From abc90da1dc8749b56a5c0ad0c0f8475f04e18213 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Fri, 26 Jul 2013 10:24:22 +0200 Subject: [PATCH] session: delay allocation of internal source Allocate the internal source when we receive a caps with the SSRC or when we see a buffer with the SSRC. --- gst/rtpmanager/rtpsession.c | 119 ++++++++++++++++++++++++++++---------------- gst/rtpmanager/rtpsession.h | 1 - 2 files changed, 76 insertions(+), 44 deletions(-) diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 56da482..8365d66 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -463,8 +463,6 @@ rtp_session_init (RTPSession * sess) { gint i; gchar *str; - guint32 ssrc; - gboolean created; g_mutex_init (&sess->lock); sess->key = g_random_int (); @@ -511,9 +509,8 @@ rtp_session_init (RTPSession * sess) gst_structure_set (sess->sdes, "tool", G_TYPE_STRING, "GStreamer", NULL); - /* create an active SSRC for this session manager */ - ssrc = rtp_session_create_new_ssrc (sess); - sess->source = obtain_internal_source (sess, ssrc, &created); + /* this is the SSRC we suggest */ + sess->suggested_ssrc = rtp_session_create_new_ssrc (sess); sess->first_rtcp = TRUE; sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE; @@ -540,7 +537,6 @@ rtp_session_finalize (GObject * object) for (i = 0; i < 32; i++) g_hash_table_destroy (sess->ssrcs[i]); - g_object_unref (sess->source); g_mutex_clear (&sess->lock); G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object); @@ -655,7 +651,8 @@ rtp_session_get_property (GObject * object, guint prop_id, g_value_set_uint (value, rtp_session_suggest_ssrc (sess)); break; case PROP_INTERNAL_SOURCE: - g_value_set_object (value, sess->source); + /* FIXME, return a random source */ + g_value_set_object (value, NULL); break; case PROP_BANDWIDTH: g_value_set_double (value, sess->bandwidth); @@ -1389,26 +1386,6 @@ obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created) return source; } -static void -rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc) -{ - if (ssrc != sess->source->ssrc) { - g_hash_table_steal (sess->ssrcs[sess->mask_idx], - GINT_TO_POINTER (sess->source->ssrc)); - - GST_DEBUG ("setting internal SSRC to %08x", ssrc); - /* After this call, any receiver of the old SSRC either in RTP or RTCP - * packets will timeout on the old SSRC, we could potentially schedule a - * BYE RTCP for the old SSRC... */ - sess->source->ssrc = ssrc; - rtp_source_reset (sess->source); - - /* rehash with the new SSRC */ - g_hash_table_insert (sess->ssrcs[sess->mask_idx], - GINT_TO_POINTER (sess->source->ssrc), sess->source); - } -} - /** * rtp_session_suggest_ssrc: * @sess: a #RTPSession @@ -1670,9 +1647,13 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer, gst_rtp_buffer_unmap (&rtp); RTP_SESSION_LOCK (sess); +#if 0 + /* FIXME, we should simply not update any stats on the BYE + * internal sources */ /* ignore more RTP packets when we left the session */ if (sess->source->marked_bye) goto ignore; +#endif /* update arrival stats */ update_arrival_stats (sess, &arrival, TRUE, buffer, current_time, @@ -1747,6 +1728,7 @@ invalid_packet: GST_DEBUG ("invalid RTP packet received"); return GST_FLOW_OK; } +#if 0 ignore: { RTP_SESSION_UNLOCK (sess); @@ -1754,6 +1736,7 @@ ignore: GST_DEBUG ("ignoring RTP packet because we are leaving"); return GST_FLOW_OK; } +#endif collision: { RTP_SESSION_UNLOCK (sess); @@ -1775,16 +1758,23 @@ rtp_session_process_rb (RTPSession * sess, RTPSource * source, guint32 ssrc, exthighestseq, jitter, lsr, dlsr; guint8 fractionlost; gint32 packetslost; + RTPSource *src; gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost, &packetslost, &exthighestseq, &jitter, &lsr, &dlsr); GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter); - if (ssrc == sess->source->ssrc) { + /* find our own source */ + src = find_source (sess, ssrc); + if (src == NULL) + continue; + + if (src->internal) { /* only deal with report blocks for our session, we update the stats of * the sender of the RTCP message. We could also compare our stats against * the other sender to see if we are better or worse. */ + /* FIXME, need to keep track who the RB block is from */ rtp_source_process_rb (source, arrival->ntpnstime, fractionlost, packetslost, exthighestseq, jitter, lsr, dlsr); } @@ -2146,19 +2136,17 @@ rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc, if (!src && sender_ssrc == 1) { GHashTableIter iter; - if (sess->stats.sender_sources > - RTP_SOURCE_IS_SENDER (sess->source) ? 2 : 1) + /* we can't find the source if there are multiple */ + if (sess->stats.sender_sources > sess->stats.internal_sender_sources + 1) return; g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]); - while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) { - if (src != sess->source && rtp_source_is_sender (src)) + if (!src->internal && rtp_source_is_sender (src)) break; src = NULL; } } - if (!src) return; @@ -2283,8 +2271,11 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer, update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1, ntpnstime); +#if 0 + /* FIXME, simply ignore RTCP for iternal sources with BYE */ if (sess->source->sent_bye) goto ignore; +#endif /* start processing the compound packet */ gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp); @@ -2369,6 +2360,7 @@ invalid_packet: gst_buffer_unref (buffer); return GST_FLOW_OK; } +#if 0 ignore: { RTP_SESSION_UNLOCK (sess); @@ -2377,6 +2369,7 @@ ignore: GST_DEBUG ("ignoring RTCP packet because we left"); return GST_FLOW_OK; } +#endif } /** @@ -2399,12 +2392,18 @@ rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps) s = gst_caps_get_structure (caps, 0); - if (gst_structure_get_uint (s, "ssrc", &ssrc)) - rtp_session_set_internal_ssrc (sess, ssrc); + if (gst_structure_get_uint (s, "ssrc", &ssrc)) { + RTPSource *source; + gboolean created; - RTP_SESSION_LOCK (sess); - rtp_source_update_caps (sess->source, caps); - RTP_SESSION_UNLOCK (sess); + RTP_SESSION_LOCK (sess); + source = obtain_internal_source (sess, ssrc, &created); + if (source) { + rtp_source_update_caps (source, caps); + g_object_unref (source); + } + RTP_SESSION_UNLOCK (sess); + } } /** @@ -2428,14 +2427,36 @@ rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list, RTPSource *source; gboolean prevsender; guint64 oldrate; + GstBuffer *buffer; + GstRTPBuffer rtp = { NULL }; + guint32 ssrc; + gboolean created; g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR); g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR); GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet"); + if (is_list) { + GstBufferList *list = GST_BUFFER_LIST_CAST (data); + + buffer = gst_buffer_list_get (list, 0); + if (!buffer) + goto no_buffer; + } else { + buffer = GST_BUFFER_CAST (data); + } + + if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)) + goto invalid_packet; + + /* get SSRC and look up in session database */ + ssrc = gst_rtp_buffer_get_ssrc (&rtp); + + gst_rtp_buffer_unmap (&rtp); + RTP_SESSION_LOCK (sess); - source = sess->source; + source = obtain_internal_source (sess, ssrc, &created); /* update last activity */ source->last_rtp_activity = current_time; @@ -2454,7 +2475,22 @@ rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list, sess->recalc_bandwidth = TRUE; RTP_SESSION_UNLOCK (sess); + g_object_unref (source); + return result; + +invalid_packet: + { + gst_mini_object_unref (GST_MINI_OBJECT_CAST (data)); + GST_DEBUG ("invalid RTP packet received"); + return GST_FLOW_OK; + } +no_buffer: + { + gst_mini_object_unref (GST_MINI_OBJECT_CAST (data)); + GST_DEBUG ("no buffer in list"); + return GST_FLOW_OK; + } } static void @@ -2640,10 +2676,7 @@ rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time) } if (sess->scheduled_bye) { - if (sess->source->sent_bye) { - GST_DEBUG ("we sent BYE already"); - interval = GST_CLOCK_TIME_NONE; - } else if (sess->stats.active_sources >= 50) { + if (sess->stats.active_sources >= 50) { GST_DEBUG ("reconsider BYE, more than 50 sources"); /* reconsider BYE if members >= 50 */ interval = calculate_rtcp_interval (sess, FALSE, TRUE); diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index 2ac3388..341bcbd 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -198,7 +198,6 @@ struct _RTPSession { guint rtcp_rr_bandwidth; guint rtcp_rs_bandwidth; - RTPSource *source; guint32 suggested_ssrc; /* for sender/receiver counting */ -- 2.7.4