/* 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #include #include #include #include #include "rtsp-media.h" #define DEFAULT_SHARED FALSE #define DEFAULT_REUSABLE FALSE #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP //#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP_MCAST #define DEFAULT_EOS_SHUTDOWN FALSE #define DEFAULT_BUFFER_SIZE 0x80000 /* define to dump received RTCP packets */ #undef DUMP_STATS enum { PROP_0, PROP_SHARED, PROP_REUSABLE, PROP_PROTOCOLS, PROP_EOS_SHUTDOWN, PROP_BUFFER_SIZE, PROP_LAST }; enum { SIGNAL_PREPARED, SIGNAL_UNPREPARED, SIGNAL_NEW_STATE, SIGNAL_LAST }; GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug); #define GST_CAT_DEFAULT rtsp_media_debug static GQuark ssrc_stream_map_key; static void gst_rtsp_media_get_property (GObject * object, guint propid, GValue * value, GParamSpec * pspec); static void gst_rtsp_media_set_property (GObject * object, guint propid, const GValue * value, GParamSpec * pspec); static void gst_rtsp_media_finalize (GObject * obj); static gpointer do_loop (GstRTSPMediaClass * klass); static gboolean default_handle_message (GstRTSPMedia * media, GstMessage * message); static gboolean default_unprepare (GstRTSPMedia * media); static void unlock_streams (GstRTSPMedia * media); static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 }; G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT); static void gst_rtsp_media_class_init (GstRTSPMediaClass * klass) { GObjectClass *gobject_class; GError *error = NULL; gobject_class = G_OBJECT_CLASS (klass); gobject_class->get_property = gst_rtsp_media_get_property; gobject_class->set_property = gst_rtsp_media_set_property; gobject_class->finalize = gst_rtsp_media_finalize; g_object_class_install_property (gobject_class, PROP_SHARED, g_param_spec_boolean ("shared", "Shared", "If this media pipeline can be shared", DEFAULT_SHARED, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_REUSABLE, g_param_spec_boolean ("reusable", "Reusable", "If this media pipeline can be reused after an unprepare", DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PROTOCOLS, g_param_spec_flags ("protocols", "Protocols", "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS, DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN, g_param_spec_boolean ("eos-shutdown", "EOS Shutdown", "Send an EOS event to the pipeline before unpreparing", DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE, g_param_spec_uint ("buffer-size", "Buffer Size", "The kernel UDP buffer size to use", 0, G_MAXUINT, DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_rtsp_media_signals[SIGNAL_PREPARED] = g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE); gst_rtsp_media_signals[SIGNAL_UNPREPARED] = g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE); gst_rtsp_media_signals[SIGNAL_NEW_STATE] = g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL, g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT); klass->context = g_main_context_new (); klass->loop = g_main_loop_new (klass->context, TRUE); GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia"); klass->thread = g_thread_create ((GThreadFunc) do_loop, klass, TRUE, &error); if (error != NULL) { g_critical ("could not start bus thread: %s", error->message); } klass->handle_message = default_handle_message; klass->unprepare = default_unprepare; ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream"); } static void gst_rtsp_media_init (GstRTSPMedia * media) { media->streams = g_array_new (FALSE, TRUE, sizeof (GstRTSPMediaStream *)); media->lock = g_mutex_new (); media->cond = g_cond_new (); media->shared = DEFAULT_SHARED; media->reusable = DEFAULT_REUSABLE; media->protocols = DEFAULT_PROTOCOLS; media->eos_shutdown = DEFAULT_EOS_SHUTDOWN; media->buffer_size = DEFAULT_BUFFER_SIZE; } void gst_rtsp_media_trans_cleanup (GstRTSPMediaTrans * trans) { if (trans->transport) { gst_rtsp_transport_free (trans->transport); trans->transport = NULL; } if (trans->rtpsource) { g_object_set_qdata (trans->rtpsource, ssrc_stream_map_key, NULL); trans->rtpsource = NULL; } } static void gst_rtsp_media_stream_free (GstRTSPMediaStream * stream) { if (stream->session) g_object_unref (stream->session); if (stream->caps) gst_caps_unref (stream->caps); if (stream->send_rtp_sink) gst_object_unref (stream->send_rtp_sink); if (stream->send_rtp_src) gst_object_unref (stream->send_rtp_src); if (stream->send_rtcp_src) gst_object_unref (stream->send_rtcp_src); if (stream->recv_rtcp_sink) gst_object_unref (stream->recv_rtcp_sink); if (stream->recv_rtp_sink) gst_object_unref (stream->recv_rtp_sink); g_list_free (stream->transports); g_free (stream); } static void gst_rtsp_media_finalize (GObject * obj) { GstRTSPMedia *media; guint i; media = GST_RTSP_MEDIA (obj); GST_INFO ("finalize media %p", media); if (media->pipeline) { unlock_streams (media); gst_element_set_state (media->pipeline, GST_STATE_NULL); gst_object_unref (media->pipeline); } for (i = 0; i < media->streams->len; i++) { GstRTSPMediaStream *stream; stream = g_array_index (media->streams, GstRTSPMediaStream *, i); gst_rtsp_media_stream_free (stream); } g_array_free (media->streams, TRUE); g_list_foreach (media->dynamic, (GFunc) gst_object_unref, NULL); g_list_free (media->dynamic); if (media->source) { g_source_destroy (media->source); g_source_unref (media->source); } g_mutex_free (media->lock); g_cond_free (media->cond); G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj); } static void gst_rtsp_media_get_property (GObject * object, guint propid, GValue * value, GParamSpec * pspec) { GstRTSPMedia *media = GST_RTSP_MEDIA (object); switch (propid) { case PROP_SHARED: g_value_set_boolean (value, gst_rtsp_media_is_shared (media)); break; case PROP_REUSABLE: g_value_set_boolean (value, gst_rtsp_media_is_reusable (media)); break; case PROP_PROTOCOLS: g_value_set_flags (value, gst_rtsp_media_get_protocols (media)); break; case PROP_EOS_SHUTDOWN: g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media)); break; case PROP_BUFFER_SIZE: g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec); } } static void gst_rtsp_media_set_property (GObject * object, guint propid, const GValue * value, GParamSpec * pspec) { GstRTSPMedia *media = GST_RTSP_MEDIA (object); switch (propid) { case PROP_SHARED: gst_rtsp_media_set_shared (media, g_value_get_boolean (value)); break; case PROP_REUSABLE: gst_rtsp_media_set_reusable (media, g_value_get_boolean (value)); break; case PROP_PROTOCOLS: gst_rtsp_media_set_protocols (media, g_value_get_flags (value)); break; case PROP_EOS_SHUTDOWN: gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value)); break; case PROP_BUFFER_SIZE: gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec); } } static gpointer do_loop (GstRTSPMediaClass * klass) { GST_INFO ("enter mainloop"); g_main_loop_run (klass->loop); GST_INFO ("exit mainloop"); return NULL; } static void collect_media_stats (GstRTSPMedia * media) { gint64 position, duration; media->range.unit = GST_RTSP_RANGE_NPT; if (media->is_live) { media->range.min.type = GST_RTSP_TIME_NOW; media->range.min.seconds = -1; media->range.max.type = GST_RTSP_TIME_END; media->range.max.seconds = -1; } else { /* get the position */ if (!gst_element_query_position (media->pipeline, GST_FORMAT_TIME, &position)) { GST_INFO ("position query failed"); position = 0; } /* get the duration */ if (!gst_element_query_duration (media->pipeline, GST_FORMAT_TIME, &duration)) { GST_INFO ("duration query failed"); duration = -1; } GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration)); if (position == -1) { media->range.min.type = GST_RTSP_TIME_NOW; media->range.min.seconds = -1; } else { media->range.min.type = GST_RTSP_TIME_SECONDS; media->range.min.seconds = ((gdouble) position) / GST_SECOND; } if (duration == -1) { media->range.max.type = GST_RTSP_TIME_END; media->range.max.seconds = -1; } else { media->range.max.type = GST_RTSP_TIME_SECONDS; media->range.max.seconds = ((gdouble) duration) / GST_SECOND; } } } /** * gst_rtsp_media_new: * * Create a new #GstRTSPMedia instance. The #GstRTSPMedia object contains the * element to produde RTP data for one or more related (audio/video/..) * streams. * * Returns: a new #GstRTSPMedia object. */ GstRTSPMedia * gst_rtsp_media_new (void) { GstRTSPMedia *result; result = g_object_new (GST_TYPE_RTSP_MEDIA, NULL); return result; } /** * gst_rtsp_media_set_shared: * @media: a #GstRTSPMedia * @shared: the new value * * Set or unset if the pipeline for @media can be shared will multiple clients. * When @shared is %TRUE, client requests for this media will share the media * pipeline. */ void gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared) { g_return_if_fail (GST_IS_RTSP_MEDIA (media)); media->shared = shared; } /** * gst_rtsp_media_is_shared: * @media: a #GstRTSPMedia * * Check if the pipeline for @media can be shared between multiple clients. * * Returns: %TRUE if the media can be shared between clients. */ gboolean gst_rtsp_media_is_shared (GstRTSPMedia * media) { g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); return media->shared; } /** * gst_rtsp_media_set_reusable: * @media: a #GstRTSPMedia * @reusable: the new value * * Set or unset if the pipeline for @media can be reused after the pipeline has * been unprepared. */ void gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable) { g_return_if_fail (GST_IS_RTSP_MEDIA (media)); media->reusable = reusable; } /** * gst_rtsp_media_is_reusable: * @media: a #GstRTSPMedia * * Check if the pipeline for @media can be reused after an unprepare. * * Returns: %TRUE if the media can be reused */ gboolean gst_rtsp_media_is_reusable (GstRTSPMedia * media) { g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); return media->reusable; } /** * gst_rtsp_media_set_protocols: * @media: a #GstRTSPMedia * @protocols: the new flags * * Configure the allowed lower transport for @media. */ void gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols) { g_return_if_fail (GST_IS_RTSP_MEDIA (media)); media->protocols = protocols; } /** * gst_rtsp_media_get_protocols: * @media: a #GstRTSPMedia * * Get the allowed protocols of @media. * * Returns: a #GstRTSPLowerTrans */ GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia * media) { g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_LOWER_TRANS_UNKNOWN); return media->protocols; } /** * gst_rtsp_media_set_eos_shutdown: * @media: a #GstRTSPMedia * @eos_shutdown: the new value * * Set or unset if an EOS event will be sent to the pipeline for @media before * it is unprepared. */ void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown) { g_return_if_fail (GST_IS_RTSP_MEDIA (media)); media->eos_shutdown = eos_shutdown; } /** * gst_rtsp_media_is_eos_shutdown: * @media: a #GstRTSPMedia * * Check if the pipeline for @media will send an EOS down the pipeline before * unpreparing. * * Returns: %TRUE if the media will send EOS before unpreparing. */ gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media) { g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); return media->eos_shutdown; } /** * gst_rtsp_media_set_buffer_size: * @media: a #GstRTSPMedia * @size: the new value * * Set the kernel UDP buffer size. */ void gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size) { g_return_if_fail (GST_IS_RTSP_MEDIA (media)); media->buffer_size = size; } /** * gst_rtsp_media_get_buffer_size: * @media: a #GstRTSPMedia * * Get the kernel UDP buffer size. * * Returns: the kernel UDP buffer size. */ guint gst_rtsp_media_get_buffer_size (GstRTSPMedia * media) { g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); return media->buffer_size; } /** * gst_rtsp_media_set_auth: * @media: a #GstRTSPMedia * @auth: a #GstRTSPAuth * * configure @auth to be used as the authentication manager of @media. */ void gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth) { GstRTSPAuth *old; g_return_if_fail (GST_IS_RTSP_MEDIA (media)); old = media->auth; if (old != auth) { if (auth) g_object_ref (auth); media->auth = auth; if (old) g_object_unref (old); } } /** * gst_rtsp_media_get_auth: * @media: a #GstRTSPMedia * * Get the #GstRTSPAuth used as the authentication manager of @media. * * Returns: the #GstRTSPAuth of @media. g_object_unref() after * usage. */ GstRTSPAuth * gst_rtsp_media_get_auth (GstRTSPMedia * media) { GstRTSPAuth *result; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL); if ((result = media->auth)) g_object_ref (result); return result; } /** * gst_rtsp_media_n_streams: * @media: a #GstRTSPMedia * * Get the number of streams in this media. * * Returns: The number of streams. */ guint gst_rtsp_media_n_streams (GstRTSPMedia * media) { g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0); return media->streams->len; } /** * gst_rtsp_media_get_stream: * @media: a #GstRTSPMedia * @idx: the stream index * * Retrieve the stream with index @idx from @media. * * Returns: the #GstRTSPMediaStream at index @idx or %NULL when a stream with * that index did not exist. */ GstRTSPMediaStream * gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx) { GstRTSPMediaStream *res; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL); if (idx < media->streams->len) res = g_array_index (media->streams, GstRTSPMediaStream *, idx); else res = NULL; return res; } /** * gst_rtsp_media_get_range_string: * @media: a #GstRTSPMedia * @play: for the PLAY request * * Get the current range as a string. * * Returns: The range as a string, g_free() after usage. */ gchar * gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play) { gchar *result; GstRTSPTimeRange range; /* make copy */ range = media->range; if (!play && media->active > 0) { range.min.type = GST_RTSP_TIME_NOW; range.min.seconds = -1; } result = gst_rtsp_range_to_string (&range); return result; } /** * gst_rtsp_media_seek: * @media: a #GstRTSPMedia * @range: a #GstRTSPTimeRange * * Seek the pipeline to @range. * * Returns: %TRUE on success. */ gboolean gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range) { GstSeekFlags flags; gboolean res; gint64 start, stop; GstSeekType start_type, stop_type; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); g_return_val_if_fail (range != NULL, FALSE); if (range->unit != GST_RTSP_RANGE_NPT) goto not_supported; /* depends on the current playing state of the pipeline. We might need to * queue this until we get EOS. */ flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT; start_type = stop_type = GST_SEEK_TYPE_NONE; switch (range->min.type) { case GST_RTSP_TIME_NOW: start = -1; break; case GST_RTSP_TIME_SECONDS: /* only seek when something changed */ if (media->range.min.seconds == range->min.seconds) { start = -1; } else { start = range->min.seconds * GST_SECOND; start_type = GST_SEEK_TYPE_SET; } break; case GST_RTSP_TIME_END: default: goto weird_type; } switch (range->max.type) { case GST_RTSP_TIME_SECONDS: /* only seek when something changed */ if (media->range.max.seconds == range->max.seconds) { stop = -1; } else { stop = range->max.seconds * GST_SECOND; stop_type = GST_SEEK_TYPE_SET; } break; case GST_RTSP_TIME_END: stop = -1; stop_type = GST_SEEK_TYPE_SET; break; case GST_RTSP_TIME_NOW: default: goto weird_type; } if (start != -1 || stop != -1) { GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (stop)); res = gst_element_seek (media->pipeline, 1.0, GST_FORMAT_TIME, flags, start_type, start, stop_type, stop); /* and block for the seek to complete */ GST_INFO ("done seeking %d", res); gst_element_get_state (media->pipeline, NULL, NULL, -1); GST_INFO ("prerolled again"); collect_media_stats (media); } else { GST_INFO ("no seek needed"); res = TRUE; } return res; /* ERRORS */ not_supported: { GST_WARNING ("seek unit %d not supported", range->unit); return FALSE; } weird_type: { GST_WARNING ("weird range type %d not supported", range->min.type); return FALSE; } } /** * gst_rtsp_media_stream_rtp: * @stream: a #GstRTSPMediaStream * @buffer: 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_media_stream_rtp (GstRTSPMediaStream * stream, GstBuffer * buffer) { GstFlowReturn ret; ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[0]), buffer); return ret; } /** * gst_rtsp_media_stream_rtcp: * @stream: a #GstRTSPMediaStream * @buffer: 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_media_stream_rtcp (GstRTSPMediaStream * stream, GstBuffer * buffer) { GstFlowReturn ret; ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[1]), buffer); return ret; } /* Allocate the udp ports and sockets */ static gboolean alloc_udp_ports (GstRTSPMedia * media, GstRTSPMediaStream * stream) { GstStateChangeReturn ret; GstElement *udpsrc0, *udpsrc1; GstElement *udpsink0, *udpsink1; gint tmp_rtp, tmp_rtcp; guint count; gint rtpport, rtcpport, sockfd; const gchar *host; udpsrc0 = NULL; udpsrc1 = NULL; udpsink0 = NULL; udpsink1 = NULL; count = 0; /* Start with random port */ tmp_rtp = 0; if (media->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); 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); 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), "sock", &sockfd, NULL); g_object_set (G_OBJECT (udpsink0), "sockfd", sockfd, NULL); g_object_set (G_OBJECT (udpsink0), "closefd", 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", media->buffer_size, NULL); } else { GST_WARNING ("multiudpsink version found without buffer-size property"); } g_object_get (G_OBJECT (udpsrc1), "sock", &sockfd, NULL); g_object_set (G_OBJECT (udpsink1), "sockfd", sockfd, NULL); g_object_set (G_OBJECT (udpsink1), "closefd", 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 configure all in configure_transport when the * server 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, GstRTSPMediaStream * stream) { gchar *capsstr; GstCaps *newcaps, *oldcaps; newcaps = gst_pad_get_current_caps (pad); oldcaps = stream->caps; stream->caps = newcaps; if (oldcaps) gst_caps_unref (oldcaps); capsstr = gst_caps_to_string (newcaps); GST_INFO ("stream %p received caps %p, %s", stream, newcaps, capsstr); g_free (capsstr); } static void dump_structure (const GstStructure * s) { gchar *sstr; sstr = gst_structure_to_string (s); GST_INFO ("structure: %s", sstr); g_free (sstr); } static GstRTSPMediaTrans * find_transport (GstRTSPMediaStream * stream, const gchar * rtcp_from) { GList *walk; GstRTSPMediaTrans *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", dest, port); for (walk = stream->transports; walk; walk = g_list_next (walk)) { GstRTSPMediaTrans *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 void on_new_ssrc (GObject * session, GObject * source, GstRTSPMediaStream * stream) { GstStructure *stats; GstRTSPMediaTrans *trans; GST_INFO ("%p: new source %p", stream, source); /* 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); } } else { GST_INFO ("%p: source %p for transport %p", stream, source, trans); } } static void on_ssrc_sdes (GObject * session, GObject * source, GstRTSPMediaStream * stream) { GST_INFO ("%p: new SDES %p", stream, source); } static void on_ssrc_active (GObject * session, GObject * source, GstRTSPMediaStream * stream) { GstRTSPMediaTrans *trans; trans = g_object_get_qdata (source, ssrc_stream_map_key); 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, GstRTSPMediaStream * stream) { GST_INFO ("%p: source %p bye", stream, source); } static void on_bye_timeout (GObject * session, GObject * source, GstRTSPMediaStream * stream) { GstRTSPMediaTrans *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, GstRTSPMediaStream * stream) { GstRTSPMediaTrans *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_buffer (GstAppSink * sink, gpointer user_data) { GList *walk; GstBuffer *buffer; GstRTSPMediaStream *stream; buffer = gst_app_sink_pull_buffer (sink); if (!buffer) return GST_FLOW_OK; stream = (GstRTSPMediaStream *) user_data; for (walk = stream->transports; walk; walk = g_list_next (walk)) { GstRTSPMediaTrans *tr = (GstRTSPMediaTrans *) 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_buffer_unref (buffer); return GST_FLOW_OK; } static GstFlowReturn handle_new_buffer_list (GstAppSink * sink, gpointer user_data) { GList *walk; GstBufferList *blist; GstRTSPMediaStream *stream; blist = gst_app_sink_pull_buffer_list (sink); if (!blist) return GST_FLOW_OK; stream = (GstRTSPMediaStream *) user_data; for (walk = stream->transports; walk; walk = g_list_next (walk)) { GstRTSPMediaTrans *tr = (GstRTSPMediaTrans *) walk->data; if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) { if (tr->send_rtp_list) tr->send_rtp_list (blist, tr->transport->interleaved.min, tr->user_data); } else { if (tr->send_rtcp_list) tr->send_rtcp_list (blist, tr->transport->interleaved.max, tr->user_data); } } gst_buffer_list_unref (blist); return GST_FLOW_OK; } static GstAppSinkCallbacks sink_cb = { NULL, /* not interested in EOS */ NULL, /* not interested in preroll buffers */ handle_new_buffer, handle_new_buffer_list }; /* prepare the pipeline objects to handle @stream in @media */ static gboolean setup_stream (GstRTSPMediaStream * stream, guint idx, GstRTSPMedia * media) { gchar *name; GstPad *pad, *teepad, *selpad; GstPadLinkReturn ret; gint i; /* allocate udp ports, we will have 4 of them, 2 for receiving RTP/RTCP and 2 * for sending RTP/RTCP. The sender and receiver ports are shared between the * elements */ if (!alloc_udp_ports (media, stream)) return FALSE; /* add the ports to the pipeline */ for (i = 0; i < 2; i++) { gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsink[i]); gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsrc[i]); } /* create elements for the TCP transfer */ for (i = 0; i < 2; i++) { stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL); 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); g_object_set (stream->appsink[i], "preroll-queue-len", 1, NULL); gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsink[i]); gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsrc[i]); gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]), &sink_cb, stream, NULL); } /* hook up the stream to the RTP session elements. */ name = g_strdup_printf ("send_rtp_sink_%d", idx); stream->send_rtp_sink = gst_element_get_request_pad (media->rtpbin, name); g_free (name); name = g_strdup_printf ("send_rtp_src_%d", idx); stream->send_rtp_src = gst_element_get_static_pad (media->rtpbin, name); g_free (name); name = g_strdup_printf ("send_rtcp_src_%d", idx); stream->send_rtcp_src = gst_element_get_request_pad (media->rtpbin, name); g_free (name); name = g_strdup_printf ("recv_rtcp_sink_%d", idx); stream->recv_rtcp_sink = gst_element_get_request_pad (media->rtpbin, name); g_free (name); name = g_strdup_printf ("recv_rtp_sink_%d", idx); stream->recv_rtp_sink = gst_element_get_request_pad (media->rtpbin, name); g_free (name); /* get the session */ g_signal_emit_by_name (media->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); /* link the RTP pad to the session manager */ ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink); if (ret != GST_PAD_LINK_OK) goto link_failed; /* make tee for RTP and link to stream */ stream->tee[0] = gst_element_factory_make ("tee", NULL); gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[0]); pad = gst_element_get_static_pad (stream->tee[0], "sink"); gst_pad_link (stream->send_rtp_src, pad); gst_object_unref (pad); /* link RTP sink, we're pretty sure this will work. */ teepad = gst_element_get_request_pad (stream->tee[0], "src%d"); pad = gst_element_get_static_pad (stream->udpsink[0], "sink"); gst_pad_link (teepad, pad); gst_object_unref (pad); gst_object_unref (teepad); teepad = gst_element_get_request_pad (stream->tee[0], "src%d"); pad = gst_element_get_static_pad (stream->appsink[0], "sink"); gst_pad_link (teepad, pad); gst_object_unref (pad); gst_object_unref (teepad); /* make tee for RTCP */ stream->tee[1] = gst_element_factory_make ("tee", NULL); gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[1]); pad = gst_element_get_static_pad (stream->tee[1], "sink"); gst_pad_link (stream->send_rtcp_src, pad); gst_object_unref (pad); /* link RTCP elements */ teepad = gst_element_get_request_pad (stream->tee[1], "src%d"); pad = gst_element_get_static_pad (stream->udpsink[1], "sink"); gst_pad_link (teepad, pad); gst_object_unref (pad); gst_object_unref (teepad); teepad = gst_element_get_request_pad (stream->tee[1], "src%d"); pad = gst_element_get_static_pad (stream->appsink[1], "sink"); gst_pad_link (teepad, pad); gst_object_unref (pad); gst_object_unref (teepad); /* make selector for the RTP receivers */ stream->selector[0] = gst_element_factory_make ("funnel", NULL); gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[0]); pad = gst_element_get_static_pad (stream->selector[0], "src"); gst_pad_link (pad, stream->recv_rtp_sink); gst_object_unref (pad); selpad = gst_element_get_request_pad (stream->selector[0], "sink%d"); pad = gst_element_get_static_pad (stream->udpsrc[0], "src"); gst_pad_link (pad, selpad); gst_object_unref (pad); gst_object_unref (selpad); selpad = gst_element_get_request_pad (stream->selector[0], "sink%d"); pad = gst_element_get_static_pad (stream->appsrc[0], "src"); gst_pad_link (pad, selpad); gst_object_unref (pad); gst_object_unref (selpad); /* make selector for the RTCP receivers */ stream->selector[1] = gst_element_factory_make ("funnel", NULL); gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[1]); pad = gst_element_get_static_pad (stream->selector[1], "src"); gst_pad_link (pad, stream->recv_rtcp_sink); gst_object_unref (pad); selpad = gst_element_get_request_pad (stream->selector[1], "sink%d"); pad = gst_element_get_static_pad (stream->udpsrc[1], "src"); gst_pad_link (pad, selpad); gst_object_unref (pad); gst_object_unref (selpad); selpad = gst_element_get_request_pad (stream->selector[1], "sink%d"); pad = gst_element_get_static_pad (stream->appsrc[1], "src"); gst_pad_link (pad, selpad); gst_object_unref (pad); gst_object_unref (selpad); /* we set and keep these to playing so that they don't cause NO_PREROLL return * values */ gst_element_set_state (stream->udpsrc[0], GST_STATE_PLAYING); gst_element_set_state (stream->udpsrc[1], GST_STATE_PLAYING); gst_element_set_locked_state (stream->udpsrc[0], TRUE); gst_element_set_locked_state (stream->udpsrc[1], TRUE); /* be notified of caps changes */ stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps", (GCallback) caps_notify, stream); stream->prepared = TRUE; return TRUE; /* ERRORS */ link_failed: { GST_WARNING ("failed to link stream %d", idx); return FALSE; } } static void unlock_streams (GstRTSPMedia * media) { guint i, n_streams; /* unlock the udp src elements */ n_streams = gst_rtsp_media_n_streams (media); for (i = 0; i < n_streams; i++) { GstRTSPMediaStream *stream; stream = gst_rtsp_media_get_stream (media, i); gst_element_set_locked_state (stream->udpsrc[0], FALSE); gst_element_set_locked_state (stream->udpsrc[1], FALSE); } } static void gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status) { g_mutex_lock (media->lock); /* never overwrite the error status */ if (media->status != GST_RTSP_MEDIA_STATUS_ERROR) media->status = status; GST_DEBUG ("setting new status to %d", status); g_cond_broadcast (media->cond); g_mutex_unlock (media->lock); } static GstRTSPMediaStatus gst_rtsp_media_get_status (GstRTSPMedia * media) { GstRTSPMediaStatus result; GTimeVal timeout; g_mutex_lock (media->lock); g_get_current_time (&timeout); g_time_val_add (&timeout, 20 * G_USEC_PER_SEC); /* while we are preparing, wait */ while (media->status == GST_RTSP_MEDIA_STATUS_PREPARING) { GST_DEBUG ("waiting for status change"); if (!g_cond_timed_wait (media->cond, media->lock, &timeout)) { GST_DEBUG ("timeout, assuming error status"); media->status = GST_RTSP_MEDIA_STATUS_ERROR; } } /* could be success or error */ result = media->status; GST_DEBUG ("got status %d", result); g_mutex_unlock (media->lock); return result; } static gboolean default_handle_message (GstRTSPMedia * media, GstMessage * message) { GstMessageType type; type = GST_MESSAGE_TYPE (message); switch (type) { case GST_MESSAGE_STATE_CHANGED: break; case GST_MESSAGE_BUFFERING: { gint percent; gst_message_parse_buffering (message, &percent); /* no state management needed for live pipelines */ if (media->is_live) break; if (percent == 100) { /* a 100% message means buffering is done */ media->buffering = FALSE; /* if the desired state is playing, go back */ if (media->target_state == GST_STATE_PLAYING) { GST_INFO ("Buffering done, setting pipeline to PLAYING"); gst_element_set_state (media->pipeline, GST_STATE_PLAYING); } else { GST_INFO ("Buffering done"); } } else { /* buffering busy */ if (media->buffering == FALSE) { if (media->target_state == GST_STATE_PLAYING) { /* we were not buffering but PLAYING, PAUSE the pipeline. */ GST_INFO ("Buffering, setting pipeline to PAUSED ..."); gst_element_set_state (media->pipeline, GST_STATE_PAUSED); } else { GST_INFO ("Buffering ..."); } } media->buffering = TRUE; } break; } case GST_MESSAGE_LATENCY: { gst_bin_recalculate_latency (GST_BIN_CAST (media->pipeline)); break; } case GST_MESSAGE_ERROR: { GError *gerror; gchar *debug; gst_message_parse_error (message, &gerror, &debug); GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug); g_error_free (gerror); g_free (debug); gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR); break; } case GST_MESSAGE_WARNING: { GError *gerror; gchar *debug; gst_message_parse_warning (message, &gerror, &debug); GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug); g_error_free (gerror); g_free (debug); break; } case GST_MESSAGE_ELEMENT: break; case GST_MESSAGE_STREAM_STATUS: break; case GST_MESSAGE_ASYNC_DONE: if (!media->adding) { /* when we are dynamically adding pads, the addition of the udpsrc will * temporarily produce ASYNC_DONE messages. We have to ignore them and * wait for the final ASYNC_DONE after everything prerolled */ GST_INFO ("%p: got ASYNC_DONE", media); collect_media_stats (media); gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED); } else { GST_INFO ("%p: ignoring ASYNC_DONE", media); } break; case GST_MESSAGE_EOS: GST_INFO ("%p: got EOS", media); if (media->eos_pending) { GST_DEBUG ("shutting down after EOS"); gst_element_set_state (media->pipeline, GST_STATE_NULL); media->eos_pending = FALSE; g_object_unref (media); } break; default: GST_INFO ("%p: got message type %s", media, gst_message_type_get_name (type)); break; } return TRUE; } static gboolean bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media) { GstRTSPMediaClass *klass; gboolean ret; klass = GST_RTSP_MEDIA_GET_CLASS (media); if (klass->handle_message) ret = klass->handle_message (media, message); else ret = FALSE; return ret; } /* called from streaming threads */ static void pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media) { GstRTSPMediaStream *stream; gchar *name; gint i; i = media->streams->len + 1; GST_INFO ("pad added %s:%s, stream %d", GST_DEBUG_PAD_NAME (pad), i); stream = g_new0 (GstRTSPMediaStream, 1); stream->payloader = element; name = g_strdup_printf ("dynpay%d", i); media->adding = TRUE; /* ghost the pad of the payloader to the element */ stream->srcpad = gst_ghost_pad_new (name, pad); gst_pad_set_active (stream->srcpad, TRUE); gst_element_add_pad (media->element, stream->srcpad); g_free (name); /* add stream now */ g_array_append_val (media->streams, stream); setup_stream (stream, i, media); for (i = 0; i < 2; i++) { gst_element_set_state (stream->udpsink[i], GST_STATE_PAUSED); gst_element_set_state (stream->appsink[i], GST_STATE_PAUSED); gst_element_set_state (stream->tee[i], GST_STATE_PAUSED); gst_element_set_state (stream->selector[i], GST_STATE_PAUSED); gst_element_set_state (stream->appsrc[i], GST_STATE_PAUSED); } media->adding = FALSE; } static void no_more_pads_cb (GstElement * element, GstRTSPMedia * media) { GST_INFO ("no more pads"); if (media->fakesink) { gst_object_ref (media->fakesink); gst_bin_remove (GST_BIN (media->pipeline), media->fakesink); gst_element_set_state (media->fakesink, GST_STATE_NULL); gst_object_unref (media->fakesink); media->fakesink = NULL; GST_INFO ("removed fakesink"); } } /** * gst_rtsp_media_prepare: * @media: a #GstRTSPMedia * * Prepare @media for streaming. This function will create the pipeline and * other objects to manage the streaming. * * It will preroll the pipeline and collect vital information about the streams * such as the duration. * * Returns: %TRUE on success. */ gboolean gst_rtsp_media_prepare (GstRTSPMedia * media) { GstStateChangeReturn ret; GstRTSPMediaStatus status; guint i, n_streams; GstRTSPMediaClass *klass; GstBus *bus; GList *walk; if (media->status == GST_RTSP_MEDIA_STATUS_PREPARED) goto was_prepared; if (!media->reusable && media->reused) goto is_reused; media->rtpbin = gst_element_factory_make ("gstrtpbin", NULL); if (media->rtpbin == NULL) goto no_gstrtpbin; GST_INFO ("preparing media %p", media); /* reset some variables */ media->is_live = FALSE; media->buffering = FALSE; /* we're preparing now */ media->status = GST_RTSP_MEDIA_STATUS_PREPARING; bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (media->pipeline)); /* add the pipeline bus to our custom mainloop */ media->source = gst_bus_create_watch (bus); gst_object_unref (bus); g_source_set_callback (media->source, (GSourceFunc) bus_message, media, NULL); klass = GST_RTSP_MEDIA_GET_CLASS (media); media->id = g_source_attach (media->source, klass->context); /* add stuff to the bin */ gst_bin_add (GST_BIN (media->pipeline), media->rtpbin); /* link streams we already have, other streams might appear when we have * dynamic elements */ n_streams = gst_rtsp_media_n_streams (media); for (i = 0; i < n_streams; i++) { GstRTSPMediaStream *stream; stream = gst_rtsp_media_get_stream (media, i); setup_stream (stream, i, media); } for (walk = media->dynamic; walk; walk = g_list_next (walk)) { GstElement *elem = walk->data; GST_INFO ("adding callbacks for dynamic element %p", elem); g_signal_connect (elem, "pad-added", (GCallback) pad_added_cb, media); g_signal_connect (elem, "no-more-pads", (GCallback) no_more_pads_cb, media); /* we add a fakesink here in order to make the state change async. We remove * the fakesink again in the no-more-pads callback. */ media->fakesink = gst_element_factory_make ("fakesink", "fakesink"); gst_bin_add (GST_BIN (media->pipeline), media->fakesink); } GST_INFO ("setting pipeline to PAUSED for media %p", media); /* first go to PAUSED */ ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED); media->target_state = GST_STATE_PAUSED; switch (ret) { case GST_STATE_CHANGE_SUCCESS: GST_INFO ("SUCCESS state change for media %p", media); break; case GST_STATE_CHANGE_ASYNC: GST_INFO ("ASYNC state change for media %p", media); break; case GST_STATE_CHANGE_NO_PREROLL: /* we need to go to PLAYING */ GST_INFO ("NO_PREROLL state change: live media %p", media); media->is_live = TRUE; ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING); if (ret == GST_STATE_CHANGE_FAILURE) goto state_failed; break; case GST_STATE_CHANGE_FAILURE: goto state_failed; } /* now wait for all pads to be prerolled */ status = gst_rtsp_media_get_status (media); if (status == GST_RTSP_MEDIA_STATUS_ERROR) goto state_failed; g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL); GST_INFO ("object %p is prerolled", media); return TRUE; /* OK */ was_prepared: { return TRUE; } /* ERRORS */ is_reused: { GST_WARNING ("can not reuse media %p", media); return FALSE; } no_gstrtpbin: { GST_WARNING ("no gstrtpbin element"); g_warning ("failed to create element 'gstrtpbin', check your installation"); return FALSE; } state_failed: { GST_WARNING ("failed to preroll pipeline"); unlock_streams (media); gst_element_set_state (media->pipeline, GST_STATE_NULL); gst_rtsp_media_unprepare (media); return FALSE; } } /** * gst_rtsp_media_unprepare: * @media: a #GstRTSPMedia * * Unprepare @media. After this call, the media should be prepared again before * it can be used again. If the media is set to be non-reusable, a new instance * must be created. * * Returns: %TRUE on success. */ gboolean gst_rtsp_media_unprepare (GstRTSPMedia * media) { GstRTSPMediaClass *klass; gboolean success; if (media->status == GST_RTSP_MEDIA_STATUS_UNPREPARED) return TRUE; GST_INFO ("unprepare media %p", media); media->target_state = GST_STATE_NULL; klass = GST_RTSP_MEDIA_GET_CLASS (media); if (klass->unprepare) success = klass->unprepare (media); else success = TRUE; media->status = GST_RTSP_MEDIA_STATUS_UNPREPARED; media->reused = TRUE; /* when the media is not reusable, this will effectively unref the media and * recreate it */ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL); return success; } static gboolean default_unprepare (GstRTSPMedia * media) { if (media->eos_shutdown) { GST_DEBUG ("sending EOS for shutdown"); /* ref so that we don't disappear */ g_object_ref (media); media->eos_pending = TRUE; gst_element_send_event (media->pipeline, gst_event_new_eos ()); /* we need to go to playing again for the EOS to propagate, normally in this * state, nothing is receiving data from us anymore so this is ok. */ gst_element_set_state (media->pipeline, GST_STATE_PLAYING); } else { GST_DEBUG ("shutting down"); gst_element_set_state (media->pipeline, GST_STATE_NULL); } return TRUE; } static void add_udp_destination (GstRTSPMedia * media, GstRTSPMediaStream * stream, gchar * dest, gint min, gint max) { 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); } static void remove_udp_destination (GstRTSPMedia * media, GstRTSPMediaStream * stream, gchar * dest, gint min, gint max) { 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); } /** * gst_rtsp_media_set_state: * @media: a #GstRTSPMedia * @state: the target state of the media * @transports: a #GArray of #GstRTSPMediaTrans pointers * * Set the state of @media to @state and for the transports in @transports. * * Returns: %TRUE on success. */ gboolean gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state, GArray * transports) { gint i; GstStateChangeReturn ret; gboolean add, remove, do_state; gint old_active; g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE); g_return_val_if_fail (transports != NULL, FALSE); /* NULL and READY are the same */ if (state == GST_STATE_READY) state = GST_STATE_NULL; add = remove = FALSE; GST_INFO ("going to state %s media %p", gst_element_state_get_name (state), media); switch (state) { case GST_STATE_NULL: /* unlock the streams so that they follow the state changes from now on */ unlock_streams (media); /* fallthrough */ case GST_STATE_PAUSED: /* we're going from PLAYING to PAUSED, READY or NULL, remove */ if (media->target_state == GST_STATE_PLAYING) remove = TRUE; break; case GST_STATE_PLAYING: /* we're going to PLAYING, add */ add = TRUE; break; default: break; } old_active = media->active; for (i = 0; i < transports->len; i++) { GstRTSPMediaTrans *tr; GstRTSPMediaStream *stream; GstRTSPTransport *trans; /* we need a non-NULL entry in the array */ tr = g_array_index (transports, GstRTSPMediaTrans *, i); if (tr == NULL) continue; /* we need a transport */ if (!(trans = tr->transport)) continue; /* get the stream and add the destinations */ stream = gst_rtsp_media_get_stream (media, tr->idx); switch (trans->lower_transport) { case GST_RTSP_LOWER_TRANS_UDP: case GST_RTSP_LOWER_TRANS_UDP_MCAST: { gchar *dest; gint min, max; dest = trans->destination; if (trans->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) { min = trans->port.min; max = trans->port.max; } else { min = trans->client_port.min; max = trans->client_port.max; } if (add && !tr->active) { add_udp_destination (media, stream, dest, min, max); stream->transports = g_list_prepend (stream->transports, tr); tr->active = TRUE; media->active++; } else if (remove && tr->active) { remove_udp_destination (media, stream, dest, min, max); stream->transports = g_list_remove (stream->transports, tr); tr->active = FALSE; media->active--; } break; } case GST_RTSP_LOWER_TRANS_TCP: if (add && !tr->active) { GST_INFO ("adding TCP %s", trans->destination); stream->transports = g_list_prepend (stream->transports, tr); tr->active = TRUE; media->active++; } else if (remove && tr->active) { GST_INFO ("removing TCP %s", trans->destination); stream->transports = g_list_remove (stream->transports, tr); tr->active = FALSE; media->active--; } break; default: GST_INFO ("Unknown transport %d", trans->lower_transport); break; } } /* we just added the first media, do the playing state change */ if (old_active == 0 && add) do_state = TRUE; /* if we have no more active media, do the downward state changes */ else if (media->active == 0) do_state = TRUE; else do_state = FALSE; GST_INFO ("state %d active %d media %p do_state %d", state, media->active, media, do_state); if (media->target_state != state) { if (do_state) { if (state == GST_STATE_NULL) { gst_rtsp_media_unprepare (media); } else { GST_INFO ("state %s media %p", gst_element_state_get_name (state), media); media->target_state = state; ret = gst_element_set_state (media->pipeline, state); } } g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state, NULL); } /* remember where we are */ if (state == GST_STATE_PAUSED || old_active != media->active) collect_media_stats (media); return TRUE; } /** * gst_rtsp_media_remove_elements: * @media: a #GstRTSPMedia * * Remove all elements and the pipeline controlled by @media. */ void gst_rtsp_media_remove_elements (GstRTSPMedia * media) { gint i, j; unlock_streams (media); for (i = 0; i < media->streams->len; i++) { GstRTSPMediaStream *stream; GST_INFO ("Removing elements of stream %d from pipeline", i); stream = g_array_index (media->streams, GstRTSPMediaStream *, i); gst_pad_unlink (stream->srcpad, stream->send_rtp_sink); g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig); for (j = 0; j < 2; j++) { gst_element_set_state (stream->udpsrc[j], GST_STATE_NULL); gst_element_set_state (stream->udpsink[j], GST_STATE_NULL); gst_element_set_state (stream->appsrc[j], GST_STATE_NULL); gst_element_set_state (stream->appsink[j], GST_STATE_NULL); gst_element_set_state (stream->tee[j], GST_STATE_NULL); gst_element_set_state (stream->selector[j], GST_STATE_NULL); gst_bin_remove (GST_BIN (media->pipeline), stream->udpsrc[j]); gst_bin_remove (GST_BIN (media->pipeline), stream->udpsink[j]); gst_bin_remove (GST_BIN (media->pipeline), stream->appsrc[j]); gst_bin_remove (GST_BIN (media->pipeline), stream->appsink[j]); gst_bin_remove (GST_BIN (media->pipeline), stream->tee[j]); gst_bin_remove (GST_BIN (media->pipeline), stream->selector[j]); } if (stream->caps) gst_caps_unref (stream->caps); stream->caps = NULL; gst_rtsp_media_stream_free (stream); } g_array_remove_range (media->streams, 0, media->streams->len); gst_element_set_state (media->rtpbin, GST_STATE_NULL); gst_bin_remove (GST_BIN (media->pipeline), media->rtpbin); gst_object_unref (media->pipeline); media->pipeline = NULL; }