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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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gst/rtpmanager/gstrtpbin.c: fix for pad name change
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_rtcp): fix for pad name change * gst/rtpmanager/gstrtpsession.c: (rtcp_thread), (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate): Fix for renamed methods. * gst/rtpmanager/rtpsession.c: (rtp_session_init), (rtp_session_finalize), (rtp_session_set_cname), (rtp_session_get_cname), (rtp_session_set_name), (rtp_session_get_name), (rtp_session_set_email), (rtp_session_get_email), (rtp_session_set_phone), (rtp_session_get_phone), (rtp_session_set_location), (rtp_session_get_location), (rtp_session_set_tool), (rtp_session_get_tool), (rtp_session_set_note), (rtp_session_get_note), (source_push_rtp), (obtain_source), (rtp_session_add_source), (rtp_session_get_source_by_ssrc), (rtp_session_create_source), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_sdes), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_get_reporting_interval), (session_report_blocks), (session_sdes), (rtp_session_perform_reporting): * gst/rtpmanager/rtpsession.h: Prepare for implementing SSRC sampling. Create SSRC for the session. Add methods to set the SDES entries. fix accounting of senders/receivers. Implement SR/RR/SDES RTCP reporting. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq), (rtp_source_process_rtp), (rtp_source_process_sr): * gst/rtpmanager/rtpsource.h: Implement extended sequence number. * gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval): * gst/rtpmanager/rtpstats.h: Rename some fields.
This commit is contained in:
Wim Taymans
Tim-Philipp Müller
parent
677b361dc3
commit
23883be047
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@ -959,7 +959,7 @@ create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
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/* get rtcp_src pad and store */
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session->rtcp_src =
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gst_element_get_request_pad (session->session, "rtcp_src");
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gst_element_get_request_pad (session->session, "send_rtcp_src");
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if (session->rtcp_src == NULL)
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goto pad_failed;
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@ -305,7 +305,7 @@ rtcp_thread (GstRTPSession * rtpsession)
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gdouble timeout;
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GstClockTime target;
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timeout = rtp_session_get_rtcp_interval (rtpsession->priv->session);
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timeout = rtp_session_get_reporting_interval (rtpsession->priv->session);
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GST_DEBUG_OBJECT (rtpsession, "next RTCP timeout: %lf", timeout);
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target = gst_clock_get_time (clock);
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@ -318,7 +318,7 @@ rtcp_thread (GstRTPSession * rtpsession)
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GST_DEBUG_OBJECT (rtpsession, "got RTCP timeout");
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/* make the session manager produce RTCP, we ignore the result. */
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rtp_session_produce_rtcp (rtpsession->priv->session);
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rtp_session_perform_reporting (rtpsession->priv->session);
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GST_RTP_SESSION_LOCK (rtpsession);
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gst_clock_id_unref (id);
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@ -472,6 +472,8 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
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rtpsession = GST_RTP_SESSION (user_data);
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priv = rtpsession->priv;
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GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
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if (rtpsession->send_rtcp_src) {
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result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
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} else {
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@ -515,6 +517,8 @@ gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
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if (!gst_structure_get_int (caps_struct, "clock-rate", &result))
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goto no_clock_rate;
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GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result);
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return result;
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/* ERRORS */
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@ -57,6 +57,9 @@ static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
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G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
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static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
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gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
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static void
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rtp_session_class_init (RTPSessionClass * klass)
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{
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@ -123,18 +126,35 @@ rtp_session_class_init (RTPSessionClass * klass)
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static void
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rtp_session_init (RTPSession * sess)
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{
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sess->lock = g_mutex_new ();
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sess->ssrcs =
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g_hash_table_new_full (NULL, NULL, NULL, (GDestroyNotify) g_object_unref);
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sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);
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gint i;
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/* create an SSRC for this session manager */
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sess->source = rtp_session_create_source (sess);
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sess->lock = g_mutex_new ();
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sess->key = g_random_int ();
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sess->mask_idx = 0;
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sess->mask = 0;
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for (i = 0; i < 32; i++) {
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sess->ssrcs[i] =
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g_hash_table_new_full (NULL, NULL, NULL,
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(GDestroyNotify) g_object_unref);
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}
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sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);
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rtp_stats_init_defaults (&sess->stats);
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/* create an active SSRC for this session manager */
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sess->source = rtp_session_create_source (sess);
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sess->stats.active_sources++;
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/* default UDP header length */
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sess->header_len = 28;
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sess->mtu = 1400;
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/* some default SDES entries */
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//sess->cname = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
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sess->cname = g_strdup_printf ("foo@%s", g_get_host_name ());
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sess->name = g_strdup (g_get_real_name ());
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sess->tool = g_strdup ("GStreamer");
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GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
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}
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@ -143,14 +163,20 @@ static void
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rtp_session_finalize (GObject * object)
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{
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RTPSession *sess;
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gint i;
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sess = RTP_SESSION_CAST (object);
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g_mutex_free (sess->lock);
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g_hash_table_destroy (sess->ssrcs);
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for (i = 0; i < 32; i++)
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g_hash_table_destroy (sess->ssrcs[i]);
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g_hash_table_destroy (sess->cnames);
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g_object_unref (sess->source);
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g_free (sess->cname);
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g_free (sess->tool);
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G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
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}
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@ -312,6 +338,230 @@ rtp_session_get_rtcp_bandwidth (RTPSession * sess)
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return sess->stats.rtcp_bandwidth;
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}
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/**
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* rtp_session_set_cname:
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* @sess: an #RTPSession
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* @cname: a CNAME for the session
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*
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* Set the CNAME for the session.
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*/
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void
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rtp_session_set_cname (RTPSession * sess, const gchar * cname)
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{
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g_return_if_fail (RTP_IS_SESSION (sess));
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g_free (sess->cname);
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sess->cname = g_strdup (cname);
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}
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/**
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* rtp_session_get_cname:
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* @sess: an #RTPSession
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*
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* Get the currently configured CNAME for the session.
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*
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* Returns: The CNAME. g_free after usage.
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*/
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gchar *
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rtp_session_get_cname (RTPSession * sess)
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{
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g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
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return g_strdup (sess->cname);
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}
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/**
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* rtp_session_set_name:
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* @sess: an #RTPSession
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* @name: a NAME for the session
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*
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* Set the NAME for the session.
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*/
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void
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rtp_session_set_name (RTPSession * sess, const gchar * name)
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{
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g_return_if_fail (RTP_IS_SESSION (sess));
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g_free (sess->name);
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sess->name = g_strdup (name);
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}
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/**
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* rtp_session_get_name:
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* @sess: an #RTPSession
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*
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* Get the currently configured NAME for the session.
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*
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* Returns: The NAME. g_free after usage.
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*/
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gchar *
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rtp_session_get_name (RTPSession * sess)
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{
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g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
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return g_strdup (sess->name);
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}
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/**
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* rtp_session_set_email:
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* @sess: an #RTPSession
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* @email: an EMAIL for the session
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*
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* Set the EMAIL the session.
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*/
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void
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rtp_session_set_email (RTPSession * sess, const gchar * email)
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{
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g_return_if_fail (RTP_IS_SESSION (sess));
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g_free (sess->email);
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sess->email = g_strdup (email);
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}
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/**
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* rtp_session_get_email:
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* @sess: an #RTPSession
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*
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* Get the currently configured EMAIL of the session.
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*
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* Returns: The EMAIL. g_free after usage.
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*/
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gchar *
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rtp_session_get_email (RTPSession * sess)
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{
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g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
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return g_strdup (sess->email);
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}
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/**
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* rtp_session_set_phone:
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* @sess: an #RTPSession
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* @phone: a PHONE for the session
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*
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* Set the PHONE the session.
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*/
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void
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rtp_session_set_phone (RTPSession * sess, const gchar * phone)
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{
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g_return_if_fail (RTP_IS_SESSION (sess));
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g_free (sess->phone);
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sess->phone = g_strdup (phone);
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}
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/**
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* rtp_session_get_location:
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* @sess: an #RTPSession
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*
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* Get the currently configured PHONE of the session.
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*
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* Returns: The PHONE. g_free after usage.
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*/
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gchar *
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rtp_session_get_phone (RTPSession * sess)
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{
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g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
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return g_strdup (sess->phone);
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}
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/**
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* rtp_session_set_location:
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* @sess: an #RTPSession
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* @location: a LOCATION for the session
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*
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* Set the LOCATION the session.
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*/
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void
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rtp_session_set_location (RTPSession * sess, const gchar * location)
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{
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g_return_if_fail (RTP_IS_SESSION (sess));
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g_free (sess->location);
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sess->location = g_strdup (location);
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}
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/**
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* rtp_session_get_location:
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* @sess: an #RTPSession
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*
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* Get the currently configured LOCATION of the session.
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*
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* Returns: The LOCATION. g_free after usage.
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*/
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gchar *
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rtp_session_get_location (RTPSession * sess)
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{
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g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
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return g_strdup (sess->location);
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}
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/**
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* rtp_session_set_tool:
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* @sess: an #RTPSession
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* @tool: a TOOL for the session
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*
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* Set the TOOL the session.
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*/
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void
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rtp_session_set_tool (RTPSession * sess, const gchar * tool)
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{
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g_return_if_fail (RTP_IS_SESSION (sess));
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g_free (sess->tool);
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sess->tool = g_strdup (tool);
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}
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/**
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* rtp_session_get_tool:
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* @sess: an #RTPSession
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*
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* Get the currently configured TOOL of the session.
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*
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* Returns: The TOOL. g_free after usage.
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*/
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gchar *
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rtp_session_get_tool (RTPSession * sess)
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{
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g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
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return g_strdup (sess->tool);
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}
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/**
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* rtp_session_set_note:
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* @sess: an #RTPSession
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* @note: a NOTE for the session
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*
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* Set the NOTE the session.
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*/
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void
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rtp_session_set_note (RTPSession * sess, const gchar * note)
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{
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g_return_if_fail (RTP_IS_SESSION (sess));
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g_free (sess->note);
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sess->note = g_strdup (note);
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}
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/**
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* rtp_session_get_note:
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* @sess: an #RTPSession
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*
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* Get the currently configured NOTE of the session.
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*
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* Returns: The NOTE. g_free after usage.
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*/
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gchar *
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rtp_session_get_note (RTPSession * sess)
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{
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g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
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return g_strdup (sess->note);
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}
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static GstFlowReturn
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source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
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{
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@ -319,6 +569,8 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
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if (source == session->source) {
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GST_DEBUG ("source %08x pushed sender RTP packet", source->ssrc);
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if (session->callbacks.send_rtp)
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result =
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session->callbacks.send_rtp (session, source, buffer,
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@ -371,7 +623,8 @@ obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
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{
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RTPSource *source;
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source = g_hash_table_lookup (sess->ssrcs, GINT_TO_POINTER (ssrc));
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source =
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g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
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if (source == NULL) {
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/* make new Source in probation and insert */
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source = rtp_source_new (ssrc);
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@ -392,7 +645,8 @@ obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
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/* configure a callback on the source */
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rtp_source_set_callbacks (source, &callbacks, sess);
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g_hash_table_insert (sess->ssrcs, GINT_TO_POINTER (ssrc), source);
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g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
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source);
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/* we have one more source now */
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sess->total_sources++;
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@ -426,9 +680,12 @@ rtp_session_add_source (RTPSession * sess, RTPSource * src)
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g_return_val_if_fail (src != NULL, FALSE);
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RTP_SESSION_LOCK (sess);
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find = g_hash_table_lookup (sess->ssrcs, GINT_TO_POINTER (src->ssrc));
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find =
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g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
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GINT_TO_POINTER (src->ssrc));
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if (find == NULL) {
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g_hash_table_insert (sess->ssrcs, GINT_TO_POINTER (src->ssrc), src);
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g_hash_table_insert (sess->ssrcs[sess->mask_idx],
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GINT_TO_POINTER (src->ssrc), src);
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/* we have one more source now */
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sess->total_sources++;
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result = TRUE;
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@ -501,7 +758,8 @@ rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
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g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
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RTP_SESSION_LOCK (sess);
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result = g_hash_table_lookup (sess->ssrcs, GINT_TO_POINTER (ssrc));
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result =
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g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
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if (result)
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g_object_ref (result);
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RTP_SESSION_UNLOCK (sess);
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@ -556,11 +814,13 @@ rtp_session_create_source (RTPSession * sess)
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ssrc = g_random_int ();
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/* see if it exists in the session, we're done if it doesn't */
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if (g_hash_table_lookup (sess->ssrcs, GINT_TO_POINTER (ssrc)) == NULL)
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if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
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GINT_TO_POINTER (ssrc)) == NULL)
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break;
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}
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source = rtp_source_new (ssrc);
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g_hash_table_insert (sess->ssrcs, GINT_TO_POINTER (ssrc), source);
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g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
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source);
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/* we have one more source now */
|
||||
sess->total_sources++;
|
||||
RTP_SESSION_UNLOCK (sess);
|
||||
|
|
@ -633,6 +893,8 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer)
|
|||
prevsender = RTP_SOURCE_IS_SENDER (source);
|
||||
prevactive = RTP_SOURCE_IS_ACTIVE (source);
|
||||
|
||||
gst_buffer_ref (buffer);
|
||||
|
||||
/* let source process the packet */
|
||||
result = rtp_source_process_rtp (source, buffer, &arrival);
|
||||
|
||||
|
|
@ -652,10 +914,11 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer)
|
|||
if (created)
|
||||
on_new_ssrc (sess, source);
|
||||
|
||||
/* for validated sources, we add the CSRCs as well */
|
||||
if (source->validated) {
|
||||
guint8 i, count;
|
||||
gboolean created;
|
||||
|
||||
/* for validated sources, we add the CSRCs as well */
|
||||
count = gst_rtp_buffer_get_csrc_count (buffer);
|
||||
|
||||
for (i = 0; i < count; i++) {
|
||||
|
|
@ -675,6 +938,8 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer)
|
|||
}
|
||||
}
|
||||
}
|
||||
gst_buffer_unref (buffer);
|
||||
|
||||
RTP_SESSION_UNLOCK (sess);
|
||||
|
||||
return result;
|
||||
|
|
@ -704,17 +969,27 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
|
|||
guint64 ntptime;
|
||||
guint count, i;
|
||||
RTPSource *source;
|
||||
gboolean created;
|
||||
gboolean created, prevsender;
|
||||
|
||||
gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
|
||||
&packet_count, &octet_count);
|
||||
|
||||
GST_DEBUG ("got SR packet: SSRC %08x", senderssrc);
|
||||
|
||||
RTP_SESSION_LOCK (sess);
|
||||
source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
|
||||
|
||||
prevsender = RTP_SOURCE_IS_SENDER (source);
|
||||
|
||||
/* first update the source */
|
||||
rtp_source_process_sr (source, ntptime, rtptime, packet_count, octet_count);
|
||||
|
||||
if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
|
||||
sess->stats.sender_sources++;
|
||||
GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
|
||||
sess->stats.sender_sources);
|
||||
}
|
||||
|
||||
if (created)
|
||||
on_new_ssrc (sess, source);
|
||||
|
||||
|
|
@ -785,36 +1060,36 @@ static void
|
|||
rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
|
||||
RTPArrivalStats * arrival)
|
||||
{
|
||||
guint chunks, i, j;
|
||||
gboolean more_chunks, more_items;
|
||||
guint items, i, j;
|
||||
gboolean more_items, more_entries;
|
||||
|
||||
chunks = gst_rtcp_packet_sdes_get_chunk_count (packet);
|
||||
GST_DEBUG ("got SDES packet with %d chunks", chunks);
|
||||
items = gst_rtcp_packet_sdes_get_item_count (packet);
|
||||
GST_DEBUG ("got SDES packet with %d items", items);
|
||||
|
||||
more_chunks = gst_rtcp_packet_sdes_first_chunk (packet);
|
||||
more_items = gst_rtcp_packet_sdes_first_item (packet);
|
||||
i = 0;
|
||||
while (more_chunks) {
|
||||
while (more_items) {
|
||||
guint32 ssrc;
|
||||
|
||||
ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
|
||||
|
||||
GST_DEBUG ("chunk %d, SSRC %08x", i, ssrc);
|
||||
GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
|
||||
|
||||
more_items = gst_rtcp_packet_sdes_first_item (packet);
|
||||
more_entries = gst_rtcp_packet_sdes_first_entry (packet);
|
||||
j = 0;
|
||||
while (more_items) {
|
||||
while (more_entries) {
|
||||
GstRTCPSDESType type;
|
||||
guint8 len;
|
||||
gchar *data;
|
||||
guint8 *data;
|
||||
|
||||
gst_rtcp_packet_sdes_get_item (packet, &type, &len, &data);
|
||||
gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
|
||||
|
||||
GST_DEBUG ("item %d, type %d, len %d, data %s", j, type, len, data);
|
||||
GST_DEBUG ("entry %d, type %d, len %d, data %s", j, type, len, data);
|
||||
|
||||
more_items = gst_rtcp_packet_sdes_next_item (packet);
|
||||
more_entries = gst_rtcp_packet_sdes_next_entry (packet);
|
||||
j++;
|
||||
}
|
||||
more_chunks = gst_rtcp_packet_sdes_next_chunk (packet);
|
||||
more_items = gst_rtcp_packet_sdes_next_item (packet);
|
||||
i++;
|
||||
}
|
||||
}
|
||||
|
|
@ -906,6 +1181,7 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer)
|
|||
GST_DEBUG ("received RTCP packet");
|
||||
|
||||
/* get packet size including header overhead */
|
||||
RTP_SESSION_LOCK (sess);
|
||||
size = GST_BUFFER_SIZE (buffer) + sess->header_len;
|
||||
|
||||
/* update average RTCP packet size */
|
||||
|
|
@ -914,6 +1190,7 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer)
|
|||
else
|
||||
sess->stats.avg_rtcp_packet_size =
|
||||
(size + (15 * sess->stats.avg_rtcp_packet_size)) >> 4;
|
||||
RTP_SESSION_UNLOCK (sess);
|
||||
|
||||
/* start processing the compound packet */
|
||||
more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
|
||||
|
|
@ -972,6 +1249,7 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer)
|
|||
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
|
||||
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
|
||||
|
||||
RTP_SESSION_LOCK (sess);
|
||||
source = sess->source;
|
||||
|
||||
prevsender = RTP_SOURCE_IS_SENDER (source);
|
||||
|
|
@ -981,20 +1259,22 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer)
|
|||
|
||||
if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
|
||||
sess->stats.sender_sources++;
|
||||
RTP_SESSION_UNLOCK (sess);
|
||||
|
||||
return result;
|
||||
}
|
||||
|
||||
/**
|
||||
* rtp_session_get_rtcp_interval:
|
||||
* rtp_session_get_reporting_interval:
|
||||
* @sess: an #RTPSession
|
||||
*
|
||||
* Get the interval for sending out the next RTCP packet
|
||||
* Get the interval for sending out the next RTCP packet and doing general
|
||||
* maintenance tasks.
|
||||
*
|
||||
* Returns: an interval in seconds.
|
||||
*/
|
||||
gdouble
|
||||
rtp_session_get_rtcp_interval (RTPSession * sess)
|
||||
rtp_session_get_reporting_interval (RTPSession * sess)
|
||||
{
|
||||
gdouble result;
|
||||
|
||||
|
|
@ -1008,8 +1288,112 @@ rtp_session_get_rtcp_interval (RTPSession * sess)
|
|||
return result;
|
||||
}
|
||||
|
||||
typedef struct
|
||||
{
|
||||
RTPSession *sess;
|
||||
GstBuffer *rtcp;
|
||||
GstRTCPPacket packet;
|
||||
} ReportData;
|
||||
|
||||
static void
|
||||
session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
|
||||
{
|
||||
RTPSession *sess = data->sess;
|
||||
RTPSource *own = sess->source;
|
||||
GstRTCPPacket *packet = &data->packet;
|
||||
|
||||
/* create a new buffer if needed */
|
||||
if (data->rtcp == NULL) {
|
||||
data->rtcp = gst_rtcp_buffer_new (sess->mtu);
|
||||
|
||||
if (RTP_SOURCE_IS_SENDER (own)) {
|
||||
/* we are a sender, create SR */
|
||||
GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
|
||||
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
|
||||
|
||||
/* fill in sender report info */
|
||||
gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
|
||||
0, 0, own->stats.packets_sent, own->stats.octets_sent);
|
||||
} else {
|
||||
/* we are only receiver, create RR */
|
||||
GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
|
||||
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
|
||||
gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
|
||||
}
|
||||
}
|
||||
if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
|
||||
/* only report about other sources */
|
||||
if (source != sess->source) {
|
||||
RTPSourceStats *stats;
|
||||
guint32 extended_max, expected;
|
||||
guint32 expected_interval, received_interval;
|
||||
guint32 lost, lost_interval, fraction;
|
||||
|
||||
stats = &source->stats;
|
||||
|
||||
extended_max = (stats->cycles << 16) + stats->max_seq;
|
||||
expected = extended_max - stats->base_seq + 1;
|
||||
|
||||
if (expected > stats->packets_received) {
|
||||
lost = expected - stats->packets_received;
|
||||
if (lost > 0x7fffff)
|
||||
lost = 0x7fffff;
|
||||
} else {
|
||||
lost = stats->packets_received - expected;
|
||||
if (lost > 0x800000)
|
||||
lost = 0x800000;
|
||||
else
|
||||
lost = -lost;
|
||||
}
|
||||
|
||||
expected_interval = expected - stats->prev_expected;
|
||||
stats->prev_expected = expected;
|
||||
received_interval = stats->packets_received - stats->prev_received;
|
||||
stats->prev_received = stats->packets_received;
|
||||
|
||||
lost_interval = expected_interval - received_interval;
|
||||
|
||||
if (expected_interval == 0 || lost_interval <= 0)
|
||||
fraction = 0;
|
||||
else
|
||||
fraction = (lost_interval << 8) / expected_interval;
|
||||
|
||||
GST_DEBUG ("add RR for SSRC %08x", source->ssrc);
|
||||
/* we scaled the jitter up for additional precision */
|
||||
GST_DEBUG ("fraction %d, lost %d, extseq %u, jitter %d", fraction, lost,
|
||||
extended_max, stats->jitter >> 4);
|
||||
|
||||
/* packet is not yet filled, add report block for this source. */
|
||||
gst_rtcp_packet_add_rb (packet, source->ssrc, fraction, lost,
|
||||
extended_max, stats->jitter >> 4, 0, 0);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
static void
|
||||
session_sdes (RTPSession * sess, GstBuffer * buffer)
|
||||
{
|
||||
GstRTCPPacket packet;
|
||||
|
||||
/* add SDES packet */
|
||||
gst_rtcp_buffer_add_packet (buffer, GST_RTCP_TYPE_SDES, &packet);
|
||||
|
||||
gst_rtcp_packet_sdes_add_item (&packet, sess->source->ssrc);
|
||||
gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_CNAME,
|
||||
strlen (sess->cname), (guint8 *) sess->cname);
|
||||
|
||||
/* other SDES items must only be added at regular intervals and only when the
|
||||
* user requests to since it might be a privacy problem */
|
||||
#if 0
|
||||
gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_NAME,
|
||||
strlen (sess->name), (guint8 *) sess->name);
|
||||
gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_TOOL,
|
||||
strlen (sess->tool), (guint8 *) sess->tool);
|
||||
#endif
|
||||
}
|
||||
|
||||
/**
|
||||
* rtp_session_produce_rtcp:
|
||||
* rtp_session_perform_reporting:
|
||||
* @sess: an #RTPSession
|
||||
*
|
||||
* Instruct the session manager to generate RTCP packets with current stats.
|
||||
|
|
@ -1019,8 +1403,39 @@ rtp_session_get_rtcp_interval (RTPSession * sess)
|
|||
* Returns: a #GstFlowReturn.
|
||||
*/
|
||||
GstFlowReturn
|
||||
rtp_session_produce_rtcp (RTPSession * sess)
|
||||
rtp_session_perform_reporting (RTPSession * sess)
|
||||
{
|
||||
/* FIXME: implement me */
|
||||
return GST_FLOW_NOT_SUPPORTED;
|
||||
GstFlowReturn result = GST_FLOW_OK;
|
||||
ReportData data;
|
||||
|
||||
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
|
||||
|
||||
data.sess = sess;
|
||||
data.rtcp = NULL;
|
||||
|
||||
RTP_SESSION_LOCK (sess);
|
||||
/* loop over all known sources and do something */
|
||||
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
|
||||
(GHFunc) session_report_blocks, &data);
|
||||
|
||||
/* add SDES for this source */
|
||||
if (data.rtcp) {
|
||||
session_sdes (sess, data.rtcp);
|
||||
sess->stats.sent_rtcp = TRUE;
|
||||
}
|
||||
|
||||
RTP_SESSION_UNLOCK (sess);
|
||||
|
||||
/* push out the RTCP packet */
|
||||
if (data.rtcp) {
|
||||
/* close the RTCP packet */
|
||||
gst_rtcp_buffer_end (data.rtcp);
|
||||
|
||||
if (sess->callbacks.send_rtcp)
|
||||
result = sess->callbacks.send_rtcp (sess, sess->source, data.rtcp,
|
||||
sess->user_data);
|
||||
else
|
||||
gst_buffer_unref (data.rtcp);
|
||||
}
|
||||
return result;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -143,9 +143,24 @@ struct _RTPSession {
|
|||
GMutex *lock;
|
||||
|
||||
guint header_len;
|
||||
guint mtu;
|
||||
|
||||
RTPSource *source;
|
||||
GHashTable *ssrcs;
|
||||
|
||||
/* info for creating reports */
|
||||
gchar *cname;
|
||||
gchar *name;
|
||||
gchar *email;
|
||||
gchar *phone;
|
||||
gchar *location;
|
||||
gchar *tool;
|
||||
gchar *note;
|
||||
|
||||
/* for sender/receiver counting */
|
||||
guint32 key;
|
||||
guint32 mask_idx;
|
||||
guint32 mask;
|
||||
GHashTable *ssrcs[32];
|
||||
GHashTable *cnames;
|
||||
guint total_sources;
|
||||
|
||||
|
|
@ -184,6 +199,21 @@ gdouble rtp_session_get_bandwidth (RTPSession *sess);
|
|||
void rtp_session_set_rtcp_fraction (RTPSession *sess, gdouble fraction);
|
||||
gdouble rtp_session_get_rtcp_fraction (RTPSession *sess);
|
||||
|
||||
void rtp_session_set_cname (RTPSession *sess, const gchar *cname);
|
||||
gchar* rtp_session_get_cname (RTPSession *sess);
|
||||
void rtp_session_set_name (RTPSession *sess, const gchar *name);
|
||||
gchar* rtp_session_get_name (RTPSession *sess);
|
||||
void rtp_session_set_email (RTPSession *sess, const gchar *email);
|
||||
gchar* rtp_session_get_email (RTPSession *sess);
|
||||
void rtp_session_set_phone (RTPSession *sess, const gchar *phone);
|
||||
gchar* rtp_session_get_phone (RTPSession *sess);
|
||||
void rtp_session_set_location (RTPSession *sess, const gchar *location);
|
||||
gchar* rtp_session_get_location (RTPSession *sess);
|
||||
void rtp_session_set_tool (RTPSession *sess, const gchar *tool);
|
||||
gchar* rtp_session_get_tool (RTPSession *sess);
|
||||
void rtp_session_set_note (RTPSession *sess, const gchar *note);
|
||||
gchar* rtp_session_get_note (RTPSession *sess);
|
||||
|
||||
/* handling sources */
|
||||
gboolean rtp_session_add_source (RTPSession *sess, RTPSource *src);
|
||||
gint rtp_session_get_num_sources (RTPSession *sess);
|
||||
|
|
@ -200,7 +230,7 @@ GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer
|
|||
GstFlowReturn rtp_session_send_rtp (RTPSession *sess, GstBuffer *buffer);
|
||||
|
||||
/* get interval for next RTCP interval */
|
||||
gdouble rtp_session_get_rtcp_interval (RTPSession *sess);
|
||||
GstFlowReturn rtp_session_produce_rtcp (RTPSession *sess);
|
||||
gdouble rtp_session_get_reporting_interval (RTPSession *sess);
|
||||
GstFlowReturn rtp_session_perform_reporting (RTPSession *sess);
|
||||
|
||||
#endif /* __RTP_SESSION_H__ */
|
||||
|
|
|
|||
|
|
@ -70,6 +70,7 @@ rtp_source_init (RTPSource * src)
|
|||
src->clock_rate = -1;
|
||||
src->packets = g_queue_new ();
|
||||
|
||||
src->stats.cycles = -1;
|
||||
src->stats.jitter = 0;
|
||||
src->stats.transit = -1;
|
||||
src->stats.curr_sr = 0;
|
||||
|
|
@ -279,6 +280,20 @@ no_clock_rate:
|
|||
}
|
||||
}
|
||||
|
||||
static void
|
||||
init_seq (RTPSource * src, guint16 seq)
|
||||
{
|
||||
src->stats.base_seq = seq;
|
||||
src->stats.max_seq = seq;
|
||||
src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
|
||||
src->stats.cycles = 0;
|
||||
src->stats.packets_received = 0;
|
||||
src->stats.octets_received = 0;
|
||||
src->stats.bytes_received = 0;
|
||||
src->stats.prev_received = 0;
|
||||
src->stats.prev_expected = 0;
|
||||
}
|
||||
|
||||
/**
|
||||
* rtp_source_process_rtp:
|
||||
* @src: an #RTPSource
|
||||
|
|
@ -293,58 +308,108 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
|
|||
RTPArrivalStats * arrival)
|
||||
{
|
||||
GstFlowReturn result = GST_FLOW_OK;
|
||||
guint16 seqnr, udelta;
|
||||
RTPSourceStats *stats;
|
||||
|
||||
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
|
||||
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
|
||||
|
||||
stats = &src->stats;
|
||||
|
||||
seqnr = gst_rtp_buffer_get_seq (buffer);
|
||||
|
||||
if (stats->cycles == -1) {
|
||||
GST_DEBUG ("first buffer");
|
||||
/* first time we heard of this source */
|
||||
init_seq (src, seqnr);
|
||||
src->stats.max_seq = seqnr - 1;
|
||||
src->probation = RTP_DEFAULT_PROBATION;
|
||||
}
|
||||
|
||||
udelta = seqnr - stats->max_seq;
|
||||
|
||||
/* if we are still on probation, check seqnum */
|
||||
if (src->probation) {
|
||||
guint16 seqnr, expected;
|
||||
guint16 expected;
|
||||
|
||||
expected = src->stats.max_seqnr + 1;
|
||||
expected = src->stats.max_seq + 1;
|
||||
|
||||
/* when in probation, we require consecutive seqnums */
|
||||
seqnr = gst_rtp_buffer_get_seq (buffer);
|
||||
if (seqnr == expected) {
|
||||
/* expected packet */
|
||||
src->probation--;
|
||||
src->stats.max_seqnr = seqnr;
|
||||
|
||||
GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
|
||||
src->probation--;
|
||||
src->stats.max_seq = seqnr;
|
||||
if (src->probation == 0) {
|
||||
GST_DEBUG ("probation done!", src->probation);
|
||||
init_seq (src, seqnr);
|
||||
} else {
|
||||
GstBuffer *q;
|
||||
|
||||
GST_DEBUG ("probation %d: queue buffer", src->probation);
|
||||
/* when still in probation, keep packets in a list. */
|
||||
g_queue_push_tail (src->packets, buffer);
|
||||
/* remove packets from queue if there are too many */
|
||||
while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
|
||||
q = g_queue_pop_head (src->packets);
|
||||
gst_object_unref (q);
|
||||
}
|
||||
goto done;
|
||||
}
|
||||
} else {
|
||||
GST_DEBUG ("probation: seqnr %d != expected %d", seqnr, expected);
|
||||
src->probation = RTP_DEFAULT_PROBATION;
|
||||
src->stats.max_seqnr = seqnr;
|
||||
src->stats.max_seq = seqnr;
|
||||
goto done;
|
||||
}
|
||||
}
|
||||
if (src->probation) {
|
||||
GstBuffer *q;
|
||||
|
||||
GST_DEBUG ("probation %d: queue buffer", src->probation);
|
||||
/* when still in probation, keep packets in a list. */
|
||||
g_queue_push_tail (src->packets, buffer);
|
||||
/* remove packets from queue if there are too many */
|
||||
while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
|
||||
q = g_queue_pop_head (src->packets);
|
||||
gst_object_unref (q);
|
||||
} else if (udelta < RTP_MAX_DROPOUT) {
|
||||
/* in order, with permissible gap */
|
||||
if (seqnr < stats->max_seq) {
|
||||
/* sequence number wrapped - count another 64K cycle. */
|
||||
stats->cycles++;
|
||||
}
|
||||
stats->max_seq = seqnr;
|
||||
} else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
|
||||
/* the sequence number made a very large jump */
|
||||
if (seqnr == stats->bad_seq) {
|
||||
/* two sequential packets -- assume that the other side
|
||||
* restarted without telling us so just re-sync
|
||||
* (i.e., pretend this was the first packet). */
|
||||
init_seq (src, seqnr);
|
||||
} else {
|
||||
/* unacceptable jump */
|
||||
stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
|
||||
goto bad_sequence;
|
||||
}
|
||||
} else {
|
||||
/* we are not in probation */
|
||||
src->stats.octetsreceived += arrival->payload_len;
|
||||
src->stats.bytesreceived += arrival->bytes;
|
||||
src->stats.packetsreceived++;
|
||||
src->is_sender = TRUE;
|
||||
|
||||
GST_DEBUG ("PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
|
||||
src->stats.packetsreceived, src->stats.octetsreceived);
|
||||
|
||||
/* calculate jitter */
|
||||
calculate_jitter (src, buffer, arrival);
|
||||
|
||||
/* we're ready to push the RTP packet now */
|
||||
result = push_packet (src, buffer);
|
||||
/* duplicate or reordered packet, will be filtered by jitterbuffer. */
|
||||
}
|
||||
|
||||
src->stats.octets_received += arrival->payload_len;
|
||||
src->stats.bytes_received += arrival->bytes;
|
||||
src->stats.packets_received++;
|
||||
/* the source that sent the packet must be a sender */
|
||||
src->is_sender = TRUE;
|
||||
src->validated = TRUE;
|
||||
|
||||
GST_DEBUG ("PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
|
||||
src->stats.packets_received, src->stats.octets_received);
|
||||
|
||||
/* calculate jitter */
|
||||
calculate_jitter (src, buffer, arrival);
|
||||
|
||||
/* we're ready to push the RTP packet now */
|
||||
result = push_packet (src, buffer);
|
||||
|
||||
done:
|
||||
return result;
|
||||
|
||||
/* ERRORS */
|
||||
bad_sequence:
|
||||
{
|
||||
GST_WARNING ("unacceptable seqnum received");
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
|
|
@ -424,6 +489,9 @@ rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
|
|||
curridx = src->stats.curr_sr ^ 1;
|
||||
curr = &src->stats.sr[curridx];
|
||||
|
||||
/* this is a sender now */
|
||||
src->is_sender = TRUE;
|
||||
|
||||
/* update current */
|
||||
curr->is_valid = TRUE;
|
||||
curr->ntptime = ntptime;
|
||||
|
|
|
|||
|
|
@ -31,6 +31,10 @@
|
|||
#define RTP_NO_PROBATION 0
|
||||
#define RTP_DEFAULT_PROBATION 2
|
||||
|
||||
#define RTP_SEQ_MOD (1 << 16)
|
||||
#define RTP_MAX_DROPOUT 3000
|
||||
#define RTP_MAX_MISORDER 100
|
||||
|
||||
typedef struct _RTPSource RTPSource;
|
||||
typedef struct _RTPSourceClass RTPSourceClass;
|
||||
|
||||
|
|
@ -69,7 +73,8 @@ typedef struct _RTPSourceClass RTPSourceClass;
|
|||
*
|
||||
* Returns: a #GstFlowReturn.
|
||||
*/
|
||||
typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, GstBuffer *buffer, gpointer user_data);
|
||||
typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, GstBuffer *buffer,
|
||||
gpointer user_data);
|
||||
|
||||
/**
|
||||
* RTPSourceClockRate:
|
||||
|
|
@ -106,19 +111,23 @@ struct _RTPSource {
|
|||
GObject object;
|
||||
|
||||
/*< private >*/
|
||||
RTPSourceCallbacks callbacks;
|
||||
gpointer user_data;
|
||||
|
||||
guint32 ssrc;
|
||||
gchar *cname;
|
||||
|
||||
gint probation;
|
||||
gboolean validated;
|
||||
gboolean received_bye;
|
||||
gchar *bye_reason;
|
||||
|
||||
gboolean is_csrc;
|
||||
gboolean is_sender;
|
||||
|
||||
gchar *cname;
|
||||
gchar *name;
|
||||
gchar *email;
|
||||
gchar *phone;
|
||||
gchar *location;
|
||||
gchar *tool;
|
||||
gchar *note;
|
||||
gboolean received_bye;
|
||||
gchar *bye_reason;
|
||||
|
||||
gboolean have_rtp_from;
|
||||
GstNetAddress rtp_from;
|
||||
gboolean have_rtcp_from;
|
||||
|
|
@ -129,6 +138,9 @@ struct _RTPSource {
|
|||
|
||||
GQueue *packets;
|
||||
|
||||
RTPSourceCallbacks callbacks;
|
||||
gpointer user_data;
|
||||
|
||||
RTPSourceStats stats;
|
||||
};
|
||||
|
||||
|
|
@ -147,6 +159,7 @@ void rtp_source_set_as_csrc (RTPSource *src);
|
|||
void rtp_source_set_rtp_from (RTPSource *src, GstNetAddress *address);
|
||||
void rtp_source_set_rtcp_from (RTPSource *src, GstNetAddress *address);
|
||||
|
||||
/* handling RTP */
|
||||
GstFlowReturn rtp_source_process_rtp (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival);
|
||||
|
||||
GstFlowReturn rtp_source_send_rtp (RTPSource *src, GstBuffer *buffer);
|
||||
|
|
|
|||
|
|
@ -65,7 +65,7 @@ rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean sender)
|
|||
GST_DEBUG ("senders: %f, receivers %f, avg_rtcp %f, sfraction %f",
|
||||
senders, receivers, avg_rtcp, sfraction);
|
||||
|
||||
if (sfraction <= stats->sender_fraction) {
|
||||
if (senders > 0 && sfraction <= stats->sender_fraction) {
|
||||
if (sender) {
|
||||
interval =
|
||||
(avg_rtcp * senders) / (stats->sender_fraction *
|
||||
|
|
|
|||
|
|
@ -92,14 +92,23 @@ typedef struct {
|
|||
* Stats about a source.
|
||||
*/
|
||||
typedef struct {
|
||||
guint64 packetsreceived;
|
||||
guint64 prevpacketsreceived;
|
||||
guint64 octetsreceived;
|
||||
guint64 bytesreceived;
|
||||
guint16 max_seqnr;
|
||||
guint64 packets_received;
|
||||
guint64 octets_received;
|
||||
guint64 bytes_received;
|
||||
|
||||
guint32 prev_expected;
|
||||
guint32 prev_received;
|
||||
|
||||
guint16 max_seq;
|
||||
guint32 cycles;
|
||||
guint32 base_seq;
|
||||
guint32 bad_seq;
|
||||
guint32 transit;
|
||||
guint32 jitter;
|
||||
|
||||
guint64 packets_sent;
|
||||
guint64 octets_sent;
|
||||
|
||||
/* when we received stuff */
|
||||
GstClockTime prev_rtptime;
|
||||
GstClockTime prev_rtcptime;
|
||||
|
|
|
|||
Loading…
Reference in a new issue