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[gst-cvs] gst-plugins-good: rtpsession: Implement sending PLI packets in response to GstForceKeyUnit
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From: <wt...@ke...> - 2011-02-01 18:00:51
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Module: gst-plugins-good Branch: master Commit: 52f95fa7ee441e842dc6ef7fb5cd28bfa1178866 URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=52f95fa7ee441e842dc6ef7fb5cd28bfa1178866 Author: Olivier Crête <oli...@co...> Date: Tue Jun 22 19:56:50 2010 -0400 rtpsession: Implement sending PLI packets in response to GstForceKeyUnit --- gst/rtpmanager/gstrtpsession.c | 73 +++++++++++++++++++++++++++++++++++++ gst/rtpmanager/rtpsession.c | 77 ++++++++++++++++++++++++++++++++++++++++ gst/rtpmanager/rtpsession.h | 6 +++ 3 files changed, 156 insertions(+), 0 deletions(-) diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index f10ab92..6d36633 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -1370,6 +1370,77 @@ gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstEvent * event) } +static gboolean +gst_rtp_session_request_remote_key_unit (GstRtpSession * rtpsession, + guint32 ssrc, guint payload, gboolean all_headers) +{ + GstCaps *caps; + gboolean requested = FALSE; + + caps = gst_rtp_session_get_caps_for_pt (rtpsession, payload); + + if (caps) { + gboolean fir, pli; + const GstStructure *s = gst_caps_get_structure (caps, 0); + + if (all_headers && + gst_structure_get_boolean (s, "rtcp-fb-nack-fir", &fir) && fir) { + /* 500 ms acceptable delay for FIR request is a guesstimate, it could + * be made configurable if needed + */ + rtp_session_request_early_rtcp (rtpsession->priv->session, + gst_clock_get_time (rtpsession->priv->sysclock), 500 * GST_MSECOND); + rtp_session_request_key_unit (rtpsession->priv->session, ssrc, TRUE); + requested = TRUE; + } else if (gst_structure_get_boolean (s, "rtcp-fb-nack-pli", &pli) && pli) { + rtp_session_request_key_unit (rtpsession->priv->session, ssrc, FALSE); + requested = TRUE; + } + gst_caps_unref (caps); + } + + return requested; +} + +static gboolean +gst_rtp_session_event_recv_rtp_src (GstPad * pad, GstEvent * event) +{ + GstRtpSession *rtpsession; + gboolean forward = TRUE; + gboolean ret = TRUE; + const GstStructure *s; + guint32 ssrc; + guint pt; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_CUSTOM_UPSTREAM: + s = gst_event_get_structure (event); + if (gst_structure_has_name (s, "GstForceKeyUnit") && + gst_structure_get_uint (s, "ssrc", &ssrc) && + gst_structure_get_uint (s, "payload", &pt)) { + gboolean all_headers = FALSE; + + gst_structure_get_boolean (s, "all-headers", &all_headers); + if (gst_rtp_session_request_remote_key_unit (rtpsession, ssrc, pt, + all_headers)) + forward = FALSE; + } + break; + default: + break; + } + + if (forward) + ret = gst_pad_push_event (rtpsession->recv_rtp_sink, event); + + gst_object_unref (rtpsession); + + return ret; +} + + static GstIterator * gst_rtp_session_iterate_internal_links (GstPad * pad) { @@ -1780,6 +1851,8 @@ create_recv_rtp_sink (GstRtpSession * rtpsession) rtpsession->recv_rtp_src = gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template, "recv_rtp_src"); + gst_pad_set_event_function (rtpsession->recv_rtp_src, + (GstPadEventFunction) gst_rtp_session_event_recv_rtp_src); gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_src, gst_rtp_session_iterate_internal_links); gst_pad_use_fixed_caps (rtpsession->recv_rtp_src); diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index ec40115..32973c0 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -102,6 +102,10 @@ static void rtp_session_set_property (GObject * object, guint prop_id, static void rtp_session_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); +static gboolean rtp_session_on_sending_rtcp (RTPSession * sess, + GstBuffer * buffer, gboolean early); + + static guint rtp_session_signals[LAST_SIGNAL] = { 0 }; G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT); @@ -403,6 +407,7 @@ rtp_session_class_init (RTPSessionClass * klass) klass->get_source_by_ssrc = GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc); + klass->on_sending_rtcp = GST_DEBUG_FUNCPTR (rtp_session_on_sending_rtcp); GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session"); } @@ -457,6 +462,8 @@ rtp_session_init (RTPSession * sess) sess->allow_early = TRUE; sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW; + sess->rtcp_pli_requests = g_array_new (FALSE, FALSE, sizeof (guint32)); + GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc); } @@ -477,6 +484,8 @@ rtp_session_finalize (GObject * object) g_hash_table_destroy (sess->cnames); g_object_unref (sess->source); + g_array_free (sess->rtcp_pli_requests, TRUE); + G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object); } @@ -2973,3 +2982,71 @@ dont_send: RTP_SESSION_UNLOCK (sess); } + +void +rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, gboolean fir) +{ + guint i; + + if (fir) + return; + + for (i = 0; i < sess->rtcp_pli_requests->len; i++) + if (ssrc == g_array_index (sess->rtcp_pli_requests, guint32, i)) + return; + + g_array_append_val (sess->rtcp_pli_requests, ssrc); +} + +static gboolean +has_pli_compare_func (gconstpointer a, gconstpointer ignored) +{ + GstRTCPPacket packet; + + packet.buffer = (GstBuffer *) a; + packet.offset = 0; + + if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB && + gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI) + return TRUE; + else + return FALSE; +} + +static gboolean +rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer, + gboolean early) +{ + gboolean ret = FALSE; + + RTP_SESSION_LOCK (sess); + + while (sess->rtcp_pli_requests->len) { + GstRTCPPacket rtcppacket; + guint media_ssrc = g_array_index (sess->rtcp_pli_requests, guint32, 0); + RTPSource *media_src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx], + GUINT_TO_POINTER (media_ssrc)); + + if (media_src && !rtp_source_has_retained (media_src, + has_pli_compare_func, NULL)) { + if (gst_rtcp_buffer_add_packet (buffer, GST_RTCP_TYPE_PSFB, &rtcppacket)) { + gst_rtcp_packet_fb_set_type (&rtcppacket, GST_RTCP_PSFB_TYPE_PLI); + gst_rtcp_packet_fb_set_sender_ssrc (&rtcppacket, + rtp_source_get_ssrc (sess->source)); + gst_rtcp_packet_fb_set_media_ssrc (&rtcppacket, media_ssrc); + ret = TRUE; + } else { + /* Break because the packet is full, will put next request in a + * further packet + */ + break; + } + } + + g_array_remove_index (sess->rtcp_pli_requests, 0); + } + + RTP_SESSION_UNLOCK (sess); + + return ret; +} diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index 6ae318e..3113247 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -203,6 +203,8 @@ struct _RTPSession { gboolean change_ssrc; gboolean favor_new; GstClockTime rtcp_feedback_retention_window; + + GArray *rtcp_pli_requests; }; /** @@ -308,5 +310,9 @@ GstFlowReturn rtp_session_on_timeout (RTPSession *sess, GstClockTi void rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time, GstClockTimeDiff max_delay); +/* Notify session of a request for a new key unit */ +void rtp_session_request_key_unit (RTPSession * sess, + guint32 ssrc, + gboolean fir); #endif /* __RTP_SESSION_H__ */ |
