/* GStreamer * Copyright (C) 2008 Wim Taymans * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #include #include #include #include #include #include "rtsp-stream.h" enum { PROP_0, PROP_LAST }; GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug); #define GST_CAT_DEFAULT rtsp_stream_debug static GQuark ssrc_stream_map_key; static void gst_rtsp_stream_finalize (GObject * obj); G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT); static void gst_rtsp_stream_class_init (GstRTSPStreamClass * klass) { GObjectClass *gobject_class; gobject_class = G_OBJECT_CLASS (klass); gobject_class->finalize = gst_rtsp_stream_finalize; GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream"); ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream"); } static void gst_rtsp_stream_init (GstRTSPStream * media) { } static void gst_rtsp_stream_finalize (GObject * obj) { GstRTSPStream *stream; stream = GST_RTSP_STREAM (obj); /* we really need to be unjoined now */ g_return_if_fail (!stream->is_joined); gst_object_unref (stream->payloader); gst_object_unref (stream->srcpad); G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj); } /** * gst_rtsp_stream_new: * @idx: an index * @srcpad: a #GstPad * @payloader: a #GstElement * * Create a new media stream with index @idx that handles RTP data on * @srcpad and has a payloader element @payloader. * * Returns: a new #GstRTSPStream */ GstRTSPStream * gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad) { GstRTSPStream *stream; g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL); g_return_val_if_fail (GST_IS_PAD (srcpad), NULL); g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL); stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL); stream->idx = idx; stream->payloader = gst_object_ref (payloader); stream->srcpad = gst_object_ref (srcpad); return stream; } /** * gst_rtsp_stream_set_mtu: * @stream: a #GstRTSPStream * @mtu: a new MTU * * Configure the mtu in the payloader of @stream to @mtu. */ void gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu) { g_return_if_fail (GST_IS_RTSP_STREAM (stream)); g_object_set (G_OBJECT (stream->payloader), "mtu", mtu, NULL); } /** * gst_rtsp_stream_get_mtu: * @stream: a #GstRTSPStream * * Get the configured MTU in the payloader of @stream. * * Returns: the MTU of the payloader. */ guint gst_rtsp_stream_get_mtu (GstRTSPStream * stream) { guint mtu; g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0); g_object_get (G_OBJECT (stream->payloader), "mtu", &mtu, NULL); return mtu; } static gboolean alloc_ports (GstRTSPStream * stream) { GstStateChangeReturn ret; GstElement *udpsrc0, *udpsrc1; GstElement *udpsink0, *udpsink1; gint tmp_rtp, tmp_rtcp; guint count; gint rtpport, rtcpport; GSocket *socket; const gchar *host; g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE); udpsrc0 = NULL; udpsrc1 = NULL; udpsink0 = NULL; udpsink1 = NULL; count = 0; /* Start with random port */ tmp_rtp = 0; if (stream->is_ipv6) host = "udp://[::0]"; else host = "udp://0.0.0.0"; /* try to allocate 2 UDP ports, the RTP port should be an even * number and the RTCP port should be the next (uneven) port */ again: udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL); if (udpsrc0 == NULL) goto no_udp_protocol; g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL); ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED); if (ret == GST_STATE_CHANGE_FAILURE) { if (tmp_rtp != 0) { tmp_rtp += 2; if (++count > 20) goto no_ports; gst_element_set_state (udpsrc0, GST_STATE_NULL); gst_object_unref (udpsrc0); goto again; } goto no_udp_protocol; } g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL); /* check if port is even */ if ((tmp_rtp & 1) != 0) { /* port not even, close and allocate another */ if (++count > 20) goto no_ports; gst_element_set_state (udpsrc0, GST_STATE_NULL); gst_object_unref (udpsrc0); tmp_rtp++; goto again; } /* allocate port+1 for RTCP now */ udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL); if (udpsrc1 == NULL) goto no_udp_rtcp_protocol; /* set port */ tmp_rtcp = tmp_rtp + 1; g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL); ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED); /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */ if (ret == GST_STATE_CHANGE_FAILURE) { if (++count > 20) goto no_ports; gst_element_set_state (udpsrc0, GST_STATE_NULL); gst_object_unref (udpsrc0); gst_element_set_state (udpsrc1, GST_STATE_NULL); gst_object_unref (udpsrc1); tmp_rtp += 2; goto again; } /* all fine, do port check */ g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL); g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL); /* this should not happen... */ if (rtpport != tmp_rtp || rtcpport != tmp_rtcp) goto port_error; udpsink0 = gst_element_factory_make ("multiudpsink", NULL); if (!udpsink0) goto no_udp_protocol; g_object_get (G_OBJECT (udpsrc0), "socket", &socket, NULL); g_object_set (G_OBJECT (udpsink0), "socket", socket, NULL); g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL); udpsink1 = gst_element_factory_make ("multiudpsink", NULL); if (!udpsink1) goto no_udp_protocol; if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0), "send-duplicates")) { g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL); g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL); } else { g_warning ("old multiudpsink version found without send-duplicates property"); } if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0), "buffer-size")) { g_object_set (G_OBJECT (udpsink0), "buffer-size", stream->buffer_size, NULL); } else { GST_WARNING ("multiudpsink version found without buffer-size property"); } g_object_get (G_OBJECT (udpsrc1), "socket", &socket, NULL); g_object_set (G_OBJECT (udpsink1), "socket", socket, NULL); g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL); g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL); g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL); g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL); g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL); g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL); g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL); /* we keep these elements, we will further configure them when the * client told us to really use the UDP ports. */ stream->udpsrc[0] = udpsrc0; stream->udpsrc[1] = udpsrc1; stream->udpsink[0] = udpsink0; stream->udpsink[1] = udpsink1; stream->server_port.min = rtpport; stream->server_port.max = rtcpport; return TRUE; /* ERRORS */ no_udp_protocol: { goto cleanup; } no_ports: { goto cleanup; } no_udp_rtcp_protocol: { goto cleanup; } port_error: { goto cleanup; } cleanup: { if (udpsrc0) { gst_element_set_state (udpsrc0, GST_STATE_NULL); gst_object_unref (udpsrc0); } if (udpsrc1) { gst_element_set_state (udpsrc1, GST_STATE_NULL); gst_object_unref (udpsrc1); } if (udpsink0) { gst_element_set_state (udpsink0, GST_STATE_NULL); gst_object_unref (udpsink0); } if (udpsink1) { gst_element_set_state (udpsink1, GST_STATE_NULL); gst_object_unref (udpsink1); } return FALSE; } } /* executed from streaming thread */ static void caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream) { GstCaps *newcaps, *oldcaps; newcaps = gst_pad_get_current_caps (pad); oldcaps = stream->caps; stream->caps = newcaps; if (oldcaps) gst_caps_unref (oldcaps); GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps, newcaps); } static void dump_structure (const GstStructure * s) { gchar *sstr; sstr = gst_structure_to_string (s); GST_INFO ("structure: %s", sstr); g_free (sstr); } static GstRTSPStreamTransport * find_transport (GstRTSPStream * stream, const gchar * rtcp_from) { GList *walk; GstRTSPStreamTransport *result = NULL; const gchar *tmp; gchar *dest; guint port; if (rtcp_from == NULL) return NULL; tmp = g_strrstr (rtcp_from, ":"); if (tmp == NULL) return NULL; port = atoi (tmp + 1); dest = g_strndup (rtcp_from, tmp - rtcp_from); GST_INFO ("finding %s:%d in %d transports", dest, port, g_list_length (stream->transports)); for (walk = stream->transports; walk; walk = g_list_next (walk)) { GstRTSPStreamTransport *trans = walk->data; gint min, max; min = trans->transport->client_port.min; max = trans->transport->client_port.max; if ((strcmp (trans->transport->destination, dest) == 0) && (min == port || max == port)) { result = trans; break; } } g_free (dest); return result; } static GstRTSPStreamTransport * check_transport (GObject * source, GstRTSPStream * stream) { GstStructure *stats; GstRTSPStreamTransport *trans; /* see if we have a stream to match with the origin of the RTCP packet */ trans = g_object_get_qdata (source, ssrc_stream_map_key); if (trans == NULL) { g_object_get (source, "stats", &stats, NULL); if (stats) { const gchar *rtcp_from; dump_structure (stats); rtcp_from = gst_structure_get_string (stats, "rtcp-from"); if ((trans = find_transport (stream, rtcp_from))) { GST_INFO ("%p: found transport %p for source %p", stream, trans, source); /* keep ref to the source */ trans->rtpsource = source; g_object_set_qdata (source, ssrc_stream_map_key, trans); } gst_structure_free (stats); } } return trans; } static void on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream) { GstRTSPStreamTransport *trans; GST_INFO ("%p: new source %p", stream, source); trans = check_transport (source, stream); if (trans) GST_INFO ("%p: source %p for transport %p", stream, source, trans); } static void on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream) { GST_INFO ("%p: new SDES %p", stream, source); } static void on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream) { GstRTSPStreamTransport *trans; trans = check_transport (source, stream); if (trans) GST_INFO ("%p: source %p in transport %p is active", stream, source, trans); if (trans && trans->keep_alive) trans->keep_alive (trans->ka_user_data); #ifdef DUMP_STATS { GstStructure *stats; g_object_get (source, "stats", &stats, NULL); if (stats) { dump_structure (stats); gst_structure_free (stats); } } #endif } static void on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream) { GST_INFO ("%p: source %p bye", stream, source); } static void on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream) { GstRTSPStreamTransport *trans; GST_INFO ("%p: source %p bye timeout", stream, source); if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) { trans->rtpsource = NULL; trans->timeout = TRUE; } } static void on_timeout (GObject * session, GObject * source, GstRTSPStream * stream) { GstRTSPStreamTransport *trans; GST_INFO ("%p: source %p timeout", stream, source); if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) { trans->rtpsource = NULL; trans->timeout = TRUE; } } static GstFlowReturn handle_new_sample (GstAppSink * sink, gpointer user_data) { GList *walk; GstSample *sample; GstBuffer *buffer; GstRTSPStream *stream; sample = gst_app_sink_pull_sample (sink); if (!sample) return GST_FLOW_OK; stream = (GstRTSPStream *) user_data; buffer = gst_sample_get_buffer (sample); for (walk = stream->transports; walk; walk = g_list_next (walk)) { GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data; if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) { if (tr->send_rtp) tr->send_rtp (buffer, tr->transport->interleaved.min, tr->user_data); } else { if (tr->send_rtcp) tr->send_rtcp (buffer, tr->transport->interleaved.max, tr->user_data); } } gst_sample_unref (sample); return GST_FLOW_OK; } static GstAppSinkCallbacks sink_cb = { NULL, /* not interested in EOS */ NULL, /* not interested in preroll samples */ handle_new_sample, }; /** * gst_rtsp_stream_join_bin: * @stream: a #GstRTSPStream * @bin: a #GstBin to join * @rtpbin: a rtpbin element in @bin * @state: the target state of the new elements * * Join the #Gstbin @bin that contains the element @rtpbin. * * @stream will link to @rtpbin, which must be inside @bin. The elements * added to @bin will be set to the state given in @state. * * Returns: %TRUE on success. */ gboolean gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin, GstElement * rtpbin, GstState state) { gint i, idx; gchar *name; GstPad *pad, *teepad, *queuepad, *selpad; GstPadLinkReturn ret; g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE); g_return_val_if_fail (GST_IS_BIN (bin), FALSE); g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE); if (stream->is_joined) return TRUE; /* create a session with the same index as the stream */ idx = stream->idx; GST_INFO ("stream %p joining bin as session %d", stream, idx); if (!alloc_ports (stream)) goto no_ports; /* get a pad for sending RTP */ name = g_strdup_printf ("send_rtp_sink_%u", idx); stream->send_rtp_sink = gst_element_get_request_pad (rtpbin, name); g_free (name); /* link the RTP pad to the session manager, it should not really fail unless * this is not really an RTP pad */ ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink); if (ret != GST_PAD_LINK_OK) goto link_failed; /* get pads from the RTP session element for sending and receiving * RTP/RTCP*/ name = g_strdup_printf ("send_rtp_src_%u", idx); stream->send_src[0] = gst_element_get_static_pad (rtpbin, name); g_free (name); name = g_strdup_printf ("send_rtcp_src_%u", idx); stream->send_src[1] = gst_element_get_request_pad (rtpbin, name); g_free (name); name = g_strdup_printf ("recv_rtp_sink_%u", idx); stream->recv_sink[0] = gst_element_get_request_pad (rtpbin, name); g_free (name); name = g_strdup_printf ("recv_rtcp_sink_%u", idx); stream->recv_sink[1] = gst_element_get_request_pad (rtpbin, name); g_free (name); /* get the session */ g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &stream->session); g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc, stream); g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes, stream); g_signal_connect (stream->session, "on-ssrc-active", (GCallback) on_ssrc_active, stream); g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc, stream); g_signal_connect (stream->session, "on-bye-timeout", (GCallback) on_bye_timeout, stream); g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout, stream); for (i = 0; i < 2; i++) { /* For the sender we create this bit of pipeline for both * RTP and RTCP. Sync and preroll are enabled on udpsink so * we need to add a queue before appsink to make the pipeline * not block. For the TCP case, we want to pump data to the * client as fast as possible anyway. * * .--------. .-----. .---------. * | rtpbin | | tee | | udpsink | * | send->sink src->sink | * '--------' | | '---------' * | | .---------. .---------. * | | | queue | | appsink | * | src->sink src->sink | * '-----' '---------' '---------' */ /* make tee for RTP/RTCP */ stream->tee[i] = gst_element_factory_make ("tee", NULL); gst_bin_add (bin, stream->tee[i]); /* and link to rtpbin send pad */ pad = gst_element_get_static_pad (stream->tee[i], "sink"); gst_pad_link (stream->send_src[i], pad); gst_object_unref (pad); /* add udpsink */ gst_bin_add (bin, stream->udpsink[i]); /* link tee to udpsink */ teepad = gst_element_get_request_pad (stream->tee[i], "src_%u"); pad = gst_element_get_static_pad (stream->udpsink[i], "sink"); gst_pad_link (teepad, pad); gst_object_unref (pad); gst_object_unref (teepad); /* make queue */ stream->appqueue[i] = gst_element_factory_make ("queue", NULL); gst_bin_add (bin, stream->appqueue[i]); /* and link to tee */ teepad = gst_element_get_request_pad (stream->tee[i], "src_%u"); pad = gst_element_get_static_pad (stream->appqueue[i], "sink"); gst_pad_link (teepad, pad); gst_object_unref (pad); gst_object_unref (teepad); /* make appsink */ stream->appsink[i] = gst_element_factory_make ("appsink", NULL); g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL); g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL); gst_bin_add (bin, stream->appsink[i]); gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]), &sink_cb, stream, NULL); /* and link to queue */ queuepad = gst_element_get_static_pad (stream->appqueue[i], "src"); pad = gst_element_get_static_pad (stream->appsink[i], "sink"); gst_pad_link (queuepad, pad); gst_object_unref (pad); gst_object_unref (queuepad); /* For the receiver we create this bit of pipeline for both * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc * and it is all funneled into the rtpbin receive pad. * * .--------. .--------. .--------. * | udpsrc | | funnel | | rtpbin | * | src->sink src->sink | * '--------' | | '--------' * .--------. | | * | appsrc | | | * | src->sink | * '--------' '--------' */ /* make funnel for the RTP/RTCP receivers */ stream->funnel[i] = gst_element_factory_make ("funnel", NULL); gst_bin_add (bin, stream->funnel[i]); pad = gst_element_get_static_pad (stream->funnel[i], "src"); gst_pad_link (pad, stream->recv_sink[i]); gst_object_unref (pad); /* add udpsrc */ gst_bin_add (bin, stream->udpsrc[i]); /* and link to the funnel */ selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u"); pad = gst_element_get_static_pad (stream->udpsrc[i], "src"); gst_pad_link (pad, selpad); gst_object_unref (pad); gst_object_unref (selpad); /* make and add appsrc */ stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL); gst_bin_add (bin, stream->appsrc[i]); /* and link to the funnel */ selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u"); pad = gst_element_get_static_pad (stream->appsrc[i], "src"); gst_pad_link (pad, selpad); gst_object_unref (pad); gst_object_unref (selpad); /* check if we need to set to a special state */ if (state != GST_STATE_NULL) { gst_element_set_state (stream->udpsink[i], state); gst_element_set_state (stream->appsink[i], state); gst_element_set_state (stream->appqueue[i], state); gst_element_set_state (stream->tee[i], state); gst_element_set_state (stream->funnel[i], state); gst_element_set_state (stream->appsrc[i], state); } /* we set and keep these to playing so that they don't cause NO_PREROLL return * values */ gst_element_set_state (stream->udpsrc[i], GST_STATE_PLAYING); gst_element_set_locked_state (stream->udpsrc[i], TRUE); } /* be notified of caps changes */ stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps", (GCallback) caps_notify, stream); stream->is_joined = TRUE; return TRUE; /* ERRORS */ no_ports: { GST_WARNING ("failed to allocate ports %d", idx); return FALSE; } link_failed: { GST_WARNING ("failed to link stream %d", idx); gst_object_unref (stream->send_rtp_sink); stream->send_rtp_sink = NULL; return FALSE; } } /** * gst_rtsp_stream_leave_bin: * @stream: a #GstRTSPStream * @bin: a #GstBin * @rtpbin: a rtpbin #GstElement * * Remove the elements of @stream from @bin. @bin must be set * to the NULL state before calling this. * * Return: %TRUE on success. */ gboolean gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin, GstElement * rtpbin) { gint i; g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE); g_return_val_if_fail (GST_IS_BIN (bin), FALSE); g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE); if (!stream->is_joined) return TRUE; /* all transports must be removed by now */ g_return_val_if_fail (stream->transports == NULL, FALSE); GST_INFO ("stream %p leaving bin", stream); gst_pad_unlink (stream->srcpad, stream->send_rtp_sink); g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig); gst_element_release_request_pad (rtpbin, stream->send_rtp_sink); gst_object_unref (stream->send_rtp_sink); stream->send_rtp_sink = NULL; for (i = 0; i < 2; i++) { /* and set udpsrc to NULL now before removing */ gst_element_set_locked_state (stream->udpsrc[i], FALSE); gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL); /* removing them should also nicely release the request * pads when they finalize */ gst_bin_remove (bin, stream->udpsrc[i]); gst_bin_remove (bin, stream->udpsink[i]); gst_bin_remove (bin, stream->appsrc[i]); gst_bin_remove (bin, stream->appsink[i]); gst_bin_remove (bin, stream->appqueue[i]); gst_bin_remove (bin, stream->tee[i]); gst_bin_remove (bin, stream->funnel[i]); gst_element_release_request_pad (rtpbin, stream->recv_sink[i]); gst_object_unref (stream->recv_sink[i]); stream->recv_sink[i] = NULL; stream->udpsrc[i] = NULL; stream->udpsink[i] = NULL; stream->appsrc[i] = NULL; stream->appsink[i] = NULL; stream->appqueue[i] = NULL; stream->tee[i] = NULL; stream->funnel[i] = NULL; } gst_object_unref (stream->send_src[0]); stream->send_src[0] = NULL; gst_element_release_request_pad (rtpbin, stream->send_src[1]); gst_object_unref (stream->send_src[1]); stream->send_src[1] = NULL; g_object_unref (stream->session); if (stream->caps) gst_caps_unref (stream->caps); stream->is_joined = FALSE; return TRUE; } /** * gst_rtsp_stream_get_rtpinfo: * @stream: a #GstRTSPStream * @rtptime: result RTP timestamp * @seq: result RTP seqnum * * Retrieve the current rtptime and seq. This is used to * construct a RTPInfo reply header. * * Returns: %TRUE when rtptime and seq could be determined. */ gboolean gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream, guint * rtptime, guint * seq) { GObjectClass *payobjclass; payobjclass = G_OBJECT_GET_CLASS (stream->payloader); if (!g_object_class_find_property (payobjclass, "seqnum") || !g_object_class_find_property (payobjclass, "timestamp")) return FALSE; g_object_get (stream->payloader, "seqnum", seq, "timestamp", rtptime, NULL); return TRUE; } /** * gst_rtsp_stream_recv_rtp: * @stream: a #GstRTSPStream * @buffer: (transfer full): a #GstBuffer * * Handle an RTP buffer for the stream. This method is usually called when a * message has been received from a client using the TCP transport. * * This function takes ownership of @buffer. * * Returns: a GstFlowReturn. */ GstFlowReturn gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer) { GstFlowReturn ret; g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR); g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR); g_return_val_if_fail (stream->is_joined, FALSE); ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[0]), buffer); return ret; } /** * gst_rtsp_stream_recv_rtcp: * @stream: a #GstRTSPStream * @buffer: (transfer full): a #GstBuffer * * Handle an RTCP buffer for the stream. This method is usually called when a * message has been received from a client using the TCP transport. * * This function takes ownership of @buffer. * * Returns: a GstFlowReturn. */ GstFlowReturn gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer) { GstFlowReturn ret; g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR); g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR); g_return_val_if_fail (stream->is_joined, FALSE); ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[1]), buffer); return ret; } static gboolean update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans, gboolean add) { GstRTSPTransport *tr; gboolean updated; updated = FALSE; tr = trans->transport; switch (tr->lower_transport) { case GST_RTSP_LOWER_TRANS_UDP: case GST_RTSP_LOWER_TRANS_UDP_MCAST: { gchar *dest; gint min, max; guint ttl = 0; dest = tr->destination; if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) { min = tr->port.min; max = tr->port.max; ttl = tr->ttl; } else { min = tr->client_port.min; max = tr->client_port.max; } if (add && !trans->active) { GST_INFO ("adding %s:%d-%d", dest, min, max); g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL); g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL); if (ttl > 0) { GST_INFO ("setting ttl-mc %d", ttl); g_object_set (G_OBJECT (stream->udpsink[0]), "ttl-mc", ttl, NULL); g_object_set (G_OBJECT (stream->udpsink[1]), "ttl-mc", ttl, NULL); } stream->transports = g_list_prepend (stream->transports, trans); trans->active = TRUE; updated = TRUE; } else if (trans->active) { GST_INFO ("removing %s:%d-%d", dest, min, max); g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL); g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL); stream->transports = g_list_remove (stream->transports, trans); trans->active = FALSE; updated = TRUE; } break; } case GST_RTSP_LOWER_TRANS_TCP: if (add && !trans->active) { GST_INFO ("adding TCP %s", tr->destination); stream->transports = g_list_prepend (stream->transports, trans); trans->active = TRUE; updated = TRUE; } else if (trans->active) { GST_INFO ("removing TCP %s", tr->destination); stream->transports = g_list_remove (stream->transports, trans); trans->active = FALSE; updated = TRUE; } break; default: GST_INFO ("Unknown transport %d", tr->lower_transport); break; } return updated; } /** * gst_rtsp_stream_add_transport: * @stream: a #GstRTSPStream * @trans: a #GstRTSPStreamTransport * * Add the transport in @trans to @stream. The media of @stream will * then also be send to the values configured in @trans. * * @stream must be joined to a bin. * * @trans must contain a valid #GstRTSPTransport. * * Returns: %TRUE if @trans was added */ gboolean gst_rtsp_stream_add_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans) { g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE); g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE); g_return_val_if_fail (stream->is_joined, FALSE); g_return_val_if_fail (trans->transport != NULL, FALSE); return update_transport (stream, trans, TRUE); } /** * gst_rtsp_stream_remove_transport: * @stream: a #GstRTSPStream * @trans: a #GstRTSPStreamTransport * * Remove the transport in @trans from @stream. The media of @stream will * not be sent to the values configured in @trans. * * @stream must be joined to a bin. * * @trans must contain a valid #GstRTSPTransport. * * Returns: %TRUE if @trans was removed */ gboolean gst_rtsp_stream_remove_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans) { g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE); g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE); g_return_val_if_fail (stream->is_joined, FALSE); g_return_val_if_fail (trans->transport != NULL, FALSE); return update_transport (stream, trans, FALSE); }