/* GStreamer * Copyright (C) <2005,2006> Wim Taymans * <2006> Lutz Mueller * * 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. */ /* * Unless otherwise indicated, Source Code is licensed under MIT license. * See further explanation attached in License Statement (distributed in the file * LICENSE). * * Permission is hereby granted, free of charge, to any person obtaining a copy of * this software and associated documentation files (the "Software"), to deal in * the Software without restriction, including without limitation the rights to * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies * of the Software, and to permit persons to whom the Software is furnished to do * so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in all * copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE * SOFTWARE. */ /** * SECTION:element-rtspsrc * * * * Makes a connection to an RTSP server and read the data. * rtspsrc strictly follows RFC 2326 and therefore does not (yet) support * RealMedia/Quicktime/Microsoft extensions. * * * RTSP supports transport over TCP or UDP in unicast or multicast mode. By * default rtspsrc will negotiate a connection in the following order: * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed * protocols can be controlled with the "protocols" property. * * * rtspsrc currently understands SDP as the format of the session description. * For each stream listed in the SDP a new rtp_stream%d pad will be created * with caps derived from the SDP media description. This is a caps of mime type * "application/x-rtp" that can be connected to any available RTP depayloader * element. * * * rtspsrc will internally instantiate an RTP session manager element * that will handle the RTCP messages to and from the server, jitter removal, * packet reordering along with providing a clock for the pipeline. * This feature is currently fully implemented with the gstrtpbin in the * gst-plugins-bad module. * * * rtspsrc acts like a live source and will therefore only generate data in the * PLAYING state. * * Example launch line * * * gst-launch rtspsrc location=rtsp://some.server/url ! fakesink * * Establish a connection to an RTSP server and send the raw RTP packets to a fakesink. * * * * Last reviewed on 2006-08-18 (0.10.5) */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #ifdef HAVE_UNISTD_H #include #endif /* HAVE_UNISTD_H */ #include #include #include #include #include #include #include #include #include "gstrtspsrc.h" #ifdef G_OS_WIN32 #include #endif GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug); #define GST_CAT_DEFAULT (rtspsrc_debug) /* elementfactory information */ static const GstElementDetails gst_rtspsrc_details = GST_ELEMENT_DETAILS ("RTSP packet receiver", "Source/Network", "Receive data over the network via RTSP (RFC 2326)", "Wim Taymans \n" "Thijs Vermeir \n" "Lutz Mueller "); static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream%d", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("application/x-rtp; application/x-rdt")); /* templates used internally */ static GstStaticPadTemplate anysrctemplate = GST_STATIC_PAD_TEMPLATE ("internalsrc%d", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS_ANY); static GstStaticPadTemplate anysinktemplate = GST_STATIC_PAD_TEMPLATE ("internalsink%d", GST_PAD_SINK, GST_PAD_SOMETIMES, GST_STATIC_CAPS_ANY); enum { /* FILL ME */ LAST_SIGNAL }; #define DEFAULT_LOCATION NULL #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP #define DEFAULT_DEBUG FALSE #define DEFAULT_RETRY 20 #define DEFAULT_TIMEOUT 5000000 #define DEFAULT_TCP_TIMEOUT 20000000 #define DEFAULT_LATENCY_MS 3000 #define DEFAULT_CONNECTION_SPEED 0 enum { PROP_0, PROP_LOCATION, PROP_PROTOCOLS, PROP_DEBUG, PROP_RETRY, PROP_TIMEOUT, PROP_TCP_TIMEOUT, PROP_LATENCY, PROP_CONNECTION_SPEED }; #define GST_TYPE_RTSP_LOWER_TRANS (gst_rtsp_lower_trans_get_type()) static GType gst_rtsp_lower_trans_get_type (void) { static GType rtsp_lower_trans_type = 0; static const GFlagsValue rtsp_lower_trans[] = { {GST_RTSP_LOWER_TRANS_UDP, "UDP Unicast Mode", "udp-unicast"}, {GST_RTSP_LOWER_TRANS_UDP_MCAST, "UDP Multicast Mode", "udp-multicast"}, {GST_RTSP_LOWER_TRANS_TCP, "TCP interleaved mode", "tcp"}, {0, NULL, NULL}, }; if (!rtsp_lower_trans_type) { rtsp_lower_trans_type = g_flags_register_static ("GstRTSPLowerTrans", rtsp_lower_trans); } return rtsp_lower_trans_type; } static void gst_rtspsrc_base_init (gpointer g_class); static void gst_rtspsrc_finalize (GObject * object); static void gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data); static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media); static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element, GstStateChange transition); static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message); static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gboolean flush); static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src); static gboolean gst_rtspsrc_open (GstRTSPSrc * src); static gboolean gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment); static gboolean gst_rtspsrc_pause (GstRTSPSrc * src); static gboolean gst_rtspsrc_close (GstRTSPSrc * src); static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri); static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src); static void gst_rtspsrc_loop (GstRTSPSrc * src); static void gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream, GstEvent * event); static void gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event); /* commands we send to out loop to notify it of events */ #define CMD_WAIT 0 #define CMD_RECONNECT 1 #define CMD_STOP 2 /*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */ static void _do_init (GType rtspsrc_type) { static const GInterfaceInfo urihandler_info = { gst_rtspsrc_uri_handler_init, NULL, NULL }; GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src"); g_type_add_interface_static (rtspsrc_type, GST_TYPE_URI_HANDLER, &urihandler_info); } GST_BOILERPLATE_FULL (GstRTSPSrc, gst_rtspsrc, GstBin, GST_TYPE_BIN, _do_init); static void gst_rtspsrc_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&rtptemplate)); gst_element_class_set_details (element_class, &gst_rtspsrc_details); } static void gst_rtspsrc_class_init (GstRTSPSrcClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBinClass *gstbin_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbin_class = (GstBinClass *) klass; gobject_class->set_property = gst_rtspsrc_set_property; gobject_class->get_property = gst_rtspsrc_get_property; gobject_class->finalize = gst_rtspsrc_finalize; g_object_class_install_property (gobject_class, PROP_LOCATION, g_param_spec_string ("location", "RTSP Location", "Location of the RTSP url to read", DEFAULT_LOCATION, G_PARAM_READWRITE)); 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_CONSTRUCT)); g_object_class_install_property (gobject_class, PROP_DEBUG, g_param_spec_boolean ("debug", "Debug", "Dump request and response messages to stdout", DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); g_object_class_install_property (gobject_class, PROP_RETRY, g_param_spec_uint ("retry", "Retry", "Max number of retries when allocating RTP ports.", 0, G_MAXUINT16, DEFAULT_RETRY, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); g_object_class_install_property (gobject_class, PROP_TIMEOUT, g_param_spec_uint64 ("timeout", "Timeout", "Retry TCP transport after UDP timeout microseconds (0 = disabled)", 0, G_MAXUINT64, DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT, g_param_spec_uint64 ("tcp-timeout", "TCP Timeout", "Fail after timeout microseconds on TCP connections (0 = disabled)", 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); g_object_class_install_property (gobject_class, PROP_LATENCY, g_param_spec_uint ("latency", "Buffer latency in ms", "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED, g_param_spec_uint ("connection-speed", "Connection Speed", "Network connection speed in kbps (0 = unknown)", 0, G_MAXINT / 1000, DEFAULT_CONNECTION_SPEED, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); gstelement_class->change_state = gst_rtspsrc_change_state; gstbin_class->handle_message = gst_rtspsrc_handle_message; gst_rtsp_ext_list_init (); } static void gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class) { #ifdef G_OS_WIN32 WSADATA wsa_data; if (WSAStartup (MAKEWORD (2, 2), &wsa_data) != 0) { GST_ERROR_OBJECT (src, "WSAStartup failed: 0x%08x", WSAGetLastError ()); } #endif src->location = g_strdup (DEFAULT_LOCATION); src->url = NULL; /* get a list of all extensions */ src->extensions = gst_rtsp_ext_list_get (); /* connect to send signal */ gst_rtsp_ext_list_connect (src->extensions, "send", (GCallback) gst_rtspsrc_send_cb, src); /* protects the streaming thread in interleaved mode or the polling * thread in UDP mode. */ src->stream_rec_lock = g_new (GStaticRecMutex, 1); g_static_rec_mutex_init (src->stream_rec_lock); /* protects our state changes from multiple invocations */ src->state_rec_lock = g_new (GStaticRecMutex, 1); g_static_rec_mutex_init (src->state_rec_lock); /* protects access to the server connection */ src->conn_rec_lock = g_new (GStaticRecMutex, 1); g_static_rec_mutex_init (src->conn_rec_lock); src->state = GST_RTSP_STATE_INVALID; } static void gst_rtspsrc_finalize (GObject * object) { GstRTSPSrc *rtspsrc; rtspsrc = GST_RTSPSRC (object); gst_rtsp_ext_list_free (rtspsrc->extensions); g_free (rtspsrc->location); g_free (rtspsrc->req_location); g_free (rtspsrc->content_base); gst_rtsp_url_free (rtspsrc->url); /* free locks */ g_static_rec_mutex_free (rtspsrc->stream_rec_lock); g_free (rtspsrc->stream_rec_lock); g_static_rec_mutex_free (rtspsrc->state_rec_lock); g_free (rtspsrc->state_rec_lock); g_static_rec_mutex_free (rtspsrc->conn_rec_lock); g_free (rtspsrc->conn_rec_lock); #ifdef G_OS_WIN32 WSACleanup (); #endif G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRTSPSrc *rtspsrc; rtspsrc = GST_RTSPSRC (object); switch (prop_id) { case PROP_LOCATION: gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc), g_value_get_string (value)); break; case PROP_PROTOCOLS: rtspsrc->protocols = g_value_get_flags (value); break; case PROP_DEBUG: rtspsrc->debug = g_value_get_boolean (value); break; case PROP_RETRY: rtspsrc->retry = g_value_get_uint (value); break; case PROP_TIMEOUT: rtspsrc->udp_timeout = g_value_get_uint64 (value); break; case PROP_TCP_TIMEOUT: { guint64 timeout = g_value_get_uint64 (value); rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC; rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC; if (timeout != 0) rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout; else rtspsrc->ptcp_timeout = NULL; break; } case PROP_LATENCY: rtspsrc->latency = g_value_get_uint (value); break; case PROP_CONNECTION_SPEED: rtspsrc->connection_speed = g_value_get_uint (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRTSPSrc *rtspsrc; rtspsrc = GST_RTSPSRC (object); switch (prop_id) { case PROP_LOCATION: g_value_set_string (value, rtspsrc->location); break; case PROP_PROTOCOLS: g_value_set_flags (value, rtspsrc->protocols); break; case PROP_DEBUG: g_value_set_boolean (value, rtspsrc->debug); break; case PROP_RETRY: g_value_set_uint (value, rtspsrc->retry); break; case PROP_TIMEOUT: g_value_set_uint64 (value, rtspsrc->udp_timeout); break; case PROP_TCP_TIMEOUT: { guint64 timeout; timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC + rtspsrc->tcp_timeout.tv_usec; g_value_set_uint64 (value, timeout); break; } case PROP_LATENCY: g_value_set_uint (value, rtspsrc->latency); break; case PROP_CONNECTION_SPEED: g_value_set_uint (value, rtspsrc->connection_speed); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gint find_stream_by_id (GstRTSPStream * stream, gconstpointer a) { gint id = GPOINTER_TO_INT (a); if (stream->id == id) return 0; return -1; } static gint find_stream_by_channel (GstRTSPStream * stream, gconstpointer a) { gint channel = GPOINTER_TO_INT (a); if (stream->channel[0] == channel || stream->channel[1] == channel) return 0; return -1; } static gint find_stream_by_pt (GstRTSPStream * stream, gconstpointer a) { gint pt = GPOINTER_TO_INT (a); if (stream->pt == pt) return 0; return -1; } static gint find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a) { GstElement *src = (GstElement *) a; if (stream->udpsrc[0] == src) return 0; if (stream->udpsrc[1] == src) return 0; return -1; } static gint find_stream_by_setup (GstRTSPStream * stream, gconstpointer a) { /* check qualified setup_url */ if (!strcmp (stream->setup_url, (gchar *) a)) return 0; /* check original control_url */ if (!strcmp (stream->control_url, (gchar *) a)) return 0; /* check if qualified setup_url ends with string */ if (g_str_has_suffix (stream->control_url, (gchar *) a)) return 0; return -1; } GstRTSPStream * find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func) { GList *lstream; /* find and get stream */ if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func))) return (GstRTSPStream *) lstream->data; return NULL; } static const GstSDPBandwidth * gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp, const GstSDPMedia * media, const gchar * type) { guint i, len; /* first look in the media specific section */ len = gst_sdp_media_bandwidths_len (media); for (i = 0; i < len; i++) { const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i); if (strcmp (bw->bwtype, type) == 0) return bw; } /* then look in the message specific section */ len = gst_sdp_message_bandwidths_len (sdp); for (i = 0; i < len; i++) { const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i); if (strcmp (bw->bwtype, type) == 0) return bw; } return NULL; } static void gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp, const GstSDPMedia * media, GstRTSPStream * stream) { const GstSDPBandwidth *bw; if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS))) stream->as_bandwidth = bw->bandwidth; else stream->as_bandwidth = -1; if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR))) stream->rr_bandwidth = bw->bandwidth; else stream->rr_bandwidth = -1; if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS))) stream->rs_bandwidth = bw->bandwidth; else stream->rs_bandwidth = -1; } static GstRTSPStream * gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx) { GstRTSPStream *stream; const gchar *control_url; const gchar *payload; const GstSDPMedia *media; /* get media, should not return NULL */ media = gst_sdp_message_get_media (sdp, idx); if (media == NULL) return NULL; stream = g_new0 (GstRTSPStream, 1); stream->parent = src; /* we mark the pad as not linked, we will mark it as OK when we add the pad to * the element. */ stream->last_ret = GST_FLOW_NOT_LINKED; stream->added = FALSE; stream->disabled = FALSE; stream->id = src->numstreams++; stream->eos = FALSE; stream->discont = TRUE; stream->seqbase = -1; stream->timebase = -1; /* collect bandwidth information for this steam */ gst_rtspsrc_collect_bandwidth (src, sdp, media, stream); /* we must have a payload. No payload means we cannot create caps */ /* FIXME, handle multiple formats. */ if ((payload = gst_sdp_media_get_format (media, 0))) { stream->pt = atoi (payload); /* convert caps */ stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media); if (stream->pt >= 96) { /* If we have a dynamic payload type, see if we have a stream with the * same payload number. If there is one, they are part of the same * container and we only need to add one pad. */ if (find_stream (src, GINT_TO_POINTER (stream->pt), (gpointer) find_stream_by_pt)) { stream->container = TRUE; } } } /* get control url to construct the setup url. The setup url is used to * configure the transport of the stream and is used to identity the stream in * the RTP-Info header field returned from PLAY. */ control_url = gst_sdp_media_get_attribute_val (media, "control"); GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream); GST_DEBUG_OBJECT (src, " pt: %d", stream->pt); GST_DEBUG_OBJECT (src, " container: %d", stream->container); GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps); GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url)); if (control_url != NULL) { stream->control_url = g_strdup (control_url); /* Build a fully qualified url using the content_base if any or by prefixing * the original request. * If the control_url starts with a '/' or a non rtsp: protocol we will most * likely build a URL that the server will fail to understand, this is ok, * we will fail then. */ if (g_str_has_prefix (control_url, "rtsp://")) stream->setup_url = g_strdup (control_url); else if (src->content_base) stream->setup_url = g_strdup_printf ("%s%s", src->content_base, control_url); else stream->setup_url = g_strdup_printf ("%s/%s", src->req_location, control_url); } GST_DEBUG_OBJECT (src, " setup: %s", GST_STR_NULL (stream->setup_url)); /* we keep track of all streams */ src->streams = g_list_append (src->streams, stream); return stream; /* ERRORS */ } static void gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream) { gint i; GST_DEBUG_OBJECT (src, "free stream %p", stream); if (stream->caps) gst_caps_unref (stream->caps); g_free (stream->control_url); g_free (stream->setup_url); for (i = 0; i < 2; i++) { GstElement *udpsrc = stream->udpsrc[i]; if (udpsrc) { GstPad *pad; /* unlink the pad */ pad = gst_element_get_pad (udpsrc, "src"); if (stream->channelpad[i]) { gst_pad_unlink (pad, stream->channelpad[i]); } gst_element_set_state (udpsrc, GST_STATE_NULL); gst_bin_remove (GST_BIN_CAST (src), udpsrc); gst_object_unref (stream->udpsrc[i]); stream->udpsrc[i] = NULL; } if (stream->channelpad[i]) { gst_object_unref (stream->channelpad[i]); stream->channelpad[i] = NULL; } } if (stream->udpsink) { gst_element_set_state (stream->udpsink, GST_STATE_NULL); gst_bin_remove (GST_BIN_CAST (src), stream->udpsink); gst_object_unref (stream->udpsink); stream->udpsink = NULL; } if (stream->srcpad) { gst_pad_set_active (stream->srcpad, FALSE); if (stream->added) { gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad); stream->added = FALSE; } stream->srcpad = NULL; } if (stream->rtcppad) { gst_object_unref (stream->rtcppad); stream->rtcppad = NULL; } g_free (stream); } static void gst_rtspsrc_cleanup (GstRTSPSrc * src) { GList *walk; GST_DEBUG_OBJECT (src, "cleanup"); for (walk = src->streams; walk; walk = g_list_next (walk)) { GstRTSPStream *stream = (GstRTSPStream *) walk->data; gst_rtspsrc_stream_free (src, stream); } g_list_free (src->streams); src->streams = NULL; if (src->session) { if (src->session_sig_id) { g_signal_handler_disconnect (src->session, src->session_sig_id); src->session_sig_id = 0; } gst_element_set_state (src->session, GST_STATE_NULL); gst_bin_remove (GST_BIN_CAST (src), src->session); src->session = NULL; } src->numstreams = 0; if (src->props) gst_structure_free (src->props); src->props = NULL; } #define PARSE_INT(p, del, res) \ G_STMT_START { \ gchar *t = p; \ p = strstr (p, del); \ if (p == NULL) \ res = -1; \ else { \ *p = '\0'; \ p++; \ res = atoi (t); \ } \ } G_STMT_END #define PARSE_STRING(p, del, res) \ G_STMT_START { \ gchar *t = p; \ p = strstr (p, del); \ if (p == NULL) { \ res = NULL; \ p = t; \ } \ else { \ *p = '\0'; \ p++; \ res = t; \ } \ } G_STMT_END #define SKIP_SPACES(p) \ while (*p && g_ascii_isspace (*p)) \ p++; /* rtpmap contains: * * /[/] */ static gboolean gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name, gint * rate, gchar ** params) { gchar *p, *t; t = p = (gchar *) rtpmap; PARSE_INT (p, " ", *payload); if (*payload == -1) return FALSE; SKIP_SPACES (p); if (*p == '\0') return FALSE; PARSE_STRING (p, "/", *name); if (*name == NULL) { GST_DEBUG ("no rate, name %s", p); /* no rate, assume -1 then, this is not supposed to happen but RealMedia * streams seem to omit the rate. */ *name = p; *rate = -1; return TRUE; } t = p; p = strstr (p, "/"); if (p == NULL) { *rate = atoi (t); return TRUE; } *p = '\0'; p++; *rate = atoi (t); t = p; if (*p == '\0') return TRUE; *params = t; return TRUE; } /* * Mapping of caps to and from SDP fields: * * m= RTP/AVP * a=rtpmap: /[/] * a=fmtp: [=];... */ static GstCaps * gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media) { GstCaps *caps; const gchar *rtpmap; const gchar *fmtp; gchar *name = NULL; gint rate = -1; gchar *params = NULL; gchar *tmp; GstStructure *s; gint payload = 0; gboolean ret; /* get and parse rtpmap */ if ((rtpmap = gst_sdp_media_get_attribute_val (media, "rtpmap"))) { ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms); if (ret) { if (payload != pt) { /* we ignore the rtpmap if the payload type is different. */ g_warning ("rtpmap of wrong payload type, ignoring"); name = NULL; rate = -1; params = NULL; } } else { /* if we failed to parse the rtpmap for a dynamic payload type, we have an * error */ if (pt >= 96) goto no_rtpmap; /* else we can ignore */ g_warning ("error parsing rtpmap, ignoring"); } } else { /* dynamic payloads need rtpmap or we fail */ if (pt >= 96) goto no_rtpmap; } /* check if we have a rate, if not, we need to look up the rate from the * default rates based on the payload types. */ if (rate == -1) { const GstRTPPayloadInfo *info; if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) { /* dynamic types, use media and encoding_name */ tmp = g_ascii_strdown (media->media, -1); info = gst_rtp_payload_info_for_name (tmp, name); g_free (tmp); } else { /* static types, use payload type */ info = gst_rtp_payload_info_for_pt (pt); } if (info) { if ((rate = info->clock_rate) == 0) rate = -1; } /* we fail if we cannot find one */ if (rate == -1) goto no_rate; } tmp = g_ascii_strdown (media->media, -1); caps = gst_caps_new_simple ("application/x-unknown", "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL); g_free (tmp); s = gst_caps_get_structure (caps, 0); gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL); /* encoding name must be upper case */ if (name != NULL) { tmp = g_ascii_strup (name, -1); gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL); g_free (tmp); } /* params must be lower case */ if (params != NULL) { tmp = g_ascii_strdown (params, -1); gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL); g_free (tmp); } /* parse optional fmtp: field */ if ((fmtp = gst_sdp_media_get_attribute_val (media, "fmtp"))) { gchar *p; gint payload = 0; p = (gchar *) fmtp; /* p is now of the format [=];... */ PARSE_INT (p, " ", payload); if (payload != -1 && payload == pt) { gchar **pairs; gint i; /* [=] are separated with ';' */ pairs = g_strsplit (p, ";", 0); for (i = 0; pairs[i]; i++) { gchar *valpos; gchar *val, *key; /* the key may not have a '=', the value can have other '='s */ valpos = strstr (pairs[i], "="); if (valpos) { /* we have a '=' and thus a value, remove the '=' with \0 */ *valpos = '\0'; /* value is everything between '=' and ';'. FIXME, strip? */ val = g_strstrip (valpos + 1); } else { /* simple ;.. is translated into =1;... */ val = "1"; } /* strip the key of spaces, convert key to lowercase but not the value. */ key = g_strstrip (pairs[i]); if (strlen (key) > 1) { tmp = g_ascii_strdown (key, -1); gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL); g_free (tmp); } } g_strfreev (pairs); } } return caps; /* ERRORS */ no_rtpmap: { g_warning ("rtpmap type not given for dynamic payload %d", pt); return NULL; } no_rate: { g_warning ("rate unknown for payload type %d", pt); return NULL; } } static gboolean gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream, gint * rtpport, gint * rtcpport) { GstRTSPSrc *src; GstStateChangeReturn ret; GstElement *tmp, *udpsrc0, *udpsrc1; gint tmp_rtp, tmp_rtcp; guint count; src = stream->parent; tmp = NULL; udpsrc0 = NULL; udpsrc1 = NULL; count = 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, "udp://0.0.0.0:0", NULL); if (udpsrc0 == NULL) goto no_udp_protocol; ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED); if (ret == GST_STATE_CHANGE_FAILURE) goto start_udp_failure; g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL); GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp); /* check if port is even */ if ((tmp_rtp & 0x01) != 0) { /* port not even, close and allocate another */ count++; if (count > src->retry) goto no_ports; GST_DEBUG_OBJECT (src, "RTP port not even, retry %d", count); /* have to keep port allocated so we can get a new one */ if (tmp != NULL) { GST_DEBUG_OBJECT (src, "free temp"); gst_element_set_state (tmp, GST_STATE_NULL); gst_object_unref (tmp); } tmp = udpsrc0; GST_DEBUG_OBJECT (src, "retry %d", count); goto again; } /* free leftover temp element/port */ if (tmp) { gst_element_set_state (tmp, GST_STATE_NULL); gst_object_unref (tmp); tmp = NULL; } /* allocate port+1 for RTCP now */ udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", 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); GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp); ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED); /* FIXME, this could fail if the next port is not free, we * should retry with another port then */ if (ret == GST_STATE_CHANGE_FAILURE) goto start_rtcp_failure; /* 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; /* we keep these elements, we configure all in configure_transport when the * server told us to really use the UDP ports. */ stream->udpsrc[0] = gst_object_ref (udpsrc0); stream->udpsrc[1] = gst_object_ref (udpsrc1); /* they are ours now */ gst_object_sink (udpsrc0); gst_object_sink (udpsrc1); return TRUE; /* ERRORS */ no_udp_protocol: { GST_DEBUG_OBJECT (src, "could not get UDP source"); goto cleanup; } start_udp_failure: { GST_DEBUG_OBJECT (src, "could not start UDP source"); goto cleanup; } no_ports: { GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries", count); goto cleanup; } no_udp_rtcp_protocol: { GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP"); goto cleanup; } start_rtcp_failure: { GST_DEBUG_OBJECT (src, "could not start UDP source for RTCP"); goto cleanup; } port_error: { GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d", tmp_rtp, *rtpport, tmp_rtcp, *rtcpport); goto cleanup; } cleanup: { if (tmp) { gst_element_set_state (tmp, GST_STATE_NULL); gst_object_unref (tmp); } 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); } return FALSE; } } static void gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush) { GstEvent *event; gint cmd, i; GstState state; GList *walk; GstClock *clock; GstClockTime base_time = GST_CLOCK_TIME_NONE; if (flush) { event = gst_event_new_flush_start (); GST_DEBUG_OBJECT (src, "start flush"); cmd = CMD_STOP; state = GST_STATE_PAUSED; } else { event = gst_event_new_flush_stop (); GST_DEBUG_OBJECT (src, "stop flush"); cmd = CMD_WAIT; state = GST_STATE_PLAYING; clock = gst_element_get_clock (GST_ELEMENT_CAST (src)); if (clock) { base_time = gst_clock_get_time (clock); gst_object_unref (clock); } } gst_rtspsrc_push_event (src, event); gst_rtspsrc_loop_send_cmd (src, cmd, flush); /* */ for (walk = src->streams; walk; walk = g_list_next (walk)) { GstRTSPStream *stream = (GstRTSPStream *) walk->data; for (i = 0; i < 2; i++) { if (stream->udpsrc[i]) { if (base_time != -1) gst_element_set_base_time (stream->udpsrc[i], base_time); gst_element_set_state (stream->udpsrc[i], state); } } } } static GstRTSPResult gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPMessage * message, GTimeVal * timeout) { GstRTSPResult ret; GST_RTSP_CONN_LOCK (src); ret = gst_rtsp_connection_send (src->connection, message, timeout); GST_RTSP_CONN_UNLOCK (src); return ret; } static GstRTSPResult gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPMessage * message, GTimeVal * timeout) { GstRTSPResult ret; GST_RTSP_CONN_LOCK (src); ret = gst_rtsp_connection_receive (src->connection, message, timeout); GST_RTSP_CONN_UNLOCK (src); return ret; } static gboolean gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment) { gboolean res; src->state = GST_RTSP_STATE_SEEKING; /* PLAY will add the range header now. */ src->need_range = TRUE; res = gst_rtspsrc_play (src, segment); return res; } static gboolean gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event) { gboolean res; gdouble rate; GstFormat format; GstSeekFlags flags; GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type; gint64 cur, stop; gboolean flush; gboolean update; GstSegment seeksegment = { 0, }; if (event) { GST_DEBUG_OBJECT (src, "doing seek with event"); gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, &stop_type, &stop); /* no negative rates yet */ if (rate < 0.0) goto negative_rate; /* we need TIME format */ if (format != src->segment.format) goto no_format; } else { GST_DEBUG_OBJECT (src, "doing seek without event"); flags = 0; cur_type = GST_SEEK_TYPE_SET; stop_type = GST_SEEK_TYPE_SET; } /* get flush flag */ flush = flags & GST_SEEK_FLAG_FLUSH; /* now we need to make sure the streaming thread is stopped. We do this by * either sending a FLUSH_START event downstream which will cause the * streaming thread to stop with a WRONG_STATE. * For a non-flushing seek we simply pause the task, which will happen as soon * as it completes one iteration (and thus might block when the sink is * blocking in preroll). */ if (flush) { GST_DEBUG_OBJECT (src, "starting flush"); gst_rtspsrc_flush (src, TRUE); } else { if (src->task) { gst_task_pause (src->task); } } /* we should now be able to grab the streaming thread because we stopped it * with the above flush/pause code */ GST_RTSP_STREAM_LOCK (src); /* stop flushing state */ gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, FALSE); GST_DEBUG_OBJECT (src, "stopped streaming"); /* copy segment, we need this because we still need the old * segment when we close the current segment. */ memcpy (&seeksegment, &src->segment, sizeof (GstSegment)); /* configure the seek parameters in the seeksegment. We will then have the * right values in the segment to perform the seek */ if (event) { GST_DEBUG_OBJECT (src, "configuring seek"); gst_segment_set_seek (&seeksegment, rate, format, flags, cur_type, cur, stop_type, stop, &update); } /* figure out the last position we need to play. If it's configured (stop != * -1), use that, else we play until the total duration of the file */ if ((stop = seeksegment.stop) == -1) stop = seeksegment.duration; res = gst_rtspsrc_do_seek (src, &seeksegment); /* prepare for streaming again */ if (flush) { /* if we started flush, we stop now */ GST_DEBUG_OBJECT (src, "stopping flush"); gst_rtspsrc_flush (src, FALSE); } else if (src->running) { /* we are running the current segment and doing a non-flushing seek, * close the segment first based on the previous last_stop. */ GST_DEBUG_OBJECT (src, "closing running segment %" G_GINT64_FORMAT " to %" G_GINT64_FORMAT, src->segment.accum, src->segment.last_stop); /* queue the segment for sending in the stream thread */ if (src->close_segment) gst_event_unref (src->close_segment); src->close_segment = gst_event_new_new_segment (TRUE, src->segment.rate, src->segment.format, src->segment.accum, src->segment.last_stop, src->segment.accum); /* keep track of our last_stop */ seeksegment.accum = src->segment.last_stop; } /* now we did the seek and can activate the new segment values */ memcpy (&src->segment, &seeksegment, sizeof (GstSegment)); /* if we're doing a segment seek, post a SEGMENT_START message */ if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) { gst_element_post_message (GST_ELEMENT_CAST (src), gst_message_new_segment_start (GST_OBJECT_CAST (src), src->segment.format, src->segment.last_stop)); } /* now create the newsegment */ GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT " to %" G_GINT64_FORMAT, src->segment.last_stop, stop); /* store the newsegment event so it can be sent from the streaming thread. */ if (src->start_segment) gst_event_unref (src->start_segment); src->start_segment = gst_event_new_new_segment (FALSE, src->segment.rate, src->segment.format, src->segment.last_stop, stop, src->segment.last_stop); /* mark discont */ GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position"); GST_RTSP_STREAM_UNLOCK (src); return TRUE; /* ERRORS */ negative_rate: { GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet."); return FALSE; } no_format: { GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted."); return FALSE; } } static gboolean gst_rtspsrc_handle_src_event (GstPad * pad, GstEvent * event) { GstRTSPSrc *src; gboolean res = FALSE; src = GST_RTSPSRC_CAST (gst_pad_get_parent (pad)); GST_DEBUG_OBJECT (src, "pad %s:%s received event %s", GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_QOS: break; case GST_EVENT_SEEK: res = gst_rtspsrc_perform_seek (src, event); break; case GST_EVENT_NAVIGATION: break; case GST_EVENT_LATENCY: break; default: break; } gst_event_unref (event); gst_object_unref (src); return res; } /* this is the final query function we receive on the internal source pad when * we deal with TCP connections */ static gboolean gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstQuery * query) { GstRTSPSrc *src; gboolean res = TRUE; src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad)); GST_DEBUG_OBJECT (src, "pad %s:%s received query %s", GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query)); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_POSITION: { /* no idea */ break; } case GST_QUERY_DURATION: { GstFormat format; gst_query_parse_duration (query, &format, NULL); switch (format) { case GST_FORMAT_TIME: gst_query_set_duration (query, format, src->segment.duration); break; default: res = FALSE; break; } break; } case GST_QUERY_LATENCY: { /* we are live with a min latency of 0 and unlimited max latency, this * result will be updated by the session manager if there is any. */ gst_query_set_latency (query, TRUE, 0, -1); break; } default: break; } return res; } /* this query is executed on the ghost source pad exposed on rtspsrc. */ static gboolean gst_rtspsrc_handle_src_query (GstPad * pad, GstQuery * query) { GstRTSPSrc *src; gboolean res = FALSE; src = GST_RTSPSRC_CAST (gst_pad_get_parent (pad)); GST_DEBUG_OBJECT (src, "pad %s:%s received query %s", GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query)); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_DURATION: { GstFormat format; gst_query_parse_duration (query, &format, NULL); switch (format) { case GST_FORMAT_TIME: gst_query_set_duration (query, format, src->segment.duration); res = TRUE; break; default: break; } break; } default: { GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)); /* forward the query to the proxy target pad */ if (target) { res = gst_pad_query (target, query); gst_object_unref (target); } break; } } gst_object_unref (src); return res; } /* callback for RTCP messages to be sent to the server when operating in TCP * mode. */ static GstFlowReturn gst_rtspsrc_sink_chain (GstPad * pad, GstBuffer * buffer) { GstRTSPSrc *src; GstRTSPStream *stream; GstFlowReturn res = GST_FLOW_OK; guint8 *data; guint size; GstRTSPResult ret; GstRTSPMessage message = { 0 }; stream = (GstRTSPStream *) gst_pad_get_element_private (pad); src = stream->parent; data = GST_BUFFER_DATA (buffer); size = GST_BUFFER_SIZE (buffer); gst_rtsp_message_init_data (&message, stream->channel[1]); /* lend the body data to the message */ gst_rtsp_message_take_body (&message, data, size); GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size); ret = gst_rtspsrc_connection_send (src, &message, NULL); GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret); /* and steal it away again because we will free it when unreffing the * buffer */ gst_rtsp_message_steal_body (&message, &data, &size); gst_rtsp_message_unset (&message); gst_buffer_unref (buffer); return res; } static void pad_unblocked (GstPad * pad, gboolean blocked, GstRTSPSrc * src) { GST_DEBUG_OBJECT (src, "pad %s:%s unblocked", GST_DEBUG_PAD_NAME (pad)); } static void pad_blocked (GstPad * pad, gboolean blocked, GstRTSPSrc * src) { GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams", GST_DEBUG_PAD_NAME (pad)); /* activate the streams */ GST_OBJECT_LOCK (src); if (!src->need_activate) goto was_ok; src->need_activate = FALSE; GST_OBJECT_UNLOCK (src); gst_rtspsrc_activate_streams (src); return; was_ok: { GST_OBJECT_UNLOCK (src); return; } } /* this callback is called when the session manager generated a new src pad with * payloaded RTP packets. We simply ghost the pad here. */ static void new_session_pad (GstElement * session, GstPad * pad, GstRTSPSrc * src) { gchar *name; GstPadTemplate *template; gint id, ssrc, pt; GList *lstream; GstRTSPStream *stream; gboolean all_added; GST_DEBUG_OBJECT (src, "got new session pad %" GST_PTR_FORMAT, pad); GST_RTSP_STATE_LOCK (src); /* find stream */ name = gst_object_get_name (GST_OBJECT_CAST (pad)); if (sscanf (name, "recv_rtp_src_%d_%d_%d", &id, &ssrc, &pt) != 3) goto unknown_stream; GST_DEBUG_OBJECT (src, "stream: %u, SSRC %d, PT %d", id, ssrc, pt); stream = find_stream (src, GINT_TO_POINTER (id), (gpointer) find_stream_by_id); if (stream == NULL) goto unknown_stream; /* create a new pad we will use to stream to */ template = gst_static_pad_template_get (&rtptemplate); stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template); gst_object_unref (template); g_free (name); stream->added = TRUE; gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event); gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query); gst_pad_set_active (stream->srcpad, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad); /* check if we added all streams */ all_added = TRUE; for (lstream = src->streams; lstream; lstream = g_list_next (lstream)) { stream = (GstRTSPStream *) lstream->data; /* a container stream only needs one pad added. Also disabled streams don't * count */ if (!stream->container && !stream->disabled && !stream->added) { all_added = FALSE; break; } } GST_RTSP_STATE_UNLOCK (src); if (all_added) { GST_DEBUG_OBJECT (src, "We added all streams"); /* when we get here, all stream are added and we can fire the no-more-pads * signal. */ gst_element_no_more_pads (GST_ELEMENT_CAST (src)); } return; /* ERRORS */ unknown_stream: { GST_DEBUG_OBJECT (src, "ignoring unknown stream"); GST_RTSP_STATE_UNLOCK (src); g_free (name); return; } } static GstCaps * request_pt_map (GstElement * sess, guint session, guint pt, GstRTSPSrc * src) { GstRTSPStream *stream; GstCaps *caps; GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session); GST_RTSP_STATE_LOCK (src); stream = find_stream (src, GINT_TO_POINTER (session), (gpointer) find_stream_by_id); if (!stream) goto unknown_stream; caps = stream->caps; if (caps) gst_caps_ref (caps); GST_RTSP_STATE_UNLOCK (src); return caps; unknown_stream: { GST_DEBUG_OBJECT (src, "unknown stream %d", session); GST_RTSP_STATE_UNLOCK (src); return NULL; } } static void gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, guint session) { GstRTSPStream *stream; GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", session); /* get stream for session */ stream = find_stream (src, GINT_TO_POINTER (session), (gpointer) find_stream_by_id); if (!stream) goto unknown_stream; if (stream->eos) goto was_eos; stream->eos = TRUE; gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ()); return; /* ERRORS */ unknown_stream: { GST_DEBUG_OBJECT (src, "unknown stream for session %u", session); return; } was_eos: { GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", session); return; } } static void on_bye_ssrc (GstElement * manager, guint session, guint32 ssrc, GstRTSPSrc * src) { GST_DEBUG_OBJECT (src, "SSRC %08x in session %u received BYE", ssrc, session); gst_rtspsrc_do_stream_eos (src, session); } static void on_timeout (GstElement * manager, guint session, guint32 ssrc, GstRTSPSrc * src) { GST_DEBUG_OBJECT (src, "SSRC %08x in session %u timed out", ssrc, session); gst_rtspsrc_do_stream_eos (src, session); } /* try to get and configure a manager */ static gboolean gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream, GstRTSPTransport * transport) { const gchar *manager; gchar *name; GstRTSPResult res; GstStateChangeReturn ret; /* find a manager */ if ((res = gst_rtsp_transport_get_manager (transport->trans, &manager, 0)) < 0) goto no_manager; if (manager) { GST_DEBUG_OBJECT (src, "using manager %s", manager); /* configure the manager */ if (src->session == NULL) { if (!(src->session = gst_element_factory_make (manager, NULL))) { /* fallback */ if ((res = gst_rtsp_transport_get_manager (transport->trans, &manager, 1)) < 0) goto no_manager; if (!manager) goto use_no_manager; if (!(src->session = gst_element_factory_make (manager, NULL))) goto manager_failed; } /* we manage this element */ gst_bin_add (GST_BIN_CAST (src), src->session); ret = gst_element_set_state (src->session, GST_STATE_PAUSED); if (ret == GST_STATE_CHANGE_FAILURE) goto start_session_failure; g_object_set (src->session, "latency", src->latency, NULL); /* connect to signals if we did not already do so */ GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p", stream); src->session_sig_id = g_signal_connect (src->session, "pad-added", (GCallback) new_session_pad, src); src->session_ptmap_id = g_signal_connect (src->session, "request-pt-map", (GCallback) request_pt_map, src); g_signal_connect (src->session, "on-bye-ssrc", (GCallback) on_bye_ssrc, src); g_signal_connect (src->session, "on-bye-timeout", (GCallback) on_timeout, src); g_signal_connect (src->session, "on-timeout", (GCallback) on_timeout, src); } /* we stream directly to the manager, get some pads. Each RTSP stream goes * into a separate RTP session. */ name = g_strdup_printf ("recv_rtp_sink_%d", stream->id); stream->channelpad[0] = gst_element_get_request_pad (src->session, name); g_free (name); name = g_strdup_printf ("recv_rtcp_sink_%d", stream->id); stream->channelpad[1] = gst_element_get_request_pad (src->session, name); g_free (name); } use_no_manager: return TRUE; /* ERRORS */ no_manager: { GST_DEBUG_OBJECT (src, "cannot get a session manager"); return FALSE; } manager_failed: { GST_DEBUG_OBJECT (src, "no session manager element %s found", manager); return FALSE; } start_session_failure: { GST_DEBUG_OBJECT (src, "could not start session"); return FALSE; } } /* free the UDP sources allocated when negotiating a transport. * This function is called when the server negotiated to a transport where the * UDP sources are not needed anymore, such as TCP or multicast. */ static void gst_rtspsrc_stream_free_udp (GstRTSPStream * stream) { gint i; for (i = 0; i < 2; i++) { if (stream->udpsrc[i]) { gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL); gst_object_unref (stream->udpsrc[i]); stream->udpsrc[i] = NULL; } } } /* for TCP, create pads to send and receive data to and from the manager and to * intercept various events and queries */ static gboolean gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream, GstRTSPTransport * transport, GstPad ** outpad) { gchar *name; GstPadTemplate *template; GstPad *pad0, *pad1; /* configure for interleaved delivery, nothing needs to be done * here, the loop function will call the chain functions of the * session manager. */ stream->channel[0] = transport->interleaved.min; stream->channel[1] = transport->interleaved.max; GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream, stream->channel[0], stream->channel[1]); /* we can remove the allocated UDP ports now */ gst_rtspsrc_stream_free_udp (stream); /* no session manager, send data to srcpad directly */ if (!stream->channelpad[0]) { GST_DEBUG_OBJECT (src, "no manager, creating pad"); /* create a new pad we will use to stream to */ name = g_strdup_printf ("stream%d", stream->id); template = gst_static_pad_template_get (&rtptemplate); stream->channelpad[0] = gst_pad_new_from_template (template, name); gst_object_unref (template); g_free (name); /* set caps and activate */ gst_pad_use_fixed_caps (stream->channelpad[0]); gst_pad_set_active (stream->channelpad[0], TRUE); *outpad = gst_object_ref (stream->channelpad[0]); } else { GST_DEBUG_OBJECT (src, "using manager source pad"); template = gst_static_pad_template_get (&anysrctemplate); /* allocate pads for sending the channel data into the manager */ pad0 = gst_pad_new_from_template (template, "internalsrc0"); gst_pad_link (pad0, stream->channelpad[0]); stream->channelpad[0] = pad0; gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query); gst_pad_set_element_private (pad0, src); gst_pad_set_active (pad0, TRUE); if (stream->channelpad[1]) { /* if we have a sinkpad for the other channel, create a pad and link to the * manager. */ pad1 = gst_pad_new_from_template (template, "internalsrc1"); gst_pad_link (pad1, stream->channelpad[1]); stream->channelpad[1] = pad1; gst_pad_set_active (pad1, TRUE); } gst_object_unref (template); } /* setup RTCP transport back to the server */ if (src->session) { GstPad *pad; template = gst_static_pad_template_get (&anysinktemplate); stream->rtcppad = gst_pad_new_from_template (template, "internalsink0"); gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain); gst_pad_set_element_private (stream->rtcppad, stream); gst_pad_set_active (stream->rtcppad, TRUE); /* get session RTCP pad */ name = g_strdup_printf ("send_rtcp_src_%d", stream->id); pad = gst_element_get_request_pad (src->session, name); g_free (name); /* and link */ if (pad) gst_pad_link (pad, stream->rtcppad); gst_object_unref (template); } return TRUE; } /* For multicast create UDP sources and join the multicast group. */ static gboolean gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream, GstRTSPTransport * transport, GstPad ** outpad) { gchar *uri; GST_DEBUG_OBJECT (src, "creating UDP sources for multicast"); /* we can remove the allocated UDP ports now */ gst_rtspsrc_stream_free_udp (stream); /* creating UDP source */ if (transport->port.min != -1) { uri = g_strdup_printf ("udp://%s:%d", transport->destination, transport->port.min); stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL); g_free (uri); if (stream->udpsrc[0] == NULL) goto no_element; /* take ownership */ gst_object_ref (stream->udpsrc[0]); gst_object_sink (stream->udpsrc[0]); /* change state */ gst_element_set_state (stream->udpsrc[0], GST_STATE_READY); } /* creating another UDP source */ if (transport->port.max != -1) { uri = g_strdup_printf ("udp://%s:%d", transport->destination, transport->port.max); stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL); g_free (uri); if (stream->udpsrc[1] == NULL) goto no_element; /* take ownership */ gst_object_ref (stream->udpsrc[1]); gst_object_sink (stream->udpsrc[1]); gst_element_set_state (stream->udpsrc[1], GST_STATE_READY); } return TRUE; /* ERRORS */ no_element: { GST_DEBUG_OBJECT (src, "no UDP source element found"); return FALSE; } } /* configure the remainder of the UDP ports */ static gboolean gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream, GstRTSPTransport * transport, GstPad ** outpad) { /* we manage the UDP elements now. For unicast, the UDP sources where * allocated in the stream when we suggested a transport. */ if (stream->udpsrc[0]) { gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]); GST_DEBUG_OBJECT (src, "setting up UDP source"); /* configure a timeout on the UDP port. When the timeout message is * posted, we assume UDP transport is not possible. We reconnect using TCP * if we can. */ g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->udp_timeout, NULL); /* get output pad of the UDP source. */ *outpad = gst_element_get_pad (stream->udpsrc[0], "src"); /* save it so we can unblock */ stream->blockedpad = *outpad; /* configure pad block on the pad. As soon as there is dataflow on the * UDP source, we know that UDP is not blocked by a firewall and we can * configure all the streams to let the application autoplug decoders. */ gst_pad_set_blocked_async (stream->blockedpad, TRUE, (GstPadBlockCallback) pad_blocked, src); if (stream->channelpad[0]) { GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager"); /* configure for UDP delivery, we need to connect the UDP pads to * the session plugin. */ gst_pad_link (*outpad, stream->channelpad[0]); gst_object_unref (*outpad); *outpad = NULL; /* we connected to pad-added signal to get pads from the manager */ } else { GST_DEBUG_OBJECT (src, "using UDP src pad as output"); } } /* RTCP port */ if (stream->udpsrc[1]) { gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]); if (stream->channelpad[1]) { GstPad *pad; GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager"); pad = gst_element_get_pad (stream->udpsrc[1], "src"); gst_pad_link (pad, stream->channelpad[1]); gst_object_unref (pad); } else { /* leave unlinked */ } } return TRUE; } /* configure the UDP sink back to the server for status reports */ static gboolean gst_rtspsrc_stream_configure_udp_sink (GstRTSPSrc * src, GstRTSPStream * stream, GstRTSPTransport * transport) { GstPad *pad; gint port, sockfd = -1; gchar *destination, *uri, *name; /* no session, we're done */ if (src->session == NULL) return TRUE; /* get host and port */ if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) port = transport->port.max; else port = transport->server_port.max; /* first take the source, then the endpoint to figure out where to send * the RTCP. */ destination = transport->source; if (destination == NULL) destination = src->connection->ip; GST_DEBUG_OBJECT (src, "configure UDP sink for %s:%d", destination, port); uri = g_strdup_printf ("udp://%s:%d", destination, port); stream->udpsink = gst_element_make_from_uri (GST_URI_SINK, uri, NULL); g_free (uri); if (stream->udpsink == NULL) goto no_sink_element; /* no sync needed */ g_object_set (G_OBJECT (stream->udpsink), "sync", FALSE, NULL); /* no async state changes needed */ g_object_set (G_OBJECT (stream->udpsink), "async", FALSE, NULL); if (stream->udpsrc[1]) { /* configure socket, we give it the same UDP socket as the udpsrc for RTCP * because some servers check the port number of where it sends RTCP to identify * the RTCP packets it receives */ g_object_get (G_OBJECT (stream->udpsrc[1]), "sock", &sockfd, NULL); GST_DEBUG_OBJECT (src, "UDP src has sock %d", sockfd); /* configure socket and make sure udpsink does not close it when shutting * down, it belongs to udpsrc after all. */ g_object_set (G_OBJECT (stream->udpsink), "sockfd", sockfd, NULL); g_object_set (G_OBJECT (stream->udpsink), "closefd", FALSE, NULL); } /* we keep this playing always */ gst_element_set_locked_state (stream->udpsink, TRUE); gst_element_set_state (stream->udpsink, GST_STATE_PLAYING); gst_object_ref (stream->udpsink); gst_bin_add (GST_BIN_CAST (src), stream->udpsink); stream->rtcppad = gst_element_get_pad (stream->udpsink, "sink"); /* get session RTCP pad */ name = g_strdup_printf ("send_rtcp_src_%d", stream->id); pad = gst_element_get_request_pad (src->session, name); g_free (name); /* and link */ if (pad) gst_pad_link (pad, stream->rtcppad); return TRUE; /* ERRORS */ no_sink_element: { GST_DEBUG_OBJECT (src, "no UDP sink element found"); return FALSE; } } /* sets up all elements needed for streaming over the specified transport. * Does not yet expose the element pads, this will be done when there is actuall * dataflow detected, which might never happen when UDP is blocked in a * firewall, for example. */ static gboolean gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream, GstRTSPTransport * transport) { GstRTSPSrc *src; GstPad *outpad = NULL; GstPadTemplate *template; gchar *name; GstStructure *s; const gchar *mime; GstRTSPResult res; src = stream->parent; GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream); s = gst_caps_get_structure (stream->caps, 0); /* get the proper mime type for this stream now */ if ((res = gst_rtsp_transport_get_mime (transport->trans, &mime)) < 0) goto unknown_transport; if (!mime) goto unknown_transport; /* configure the final mime type */ GST_DEBUG_OBJECT (src, "setting mime to %s", mime); gst_structure_set_name (s, mime); /* try to get and configure a manager, channelpad[0-1] will be configured with * the pads for the manager, or NULL when no manager is needed. */ if (!gst_rtspsrc_stream_configure_manager (src, stream, transport)) goto no_manager; switch (transport->lower_transport) { case GST_RTSP_LOWER_TRANS_TCP: if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad)) goto transport_failed; break; case GST_RTSP_LOWER_TRANS_UDP_MCAST: if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad)) goto transport_failed; /* fallthrough, the rest is the same for UDP and MCAST */ case GST_RTSP_LOWER_TRANS_UDP: if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad)) goto transport_failed; /* configure udpsink back to the server for RTCP messages. */ if (!gst_rtspsrc_stream_configure_udp_sink (src, stream, transport)) goto transport_failed; break; default: goto unknown_transport; } if (outpad) { GST_DEBUG_OBJECT (src, "creating ghostpad"); gst_pad_use_fixed_caps (outpad); /* create ghostpad, don't add just yet, this will be done when we activate * the stream. */ name = g_strdup_printf ("stream%d", stream->id); template = gst_static_pad_template_get (&rtptemplate); stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template); gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event); gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query); gst_object_unref (template); g_free (name); gst_object_unref (outpad); } /* mark pad as ok */ stream->last_ret = GST_FLOW_OK; return TRUE; /* ERRORS */ transport_failed: { GST_DEBUG_OBJECT (src, "failed to configure transport"); return FALSE; } unknown_transport: { GST_DEBUG_OBJECT (src, "unknown transport"); return FALSE; } no_manager: { GST_DEBUG_OBJECT (src, "cannot get a session manager"); return FALSE; } } /* Adds the source pads of all configured streams to the element. * This code is performed when we detected dataflow. * * We detect dataflow from either the _loop function or with pad probes on the * udp sources. */ static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src) { GList *walk; GST_DEBUG_OBJECT (src, "activating streams"); for (walk = src->streams; walk; walk = g_list_next (walk)) { GstRTSPStream *stream = (GstRTSPStream *) walk->data; if (stream->udpsrc[0]) { /* remove timeout, we are streaming now and timeouts will be handled by * the session manager and jitter buffer */ g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL); } if (stream->srcpad) { /* if we don't have a session manager, set the caps now. If we have a * session, we will get a notification of the pad and the caps. */ if (!src->session) { GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream); gst_pad_set_caps (stream->srcpad, stream->caps); } GST_DEBUG_OBJECT (src, "activating stream pad %p", stream); gst_pad_set_active (stream->srcpad, TRUE); /* add the pad */ if (!stream->added) { GST_DEBUG_OBJECT (src, "adding stream pad %p", stream); gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad); stream->added = TRUE; } } } /* unblock all pads */ for (walk = src->streams; walk; walk = g_list_next (walk)) { GstRTSPStream *stream = (GstRTSPStream *) walk->data; if (stream->blockedpad) { GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream); gst_pad_set_blocked_async (stream->blockedpad, FALSE, (GstPadBlockCallback) pad_unblocked, src); stream->blockedpad = NULL; } } return TRUE; } static void gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment) { GList *walk; guint64 start, stop; gdouble play_speed, play_scale; GST_DEBUG_OBJECT (src, "configuring stream caps"); start = segment->last_stop; stop = segment->duration; play_speed = segment->rate; play_scale = segment->applied_rate; for (walk = src->streams; walk; walk = g_list_next (walk)) { GstRTSPStream *stream = (GstRTSPStream *) walk->data; GstCaps *caps; if ((caps = stream->caps)) { caps = gst_caps_make_writable (caps); /* update caps */ if (stream->timebase != -1) gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT, (guint) stream->timebase, NULL); if (stream->seqbase != -1) gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT, (guint) stream->seqbase, NULL); gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL); if (stop != -1) gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL); gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL); gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL); stream->caps = caps; } GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps); } if (src->session) { GST_DEBUG_OBJECT (src, "clear session"); g_signal_emit_by_name (src->session, "clear-pt-map", NULL); } } static GstFlowReturn gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream, GstFlowReturn ret) { GList *streams; /* store the value */ stream->last_ret = ret; /* if it's success we can return the value right away */ if (GST_FLOW_IS_SUCCESS (ret)) goto done; /* any other error that is not-linked can be returned right * away */ if (ret != GST_FLOW_NOT_LINKED) goto done; /* only return NOT_LINKED if all other pads returned NOT_LINKED */ for (streams = src->streams; streams; streams = g_list_next (streams)) { GstRTSPStream *ostream = (GstRTSPStream *) streams->data; ret = ostream->last_ret; /* some other return value (must be SUCCESS but we can return * other values as well) */ if (ret != GST_FLOW_NOT_LINKED) goto done; } /* if we get here, all other pads were unlinked and we return * NOT_LINKED then */ done: return ret; } static void gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream, GstEvent * event) { /* only streams that have a connection to the outside world */ if (stream->srcpad == NULL) goto done; if (stream->channelpad[0]) { gst_event_ref (event); if (GST_PAD_IS_SRC (stream->channelpad[0])) gst_pad_push_event (stream->channelpad[0], event); else gst_pad_send_event (stream->channelpad[0], event); } if (stream->channelpad[1]) { gst_event_ref (event); if (GST_PAD_IS_SRC (stream->channelpad[1])) gst_pad_push_event (stream->channelpad[1], event); else gst_pad_send_event (stream->channelpad[1], event); } done: gst_event_unref (event); } static void gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event) { GList *streams; for (streams = src->streams; streams; streams = g_list_next (streams)) { GstRTSPStream *ostream = (GstRTSPStream *) streams->data; gst_event_ref (event); gst_rtspsrc_stream_push_event (src, ostream, event); } gst_event_unref (event); } /* FIXME, handle server request, reply with OK, for now */ static GstRTSPResult gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPMessage * request) { GstRTSPMessage response = { 0 }; GstRTSPResult res; GST_DEBUG_OBJECT (src, "got server request message"); if (src->debug) gst_rtsp_message_dump (request); res = gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK", request); if (res < 0) goto send_error; GST_DEBUG_OBJECT (src, "replying with OK"); if (src->debug) gst_rtsp_message_dump (&response); res = gst_rtspsrc_connection_send (src, &response, NULL); if (res < 0) goto send_error; return GST_RTSP_OK; /* ERRORS */ send_error: { return res; } } /* send server keep-alive */ static GstRTSPResult gst_rtspsrc_send_keep_alive (GstRTSPSrc * src) { GstRTSPMessage request = { 0 }; GstRTSPResult res; GstRTSPMethod method; GST_DEBUG_OBJECT (src, "creating server keep-alive"); /* find a method to use for keep-alive */ if (src->methods & GST_RTSP_GET_PARAMETER) method = GST_RTSP_GET_PARAMETER; else method = GST_RTSP_OPTIONS; res = gst_rtsp_message_init_request (&request, method, src->req_location); if (res < 0) goto send_error; if (src->debug) gst_rtsp_message_dump (&request); res = gst_rtspsrc_connection_send (src, &request, NULL); if (res < 0) goto send_error; gst_rtsp_connection_reset_timeout (src->connection); gst_rtsp_message_unset (&request); return GST_RTSP_OK; /* ERRORS */ send_error: { gchar *str = gst_rtsp_strresult (res); gst_rtsp_message_unset (&request); GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL), ("Could not send keep-alive. (%s)", str)); g_free (str); return res; } } static GstFlowReturn gst_rtspsrc_loop_interleaved (GstRTSPSrc * src) { GstRTSPMessage message = { 0 }; GstRTSPResult res; gint channel; GstRTSPStream *stream; GstPad *outpad = NULL; guint8 *data; guint size; GstFlowReturn ret = GST_FLOW_OK; GstBuffer *buf; gboolean is_rtcp, have_data; /* here we are only interested in data messages */ have_data = FALSE; do { GTimeVal tv_timeout; /* get the next timeout interval */ gst_rtsp_connection_next_timeout (src->connection, &tv_timeout); /* see if the timeout period expired */ if ((tv_timeout.tv_usec | tv_timeout.tv_usec) == 0) { GST_DEBUG_OBJECT (src, "timout, sending keep-alive"); /* send keep-alive, ignore the result, a warning will be posted. */ res = gst_rtspsrc_send_keep_alive (src); } GST_DEBUG_OBJECT (src, "doing receive"); /* We need to check if playback has been paused while we have been * doing something else in our own GstTask (e.g. pushing buffer). There * is a slight chance that we have just received data buffer when PAUSE * state change happens (in another thread). In this case we well be * totally ignorant of that unless we explicitly check it here. */ GST_RTSP_STATE_LOCK (src); if (src->state == GST_RTSP_STATE_READY) { /* We are looping in a paused mode */ GST_RTSP_STATE_UNLOCK (src); goto already_paused; } /* protect the connection with the connection lock so that we can see when * we are finished doing server communication */ res = gst_rtspsrc_connection_receive (src, &message, src->ptcp_timeout); GST_RTSP_STATE_UNLOCK (src); switch (res) { case GST_RTSP_OK: GST_DEBUG_OBJECT (src, "we received a server message"); break; case GST_RTSP_EINTR: /* we got interrupted this means we need to stop */ goto interrupt; case GST_RTSP_ETIMEOUT: /* no reply, go EOS */ goto timeout; case GST_RTSP_EEOF: /* go EOS when the server closed the connection */ goto server_eof; default: goto receive_error; } switch (message.type) { case GST_RTSP_MESSAGE_REQUEST: /* server sends us a request message, handle it */ if ((res = gst_rtspsrc_handle_request (src, &message)) < 0) goto handle_request_failed; break; case GST_RTSP_MESSAGE_RESPONSE: /* we ignore response messages */ GST_DEBUG_OBJECT (src, "ignoring response message"); if (src->debug) gst_rtsp_message_dump (&message); break; case GST_RTSP_MESSAGE_DATA: GST_DEBUG_OBJECT (src, "got data message"); have_data = TRUE; break; default: GST_WARNING_OBJECT (src, "ignoring unknown message type %d", message.type); break; } } while (!have_data); channel = message.type_data.data.channel; stream = find_stream (src, GINT_TO_POINTER (channel), (gpointer) find_stream_by_channel); if (!stream) goto unknown_stream; if (channel == stream->channel[0]) { outpad = stream->channelpad[0]; is_rtcp = FALSE; } else if (channel == stream->channel[1]) { outpad = stream->channelpad[1]; is_rtcp = TRUE; } else { is_rtcp = FALSE; } /* take a look at the body to figure out what we have */ gst_rtsp_message_get_body (&message, &data, &size); if (size < 2) goto invalid_length; /* channels are not correct on some servers, do extra check */ if (data[1] >= 200 && data[1] <= 204) { /* hmm RTCP message switch to the RTCP pad of the same stream. */ outpad = stream->channelpad[1]; is_rtcp = TRUE; } /* we have no clue what this is, just ignore then. */ if (outpad == NULL) goto unknown_stream; /* take the message body for further processing */ gst_rtsp_message_steal_body (&message, &data, &size); /* strip the trailing \0 */ size -= 1; buf = gst_buffer_new (); GST_BUFFER_DATA (buf) = data; GST_BUFFER_MALLOCDATA (buf) = data; GST_BUFFER_SIZE (buf) = size; /* don't need message anymore */ gst_rtsp_message_unset (&message); GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size, channel); if (src->need_activate) { gst_rtspsrc_activate_streams (src); src->need_activate = FALSE; } if (!src->session) { /* set stream caps on buffer when we don't have a session manager to do it * for us */ gst_buffer_set_caps (buf, stream->caps); } if (src->base_time == -1) { /* Take current running_time. This timestamp will be put on * the first buffer of each stream because we are a live source and so we * timestamp with the running_time. When we are dealing with TCP, we also * only timestamp the first buffer (using the DISCONT flag) because a server * typically bursts data, for which we don't want to compensate by speeding * up the media. The other timestamps will be interpollated from this one * using the RTP timestamps. */ GST_OBJECT_LOCK (src); if (GST_ELEMENT_CLOCK (src)) { GstClockTime now = gst_clock_get_time (GST_ELEMENT_CLOCK (src)); src->base_time = now - GST_ELEMENT_CAST (src)->base_time; } GST_OBJECT_UNLOCK (src); } if (stream->discont && !is_rtcp) { /* mark first RTP buffer as discont */ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); stream->discont = FALSE; /* first buffer gets the timestamp, other buffers are not timestamped and * their presentation time will be interpollated from the rtp timestamps. */ GST_BUFFER_TIMESTAMP (buf) = src->base_time; } /* chain to the peer pad */ if (GST_PAD_IS_SINK (outpad)) ret = gst_pad_chain (outpad, buf); else ret = gst_pad_push (outpad, buf); if (!is_rtcp) { /* combine all stream flows for the data transport */ ret = gst_rtspsrc_combine_flows (src, stream, ret); } return ret; /* ERRORS */ unknown_stream: { GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel); gst_rtsp_message_unset (&message); return GST_FLOW_OK; } timeout: { GST_DEBUG_OBJECT (src, "we got a timeout"); GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL), ("Timeout while waiting for server message.")); gst_rtsp_message_unset (&message); return GST_FLOW_UNEXPECTED; } server_eof: { GST_DEBUG_OBJECT (src, "we got an eof from the server"); GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL), ("The server closed the connection.")); return GST_FLOW_UNEXPECTED; } interrupt: { gst_rtsp_message_unset (&message); GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush"); /* unset flushing so we can do something else */ gst_rtsp_connection_flush (src->connection, FALSE); return GST_FLOW_WRONG_STATE; } already_paused: { GST_DEBUG_OBJECT (src, "got interrupted: playback already paused"); return GST_FLOW_WRONG_STATE; } receive_error: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Could not receive message. (%s)", str)); g_free (str); gst_rtsp_message_unset (&message); return GST_FLOW_ERROR; } handle_request_failed: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL), ("Could not handle server message. (%s)", str)); g_free (str); gst_rtsp_message_unset (&message); return GST_FLOW_ERROR; } invalid_length: { GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL), ("Short message received, ignoring.")); gst_rtsp_message_unset (&message); return GST_FLOW_OK; } } static GstFlowReturn gst_rtspsrc_loop_udp (GstRTSPSrc * src) { gboolean restart = FALSE; GstRTSPResult res; GST_OBJECT_LOCK (src); if (src->loop_cmd == CMD_STOP) goto stopping; while (src->loop_cmd == CMD_WAIT) { GST_OBJECT_UNLOCK (src); while (TRUE) { GstRTSPMessage message = { 0 }; GTimeVal tv_timeout; /* get the next timeout interval */ gst_rtsp_connection_next_timeout (src->connection, &tv_timeout); GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds", (gint) tv_timeout.tv_sec); /* we should continue reading the TCP socket because the server might * send us requests. When the session timeout expires, we need to send a * keep-alive request to keep the session open. */ res = gst_rtspsrc_connection_receive (src, &message, &tv_timeout); switch (res) { case GST_RTSP_OK: GST_DEBUG_OBJECT (src, "we received a server message"); break; case GST_RTSP_EINTR: /* we got interrupted, see what we have to do */ GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush"); /* unset flushing so we can do something else */ gst_rtsp_connection_flush (src->connection, FALSE); goto interrupt; case GST_RTSP_ETIMEOUT: /* send keep-alive, ignore the result, a warning will be posted. */ GST_DEBUG_OBJECT (src, "timout, sending keep-alive"); res = gst_rtspsrc_send_keep_alive (src); continue; case GST_RTSP_EEOF: /* server closed the connection. not very fatal for UDP, reconnect and * see what happens. */ GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL), ("The server closed the connection.")); gst_rtsp_connection_close (src->connection); gst_rtsp_connection_connect (src->connection, src->ptcp_timeout); continue; default: goto receive_error; } switch (message.type) { case GST_RTSP_MESSAGE_REQUEST: /* server sends us a request message, handle it */ if ((res = gst_rtspsrc_handle_request (src, &message)) < 0) goto handle_request_failed; break; case GST_RTSP_MESSAGE_RESPONSE: /* we ignore response and data messages */ GST_DEBUG_OBJECT (src, "ignoring response message"); if (src->debug) gst_rtsp_message_dump (&message); break; case GST_RTSP_MESSAGE_DATA: /* we ignore response and data messages */ GST_DEBUG_OBJECT (src, "ignoring data message"); break; default: GST_WARNING_OBJECT (src, "ignoring unknown message type %d", message.type); break; } } interrupt: GST_OBJECT_LOCK (src); GST_DEBUG_OBJECT (src, "we have command %d", src->loop_cmd); if (src->loop_cmd == CMD_STOP) goto stopping; } if (src->loop_cmd == CMD_RECONNECT) { /* when we get here we have to reconnect using tcp */ src->loop_cmd = CMD_WAIT; /* only restart when the pads were not yet activated, else we were * streaming over UDP */ restart = src->need_activate; } GST_OBJECT_UNLOCK (src); /* no need to restart, we're done */ if (!restart) goto done; /* We post a warning message now to inform the user * that nothing happened. It's most likely a firewall thing. */ GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL), ("Could not receive any UDP packets for %.4f seconds, maybe your " "firewall is blocking it. Retrying using a TCP connection.", gst_guint64_to_gdouble (src->udp_timeout / 1000000))); /* we can try only TCP now */ src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP; /* pause to prepare for a restart */ gst_rtspsrc_pause (src); if (src->task) { /* stop task, we cannot join as this would deadlock, the task will stop when * we exit this function below. */ gst_task_stop (src->task); /* and free the task so that _close will not stop/join it again. */ gst_object_unref (GST_OBJECT (src->task)); src->task = NULL; } /* close and cleanup our state */ gst_rtspsrc_close (src); /* see if we have TCP left to try */ if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP)) goto no_protocols; /* open new connection using tcp */ if (!gst_rtspsrc_open (src)) goto open_failed; /* start playback */ if (!gst_rtspsrc_play (src, &src->segment)) goto play_failed; done: return GST_FLOW_OK; /* ERRORS */ stopping: { GST_DEBUG_OBJECT (src, "we are stopping"); GST_OBJECT_UNLOCK (src); return GST_FLOW_WRONG_STATE; } receive_error: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Could not receive message. (%s)", str)); g_free (str); return GST_FLOW_ERROR; } handle_request_failed: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL), ("Could not handle server message. (%s)", str)); g_free (str); return GST_FLOW_ERROR; } no_protocols: { src->cur_protocols = 0; /* no transport possible, post an error and stop */ GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Could not connect to server, no protocols left")); return GST_FLOW_ERROR; } open_failed: { GST_DEBUG_OBJECT (src, "open failed"); return GST_FLOW_OK; } play_failed: { GST_DEBUG_OBJECT (src, "play failed"); return GST_FLOW_OK; } } static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gboolean flush) { GST_OBJECT_LOCK (src); src->loop_cmd = cmd; if (flush) { GST_DEBUG_OBJECT (src, "start connection flush"); gst_rtsp_connection_flush (src->connection, TRUE); } else { GST_DEBUG_OBJECT (src, "stop connection flush"); gst_rtsp_connection_flush (src->connection, FALSE); } GST_OBJECT_UNLOCK (src); } static void gst_rtspsrc_loop (GstRTSPSrc * src) { GstFlowReturn ret; if (src->interleaved) ret = gst_rtspsrc_loop_interleaved (src); else ret = gst_rtspsrc_loop_udp (src); if (ret != GST_FLOW_OK) goto pause; return; /* ERRORS */ pause: { const gchar *reason = gst_flow_get_name (ret); GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason); src->running = FALSE; gst_task_pause (src->task); if (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED) { if (ret == GST_FLOW_UNEXPECTED) { /* perform EOS logic */ if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) { gst_element_post_message (GST_ELEMENT_CAST (src), gst_message_new_segment_done (GST_OBJECT_CAST (src), src->segment.format, src->segment.last_stop)); } else { gst_rtspsrc_push_event (src, gst_event_new_eos ()); } } else { /* for fatal errors we post an error message, post the error before the * EOS so the app knows about the error first. */ GST_ELEMENT_ERROR (src, STREAM, FAILED, ("Internal data flow error."), ("streaming task paused, reason %s (%d)", reason, ret)); gst_rtspsrc_push_event (src, gst_event_new_eos ()); } } return; } } #ifndef GST_DISABLE_GST_DEBUG const gchar * gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method) { gint index = 0; while (method != 0) { index++; method >>= 1; } switch (index) { case 0: return "None"; case 1: return "Basic"; case 2: return "Digest"; } return "Unknown"; } #endif /* Parse a WWW-Authenticate Response header and determine the * available authentication methods * FIXME: To implement digest or other auth types, we should extract * the authentication tokens that the server provided for each method * into an array of structures and give those to the connection object. * * This code should also cope with the fact that each WWW-Authenticate * header can contain multiple challenge methods + tokens * * At the moment, we just do a minimal check for Basic auth and don't * even parse out the realm */ static void gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods) { gchar *start; g_return_if_fail (hdr != NULL); g_return_if_fail (methods != NULL); /* Skip whitespace at the start of the string */ for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++); if (g_ascii_strncasecmp (start, "basic", 5) == 0) *methods |= GST_RTSP_AUTH_BASIC; } /** * gst_rtspsrc_setup_auth: * @src: the rtsp source * * Configure a username and password and auth method on the * connection object based on a response we received from the * peer. * * Currently, this requires that a username and password were supplied * in the uri. In the future, they may be requested on demand by sending * a message up the bus. * * Returns: TRUE if authentication information could be set up correctly. */ static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response) { gchar *user = NULL; gchar *pass = NULL; GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE; GstRTSPAuthMethod method; GstRTSPResult auth_result; gchar *hdr; /* Identify the available auth methods and see if any are supported */ if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE, &hdr, 0) == GST_RTSP_OK) { gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods); } if (avail_methods == GST_RTSP_AUTH_NONE) goto no_auth_available; /* FIXME: For digest auth, if the response indicates that the session * data are stale, we just update them in the connection object and * return TRUE to retry the request */ /* Do we have username and password available? */ if (src->url != NULL && !src->tried_url_auth) { user = src->url->user; pass = src->url->passwd; src->tried_url_auth = TRUE; GST_DEBUG_OBJECT (src, "Attempting authentication using credentials from the URL"); } /* FIXME: If the url didn't contain username and password or we tried them * already, request a username and passwd from the application via some kind * of credentials request message */ /* If we don't have a username and passwd at this point, bail out. */ if (user == NULL || pass == NULL) goto no_user_pass; /* Try to configure for each available authentication method, strongest to * weakest */ for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) { /* Check if this method is available on the server */ if ((method & avail_methods) == 0) continue; /* Pass the credentials to the connection to try on the next request */ auth_result = gst_rtsp_connection_set_auth (src->connection, method, user, pass); /* INVAL indicates an invalid username/passwd were supplied, so we'll just * ignore it and end up retrying later */ if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) { GST_DEBUG_OBJECT (src, "Attempting %s authentication", gst_rtsp_auth_method_to_string (method)); break; } } if (method == GST_RTSP_AUTH_NONE) goto no_auth_available; return TRUE; no_auth_available: { /* Output an error indicating that we couldn't connect because there were * no supported authentication protocols */ GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("No supported authentication protocol was found")); return FALSE; } no_user_pass: { /* We don't fire an error message, we just return FALSE and let the * normal NOT_AUTHORIZED error be propagated */ return FALSE; } } static GstRTSPResult gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPStatusCode * code) { GstRTSPResult res; GstRTSPStatusCode thecode; gchar *content_base = NULL; gint try = 0; again: gst_rtsp_ext_list_before_send (src->extensions, request); GST_DEBUG_OBJECT (src, "sending message"); if (src->debug) gst_rtsp_message_dump (request); res = gst_rtspsrc_connection_send (src, request, src->ptcp_timeout); if (res < 0) goto send_error; gst_rtsp_connection_reset_timeout (src->connection); next: res = gst_rtspsrc_connection_receive (src, response, src->ptcp_timeout); if (res < 0) goto receive_error; if (src->debug) gst_rtsp_message_dump (response); switch (response->type) { case GST_RTSP_MESSAGE_REQUEST: if ((res = gst_rtspsrc_handle_request (src, response)) < 0) goto handle_request_failed; goto next; case GST_RTSP_MESSAGE_RESPONSE: /* ok, a response is good */ GST_DEBUG_OBJECT (src, "received response message"); break; default: case GST_RTSP_MESSAGE_DATA: /* get next response */ GST_DEBUG_OBJECT (src, "ignoring data response message"); goto next; } thecode = response->type_data.response.code; GST_DEBUG_OBJECT (src, "got response message %d", thecode); /* if the caller wanted the result code, we store it. */ if (code) *code = thecode; /* If the request didn't succeed, bail out before doing any more */ if (thecode != GST_RTSP_STS_OK) return GST_RTSP_OK; /* store new content base if any */ gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE, &content_base, 0); if (content_base) { g_free (src->content_base); src->content_base = g_strdup (content_base); } gst_rtsp_ext_list_after_send (src->extensions, request, response); return GST_RTSP_OK; /* ERRORS */ send_error: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL), ("Could not send message. (%s)", str)); g_free (str); return res; } receive_error: { switch (res) { case GST_RTSP_EEOF: GST_WARNING_OBJECT (src, "server closed connection, doing reconnect"); if (try == 0) { gst_rtsp_connection_close (src->connection); try++; /* if reconnect succeeds, try again */ if ((res = gst_rtsp_connection_connect (src->connection, src->ptcp_timeout)) == 0) goto again; } /* only try once after reconnect, then fallthrough and error out */ default: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Could not receive message. (%s)", str)); g_free (str); break; } } return res; } handle_request_failed: { /* ERROR was posted */ return res; } } /** * gst_rtspsrc_send: * @src: the rtsp source * @request: must point to a valid request * @response: must point to an empty #GstRTSPMessage * * send @request and retrieve the response in @response. optionally @code can be * non-NULL in which case it will contain the status code of the response. * * If This function returns #GST_RTSP_OK, @response will contain a valid response * message that should be cleaned with gst_rtsp_message_unset() after usage. * * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid * @response message) if the response code was not 200 (OK). * * If the attempt results in an authentication failure, then this will attempt * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry * the request. * * Returns: #GST_RTSP_OK if the processing was successful. */ static GstRTSPResult gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPStatusCode * code) { GstRTSPStatusCode int_code = GST_RTSP_STS_OK; GstRTSPResult res = GST_RTSP_ERROR; gint count; gboolean retry; GstRTSPMethod method = GST_RTSP_INVALID; count = 0; do { retry = FALSE; /* make sure we don't loop forever */ if (count++ > 8) break; /* save method so we can disable it when the server complains */ method = request->type_data.request.method; if ((res = gst_rtspsrc_try_send (src, request, response, &int_code)) < 0) goto error; switch (int_code) { case GST_RTSP_STS_UNAUTHORIZED: if (gst_rtspsrc_setup_auth (src, response)) { /* Try the request/response again after configuring the auth info * and loop again */ retry = TRUE; } break; default: break; } } while (retry == TRUE); /* If the user requested the code, let them handle errors, otherwise * post an error below */ if (code != NULL) *code = int_code; else if (int_code != GST_RTSP_STS_OK) goto error_response; return res; /* ERRORS */ error: { GST_DEBUG_OBJECT (src, "got error %d", res); return res; } error_response: { res = GST_RTSP_ERROR; switch (response->type_data.response.code) { case GST_RTSP_STS_NOT_FOUND: GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s", response->type_data.response.reason)); break; case GST_RTSP_STS_MOVED_PERMANENTLY: case GST_RTSP_STS_MOVE_TEMPORARILY: { gchar *new_location; GST_DEBUG_OBJECT (src, "got redirection"); /* if we don't have a Location Header, we must error */ if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION, &new_location, 0) < 0) break; /* When we receive a redirect result, we go back to the INIT state after * parsing the new URI. The caller should do the needed steps to issue * a new setup when it detects this state change. */ GST_DEBUG_OBJECT (src, "redirection to %s", new_location); gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location); src->need_redirect = TRUE; src->state = GST_RTSP_STATE_INIT; res = GST_RTSP_OK; break; } case GST_RTSP_STS_NOT_ACCEPTABLE: case GST_RTSP_STS_NOT_IMPLEMENTED: case GST_RTSP_STS_METHOD_NOT_ALLOWED: GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s", gst_rtsp_method_as_text (method)); src->methods &= ~method; res = GST_RTSP_OK; break; default: GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Got error response: %d (%s).", response->type_data.response.code, response->type_data.response.reason)); break; } /* if we return ERROR we should unset the response ourselves */ if (res == GST_RTSP_ERROR) gst_rtsp_message_unset (response); return res; } } static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src) { return gst_rtspsrc_send (src, request, response, NULL); } /* parse the response and collect all the supported methods. We need this * information so that we don't try to send an unsupported request to the * server. */ static gboolean gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response) { GstRTSPHeaderField field; gchar *respoptions; gchar **options; gint indx = 0; gint i; /* reset supported methods */ src->methods = 0; /* Try Allow Header first */ field = GST_RTSP_HDR_ALLOW; while (TRUE) { respoptions = NULL; gst_rtsp_message_get_header (response, field, &respoptions, indx); if (indx == 0 && !respoptions) { /* if no Allow header was found then try the Public header... */ field = GST_RTSP_HDR_PUBLIC; gst_rtsp_message_get_header (response, field, &respoptions, indx); } if (!respoptions) break; /* If we get here, the server gave a list of supported methods, parse * them here. The string is like: * * OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ... */ options = g_strsplit (respoptions, ",", 0); for (i = 0; options[i]; i++) { gchar *stripped; gint method; stripped = g_strstrip (options[i]); method = gst_rtsp_find_method (stripped); /* keep bitfield of supported methods */ if (method != GST_RTSP_INVALID) src->methods |= method; } g_strfreev (options); indx++; } if (src->methods == 0) { /* neither Allow nor Public are required, assume the server supports * at least DESCRIBE, SETUP, we always assume it supports PLAY and PAUSE as * well. */ GST_DEBUG_OBJECT (src, "could not get OPTIONS"); src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP; } /* always assume PLAY and PAUSED, FIXME, extensions should be able to override * this */ src->methods |= GST_RTSP_PLAY | GST_RTSP_PAUSE; /* we need describe and setup */ if (!(src->methods & GST_RTSP_DESCRIBE)) goto no_describe; if (!(src->methods & GST_RTSP_SETUP)) goto no_setup; return TRUE; /* ERRORS */ no_describe: { GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("Server does not support DESCRIBE.")); return FALSE; } no_setup: { GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("Server does not support SETUP.")); return FALSE; } } static GstRTSPResult gst_rtspsrc_create_transports_string (GstRTSPSrc * src, GstRTSPLowerTrans protocols, gchar ** transports) { gchar *result; GstRTSPResult res; *transports = NULL; res = gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports); if (res < 0) goto failed; GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports)); /* extension listed transports, use those */ if (*transports != NULL) return GST_RTSP_OK; /* the default RTSP transports */ result = g_strdup (""); if (protocols & GST_RTSP_LOWER_TRANS_UDP) { gchar *new; GST_DEBUG_OBJECT (src, "adding UDP unicast"); new = g_strconcat (result, "RTP/AVP/UDP;unicast;client_port=%%u1-%%u2", NULL); g_free (result); result = new; } if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) { gchar *new; GST_DEBUG_OBJECT (src, "adding UDP multicast"); /* we don't have to allocate any UDP ports yet, if the selected transport * turns out to be multicast we can create them and join the multicast * group indicated in the transport reply */ new = g_strconcat (result, result[0] ? "," : "", "RTP/AVP/UDP;multicast", NULL); g_free (result); result = new; } if (protocols & GST_RTSP_LOWER_TRANS_TCP) { gchar *new; GST_DEBUG_OBJECT (src, "adding TCP"); new = g_strconcat (result, result[0] ? "," : "", "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2", NULL); g_free (result); result = new; } *transports = result; return GST_RTSP_OK; /* ERRORS */ failed: { return res; } } static GstRTSPResult gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports) { GstRTSPSrc *src; gint nr_udp, nr_int; gchar *next, *p; gint rtpport = 0, rtcpport = 0; GString *str; src = stream->parent; /* find number of placeholders first */ if (strstr (*transports, "%%i2")) nr_int = 2; else if (strstr (*transports, "%%i1")) nr_int = 1; else nr_int = 0; if (strstr (*transports, "%%u2")) nr_udp = 2; else if (strstr (*transports, "%%u1")) nr_udp = 1; else nr_udp = 0; if (nr_udp == 0 && nr_int == 0) goto done; if (nr_udp > 0) { if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport)) goto failed; } str = g_string_new (""); p = *transports; while ((next = strstr (p, "%%"))) { g_string_append_len (str, p, next - p); if (next[2] == 'u') { if (next[3] == '1') g_string_append_printf (str, "%d", rtpport); else if (next[3] == '2') g_string_append_printf (str, "%d", rtcpport); } if (next[2] == 'i') { if (next[3] == '1') g_string_append_printf (str, "%d", src->free_channel); else if (next[3] == '2') g_string_append_printf (str, "%d", src->free_channel + 1); } p = next + 4; } /* append final part */ g_string_append (str, p); g_free (*transports); *transports = g_string_free (str, FALSE); done: return GST_RTSP_OK; /* ERRORS */ failed: { return GST_RTSP_ERROR; } } /* Perform the SETUP request for all the streams. * * We ask the server for a specific transport, which initially includes all the * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate * two local UDP ports that we send to the server. * * Once the server replied with a transport, we configure the other streams * with the same transport. * * This function will also configure the stream for the selected transport, * which basically means creating the pipeline. */ static gboolean gst_rtspsrc_setup_streams (GstRTSPSrc * src) { GList *walk; GstRTSPResult res; GstRTSPMessage request = { 0 }; GstRTSPMessage response = { 0 }; GstRTSPStream *stream = NULL; GstRTSPLowerTrans protocols; GstRTSPStatusCode code; /* we initially allow all configured lower transports. based on the URL * transports and the replies from the server we narrow them down. */ protocols = src->url->transports & src->cur_protocols; if (protocols == 0) goto no_protocols; /* reset some state */ src->free_channel = 0; src->interleaved = FALSE; src->need_activate = FALSE; for (walk = src->streams; walk; walk = g_list_next (walk)) { gchar *transports; stream = (GstRTSPStream *) walk->data; /* see if we need to configure this stream */ if (!gst_rtsp_ext_list_configure_stream (src->extensions, stream->caps)) { GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension", stream); stream->disabled = TRUE; continue; } /* merge/overwrite global caps */ if (stream->caps) { guint j, num; GstStructure *s; s = gst_caps_get_structure (stream->caps, 0); num = gst_structure_n_fields (src->props); for (j = 0; j < num; j++) { const gchar *name; const GValue *val; name = gst_structure_nth_field_name (src->props, j); val = gst_structure_get_value (src->props, name); gst_structure_set_value (s, name, val); GST_DEBUG_OBJECT (src, "copied %s", name); } } /* skip setup if we have no URL for it */ if (stream->setup_url == NULL) { GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream); continue; } GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream, stream->setup_url); /* create a string with all the transports */ res = gst_rtspsrc_create_transports_string (src, protocols, &transports); if (res < 0) goto setup_transport_failed; /* replace placeholders with real values, this function will optionally * allocate UDP ports and other info needed to execute the setup request */ res = gst_rtspsrc_prepare_transports (stream, &transports); if (res < 0) goto setup_transport_failed; /* create SETUP request */ res = gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, stream->setup_url); if (res < 0) goto create_request_failed; /* select transport, copy is made when adding to header so we can free it. */ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transports); g_free (transports); /* handle the code ourselves */ if ((res = gst_rtspsrc_send (src, &request, &response, &code) < 0)) goto send_error; switch (code) { case GST_RTSP_STS_OK: break; case GST_RTSP_STS_UNSUPPORTED_TRANSPORT: gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); /* cleanup of leftover transport and move to the next stream */ gst_rtspsrc_stream_free_udp (stream); continue; default: goto response_error; } /* parse response transport */ { gchar *resptrans = NULL; GstRTSPTransport transport = { 0 }; gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT, &resptrans, 0); if (!resptrans) goto no_transport; /* parse transport, go to next stream on parse error */ if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) { GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans); goto next; } /* update allowed transports for other streams. once the transport of * one stream has been determined, we make sure that all other streams * are configured in the same way */ switch (transport.lower_transport) { case GST_RTSP_LOWER_TRANS_TCP: GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream); protocols = GST_RTSP_LOWER_TRANS_TCP; src->interleaved = TRUE; /* update free channels */ src->free_channel = MAX (transport.interleaved.min, src->free_channel); src->free_channel = MAX (transport.interleaved.max, src->free_channel); src->free_channel++; break; case GST_RTSP_LOWER_TRANS_UDP_MCAST: /* only allow multicast for other streams */ GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream); protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST; break; case GST_RTSP_LOWER_TRANS_UDP: /* only allow unicast for other streams */ GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream); protocols = GST_RTSP_LOWER_TRANS_UDP; break; default: GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream, transport.lower_transport); break; } if (!stream->container || !src->interleaved) { /* now configure the stream with the selected transport */ if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) { GST_DEBUG_OBJECT (src, "could not configure stream %p transport, skipping stream", stream); goto next; } } /* we need to activate at least one streams when we detect activity */ src->need_activate = TRUE; next: /* clean up our transport struct */ gst_rtsp_transport_init (&transport); /* clean up used RTSP messages */ gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); } } gst_rtsp_ext_list_stream_select (src->extensions, src->url); /* if there is nothing to activate, error out */ if (!src->need_activate) goto nothing_to_activate; return TRUE; /* ERRORS */ no_protocols: { /* no transport possible, post an error and stop */ GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Could not connect to server, no protocols left")); return FALSE; } create_request_failed: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL), ("Could not create request. (%s)", str)); g_free (str); goto cleanup_error; } setup_transport_failed: { GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL), ("Could not setup transport.")); goto cleanup_error; } response_error: { const gchar *str = gst_rtsp_status_as_text (code); GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL), ("Error (%d): %s", code, GST_STR_NULL (str))); goto cleanup_error; } send_error: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL), ("Could not send message. (%s)", str)); g_free (str); goto cleanup_error; } no_transport: { GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL), ("Server did not select transport.")); goto cleanup_error; } nothing_to_activate: { GST_ELEMENT_ERROR (src, STREAM, FORMAT, (NULL), ("No supported stream was found.")); return FALSE; } cleanup_error: { gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); return FALSE; } } static void gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range, GstSegment * segment) { GstRTSPTimeRange *therange; if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) { gint64 seconds; GST_DEBUG_OBJECT (src, "range: '%s', min %f - max %f ", GST_STR_NULL (range), therange->min.seconds, therange->max.seconds); if (therange->min.type == GST_RTSP_TIME_NOW) seconds = 0; else if (therange->min.type == GST_RTSP_TIME_END) seconds = 0; else seconds = therange->min.seconds * GST_SECOND; GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT, GST_TIME_ARGS (seconds)); gst_segment_set_last_stop (segment, GST_FORMAT_TIME, seconds); if (therange->max.type == GST_RTSP_TIME_NOW) seconds = -1; else if (therange->max.type == GST_RTSP_TIME_END) seconds = -1; else seconds = therange->max.seconds * GST_SECOND; GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT, GST_TIME_ARGS (seconds)); /* don't change duration with unknown value, we might have a valid value * there that we want to keep. */ if (seconds != -1) gst_segment_set_duration (segment, GST_FORMAT_TIME, seconds); gst_rtsp_range_free (therange); } else { GST_WARNING_OBJECT (src, "could not parse range: '%s'", range); } } static gboolean gst_rtspsrc_open (GstRTSPSrc * src) { GstRTSPResult res; GstRTSPMessage request = { 0 }; GstRTSPMessage response = { 0 }; guint8 *data; guint size; gint i, n_streams; GstSDPMessage sdp = { 0 }; GstRTSPStream *stream = NULL; gchar *respcont = NULL; GST_RTSP_STATE_LOCK (src); restart: /* reset our state */ gst_segment_init (&src->segment, GST_FORMAT_TIME); src->need_range = TRUE; src->need_redirect = FALSE; /* can't continue without a valid url */ if (G_UNLIKELY (src->url == NULL)) goto no_url; src->tried_url_auth = FALSE; /* create connection */ GST_DEBUG_OBJECT (src, "creating connection (%s)...", src->req_location); if ((res = gst_rtsp_connection_create (src->url, &src->connection)) < 0) goto could_not_create; /* connect */ GST_DEBUG_OBJECT (src, "connecting (%s)...", src->req_location); if ((res = gst_rtsp_connection_connect (src->connection, src->ptcp_timeout)) < 0) goto could_not_connect; /* create OPTIONS */ GST_DEBUG_OBJECT (src, "create options..."); res = gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS, src->req_location); if (res < 0) goto create_request_failed; /* send OPTIONS */ GST_DEBUG_OBJECT (src, "send options..."); if ((res = gst_rtspsrc_send (src, &request, &response, NULL)) < 0) goto send_error; /* parse OPTIONS */ if (!gst_rtspsrc_parse_methods (src, &response)) goto methods_error; /* create DESCRIBE */ GST_DEBUG_OBJECT (src, "create describe..."); res = gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE, src->req_location); if (res < 0) goto create_request_failed; /* we only accept SDP for now */ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT, "application/sdp"); /* prepare global stream caps properties */ if (src->props) gst_structure_remove_all_fields (src->props); else src->props = gst_structure_empty_new ("RTSPProperties"); /* send DESCRIBE */ GST_DEBUG_OBJECT (src, "send describe..."); if ((res = gst_rtspsrc_send (src, &request, &response, NULL)) < 0) goto send_error; /* we only perform redirect for the describe, currently */ if (src->need_redirect) { /* close connection, we don't have to send a TEARDOWN yet, ignore the * result. */ gst_rtsp_connection_close (src->connection); gst_rtsp_connection_free (src->connection); src->connection = NULL; gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); /* and now retry */ goto restart; } /* check if reply is SDP */ gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont, 0); /* could not be set but since the request returned OK, we assume it * was SDP, else check it. */ if (respcont) { if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0) goto wrong_content_type; } /* get message body and parse as SDP */ gst_rtsp_message_get_body (&response, &data, &size); GST_DEBUG_OBJECT (src, "parse SDP..."); gst_sdp_message_init (&sdp); gst_sdp_message_parse_buffer (data, size, &sdp); if (src->debug) gst_sdp_message_dump (&sdp); gst_rtsp_ext_list_parse_sdp (src->extensions, &sdp, src->props); /* parse range for duration reporting. */ { const gchar *range; range = gst_sdp_message_get_attribute_val (&sdp, "range"); if (range) gst_rtspsrc_parse_range (src, range, &src->segment); } /* create streams */ n_streams = gst_sdp_message_medias_len (&sdp); for (i = 0; i < n_streams; i++) { stream = gst_rtspsrc_create_stream (src, &sdp, i); } src->state = GST_RTSP_STATE_INIT; /* setup streams */ if (!gst_rtspsrc_setup_streams (src)) goto setup_failed; src->state = GST_RTSP_STATE_READY; GST_RTSP_STATE_UNLOCK (src); /* clean up any messages */ gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); gst_sdp_message_uninit (&sdp); return TRUE; /* ERRORS */ no_url: { GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("No valid RTSP URL was provided")); goto cleanup_error; } could_not_create: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL), ("Could not create connection. (%s)", str)); g_free (str); goto cleanup_error; } could_not_connect: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL), ("Could not connect to server. (%s)", str)); g_free (str); goto cleanup_error; } create_request_failed: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL), ("Could not create request. (%s)", str)); g_free (str); goto cleanup_error; } send_error: { /* Don't post a message - the rtsp_send method will have * taken care of it because we passed NULL for the response code */ goto cleanup_error; } methods_error: { /* error was posted */ goto cleanup_error; } wrong_content_type: { GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL), ("Server does not support SDP, got %s.", respcont)); goto cleanup_error; } setup_failed: { /* error was posted */ goto cleanup_error; } cleanup_error: { if (src->connection) { gst_rtsp_connection_free (src->connection); src->connection = NULL; } GST_RTSP_STATE_UNLOCK (src); gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); gst_sdp_message_uninit (&sdp); return FALSE; } } #if 0 static gboolean gst_rtspsrc_async_open (GstRTSPSrc * src) { GError *error = NULL; gboolean res = TRUE; src->thread = g_thread_create ((GThreadFunc) gst_rtspsrc_open, src, TRUE, &error); if (error != NULL) { GST_ELEMENT_ERROR (src, RESOURCE, INIT, (NULL), ("Could not start async thread (%s).", error->message)); } return res; } #endif static gboolean gst_rtspsrc_close (GstRTSPSrc * src) { GstRTSPMessage request = { 0 }; GstRTSPMessage response = { 0 }; GstRTSPResult res; GST_DEBUG_OBJECT (src, "TEARDOWN..."); GST_RTSP_STATE_LOCK (src); gst_rtspsrc_loop_send_cmd (src, CMD_STOP, TRUE); /* stop task if any */ if (src->task) { gst_task_stop (src->task); /* make sure it is not running */ GST_RTSP_STREAM_LOCK (src); GST_RTSP_STREAM_UNLOCK (src); /* now wait for the task to finish */ gst_task_join (src->task); /* and free the task */ gst_object_unref (GST_OBJECT (src->task)); src->task = NULL; } GST_DEBUG_OBJECT (src, "stop connection flush"); gst_rtsp_connection_flush (src->connection, FALSE); if (src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)) { /* do TEARDOWN */ res = gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, src->req_location); if (res < 0) goto create_request_failed; if ((res = gst_rtspsrc_send (src, &request, &response, NULL)) < 0) goto send_error; /* FIXME, parse result? */ gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); } else { GST_DEBUG_OBJECT (src, "TEARDOWN and PLAY not supported, can't do TEARDOWN"); } /* close connection */ GST_DEBUG_OBJECT (src, "closing connection..."); if ((res = gst_rtsp_connection_close (src->connection)) < 0) goto close_failed; /* free connection */ gst_rtsp_connection_free (src->connection); src->connection = NULL; /* cleanup */ gst_rtspsrc_cleanup (src); src->state = GST_RTSP_STATE_INVALID; GST_RTSP_STATE_UNLOCK (src); return TRUE; /* ERRORS */ create_request_failed: { GST_RTSP_STATE_UNLOCK (src); GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL), ("Could not create request.")); return FALSE; } send_error: { GST_RTSP_STATE_UNLOCK (src); gst_rtsp_message_unset (&request); GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL), ("Could not send message.")); return FALSE; } close_failed: { GST_RTSP_STATE_UNLOCK (src); GST_ELEMENT_ERROR (src, RESOURCE, CLOSE, (NULL), ("Close failed.")); return FALSE; } } /* RTP-Info is of the format: * * url=;[seq=;rtptime=] [, url=...] * * rtptime corresponds to the timestamp for the NPT time given in the header * seqbase corresponds to the next sequence number we received. This number * indicates the first seqnum after the seek and should be used to discard * packets that are from before the seek. */ static gboolean gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo) { gchar **infos; gint i, j; GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo); infos = g_strsplit (rtpinfo, ",", 0); for (i = 0; infos[i]; i++) { gchar **fields; GstRTSPStream *stream; gint32 seqbase; gint64 timebase; GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]); /* init values, types of seqbase and timebase are bigger than needed so we * can store -1 as uninitialized values */ stream = NULL; seqbase = -1; timebase = -1; /* parse url, find stream for url. * parse seq and rtptime. The seq number should be configured in the rtp * depayloader or session manager to detect gaps. Same for the rtptime, it * should be used to create an initial time newsegment. */ fields = g_strsplit (infos[i], ";", 0); for (j = 0; fields[j]; j++) { GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]); /* remove leading whitespace */ fields[j] = g_strchug (fields[j]); if (g_str_has_prefix (fields[j], "url=")) { /* get the url and the stream */ stream = find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup); } else if (g_str_has_prefix (fields[j], "seq=")) { seqbase = atoi (fields[j] + 4); } else if (g_str_has_prefix (fields[j], "rtptime=")) { timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10); } } g_strfreev (fields); /* now we need to store the values for the caps of the stream */ if (stream != NULL) { GST_DEBUG_OBJECT (src, "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT, stream, seqbase, timebase); /* we have a stream, configure detected params */ stream->seqbase = seqbase; stream->timebase = timebase; } } g_strfreev (infos); return TRUE; } #define USE_POSIX_LOCALE { \ gchar *__old_locale = g_strdup (setlocale (LC_NUMERIC, NULL)); \ setlocale (LC_NUMERIC, "POSIX"); #define RESTORE_LOCALE \ setlocale (LC_NUMERIC, __old_locale); \ g_free (__old_locale);} static gchar * gst_rtspsrc_dup_printf (const gchar * format, ...) { gchar *result; va_list varargs; USE_POSIX_LOCALE va_start (varargs, format); result = g_strdup_vprintf (format, varargs); va_end (varargs); RESTORE_LOCALE return result; } static gint gst_rtspsrc_get_float (const char *str, gfloat * val) { gint result; USE_POSIX_LOCALE result = sscanf (str, "%f", val); RESTORE_LOCALE return result; } static gboolean gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment) { GstRTSPMessage request = { 0 }; GstRTSPMessage response = { 0 }; GstRTSPResult res; gchar *hval; GST_RTSP_STATE_LOCK (src); GST_DEBUG_OBJECT (src, "PLAY..."); if (!(src->methods & GST_RTSP_PLAY)) goto not_supported; if (src->state == GST_RTSP_STATE_PLAYING) goto was_playing; /* do play */ res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, src->req_location); if (res < 0) goto create_request_failed; if (src->need_range) { if (segment->last_stop == 0) hval = g_strdup_printf ("npt=0-"); else hval = gst_rtspsrc_dup_printf ("npt=%f-", ((gdouble) segment->last_stop) / GST_SECOND); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval); g_free (hval); src->need_range = FALSE; } if (segment->rate != 1.0) { hval = gst_rtspsrc_dup_printf ("%f", segment->rate); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval); g_free (hval); } if (segment->applied_rate != 1.0) { hval = gst_rtspsrc_dup_printf ("%f", segment->applied_rate); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval); g_free (hval); } if ((res = gst_rtspsrc_send (src, &request, &response, NULL)) < 0) goto send_error; gst_rtsp_message_unset (&request); /* parse RTP npt field. This is the current position in the stream (Normal * Play Time) and should be put in the NEWSEGMENT position field. */ if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval, 0) == GST_RTSP_OK) gst_rtspsrc_parse_range (src, hval, segment); /* parse Speed header. This is the intended playback rate of the stream * and should be put in the NEWSEGMENT rate field. */ if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval, 0) == GST_RTSP_OK) { gfloat fval; if (gst_rtspsrc_get_float (hval, &fval) > 0) segment->rate = fval; } else { segment->rate = 1.0; } /* parse Scale header. This is the playback rate as sent by the server * and should be put in the NEWSEGMENT applied_rate field. */ if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE, &hval, 0) == GST_RTSP_OK) { gfloat fval; if (gst_rtspsrc_get_float (hval, &fval) > 0) segment->applied_rate = fval; } else { segment->applied_rate = 1.0; } /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp * for the RTP packets. If this is not present, we assume all starts from 0... * This is info for the RTP session manager that we pass to it in caps. */ if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO, &hval, 0) == GST_RTSP_OK) gst_rtspsrc_parse_rtpinfo (src, hval); gst_rtsp_message_unset (&response); /* configure the caps of the streams after we parsed all headers. */ gst_rtspsrc_configure_caps (src, segment); /* for interleaved transport, we receive the data on the RTSP connection * instead of UDP. We start a task to select and read from that connection. * For UDP we start the task as well to look for server info and UDP timeouts. */ if (src->task == NULL) { src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_loop, src); gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src)); } src->running = TRUE; src->base_time = -1; src->state = GST_RTSP_STATE_PLAYING; gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, FALSE); gst_task_start (src->task); done: GST_RTSP_STATE_UNLOCK (src); return TRUE; /* ERRORS */ not_supported: { GST_DEBUG_OBJECT (src, "PLAY is not supported"); goto done; } was_playing: { GST_DEBUG_OBJECT (src, "we were already PLAYING"); goto done; } create_request_failed: { GST_RTSP_STATE_UNLOCK (src); GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL), ("Could not create request.")); return FALSE; } send_error: { GST_RTSP_STATE_UNLOCK (src); gst_rtsp_message_unset (&request); GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL), ("Could not send message.")); return FALSE; } } static gboolean gst_rtspsrc_pause (GstRTSPSrc * src) { GstRTSPMessage request = { 0 }; GstRTSPMessage response = { 0 }; GstRTSPResult res; GST_RTSP_STATE_LOCK (src); GST_DEBUG_OBJECT (src, "PAUSE..."); if (!(src->methods & GST_RTSP_PAUSE)) goto not_supported; if (src->state == GST_RTSP_STATE_READY) goto was_paused; /* waiting for connection idle, we were flushing so any attempt at doing data * transfer will result in pausing the tasks. */ GST_DEBUG_OBJECT (src, "wait for connection idle"); GST_RTSP_CONN_LOCK (src); GST_DEBUG_OBJECT (src, "connection is idle now"); GST_RTSP_CONN_UNLOCK (src); GST_DEBUG_OBJECT (src, "stop connection flush"); gst_rtsp_connection_flush (src->connection, FALSE); /* do pause */ res = gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE, src->req_location); if (res < 0) goto create_request_failed; if ((res = gst_rtspsrc_send (src, &request, &response, NULL)) < 0) goto send_error; gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); src->state = GST_RTSP_STATE_READY; done: GST_RTSP_STATE_UNLOCK (src); return TRUE; /* ERRORS */ not_supported: { GST_DEBUG_OBJECT (src, "PAUSE is not supported"); goto done; } was_paused: { GST_DEBUG_OBJECT (src, "we were already PAUSED"); goto done; } create_request_failed: { GST_RTSP_STATE_UNLOCK (src); GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL), ("Could not create request.")); return FALSE; } send_error: { GST_RTSP_STATE_UNLOCK (src); gst_rtsp_message_unset (&request); GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL), ("Could not send message.")); return FALSE; } } static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message) { GstRTSPSrc *rtspsrc; rtspsrc = GST_RTSPSRC (bin); switch (GST_MESSAGE_TYPE (message)) { case GST_MESSAGE_ELEMENT: { const GstStructure *s = gst_message_get_structure (message); if (gst_structure_has_name (s, "GstUDPSrcTimeout")) { gboolean ignore_timeout; GST_DEBUG_OBJECT (bin, "timeout on UDP port"); GST_OBJECT_LOCK (rtspsrc); ignore_timeout = rtspsrc->ignore_timeout; rtspsrc->ignore_timeout = TRUE; GST_OBJECT_UNLOCK (rtspsrc); /* we only act on the first udp timeout message, others are irrelevant * and can be ignored. */ if (ignore_timeout) gst_message_unref (message); else gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, TRUE); return; } GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } case GST_MESSAGE_ERROR: { GstObject *udpsrc; GstRTSPStream *stream; GstFlowReturn ret; udpsrc = GST_MESSAGE_SRC (message); GST_DEBUG_OBJECT (rtspsrc, "got error from %s", GST_ELEMENT_NAME (udpsrc)); stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc); if (!stream) goto forward; /* we ignore the RTCP udpsrc */ if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc)) goto done; /* if we get error messages from the udp sources, that's not a problem as * long as not all of them error out. We also don't really know what the * problem is, the message does not give enough detail... */ ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED); GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret)); if (ret != GST_FLOW_OK) goto forward; done: gst_message_unref (message); break; forward: /* fatal but not our message, forward */ GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } default: { GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } } } static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element, GstStateChange transition) { GstRTSPSrc *rtspsrc; GstStateChangeReturn ret; rtspsrc = GST_RTSPSRC (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: rtspsrc->cur_protocols = rtspsrc->protocols; /* first attempt, don't ignore timeouts */ rtspsrc->ignore_timeout = FALSE; if (!gst_rtspsrc_open (rtspsrc)) goto open_failed; break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: GST_DEBUG_OBJECT (rtspsrc, "PAUSED->PLAYING: stop connection flush"); gst_rtsp_connection_flush (rtspsrc->connection, FALSE); /* FIXME, the server might send UDP packets before we activate the UDP * ports */ gst_rtspsrc_play (rtspsrc, &rtspsrc->segment); break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: case GST_STATE_CHANGE_PAUSED_TO_READY: GST_DEBUG_OBJECT (rtspsrc, "shutdown: sending stop command"); gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_STOP, TRUE); break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); if (ret == GST_STATE_CHANGE_FAILURE) goto done; switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: ret = GST_STATE_CHANGE_NO_PREROLL; break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: gst_rtspsrc_pause (rtspsrc); ret = GST_STATE_CHANGE_NO_PREROLL; break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_rtspsrc_close (rtspsrc); break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } done: return ret; open_failed: { return GST_STATE_CHANGE_FAILURE; } } /*** GSTURIHANDLER INTERFACE *************************************************/ static GstURIType gst_rtspsrc_uri_get_type (void) { return GST_URI_SRC; } static gchar ** gst_rtspsrc_uri_get_protocols (void) { static gchar *protocols[] = { "rtsp", "rtspu", "rtspt", NULL }; return protocols; } static const gchar * gst_rtspsrc_uri_get_uri (GstURIHandler * handler) { GstRTSPSrc *src = GST_RTSPSRC (handler); /* should not dup */ return src->location; } static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri) { GstRTSPSrc *src; GstRTSPResult res; GstRTSPUrl *newurl; src = GST_RTSPSRC (handler); /* same URI, we're fine */ if (src->location && uri && !strcmp (uri, src->location)) goto was_ok; /* try to parse */ if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0) goto parse_error; /* if worked, free previous and store new url object along with the original * location. */ gst_rtsp_url_free (src->url); src->url = newurl; g_free (src->location); g_free (src->req_location); src->location = g_strdup (uri); src->req_location = gst_rtsp_url_get_request_uri (src->url); GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri)); GST_DEBUG_OBJECT (src, "request uri is: %s", GST_STR_NULL (src->req_location)); return TRUE; /* Special cases */ was_ok: { GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri)); return TRUE; } parse_error: { GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)", GST_STR_NULL (uri), res); return FALSE; } } static void gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data) { GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface; iface->get_type = gst_rtspsrc_uri_get_type; iface->get_protocols = gst_rtspsrc_uri_get_protocols; iface->get_uri = gst_rtspsrc_uri_get_uri; iface->set_uri = gst_rtspsrc_uri_set_uri; }