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Diffstat (limited to 'gst/rtpmanager/gstrtpbin.c')
| -rw-r--r-- | gst/rtpmanager/gstrtpbin.c | 2458 |
1 files changed, 0 insertions, 2458 deletions
diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c deleted file mode 100644 index c09b0ab9..00000000 --- a/gst/rtpmanager/gstrtpbin.c +++ /dev/null @@ -1,2458 +0,0 @@ -/* GStreamer - * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com> - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/** - * SECTION:element-gstrtpbin - * @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux - * - * RTP bin combines the functions of #GstRtpSession, #GstRtpsSrcDemux, - * #GstRtpJitterBuffer and #GstRtpPtDemux in one element. It allows for multiple - * RTP sessions that will be synchronized together using RTCP SR packets. - * - * #GstRtpBin is configured with a number of request pads that define the - * functionality that is activated, similar to the #GstRtpSession element. - * - * To use #GstRtpBin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session - * number must be specified in the pad name. - * Data received on the recv_rtp_sink_%%d pad will be processed in the gstrtpsession - * manager and after being validated forwarded on #GstRtpsSrcDemux element. Each - * RTP stream is demuxed based on the SSRC and send to a #GstRtpJitterBuffer. After - * the packets are released from the jitterbuffer, they will be forwarded to a - * #GstRtpsSrcDemux element. The #GstRtpsSrcDemux element will demux the packets based - * on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on - * gstrtpbin with the session number, SSRC and payload type respectively as the pad - * name. - * - * To also use #GstRtpBin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The - * session number must be specified in the pad name. - * - * If you want the session manager to generate and send RTCP packets, request - * the send_rtcp_src_%%d pad with the session number in the pad name. Packet pushed - * on this pad contain SR/RR RTCP reports that should be sent to all participants - * in the session. - * - * To use #GstRtpBin as a sender, request a send_rtp_sink_%%d pad, which will - * automatically create a send_rtp_src_%%d pad. If the session number is not provided, - * the pad from the lowest available session will be returned. 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_%%d 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 - * mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map - * signal. - * - * <refsect2> - * <title>Example pipelines</title> - * |[ - * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \ - * gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink - * ]| Receive RTP data from port 5000 and send to the session 0 in gstrtpbin. - * |[ - * gst-launch gstrtpbin name=rtpbin \ - * v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ - * rtpbin.send_rtp_src_0 ! udpsink port=5000 \ - * rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \ - * udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ - * audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \ - * rtpbin.send_rtp_src_1 ! udpsink port=5002 \ - * rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \ - * udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 - * ]| Encode and payload H263 video captured from a v4l2src. Encode and payload AMR - * audio generated from audiotestsrc. The video is sent to session 0 in rtpbin - * and the audio is sent to session 1. Video packets are sent on UDP port 5000 - * and audio packets on port 5002. The video RTCP packets for session 0 are sent - * on port 5001 and the audio RTCP packets for session 0 are sent on port 5003. - * RTCP packets for session 0 are received on port 5005 and RTCP for session 1 - * is received on port 5007. Since RTCP packets from the sender should be sent - * as soon as possible and do not participate in preroll, sync=false and - * async=false is configured on udpsink - * |[ - * gst-launch -v gstrtpbin name=rtpbin \ - * udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \ - * port=5000 ! rtpbin.recv_rtp_sink_0 \ - * rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \ - * udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ - * rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ - * udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" \ - * port=5002 ! rtpbin.recv_rtp_sink_1 \ - * rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink \ - * udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ - * rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false - * ]| Receive H263 on port 5000, send it through rtpbin in session 0, depayload, - * decode and display the video. - * Receive AMR on port 5002, send it through rtpbin in session 1, depayload, - * decode and play the audio. - * Receive server RTCP packets for session 0 on port 5001 and RTCP packets for - * session 1 on port 5003. These packets will be used for session management and - * synchronisation. - * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1 - * on port 5007. - * </refsect2> - * - * Last reviewed on 2007-08-30 (0.10.6) - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif -#include <string.h> - -#include <gst/rtp/gstrtpbuffer.h> -#include <gst/rtp/gstrtcpbuffer.h> - -#include "gstrtpbin-marshal.h" -#include "gstrtpbin.h" -#include "rtpsession.h" -#include "gstrtpsession.h" -#include "gstrtpjitterbuffer.h" - -GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug); -#define GST_CAT_DEFAULT gst_rtp_bin_debug - -/* elementfactory information */ -static const GstElementDetails rtpbin_details = GST_ELEMENT_DETAILS ("RTP Bin", - "Filter/Network/RTP", - "Implement an RTP bin", - "Wim Taymans <wim.taymans@gmail.com>"); - -/* sink pads */ -static GstStaticPadTemplate rtpbin_recv_rtp_sink_template = -GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%d", - GST_PAD_SINK, - GST_PAD_REQUEST, - GST_STATIC_CAPS ("application/x-rtp") - ); - -static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template = -GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%d", - GST_PAD_SINK, - GST_PAD_REQUEST, - GST_STATIC_CAPS ("application/x-rtcp") - ); - -static GstStaticPadTemplate rtpbin_send_rtp_sink_template = -GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%d", - GST_PAD_SINK, - GST_PAD_REQUEST, - GST_STATIC_CAPS ("application/x-rtp") - ); - -/* src pads */ -static GstStaticPadTemplate rtpbin_recv_rtp_src_template = -GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%d_%d_%d", - GST_PAD_SRC, - GST_PAD_SOMETIMES, - GST_STATIC_CAPS ("application/x-rtp") - ); - -static GstStaticPadTemplate rtpbin_send_rtcp_src_template = -GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%d", - GST_PAD_SRC, - GST_PAD_REQUEST, - GST_STATIC_CAPS ("application/x-rtcp") - ); - -static GstStaticPadTemplate rtpbin_send_rtp_src_template = -GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%d", - GST_PAD_SRC, - GST_PAD_SOMETIMES, - GST_STATIC_CAPS ("application/x-rtp") - ); - -#define GST_RTP_BIN_GET_PRIVATE(obj) \ - (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRtpBinPrivate)) - -#define GST_RTP_BIN_LOCK(bin) g_mutex_lock ((bin)->priv->bin_lock) -#define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock ((bin)->priv->bin_lock) - -/* lock to protect dynamic callbacks, like pad-added and new ssrc. */ -#define GST_RTP_BIN_DYN_LOCK(bin) g_mutex_lock ((bin)->priv->dyn_lock) -#define GST_RTP_BIN_DYN_UNLOCK(bin) g_mutex_unlock ((bin)->priv->dyn_lock) - -/* lock for shutdown */ -#define GST_RTP_BIN_SHUTDOWN_LOCK(bin,label) \ -G_STMT_START { \ - if (g_atomic_int_get (&bin->priv->shutdown)) \ - goto label; \ - GST_RTP_BIN_DYN_LOCK (bin); \ - if (g_atomic_int_get (&bin->priv->shutdown)) { \ - GST_RTP_BIN_DYN_UNLOCK (bin); \ - goto label; \ - } \ -} G_STMT_END - -/* unlock for shutdown */ -#define GST_RTP_BIN_SHUTDOWN_UNLOCK(bin) \ - GST_RTP_BIN_DYN_UNLOCK (bin); \ - -struct _GstRtpBinPrivate -{ - GMutex *bin_lock; - - /* lock protecting dynamic adding/removing */ - GMutex *dyn_lock; - - /* the time when we went to playing */ - GstClockTime ntp_ns_base; - - /* if we are shutting down or not */ - gint shutdown; -}; - -/* signals and args */ -enum -{ - SIGNAL_REQUEST_PT_MAP, - SIGNAL_CLEAR_PT_MAP, - SIGNAL_RESET_SYNC, - SIGNAL_GET_INTERNAL_SESSION, - - SIGNAL_ON_NEW_SSRC, - SIGNAL_ON_SSRC_COLLISION, - SIGNAL_ON_SSRC_VALIDATED, - SIGNAL_ON_SSRC_ACTIVE, - SIGNAL_ON_SSRC_SDES, - SIGNAL_ON_BYE_SSRC, - SIGNAL_ON_BYE_TIMEOUT, - SIGNAL_ON_TIMEOUT, - SIGNAL_ON_SENDER_TIMEOUT, - SIGNAL_ON_NPT_STOP, - LAST_SIGNAL -}; - -#define DEFAULT_LATENCY_MS 200 -#define DEFAULT_SDES NULL -#define DEFAULT_DO_LOST FALSE - -enum -{ - PROP_0, - PROP_LATENCY, - PROP_SDES, - PROP_DO_LOST, - PROP_LAST -}; - -/* helper objects */ -typedef struct _GstRtpBinSession GstRtpBinSession; -typedef struct _GstRtpBinStream GstRtpBinStream; -typedef struct _GstRtpBinClient GstRtpBinClient; - -static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 }; - -static GstCaps *pt_map_requested (GstElement * element, guint pt, - GstRtpBinSession * session); -static void free_stream (GstRtpBinStream * stream); - -/* Manages the RTP stream for one SSRC. - * - * We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer. - * If we see an SDES RTCP packet that links multiple SSRCs together based on a - * common CNAME, we create a GstRtpBinClient structure to group the SSRCs - * together (see below). - */ -struct _GstRtpBinStream -{ - /* the SSRC of this stream */ - guint32 ssrc; - - /* parent bin */ - GstRtpBin *bin; - - /* the session this SSRC belongs to */ - GstRtpBinSession *session; - - /* the jitterbuffer of the SSRC */ - GstElement *buffer; - gulong buffer_handlesync_sig; - gulong buffer_ptreq_sig; - gulong buffer_ntpstop_sig; - - /* the PT demuxer of the SSRC */ - GstElement *demux; - gulong demux_newpad_sig; - gulong demux_padremoved_sig; - gulong demux_ptreq_sig; - gulong demux_pt_change_sig; - - /* if we have calculated a valid unix_delta for this stream */ - gboolean have_sync; - /* mapping to local RTP and NTP time */ - gint64 unix_delta; -}; - -#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->lock) -#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->lock) - -/* Manages the receiving end of the packets. - * - * There is one such structure for each RTP session (audio/video/...). - * We get the RTP/RTCP packets and stuff them into the session manager. From - * there they are pushed into an SSRC demuxer that splits the stream based on - * SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with - * the GstRtpBinStream above). - */ -struct _GstRtpBinSession -{ - /* session id */ - gint id; - /* the parent bin */ - GstRtpBin *bin; - /* the session element */ - GstElement *session; - /* the SSRC demuxer */ - GstElement *demux; - gulong demux_newpad_sig; - gulong demux_padremoved_sig; - - GMutex *lock; - - /* list of GstRtpBinStream */ - GSList *streams; - - /* mapping of payload type to caps */ - GHashTable *ptmap; - - /* the pads of the session */ - GstPad *recv_rtp_sink; - GstPad *recv_rtp_sink_ghost; - GstPad *recv_rtp_src; - GstPad *recv_rtcp_sink; - GstPad *recv_rtcp_sink_ghost; - GstPad *sync_src; - GstPad *send_rtp_sink; - GstPad *send_rtp_sink_ghost; - GstPad *send_rtp_src; - GstPad *send_rtp_src_ghost; - GstPad *send_rtcp_src; - GstPad *send_rtcp_src_ghost; -}; - -/* Manages the RTP streams that come from one client and should therefore be - * synchronized. - */ -struct _GstRtpBinClient -{ - /* the common CNAME for the streams */ - gchar *cname; - guint cname_len; - - /* the streams */ - guint nstreams; - GSList *streams; -}; - -/* find a session with the given id. Must be called with RTP_BIN_LOCK */ -static GstRtpBinSession * -find_session_by_id (GstRtpBin * rtpbin, gint id) -{ - GSList *walk; - - for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) { - GstRtpBinSession *sess = (GstRtpBinSession *) walk->data; - - if (sess->id == id) - return sess; - } - return NULL; -} - -/* find a session with the given request pad. Must be called with RTP_BIN_LOCK */ -static GstRtpBinSession * -find_session_by_pad (GstRtpBin * rtpbin, GstPad * pad) -{ - GSList *walk; - - for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) { - GstRtpBinSession *sess = (GstRtpBinSession *) walk->data; - - if ((sess->recv_rtp_sink_ghost == pad) || - (sess->recv_rtcp_sink_ghost == pad) || - (sess->send_rtp_sink_ghost == pad) - || (sess->send_rtcp_src_ghost == pad)) - return sess; - } - return NULL; -} - -static void -on_new_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) -{ - g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC], 0, - sess->id, ssrc); -} - -static void -on_ssrc_collision (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) -{ - g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION], 0, - sess->id, ssrc); -} - -static void -on_ssrc_validated (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) -{ - g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED], 0, - sess->id, ssrc); -} - -static void -on_ssrc_active (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) -{ - g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE], 0, - sess->id, ssrc); -} - -static void -on_ssrc_sdes (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) -{ - g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES], 0, - sess->id, ssrc); -} - -static void -on_bye_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) -{ - g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC], 0, - sess->id, ssrc); -} - -static void -on_bye_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) -{ - g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT], 0, - sess->id, ssrc); -} - -static void -on_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) -{ - g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT], 0, - sess->id, ssrc); -} - -static void -on_sender_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) -{ - g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT], 0, - sess->id, ssrc); -} - -static void -on_npt_stop (GstElement * jbuf, GstRtpBinStream * stream) -{ - g_signal_emit (stream->bin, gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP], 0, - stream->session->id, stream->ssrc); -} - -/* must be called with the SESSION lock */ -static GstRtpBinStream * -find_stream_by_ssrc (GstRtpBinSession * session, guint32 ssrc) -{ - GSList *walk; - - for (walk = session->streams; walk; walk = g_slist_next (walk)) { - GstRtpBinStream *stream = (GstRtpBinStream *) walk->data; - - if (stream->ssrc == ssrc) - return stream; - } - return NULL; -} - -static void -ssrc_demux_pad_removed (GstElement * element, guint ssrc, GstPad * pad, - GstRtpBinSession * session) -{ - GstRtpBinStream *stream = NULL; - - GST_RTP_SESSION_LOCK (session); - if ((stream = find_stream_by_ssrc (session, ssrc))) - session->streams = g_slist_remove (session->streams, stream); - GST_RTP_SESSION_UNLOCK (session); - - if (stream) - free_stream (stream); -} - -/* create a session with the given id. Must be called with RTP_BIN_LOCK */ -static GstRtpBinSession * -create_session (GstRtpBin * rtpbin, gint id) -{ - GstRtpBinSession *sess; - GstElement *session, *demux; - GstState target; - - if (!(session = gst_element_factory_make ("gstrtpsession", NULL))) - goto no_session; - - if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL))) - goto no_demux; - - sess = g_new0 (GstRtpBinSession, 1); - sess->lock = g_mutex_new (); - sess->id = id; - sess->bin = rtpbin; - sess->session = session; - sess->demux = demux; - sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL, - (GDestroyNotify) gst_caps_unref); - rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess); - - /* set NTP base or new session */ - g_object_set (session, "ntp-ns-base", rtpbin->priv->ntp_ns_base, NULL); - /* configure SDES items */ - GST_OBJECT_LOCK (rtpbin); - g_object_set (session, "sdes", rtpbin->sdes, NULL); - GST_OBJECT_UNLOCK (rtpbin); - - /* provide clock_rate to the session manager when needed */ - g_signal_connect (session, "request-pt-map", - (GCallback) pt_map_requested, sess); - - g_signal_connect (sess->session, "on-new-ssrc", - (GCallback) on_new_ssrc, sess); - g_signal_connect (sess->session, "on-ssrc-collision", - (GCallback) on_ssrc_collision, sess); - g_signal_connect (sess->session, "on-ssrc-validated", - (GCallback) on_ssrc_validated, sess); - g_signal_connect (sess->session, "on-ssrc-active", - (GCallback) on_ssrc_active, sess); - g_signal_connect (sess->session, "on-ssrc-sdes", - (GCallback) on_ssrc_sdes, sess); - g_signal_connect (sess->session, "on-bye-ssrc", - (GCallback) on_bye_ssrc, sess); - g_signal_connect (sess->session, "on-bye-timeout", - (GCallback) on_bye_timeout, sess); - g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess); - g_signal_connect (sess->session, "on-sender-timeout", - (GCallback) on_sender_timeout, sess); - - gst_bin_add (GST_BIN_CAST (rtpbin), session); - gst_bin_add (GST_BIN_CAST (rtpbin), demux); - - GST_OBJECT_LOCK (rtpbin); - target = GST_STATE_TARGET (rtpbin); - GST_OBJECT_UNLOCK (rtpbin); - - /* change state only to what's needed */ - gst_element_set_state (demux, target); - gst_element_set_state (session, target); - - return sess; - - /* ERRORS */ -no_session: - { - g_warning ("gstrtpbin: could not create gstrtpsession element"); - return NULL; - } -no_demux: - { - gst_object_unref (session); - g_warning ("gstrtpbin: could not create gstrtpssrcdemux element"); - return NULL; - } -} - -static void -free_session (GstRtpBinSession * sess, GstRtpBin * bin) -{ - GST_DEBUG_OBJECT (bin, "freeing session %p", sess); - - gst_element_set_state (sess->demux, GST_STATE_NULL); - gst_element_set_state (sess->session, GST_STATE_NULL); - - if (sess->recv_rtp_sink != NULL) { - gst_element_release_request_pad (sess->session, sess->recv_rtp_sink); - gst_object_unref (sess->recv_rtp_sink); - } - if (sess->recv_rtp_src != NULL) - gst_object_unref (sess->recv_rtp_src); - if (sess->recv_rtcp_sink != NULL) { - gst_element_release_request_pad (sess->session, sess->recv_rtcp_sink); - gst_object_unref (sess->recv_rtcp_sink); - } - if (sess->sync_src != NULL) - gst_object_unref (sess->sync_src); - if (sess->send_rtp_sink != NULL) { - gst_element_release_request_pad (sess->session, sess->send_rtp_sink); - gst_object_unref (sess->send_rtp_sink); - } - if (sess->send_rtp_src != NULL) - gst_object_unref (sess->send_rtp_src); - if (sess->send_rtcp_src != NULL) { - gst_element_release_request_pad (sess->session, sess->send_rtcp_src); - gst_object_unref (sess->send_rtcp_src); - } - - gst_bin_remove (GST_BIN_CAST (bin), sess->session); - gst_bin_remove (GST_BIN_CAST (bin), sess->demux); - - g_slist_foreach (sess->streams, (GFunc) free_stream, NULL); - g_slist_free (sess->streams); - - g_mutex_free (sess->lock); - g_hash_table_destroy (sess->ptmap); - - g_free (sess); -} - -/* get the payload type caps for the specific payload @pt in @session */ -static GstCaps * -get_pt_map (GstRtpBinSession * session, guint pt) -{ - GstCaps *caps = NULL; - GstRtpBin *bin; - GValue ret = { 0 }; - GValue args[3] = { {0}, {0}, {0} }; - - GST_DEBUG ("searching pt %d in cache", pt); - - GST_RTP_SESSION_LOCK (session); - - /* first look in the cache */ - caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt)); - if (caps) { - gst_caps_ref (caps); - goto done; - } - - bin = session->bin; - - GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id); - - /* not in cache, send signal to request caps */ - g_value_init (&args[0], GST_TYPE_ELEMENT); - g_value_set_object (&args[0], bin); - g_value_init (&args[1], G_TYPE_UINT); - g_value_set_uint (&args[1], session->id); - g_value_init (&args[2], G_TYPE_UINT); - g_value_set_uint (&args[2], pt); - - g_value_init (&ret, GST_TYPE_CAPS); - g_value_set_boxed (&ret, NULL); - - GST_RTP_SESSION_UNLOCK (session); - - g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret); - - GST_RTP_SESSION_LOCK (session); - - g_value_unset (&args[0]); - g_value_unset (&args[1]); - g_value_unset (&args[2]); - - /* look in the cache again because we let the lock go */ - caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt)); - if (caps) { - gst_caps_ref (caps); - g_value_unset (&ret); - goto done; - } - - caps = (GstCaps *) g_value_dup_boxed (&ret); - g_value_unset (&ret); - if (!caps) - goto no_caps; - - GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps); - - /* store in cache, take additional ref */ - g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt), - gst_caps_ref (caps)); - -done: - GST_RTP_SESSION_UNLOCK (session); - - return caps; - - /* ERRORS */ -no_caps: - { - GST_RTP_SESSION_UNLOCK (session); - GST_DEBUG ("no pt map could be obtained"); - return NULL; - } -} - -static gboolean -return_true (gpointer key, gpointer value, gpointer user_data) -{ - return TRUE; -} - -static void -gst_rtp_bin_reset_sync (GstRtpBin * rtpbin) -{ - GSList *clients, *streams; - - GST_DEBUG_OBJECT (rtpbin, "Reset sync on all clients"); - - GST_RTP_BIN_LOCK (rtpbin); - for (clients = rtpbin->clients; clients; clients = g_slist_next (clients)) { - GstRtpBinClient *client = (GstRtpBinClient *) clients->data; - - /* reset sync on all streams for this client */ - for (streams = client->streams; streams; streams = g_slist_next (streams)) { - GstRtpBinStream *stream = (GstRtpBinStream *) streams->data; - - /* make use require a new SR packet for this stream before we attempt new - * lip-sync */ - stream->have_sync = FALSE; - stream->unix_delta = 0; - } - } - GST_RTP_BIN_UNLOCK (rtpbin); -} - -static void -gst_rtp_bin_clear_pt_map (GstRtpBin * bin) -{ - GSList *sessions, *streams; - - GST_RTP_BIN_LOCK (bin); - GST_DEBUG_OBJECT (bin, "clearing pt map"); - for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) { - GstRtpBinSession *session = (GstRtpBinSession *) sessions->data; - - GST_DEBUG_OBJECT (bin, "clearing session %p", session); - g_signal_emit_by_name (session->session, "clear-pt-map", NULL); - - GST_RTP_SESSION_LOCK (session); - g_hash_table_foreach_remove (session->ptmap, return_true, NULL); - - for (streams = session->streams; streams; streams = g_slist_next (streams)) { - GstRtpBinStream *stream = (GstRtpBinStream *) streams->data; - - GST_DEBUG_OBJECT (bin, "clearing stream %p", stream); - g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL); - g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL); - } - GST_RTP_SESSION_UNLOCK (session); - } - GST_RTP_BIN_UNLOCK (bin); - - /* reset sync too */ - gst_rtp_bin_reset_sync (bin); -} - -static RTPSession * -gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id) -{ - RTPSession *internal_session = NULL; - GstRtpBinSession *session; - - GST_RTP_BIN_LOCK (bin); - GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %d", - session_id); - session = find_session_by_id (bin, (gint) session_id); - if (session) { - g_object_get (session->session, "internal-session", &internal_session, - NULL); - } - GST_RTP_BIN_UNLOCK (bin); - - return internal_session; -} - -static void -gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin, - const gchar * name, const GValue * value) -{ - GSList *sessions, *streams; - - GST_RTP_BIN_LOCK (bin); - for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) { - GstRtpBinSession *session = (GstRtpBinSession *) sessions->data; - - GST_RTP_SESSION_LOCK (session); - for (streams = session->streams; streams; streams = g_slist_next (streams)) { - GstRtpBinStream *stream = (GstRtpBinStream *) streams->data; - - g_object_set_property (G_OBJECT (stream->buffer), name, value); - } - GST_RTP_SESSION_UNLOCK (session); - } - GST_RTP_BIN_UNLOCK (bin); -} - -/* get a client with the given SDES name. Must be called with RTP_BIN_LOCK */ -static GstRtpBinClient * -get_client (GstRtpBin * bin, guint8 len, guint8 * data, gboolean * created) -{ - GstRtpBinClient *result = NULL; - GSList *walk; - - for (walk = bin->clients; walk; walk = g_slist_next (walk)) { - GstRtpBinClient *client = (GstRtpBinClient *) walk->data; - - if (len != client->cname_len) - continue; - - if (!strncmp ((gchar *) data, client->cname, client->cname_len)) { - GST_DEBUG_OBJECT (bin, "found existing client %p with CNAME %s", client, - client->cname); - result = client; - break; - } - } - - /* nothing found, create one */ - if (result == NULL) { - result = g_new0 (GstRtpBinClient, 1); - result->cname = g_strndup ((gchar *) data, len); - result->cname_len = len; - bin->clients = g_slist_prepend (bin->clients, result); - GST_DEBUG_OBJECT (bin, "created new client %p with CNAME %s", result, - result->cname); - } - return result; -} - -static void -free_client (GstRtpBinClient * client, GstRtpBin * bin) -{ - GST_DEBUG_OBJECT (bin, "freeing client %p", client); - g_slist_free (client->streams); - g_free (client->cname); - g_free (client); -} - -/* associate a stream to the given CNAME. This will make sure all streams for - * that CNAME are synchronized together. - * Must be called with GST_RTP_BIN_LOCK */ -static void -gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len, - guint8 * data, guint64 last_unix, guint64 last_extrtptime, - guint64 clock_base, guint64 clock_base_time, guint clock_rate) -{ - GstRtpBinClient *client; - gboolean created; - GSList *walk; - guint64 local_unix; - guint64 local_rtp; - - /* first find or create the CNAME */ - client = get_client (bin, len, data, &created); - - /* find stream in the client */ - for (walk = client->streams; walk; walk = g_slist_next (walk)) { - GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data; - - if (ostream == stream) - break; - } - /* not found, add it to the list */ - if (walk == NULL) { - GST_DEBUG_OBJECT (bin, - "new association of SSRC %08x with client %p with CNAME %s", - stream->ssrc, client, client->cname); - client->streams = g_slist_prepend (client->streams, stream); - client->nstreams++; - } else { - GST_DEBUG_OBJECT (bin, - "found association of SSRC %08x with client %p with CNAME %s", - stream->ssrc, client, client->cname); - } - - /* take the extended rtptime we found in the SR packet and map it to the - * local rtptime. The local rtp time is used to construct timestamps on the - * buffers. */ - local_rtp = last_extrtptime - clock_base; - - GST_DEBUG_OBJECT (bin, - "base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT - ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d", clock_base, - last_extrtptime, local_rtp, clock_rate); - - /* calculate local NTP time in gstreamer timestamp, we essentially perform the - * same conversion that a jitterbuffer would use to convert an rtp timestamp - * into a corresponding gstreamer timestamp. */ - local_unix = gst_util_uint64_scale_int (local_rtp, GST_SECOND, clock_rate); - local_unix += clock_base_time; - - /* calculate delta between server and receiver. last_unix is created by - * converting the ntptime in the last SR packet to a gstreamer timestamp. This - * delta expresses the difference to our timeline and the server timeline. */ - stream->unix_delta = last_unix - local_unix; - stream->have_sync = TRUE; - - GST_DEBUG_OBJECT (bin, - "local UNIX %" G_GUINT64_FORMAT ", remote UNIX %" G_GUINT64_FORMAT - ", delta %" G_GINT64_FORMAT, local_unix, last_unix, stream->unix_delta); - - /* recalc inter stream playout offset, but only if there is more than one - * stream. */ - if (client->nstreams > 1) { - gint64 min; - - /* calculate the min of all deltas, ignoring streams that did not yet have a - * valid unix_delta because we did not yet receive an SR packet for those - * streams. - * We calculate the mininum because we would like to only apply positive - * offsets to streams, delaying their playback instead of trying to speed up - * other streams (which might be imposible when we have to create negative - * latencies). - * The stream that has the smallest diff is selected as the reference stream, - * all other streams will have a positive offset to this difference. */ - min = G_MAXINT64; - for (walk = client->streams; walk; walk = g_slist_next (walk)) { - GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data; - - if (!ostream->have_sync) - continue; - - if (ostream->unix_delta < min) - min = ostream->unix_delta; - } - - GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT, client, - min); - - /* calculate offsets for each stream */ - for (walk = client->streams; walk; walk = g_slist_next (walk)) { - GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data; - gint64 ts_offset, prev_ts_offset; - - /* ignore streams for which we didn't receive an SR packet yet, we - * can't synchronize them yet. We can however sync other streams just - * fine. */ - if (!ostream->have_sync) - continue; - - /* calculate offset to our reference stream, this should always give a - * positive number. */ - ts_offset = ostream->unix_delta - min; - - g_object_get (ostream->buffer, "ts-offset", &prev_ts_offset, NULL); - - /* delta changed, see how much */ - if (prev_ts_offset != ts_offset) { - gint64 diff; - - if (prev_ts_offset > ts_offset) - diff = prev_ts_offset - ts_offset; - else - diff = ts_offset - prev_ts_offset; - - GST_DEBUG_OBJECT (bin, - "ts-offset %" G_GUINT64_FORMAT ", prev %" G_GUINT64_FORMAT - ", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff); - - /* only change diff when it changed more than 4 milliseconds. This - * compensates for rounding errors in NTP to RTP timestamp - * conversions */ - if (diff > 4 * GST_MSECOND && diff < (3 * GST_SECOND)) { - g_object_set (ostream->buffer, "ts-offset", ts_offset, NULL); - } - } - GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT, - ostream->ssrc, ts_offset); - } - } - return; -} - -#define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \ - for ((b) = gst_rtcp_buffer_get_first_packet ((buffer), (packet)); (b); \ - (b) = gst_rtcp_packet_move_to_next ((packet))) - -#define GST_RTCP_SDES_FOR_ITEMS(b,packet) \ - for ((b) = gst_rtcp_packet_sdes_first_item ((packet)); (b); \ - (b) = gst_rtcp_packet_sdes_next_item ((packet))) - -#define GST_RTCP_SDES_FOR_ENTRIES(b,packet) \ - for ((b) = gst_rtcp_packet_sdes_first_entry ((packet)); (b); \ - (b) = gst_rtcp_packet_sdes_next_entry ((packet))) - -static void -gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s, - GstRtpBinStream * stream) -{ - GstRtpBin *bin; - GstRTCPPacket packet; - guint32 ssrc; - guint64 ntptime; - gboolean have_sr, have_sdes; - gboolean more; - guint64 clock_base; - guint64 clock_base_time; - guint clock_rate; - guint64 extrtptime; - GstBuffer *buffer; - - bin = stream->bin; - - GST_DEBUG_OBJECT (bin, "sync handler called"); - - /* get the last relation between the rtp timestamps and the gstreamer - * timestamps. We get this info directly from the jitterbuffer which - * constructs gstreamer timestamps from rtp timestamps and so it know exactly - * what the current situation is. */ - clock_base = g_value_get_uint64 (gst_structure_get_value (s, "base-rtptime")); - clock_base_time = - g_value_get_uint64 (gst_structure_get_value (s, "base-time")); - clock_rate = g_value_get_uint (gst_structure_get_value (s, "clock-rate")); - extrtptime = - g_value_get_uint64 (gst_structure_get_value (s, "sr-ext-rtptime")); - buffer = gst_value_get_buffer (gst_structure_get_value (s, "sr-buffer")); - - have_sr = FALSE; - have_sdes = FALSE; - GST_RTCP_BUFFER_FOR_PACKETS (more, buffer, &packet) { - /* first packet must be SR or RR or else the validate would have failed */ - switch (gst_rtcp_packet_get_type (&packet)) { - case GST_RTCP_TYPE_SR: - /* only parse first. There is only supposed to be one SR in the packet - * but we will deal with malformed packets gracefully */ - if (have_sr) - break; - /* get NTP and RTP times */ - gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, NULL, - NULL, NULL); - - GST_DEBUG_OBJECT (bin, "received sync packet from SSRC %08x", ssrc); - /* ignore SR that is not ours */ - if (ssrc != stream->ssrc) - continue; - - have_sr = TRUE; - break; - case GST_RTCP_TYPE_SDES: - { - gboolean more_items, more_entries; - - /* only deal with first SDES, there is only supposed to be one SDES in - * the RTCP packet but we deal with bad packets gracefully. Also bail - * out if we have not seen an SR item yet. */ - if (have_sdes || !have_sr) - break; - - GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) { - /* skip items that are not about the SSRC of the sender */ - if (gst_rtcp_packet_sdes_get_ssrc (&packet) != ssrc) - continue; - - /* find the CNAME entry */ - GST_RTCP_SDES_FOR_ENTRIES (more_entries, &packet) { - GstRTCPSDESType type; - guint8 len; - guint8 *data; - - gst_rtcp_packet_sdes_get_entry (&packet, &type, &len, &data); - - if (type == GST_RTCP_SDES_CNAME) { - GST_RTP_BIN_LOCK (bin); - /* associate the stream to CNAME */ - gst_rtp_bin_associate (bin, stream, len, data, - gst_rtcp_ntp_to_unix (ntptime), extrtptime, - clock_base, clock_base_time, clock_rate); - GST_RTP_BIN_UNLOCK (bin); - } - } - } - have_sdes = TRUE; - break; - } - default: - /* we can ignore these packets */ - break; - } - } -} - -/* create a new stream with @ssrc in @session. Must be called with - * RTP_SESSION_LOCK. */ -static GstRtpBinStream * -create_stream (GstRtpBinSession * session, guint32 ssrc) -{ - GstElement *buffer, *demux; - GstRtpBinStream *stream; - GstRtpBin *rtpbin; - GstState target; - - if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL))) - goto no_jitterbuffer; - - if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL))) - goto no_demux; - - rtpbin = session->bin; - - stream = g_new0 (GstRtpBinStream, 1); - stream->ssrc = ssrc; - stream->bin = rtpbin; - stream->session = session; - stream->buffer = buffer; - stream->demux = demux; - stream->have_sync = FALSE; - stream->unix_delta = 0; - session->streams = g_slist_prepend (session->streams, stream); - - /* provide clock_rate to the jitterbuffer when needed */ - stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map", - (GCallback) pt_map_requested, session); - stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop", - (GCallback) on_npt_stop, stream); - - /* configure latency and packet lost */ - g_object_set (buffer, "latency", rtpbin->latency, NULL); - g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL); - - gst_bin_add (GST_BIN_CAST (rtpbin), demux); - gst_bin_add (GST_BIN_CAST (rtpbin), buffer); - - /* link stuff */ - gst_element_link (buffer, demux); - - GST_OBJECT_LOCK (rtpbin); - target = GST_STATE_TARGET (rtpbin); - GST_OBJECT_UNLOCK (rtpbin); - - /* from sink to source */ - gst_element_set_state (demux, target); - gst_element_set_state (buffer, target); - - return stream; - - /* ERRORS */ -no_jitterbuffer: - { - g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element"); - return NULL; - } -no_demux: - { - gst_object_unref (buffer); - g_warning ("gstrtpbin: could not create gstrtpptdemux element"); - return NULL; - } -} - -static void -free_stream (GstRtpBinStream * stream) -{ - GstRtpBinSession *session; - - session = stream->session; - - g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig); - g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig); - g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig); - g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig); - g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig); - - gst_element_set_state (stream->demux, GST_STATE_NULL); - gst_element_set_state (stream->buffer, GST_STATE_NULL); - - /* now remove this signal, we need this while going to NULL because it to - * do some cleanups */ - g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig); - - gst_bin_remove (GST_BIN_CAST (session->bin), stream->buffer); - gst_bin_remove (GST_BIN_CAST (session->bin), stream->demux); - - g_free (stream); -} - -/* GObject vmethods */ -static void gst_rtp_bin_dispose (GObject * object); -static void gst_rtp_bin_finalize (GObject * object); -static void gst_rtp_bin_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_rtp_bin_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -/* GstElement vmethods */ -static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element, - GstStateChange transition); -static GstPad *gst_rtp_bin_request_new_pad (GstElement * element, - GstPadTemplate * templ, const gchar * name); -static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad); -static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message); -static void gst_rtp_bin_clear_pt_map (GstRtpBin * bin); - -GST_BOILERPLATE (GstRtpBin, gst_rtp_bin, GstBin, GST_TYPE_BIN); - -static void -gst_rtp_bin_base_init (gpointer klass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - - /* sink pads */ - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&rtpbin_send_rtp_sink_template)); - - /* src pads */ - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&rtpbin_recv_rtp_src_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&rtpbin_send_rtcp_src_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&rtpbin_send_rtp_src_template)); - - gst_element_class_set_details (element_class, &rtpbin_details); -} - -static void -gst_rtp_bin_class_init (GstRtpBinClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - GstBinClass *gstbin_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; - gstbin_class = (GstBinClass *) klass; - - g_type_class_add_private (klass, sizeof (GstRtpBinPrivate)); - - gobject_class->dispose = gst_rtp_bin_dispose; - gobject_class->finalize = gst_rtp_bin_finalize; - gobject_class->set_property = gst_rtp_bin_set_property; - gobject_class->get_property = gst_rtp_bin_get_property; - - g_object_class_install_property (gobject_class, PROP_LATENCY, - g_param_spec_uint ("latency", "Buffer latency in ms", - "Default amount of ms to buffer in the jitterbuffers", 0, - G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE)); - - /** - * GstRtpBin::request-pt-map: - * @rtpbin: the object which received the signal - * @session: the session - * @pt: the pt - * - * Request the payload type as #GstCaps for @pt in @session. - */ - gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] = - g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_pt_map), - NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT_UINT, GST_TYPE_CAPS, 2, - G_TYPE_UINT, G_TYPE_UINT); - /** - * GstRtpBin::clear-pt-map: - * @rtpbin: the object which received the signal - * - * Clear all previously cached pt-mapping obtained with - * #GstRtpBin::request-pt-map. - */ - gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] = - g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass, - clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, - 0, G_TYPE_NONE); - /** - * GstRtpBin::reset-sync: - * @rtpbin: the object which received the signal - * - * Reset all currently configured lip-sync parameters and require new SR - * packets for all streams before lip-sync is attempted again. - */ - gst_rtp_bin_signals[SIGNAL_RESET_SYNC] = - g_signal_new ("reset-sync", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass, - reset_sync), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, - 0, G_TYPE_NONE); - - /** - * GstRtpBin::get-internal-session: - * @rtpbin: the object which received the signal - * @id: the session id - * - * Request the internal RTPSession object as #GObject in session @id. - */ - gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_SESSION] = - g_signal_new ("get-internal-session", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass, - get_internal_session), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT, - RTP_TYPE_SESSION, 1, G_TYPE_UINT); - - /** - * GstRtpBin::on-new-ssrc: - * @rtpbin: the object which received the signal - * @session: the session - * @ssrc: the SSRC - * - * Notify of a new SSRC that entered @session. - */ - gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] = - g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc), - NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2, - G_TYPE_UINT, G_TYPE_UINT); - /** - * GstRtpBin::on-ssrc-collision: - * @rtpbin: the object which received the signal - * @session: the session - * @ssrc: the SSRC - * - * Notify when we have an SSRC collision - */ - gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] = - g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_collision), - NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2, - G_TYPE_UINT, G_TYPE_UINT); - /** - * GstRtpBin::on-ssrc-validated: - * @rtpbin: the object which received the signal - * @session: the session - * @ssrc: the SSRC - * - * Notify of a new SSRC that became validated. - */ - gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] = - g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_validated), - NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2, - G_TYPE_UINT, G_TYPE_UINT); - /** - * GstRtpBin::on-ssrc-active: - * @rtpbin: the object which received the signal - * @session: the session - * @ssrc: the SSRC - * - * Notify of a SSRC that is active, i.e., sending RTCP. - */ - gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE] = - g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_active), - NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2, - G_TYPE_UINT, G_TYPE_UINT); - /** - * GstRtpBin::on-ssrc-sdes: - * @rtpbin: the object which received the signal - * @session: the session - * @ssrc: the SSRC - * - * Notify of a SSRC that is active, i.e., sending RTCP. - */ - gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES] = - g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_sdes), - NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2, - G_TYPE_UINT, G_TYPE_UINT); - - /** - * GstRtpBin::on-bye-ssrc: - * @rtpbin: the object which received the signal - * @session: the session - * @ssrc: the SSRC - * - * Notify of an SSRC that became inactive because of a BYE packet. - */ - gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] = - g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_ssrc), - NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2, - G_TYPE_UINT, G_TYPE_UINT); - /** - * GstRtpBin::on-bye-timeout: - * @rtpbin: the object which received the signal - * @session: the session - * @ssrc: the SSRC - * - * Notify of an SSRC that has timed out because of BYE - */ - gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] = - g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_timeout), - NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2, - G_TYPE_UINT, G_TYPE_UINT); - /** - * GstRtpBin::on-timeout: - * @rtpbin: the object which received the signal - * @session: the session - * @ssrc: the SSRC - * - * Notify of an SSRC that has timed out - */ - gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] = - g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout), - NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2, - G_TYPE_UINT, G_TYPE_UINT); - /** - * GstRtpBin::on-sender-timeout: - * @rtpbin: the object which received the signal - * @session: the session - * @ssrc: the SSRC - * - * Notify of a sender SSRC that has timed out and became a receiver - */ - gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT] = - g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_sender_timeout), - NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2, - G_TYPE_UINT, G_TYPE_UINT); - - /** - * GstRtpBin::on-npt-stop: - * @rtpbin: the object which received the signal - * @session: the session - * @ssrc: the SSRC - * - * Notify that SSRC sender has sent data up to the configured NPT stop time. - */ - gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP] = - g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass), - G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_npt_stop), - NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2, - G_TYPE_UINT, G_TYPE_UINT); - - g_object_class_install_property (gobject_class, PROP_SDES, - g_param_spec_boxed ("sdes", "SDES", - "The SDES items of this session", - GST_TYPE_STRUCTURE, G_PARAM_READWRITE)); - - g_object_class_install_property (gobject_class, PROP_DO_LOST, - g_param_spec_boolean ("do-lost", "Do Lost", - "Send an event downstream when a packet is lost", DEFAULT_DO_LOST, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state); - gstelement_class->request_new_pad = - GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad); - gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad); - - gstbin_class->handle_message = GST_DEBUG_FUNCPTR (gst_rtp_bin_handle_message); - - klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map); - klass->reset_sync = GST_DEBUG_FUNCPTR (gst_rtp_bin_reset_sync); - klass->get_internal_session = - GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session); - - GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin"); -} - -static void -gst_rtp_bin_init (GstRtpBin * rtpbin, GstRtpBinClass * klass) -{ - gchar *str; - - rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin); - rtpbin->priv->bin_lock = g_mutex_new (); - rtpbin->priv->dyn_lock = g_mutex_new (); - - rtpbin->latency = DEFAULT_LATENCY_MS; - rtpbin->do_lost = DEFAULT_DO_LOST; - - /* some default SDES entries */ - str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ()); - rtpbin->sdes = gst_structure_new ("application/x-rtp-source-sdes", - "cname", G_TYPE_STRING, str, - "name", G_TYPE_STRING, g_get_real_name (), - "tool", G_TYPE_STRING, "GStreamer", NULL); - g_free (str); -} - -static void -gst_rtp_bin_dispose (GObject * object) -{ - GstRtpBin *rtpbin; - - rtpbin = GST_RTP_BIN (object); - - GST_DEBUG_OBJECT (object, "freeing sessions"); - g_slist_foreach (rtpbin->sessions, (GFunc) free_session, rtpbin); - g_slist_free (rtpbin->sessions); - rtpbin->sessions = NULL; - GST_DEBUG_OBJECT (object, "freeing clients"); - g_slist_foreach (rtpbin->clients, (GFunc) free_client, rtpbin); - g_slist_free (rtpbin->clients); - rtpbin->clients = NULL; - - G_OBJECT_CLASS (parent_class)->dispose (object); -} - -static void -gst_rtp_bin_finalize (GObject * object) -{ - GstRtpBin *rtpbin; - - rtpbin = GST_RTP_BIN (object); - - if (rtpbin->sdes) - gst_structure_free (rtpbin->sdes); - - g_mutex_free (rtpbin->priv->bin_lock); - g_mutex_free (rtpbin->priv->dyn_lock); - - G_OBJECT_CLASS (parent_class)->finalize (object); -} - - -static void -gst_rtp_bin_set_sdes_struct (GstRtpBin * bin, const GstStructure * sdes) -{ - GSList *item; - - if (sdes == NULL) - return; - - GST_RTP_BIN_LOCK (bin); - - GST_OBJECT_LOCK (bin); - if (bin->sdes) - gst_structure_free (bin->sdes); - bin->sdes = gst_structure_copy (sdes); - - /* store in all sessions */ - for (item = bin->sessions; item; item = g_slist_next (item)) - g_object_set (item->data, "sdes", sdes, NULL); - GST_OBJECT_UNLOCK (bin); - - GST_RTP_BIN_UNLOCK (bin); -} - -static GstStructure * -gst_rtp_bin_get_sdes_struct (GstRtpBin * bin) -{ - GstStructure *result; - - GST_OBJECT_LOCK (bin); - result = gst_structure_copy (bin->sdes); - GST_OBJECT_UNLOCK (bin); - - return result; -} - -static void -gst_rtp_bin_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstRtpBin *rtpbin; - - rtpbin = GST_RTP_BIN (object); - - switch (prop_id) { - case PROP_LATENCY: - GST_RTP_BIN_LOCK (rtpbin); - rtpbin->latency = g_value_get_uint (value); - GST_RTP_BIN_UNLOCK (rtpbin); - /* propegate the property down to the jitterbuffer */ - gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value); - break; - case PROP_SDES: - gst_rtp_bin_set_sdes_struct (rtpbin, g_value_get_boxed (value)); - break; - case PROP_DO_LOST: - GST_RTP_BIN_LOCK (rtpbin); - rtpbin->do_lost = g_value_get_boolean (value); - GST_RTP_BIN_UNLOCK (rtpbin); - gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "do-lost", value); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_rtp_bin_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstRtpBin *rtpbin; - - rtpbin = GST_RTP_BIN (object); - - switch (prop_id) { - case PROP_LATENCY: - GST_RTP_BIN_LOCK (rtpbin); - g_value_set_uint (value, rtpbin->latency); - GST_RTP_BIN_UNLOCK (rtpbin); - break; - case PROP_SDES: - g_value_take_boxed (value, gst_rtp_bin_get_sdes_struct (rtpbin)); - break; - case PROP_DO_LOST: - GST_RTP_BIN_LOCK (rtpbin); - g_value_set_boolean (value, rtpbin->do_lost); - GST_RTP_BIN_UNLOCK (rtpbin); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message) -{ - GstRtpBin *rtpbin; - - rtpbin = GST_RTP_BIN (bin); - - switch (GST_MESSAGE_TYPE (message)) { - case GST_MESSAGE_ELEMENT: - { - const GstStructure *s = gst_message_get_structure (message); - - /* we change the structure name and add the session ID to it */ - if (gst_structure_has_name (s, "application/x-rtp-source-sdes")) { - GSList *walk; - - /* find the session, the message source has it */ - for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) { - GstRtpBinSession *sess = (GstRtpBinSession *) walk->data; - - /* if we found the session, change message. else we exit the loop and - * leave the message unchanged */ - if (GST_OBJECT_CAST (sess->session) == GST_MESSAGE_SRC (message)) { - message = gst_message_make_writable (message); - s = gst_message_get_structure (message); - - gst_structure_set ((GstStructure *) s, "session", G_TYPE_UINT, - sess->id, NULL); - break; - } - } - } - /* fallthrough to forward the modified message to the parent */ - } - default: - { - GST_BIN_CLASS (parent_class)->handle_message (bin, message); - break; - } - } -} - -static void -calc_ntp_ns_base (GstRtpBin * bin) -{ - GstClockTime now; - GTimeVal current; - GSList *walk; - - /* get the current time and convert it to NTP time in nanoseconds */ - g_get_current_time (¤t); - now = GST_TIMEVAL_TO_TIME (current); - now += (2208988800LL * GST_SECOND); - - GST_RTP_BIN_LOCK (bin); - bin->priv->ntp_ns_base = now; - for (walk = bin->sessions; walk; walk = g_slist_next (walk)) { - GstRtpBinSession *session = (GstRtpBinSession *) walk->data; - - g_object_set (session->session, "ntp-ns-base", now, NULL); - } - GST_RTP_BIN_UNLOCK (bin); - - return; -} - -static GstStateChangeReturn -gst_rtp_bin_change_state (GstElement * element, GstStateChange transition) -{ - GstStateChangeReturn res; - GstRtpBin *rtpbin; - GstRtpBinPrivate *priv; - - rtpbin = GST_RTP_BIN (element); - priv = rtpbin->priv; - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - break; - case GST_STATE_CHANGE_READY_TO_PAUSED: - GST_LOG_OBJECT (rtpbin, "clearing shutdown flag"); - g_atomic_int_set (&priv->shutdown, 0); - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: - calc_ntp_ns_base (rtpbin); - break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - GST_LOG_OBJECT (rtpbin, "setting shutdown flag"); - g_atomic_int_set (&priv->shutdown, 1); - /* wait for all callbacks to end by taking the lock. No new callbacks will - * be able to happen as we set the shutdown flag. */ - GST_RTP_BIN_DYN_LOCK (rtpbin); - GST_LOG_OBJECT (rtpbin, "dynamic lock taken, we can continue shutdown"); - GST_RTP_BIN_DYN_UNLOCK (rtpbin); - break; - default: - break; - } - - res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - switch (transition) { - case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - break; - case GST_STATE_CHANGE_READY_TO_NULL: - break; - default: - break; - } - return res; -} - -/* a new pad (SSRC) was created in @session. This signal is emited from the - * payload demuxer. */ -static void -new_payload_found (GstElement * element, guint pt, GstPad * pad, - GstRtpBinStream * stream) -{ - GstRtpBin *rtpbin; - GstElementClass *klass; - GstPadTemplate *templ; - gchar *padname; - GstPad *gpad; - - rtpbin = stream->bin; - - GST_DEBUG ("new payload pad %d", pt); - - GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown); - - /* ghost the pad to the parent */ - klass = GST_ELEMENT_GET_CLASS (rtpbin); - templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%d_%d_%d"); - padname = g_strdup_printf ("recv_rtp_src_%d_%u_%d", - stream->session->id, stream->ssrc, pt); - gpad = gst_ghost_pad_new_from_template (padname, pad, templ); - g_free (padname); - g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", gpad); - - gst_pad_set_caps (gpad, GST_PAD_CAPS (pad)); - gst_pad_set_active (gpad, TRUE); - gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad); - GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin); - - return; - -shutdown: - { - GST_DEBUG ("ignoring, we are shutting down"); - return; - } -} - -static void -payload_pad_removed (GstElement * element, GstPad * pad, - GstRtpBinStream * stream) -{ - GstRtpBin *rtpbin; - GstPad *gpad; - - rtpbin = stream->bin; - - GST_DEBUG ("payload pad removed"); - - GST_RTP_BIN_DYN_LOCK (rtpbin); - if ((gpad = g_object_get_data (G_OBJECT (pad), "GstRTPBin.ghostpad"))) { - g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", NULL); - - gst_pad_set_active (gpad, FALSE); - gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), gpad); - } - GST_RTP_BIN_DYN_UNLOCK (rtpbin); -} - -static GstCaps * -pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session) -{ - GstRtpBin *rtpbin; - GstCaps *caps; - - rtpbin = session->bin; - - GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt, - session->id); - - caps = get_pt_map (session, pt); - if (!caps) - goto no_caps; - - return caps; - - /* ERRORS */ -no_caps: - { - GST_DEBUG_OBJECT (rtpbin, "could not get caps"); - return NULL; - } -} - -/* emited when caps changed for the session */ -static void -caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session) -{ - GstRtpBin *bin; - GstCaps *caps; - gint payload; - const GstStructure *s; - - bin = session->bin; - - g_object_get (pad, "caps", &caps, NULL); - - if (caps == NULL) - return; - - GST_DEBUG_OBJECT (bin, "got caps %" GST_PTR_FORMAT, caps); - - s = gst_caps_get_structure (caps, 0); - - /* get payload, finish when it's not there */ - if (!gst_structure_get_int (s, "payload", &payload)) - return; - - GST_RTP_SESSION_LOCK (session); - GST_DEBUG_OBJECT (bin, "insert caps for payload %d", payload); - g_hash_table_insert (session->ptmap, GINT_TO_POINTER (payload), caps); - GST_RTP_SESSION_UNLOCK (session); -} - -/* a new pad (SSRC) was created in @session */ -static void -new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad, - GstRtpBinSession * session) -{ - GstRtpBin *rtpbin; - GstRtpBinStream *stream; - GstPad *sinkpad, *srcpad; - gchar *padname; - - rtpbin = session->bin; - - GST_DEBUG_OBJECT (rtpbin, "new SSRC pad %08x, %s:%s", ssrc, - GST_DEBUG_PAD_NAME (pad)); - - GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown); - - GST_RTP_SESSION_LOCK (session); - - /* create new stream */ - stream = create_stream (session, ssrc); - if (!stream) - goto no_stream; - - /* get pad and link */ - GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTP"); - padname = g_strdup_printf ("src_%d", ssrc); - srcpad = gst_element_get_static_pad (element, padname); - g_free (padname); - sinkpad = gst_element_get_static_pad (stream->buffer, "sink"); - gst_pad_link (srcpad, sinkpad); - gst_object_unref (sinkpad); - gst_object_unref (srcpad); - - GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP"); - padname = g_strdup_printf ("rtcp_src_%d", ssrc); - srcpad = gst_element_get_static_pad (element, padname); - g_free (padname); - sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp"); - gst_pad_link (srcpad, sinkpad); - gst_object_unref (sinkpad); - gst_object_unref (srcpad); - - /* connect to the RTCP sync signal from the jitterbuffer */ - GST_DEBUG_OBJECT (rtpbin, "connecting sync signal"); - stream->buffer_handlesync_sig = g_signal_connect (stream->buffer, - "handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream); - - /* connect to the new-pad signal of the payload demuxer, this will expose the - * new pad by ghosting it. */ - stream->demux_newpad_sig = g_signal_connect (stream->demux, - "new-payload-type", (GCallback) new_payload_found, stream); - stream->demux_padremoved_sig = g_signal_connect (stream->demux, - "pad-removed", (GCallback) payload_pad_removed, stream); - - /* connect to the request-pt-map signal. This signal will be emited by the - * demuxer so that it can apply a proper caps on the buffers for the - * depayloaders. */ - stream->demux_ptreq_sig = g_signal_connect (stream->demux, - "request-pt-map", (GCallback) pt_map_requested, session); - - GST_RTP_SESSION_UNLOCK (session); - GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin); - - return; - - /* ERRORS */ -shutdown: - { - GST_DEBUG_OBJECT (rtpbin, "we are shutting down"); - return; - } -no_stream: - { - GST_RTP_SESSION_UNLOCK (session); - GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin); - GST_DEBUG_OBJECT (rtpbin, "could not create stream"); - return; - } -} - -/* Create a pad for receiving RTP for the session in @name. Must be called with - * RTP_BIN_LOCK. - */ -static GstPad * -create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name) -{ - GstPad *sinkdpad; - guint sessid; - GstRtpBinSession *session; - GstPadLinkReturn lres; - - /* first get the session number */ - if (name == NULL || sscanf (name, "recv_rtp_sink_%d", &sessid) != 1) - goto no_name; - - GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid); - - /* get or create session */ - session = find_session_by_id (rtpbin, sessid); - if (!session) { - GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid); - /* create session now */ - session = create_session (rtpbin, sessid); - if (session == NULL) - goto create_error; - } - - /* check if pad was requested */ - if (session->recv_rtp_sink_ghost != NULL) - return session->recv_rtp_sink_ghost; - - GST_DEBUG_OBJECT (rtpbin, "getting RTP sink pad"); - /* get recv_rtp pad and store */ - session->recv_rtp_sink = - gst_element_get_request_pad (session->session, "recv_rtp_sink"); - if (session->recv_rtp_sink == NULL) - goto pad_failed; - - g_signal_connect (session->recv_rtp_sink, "notify::caps", - (GCallback) caps_changed, session); - - GST_DEBUG_OBJECT (rtpbin, "getting RTP src pad"); - /* get srcpad, link to SSRCDemux */ - session->recv_rtp_src = - gst_element_get_static_pad (session->session, "recv_rtp_src"); - if (session->recv_rtp_src == NULL) - goto pad_failed; - - GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad"); - sinkdpad = gst_element_get_static_pad (session->demux, "sink"); - GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad"); - lres = gst_pad_link (session->recv_rtp_src, sinkdpad); - gst_object_unref (sinkdpad); - if (lres != GST_PAD_LINK_OK) - goto link_failed; - - /* connect to the new-ssrc-pad signal of the SSRC demuxer */ - session->demux_newpad_sig = g_signal_connect (session->demux, - "new-ssrc-pad", (GCallback) new_ssrc_pad_found, session); - session->demux_padremoved_sig = g_signal_connect (session->demux, - "removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session); - - GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad"); - session->recv_rtp_sink_ghost = - gst_ghost_pad_new_from_template (name, session->recv_rtp_sink, templ); - gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE); - gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost); - - return session->recv_rtp_sink_ghost; - - /* ERRORS */ -no_name: - { - g_warning ("gstrtpbin: invalid name given"); - return NULL; - } -create_error: - { - /* create_session already warned */ - return NULL; - } -pad_failed: - { - g_warning ("gstrtpbin: failed to get session pad"); - return NULL; - } -link_failed: - { - g_warning ("gstrtpbin: failed to link pads"); - return NULL; - } -} - -static void -remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session) -{ - if (session->demux_newpad_sig) { - g_signal_handler_disconnect (session->demux, session->demux_newpad_sig); - session->demux_newpad_sig = 0; - } - if (session->demux_padremoved_sig) { - g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig); - session->demux_padremoved_sig = 0; - } - if (session->recv_rtp_src) { - gst_object_unref (session->recv_rtp_src); - session->recv_rtp_src = NULL; - } - if (session->recv_rtp_sink) { - gst_element_release_request_pad (session->session, session->recv_rtp_sink); - gst_object_unref (session->recv_rtp_sink); - session->recv_rtp_sink = NULL; - } - if (session->recv_rtp_sink_ghost) { - gst_pad_set_active (session->recv_rtp_sink_ghost, FALSE); - gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), - session->recv_rtp_sink_ghost); - session->recv_rtp_sink_ghost = NULL; - } -} - -/* Create a pad for receiving RTCP for the session in @name. Must be called with - * RTP_BIN_LOCK. - */ -static GstPad * -create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, - const gchar * name) -{ - guint sessid; - GstRtpBinSession *session; - GstPad *sinkdpad; - GstPadLinkReturn lres; - - /* first get the session number */ - if (name == NULL || sscanf (name, "recv_rtcp_sink_%d", &sessid) != 1) - goto no_name; - - GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid); - - /* get or create the session */ - session = find_session_by_id (rtpbin, sessid); - if (!session) { - GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid); - /* create session now */ - session = create_session (rtpbin, sessid); - if (session == NULL) - goto create_error; - } - - /* check if pad was requested */ - if (session->recv_rtcp_sink_ghost != NULL) - return session->recv_rtcp_sink_ghost; - - /* get recv_rtp pad and store */ - GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad"); - session->recv_rtcp_sink = - gst_element_get_request_pad (session->session, "recv_rtcp_sink"); - if (session->recv_rtcp_sink == NULL) - goto pad_failed; - - /* get srcpad, link to SSRCDemux */ - GST_DEBUG_OBJECT (rtpbin, "getting sync src pad"); - session->sync_src = gst_element_get_static_pad (session->session, "sync_src"); - if (session->sync_src == NULL) - goto pad_failed; - - GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad"); - sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink"); - lres = gst_pad_link (session->sync_src, sinkdpad); - gst_object_unref (sinkdpad); - if (lres != GST_PAD_LINK_OK) - goto link_failed; - - session->recv_rtcp_sink_ghost = - gst_ghost_pad_new_from_template (name, session->recv_rtcp_sink, templ); - gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE); - gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), - session->recv_rtcp_sink_ghost); - - return session->recv_rtcp_sink_ghost; - - /* ERRORS */ -no_name: - { - g_warning ("gstrtpbin: invalid name given"); - return NULL; - } -create_error: - { - /* create_session already warned */ - return NULL; - } -pad_failed: - { - g_warning ("gstrtpbin: failed to get session pad"); - return NULL; - } -link_failed: - { - g_warning ("gstrtpbin: failed to link pads"); - return NULL; - } -} - -static void -remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session) -{ - if (session->recv_rtcp_sink_ghost) { - gst_pad_set_active (session->recv_rtcp_sink_ghost, FALSE); - gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), - session->recv_rtcp_sink_ghost); - session->recv_rtcp_sink_ghost = NULL; - } - if (session->sync_src) { - /* releasing the request pad should also unref the sync pad */ - gst_object_unref (session->sync_src); - session->sync_src = NULL; - } - if (session->recv_rtcp_sink) { - gst_element_release_request_pad (session->session, session->recv_rtcp_sink); - gst_object_unref (session->recv_rtcp_sink); - session->recv_rtcp_sink = NULL; - } -} - -/* Create a pad for sending RTP for the session in @name. Must be called with - * RTP_BIN_LOCK. - */ -static GstPad * -create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name) -{ - gchar *gname; - guint sessid; - GstRtpBinSession *session; - GstElementClass *klass; - - /* first get the session number */ - if (name == NULL || sscanf (name, "send_rtp_sink_%d", &sessid) != 1) - goto no_name; - - /* get or create session */ - session = find_session_by_id (rtpbin, sessid); - if (!session) { - /* create session now */ - session = create_session (rtpbin, sessid); - if (session == NULL) - goto create_error; - } - - /* check if pad was requested */ - if (session->send_rtp_sink_ghost != NULL) - return session->send_rtp_sink_ghost; - - /* get send_rtp pad and store */ - session->send_rtp_sink = - gst_element_get_request_pad (session->session, "send_rtp_sink"); - if (session->send_rtp_sink == NULL) - goto pad_failed; - - session->send_rtp_sink_ghost = - gst_ghost_pad_new_from_template (name, session->send_rtp_sink, templ); - gst_pad_set_active (session->send_rtp_sink_ghost, TRUE); - gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_sink_ghost); - - /* get srcpad */ - session->send_rtp_src = - gst_element_get_static_pad (session->session, "send_rtp_src"); - if (session->send_rtp_src == NULL) - goto no_srcpad; - - /* ghost the new source pad */ - klass = GST_ELEMENT_GET_CLASS (rtpbin); - gname = g_strdup_printf ("send_rtp_src_%d", sessid); - templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%d"); - session->send_rtp_src_ghost = - gst_ghost_pad_new_from_template (gname, session->send_rtp_src, templ); - gst_pad_set_active (session->send_rtp_src_ghost, TRUE); - gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost); - g_free (gname); - - return session->send_rtp_sink_ghost; - - /* ERRORS */ -no_name: - { - g_warning ("gstrtpbin: invalid name given"); - return NULL; - } -create_error: - { - /* create_session already warned */ - return NULL; - } -pad_failed: - { - g_warning ("gstrtpbin: failed to get session pad for session %d", sessid); - return NULL; - } -no_srcpad: - { - g_warning ("gstrtpbin: failed to get rtp source pad for session %d", - sessid); - return NULL; - } -} - -static void -remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session) -{ - if (session->send_rtp_src_ghost) { - gst_pad_set_active (session->send_rtp_src_ghost, FALSE); - gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), - session->send_rtp_src_ghost); - session->send_rtp_src_ghost = NULL; - } - if (session->send_rtp_src) { - gst_object_unref (session->send_rtp_src); - session->send_rtp_src = NULL; - } - if (session->send_rtp_sink) { - gst_element_release_request_pad (GST_ELEMENT_CAST (session->session), - session->send_rtp_sink); - gst_object_unref (session->send_rtp_sink); - session->send_rtp_sink = NULL; - } - if (session->send_rtp_sink_ghost) { - gst_pad_set_active (session->send_rtp_sink_ghost, FALSE); - gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), - session->send_rtp_sink_ghost); - session->send_rtp_sink_ghost = NULL; - } -} - -/* Create a pad for sending RTCP for the session in @name. Must be called with - * RTP_BIN_LOCK. - */ -static GstPad * -create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name) -{ - guint sessid; - GstRtpBinSession *session; - - /* first get the session number */ - if (name == NULL || sscanf (name, "send_rtcp_src_%d", &sessid) != 1) - goto no_name; - - /* get or create session */ - session = find_session_by_id (rtpbin, sessid); - if (!session) - goto no_session; - - /* check if pad was requested */ - if (session->send_rtcp_src_ghost != NULL) - return session->send_rtcp_src_ghost; - - /* get rtcp_src pad and store */ - session->send_rtcp_src = - gst_element_get_request_pad (session->session, "send_rtcp_src"); - if (session->send_rtcp_src == NULL) - goto pad_failed; - - session->send_rtcp_src_ghost = - gst_ghost_pad_new_from_template (name, session->send_rtcp_src, templ); - gst_pad_set_active (session->send_rtcp_src_ghost, TRUE); - gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtcp_src_ghost); - - return session->send_rtcp_src_ghost; - - /* ERRORS */ -no_name: - { - g_warning ("gstrtpbin: invalid name given"); - return NULL; - } -no_session: - { - g_warning ("gstrtpbin: session with id %d does not exist", sessid); - return NULL; - } -pad_failed: - { - g_warning ("gstrtpbin: failed to get rtcp pad for session %d", sessid); - return NULL; - } -} - -static void -remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session) -{ - if (session->send_rtcp_src_ghost) { - gst_pad_set_active (session->send_rtcp_src_ghost, FALSE); - gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), - session->send_rtcp_src_ghost); - session->send_rtcp_src_ghost = NULL; - } - if (session->send_rtcp_src) { - gst_element_release_request_pad (session->session, session->send_rtcp_src); - gst_object_unref (session->send_rtcp_src); - session->send_rtcp_src = NULL; - } -} - -/* If the requested name is NULL we should create a name with - * the session number assuming we want the lowest posible session - * with a free pad like the template */ -static gchar * -gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ) -{ - gboolean name_found = FALSE; - gint session = 0; - GstPad *pad = NULL; - GstIterator *pad_it = NULL; - gchar *pad_name = NULL; - - GST_DEBUG_OBJECT (element, "find a free pad name for template"); - while (!name_found) { - g_free (pad_name); - pad_name = g_strdup_printf (templ->name_template, session++); - pad_it = gst_element_iterate_pads (GST_ELEMENT (element)); - name_found = TRUE; - while (gst_iterator_next (pad_it, (gpointer) & pad) == GST_ITERATOR_OK) { - gchar *name; - - name = gst_pad_get_name (pad); - if (strcmp (name, pad_name) == 0) - name_found = FALSE; - g_free (name); - } - gst_iterator_free (pad_it); - } - - GST_DEBUG_OBJECT (element, "free pad name found: '%s'", pad_name); - return pad_name; -} - -/* - */ -static GstPad * -gst_rtp_bin_request_new_pad (GstElement * element, - GstPadTemplate * templ, const gchar * name) -{ - GstRtpBin *rtpbin; - GstElementClass *klass; - GstPad *result; - - gchar *pad_name = NULL; - - g_return_val_if_fail (templ != NULL, NULL); - g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL); - - rtpbin = GST_RTP_BIN (element); - klass = GST_ELEMENT_GET_CLASS (element); - - GST_RTP_BIN_LOCK (rtpbin); - - if (name == NULL) { - /* use a free pad name */ - pad_name = gst_rtp_bin_get_free_pad_name (element, templ); - } else { - /* use the provided name */ - pad_name = g_strdup (name); - } - - GST_DEBUG_OBJECT (rtpbin, "Trying to request a pad with name %s", pad_name); - - /* figure out the template */ - if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%d")) { - result = create_recv_rtp (rtpbin, templ, pad_name); - } else if (templ == gst_element_class_get_pad_template (klass, - "recv_rtcp_sink_%d")) { - result = create_recv_rtcp (rtpbin, templ, pad_name); - } else if (templ == gst_element_class_get_pad_template (klass, - "send_rtp_sink_%d")) { - result = create_send_rtp (rtpbin, templ, pad_name); - } else if (templ == gst_element_class_get_pad_template (klass, - "send_rtcp_src_%d")) { - result = create_rtcp (rtpbin, templ, pad_name); - } else - goto wrong_template; - - g_free (pad_name); - GST_RTP_BIN_UNLOCK (rtpbin); - - return result; - - /* ERRORS */ -wrong_template: - { - g_free (pad_name); - GST_RTP_BIN_UNLOCK (rtpbin); - g_warning ("gstrtpbin: this is not our template"); - return NULL; - } -} - -static void -gst_rtp_bin_release_pad (GstElement * element, GstPad * pad) -{ - GstRtpBinSession *session; - GstRtpBin *rtpbin; - - g_return_if_fail (GST_IS_GHOST_PAD (pad)); - g_return_if_fail (GST_IS_RTP_BIN (element)); - - rtpbin = GST_RTP_BIN (element); - - GST_RTP_BIN_LOCK (rtpbin); - GST_DEBUG_OBJECT (rtpbin, "Trying to release pad %s:%s", - GST_DEBUG_PAD_NAME (pad)); - - if (!(session = find_session_by_pad (rtpbin, pad))) - goto unknown_pad; - - if (session->recv_rtp_sink_ghost == pad) { - remove_recv_rtp (rtpbin, session); - } else if (session->recv_rtcp_sink_ghost == pad) { - remove_recv_rtcp (rtpbin, session); - } else if (session->send_rtp_sink_ghost == pad) { - remove_send_rtp (rtpbin, session); - } else if (session->send_rtcp_src_ghost == pad) { - remove_rtcp (rtpbin, session); - } - - /* no more request pads, free the complete session */ - if (session->recv_rtp_sink_ghost == NULL - && session->recv_rtcp_sink_ghost == NULL - && session->send_rtp_sink_ghost == NULL - && session->send_rtcp_src_ghost == NULL) { - GST_DEBUG_OBJECT (rtpbin, "no more pads for session %p", session); - rtpbin->sessions = g_slist_remove (rtpbin->sessions, session); - free_session (session, rtpbin); - } - GST_RTP_BIN_UNLOCK (rtpbin); - - return; - - /* ERROR */ -unknown_pad: - { - GST_RTP_BIN_UNLOCK (rtpbin); - g_warning ("gstrtpbin: %s:%s is not one of our request pads", - GST_DEBUG_PAD_NAME (pad)); - return; - } -} |