priv->session, &is_random);
/* we can basically accept anything but we prefer to receive packets with our
* internal SSRC so that we don't have to patch it. Create a structure with
- * the SSRC and another one without. */
- s1 = gst_structure_new ("application/x-rtp", "ssrc", G_TYPE_UINT, ssrc, NULL);
- s2 = gst_structure_new_empty ("application/x-rtp");
-
- result = gst_caps_new_full (s1, s2, NULL);
+ * the SSRC and another one without.
+ * Only do this if the session actually decided on an ssrc already,
+ * otherwise we give upstream the opportunity to select an ssrc itself */
+ if (!is_random) {
+ s1 = gst_structure_new ("application/x-rtp", "ssrc", G_TYPE_UINT, ssrc,
+ NULL);
+ s2 = gst_structure_new_empty ("application/x-rtp");
+
+ result = gst_caps_new_full (s1, s2, NULL);
+ } else {
+ result = gst_caps_new_empty_simple ("application/x-rtp");
+ }
if (filter) {
GstCaps *caps = result;
GstCaps * caps)
{
GstRtpSessionPrivate *priv;
- GstStructure *s = gst_caps_get_structure (caps, 0);
- guint ssrc;
priv = rtpsession->priv;
- if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
- GST_DEBUG_OBJECT (rtpsession, "setting internal SSRC to %08x", ssrc);
- rtp_session_set_internal_ssrc (priv->session, ssrc);
- }
rtp_session_update_send_caps (priv->session, caps);
return TRUE;
}
-/* Recieve an RTP packet or a list of packets to be send to the receivers,
+/* Receive an RTP packet or a list of packets to be sent to the receivers,
* send to RTP session manager and forward to send_rtp_src.
*/
static GstFlowReturn
if (is_list) {
GstBuffer *buffer = NULL;
- /* All groups in an list have the same timestamp.
- * So, just take it from the first group. */
+ /* All buffers in a list have the same timestamp.
+ * So, just take it from the first buffer. */
buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0);
if (buffer)
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ timestamp = GST_BUFFER_PTS (buffer);
else
timestamp = -1;
} else {
- timestamp = GST_BUFFER_TIMESTAMP (GST_BUFFER_CAST (data));
+ timestamp = GST_BUFFER_PTS (GST_BUFFER_CAST (data));
}
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
running_time =
gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME,
timestamp);
- running_time += priv->send_latency;
+ if (priv->rtcp_sync_send_time)
+ running_time += priv->send_latency;
} else {
/* no timestamp. */
running_time = -1;
gst_rtp_session_event_recv_rtp_sink);
gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_sink,
gst_rtp_session_iterate_internal_links);
+ GST_PAD_SET_PROXY_ALLOCATION (rtpsession->recv_rtp_sink);
gst_pad_set_active (rtpsession->recv_rtp_sink, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtp_sink);
gst_rtp_session_event_send_rtp_sink);
gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_sink,
gst_rtp_session_iterate_internal_links);
+ GST_PAD_SET_PROXY_CAPS (rtpsession->send_rtp_sink);
+ GST_PAD_SET_PROXY_ALLOCATION (rtpsession->send_rtp_sink);
gst_pad_set_active (rtpsession->send_rtp_sink, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->send_rtp_sink);
gst_rtp_session_iterate_internal_links);
gst_pad_set_event_function (rtpsession->send_rtp_src,
gst_rtp_session_event_send_rtp_src);
+ GST_PAD_SET_PROXY_CAPS (rtpsession->send_rtp_src);
gst_pad_set_active (rtpsession->send_rtp_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);
wrong_template:
{
GST_RTP_SESSION_UNLOCK (rtpsession);
- g_warning ("gstrtpsession: this is not our template");
+ g_warning ("rtpsession: this is not our template");
return NULL;
}
exists:
{
GST_RTP_SESSION_UNLOCK (rtpsession);
- g_warning ("gstrtpsession: pad already requested");
+ g_warning ("rtpsession: pad already requested");
return NULL;
}
}
wrong_pad:
{
GST_RTP_SESSION_UNLOCK (rtpsession);
- g_warning ("gstrtpsession: asked to release an unknown pad");
+ g_warning ("rtpsession: asked to release an unknown pad");
return;
}
}
static void
gst_rtp_session_request_key_unit (RTPSession * sess,
- gboolean all_headers, gpointer user_data)
+ guint32 ssrc, gboolean all_headers, gpointer user_data)
{
GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
GstEvent *event;
if (send_rtp_sink) {
event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
- gst_structure_new ("GstForceKeyUnit",
+ gst_structure_new ("GstForceKeyUnit", "ssrc", G_TYPE_UINT, ssrc,
"all-headers", G_TYPE_BOOLEAN, all_headers, NULL));
gst_pad_push_event (send_rtp_sink, event);
gst_object_unref (send_rtp_sink);
return gst_clock_get_time (rtpsession->priv->sysclock);
}
+
+static void
+gst_rtp_session_notify_nack (RTPSession * sess, guint16 seqnum,
+ guint16 blp, guint32 ssrc, gpointer user_data)
+{
+ GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
+ GstEvent *event;
+ GstPad *send_rtp_sink;
+
+ GST_RTP_SESSION_LOCK (rtpsession);
+ if ((send_rtp_sink = rtpsession->send_rtp_sink))
+ gst_object_ref (send_rtp_sink);
+ GST_RTP_SESSION_UNLOCK (rtpsession);
+
+ if (send_rtp_sink) {
+ while (TRUE) {
+ event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
+ gst_structure_new ("GstRTPRetransmissionRequest",
+ "seqnum", G_TYPE_UINT, (guint) seqnum,
+ "ssrc", G_TYPE_UINT, (guint) ssrc, NULL));
+ gst_pad_push_event (send_rtp_sink, event);
+
+ GST_RTP_SESSION_LOCK (rtpsession);
+ rtpsession->priv->sent_rtx_req_count++;
+ GST_RTP_SESSION_UNLOCK (rtpsession);
+
+ if (blp == 0)
+ break;
+
+ seqnum++;
+ while ((blp & 1) == 0) {
+ seqnum++;
+ blp >>= 1;
+ }
+ blp >>= 1;
+ }
+ gst_object_unref (send_rtp_sink);
+ }
+}
+
+static void
+gst_rtp_session_reconfigure (RTPSession * sess, gpointer user_data)
+{
+ GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
+ GstPad *send_rtp_sink;
+
+ GST_RTP_SESSION_LOCK (rtpsession);
+ if ((send_rtp_sink = rtpsession->send_rtp_sink))
+ gst_object_ref (send_rtp_sink);
+ GST_RTP_SESSION_UNLOCK (rtpsession);
+
+ if (send_rtp_sink) {
+ gst_pad_push_event (send_rtp_sink, gst_event_new_reconfigure ());
+ gst_object_unref (send_rtp_sink);
+ }
+}
+
+static void
+gst_rtp_session_notify_early_rtcp (RTPSession * sess, gpointer user_data)
+{
+ GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
+
+ GST_DEBUG_OBJECT (rtpsession, "Notified of early RTCP");
+ /* with an early RTCP request, we might have to start the RTCP thread */
+ GST_RTP_SESSION_LOCK (rtpsession);
+ signal_waiting_rtcp_thread_unlocked (rtpsession);
+ GST_RTP_SESSION_UNLOCK (rtpsession);
+}
projects / platform / upstream / gst-plugins-good.git / blobdiff