/* GStreamer * Copyright (C) <2005,2006> Wim Taymans * <2006> Lutz Mueller * <2015> Jan Schmidt * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /* * 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-rtspclientsink * * Makes a connection to an RTSP server and send data via RTSP RECORD. * rtspclientsink strictly follows RFC 2326 * * RTSP supports transport over TCP or UDP in unicast or multicast mode. By * default rtspclientsink 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 #GstRTSPClientSink:protocols property. * * rtspclientsink will internally instantiate an RTP session manager element * that will handle the RTCP messages to and from the server, jitter removal, * and packet reordering. * This feature is implemented using the gstrtpbin element. * * rtspclientsink accepts any stream for which there is an installed payloader, * creates the payloader and manages payload-types, as well as RTX setup. * The new-payloader signal is fired when a payloader is created, in case * an app wants to do custom configuration (such as for MTU). * * ## Example launch line * * |[ * gst-launch-1.0 videotestsrc ! jpegenc ! rtspclientsink location=rtsp://some.server/url * ]| Establish a connection to an RTSP server and send JPEG encoded video packets */ /* FIXMEs * - Handle EOS properly and shutdown. The problem with EOS is we don't know * when the server has received all data, so we don't know when to do teardown. * At the moment, we forward EOS to the app as soon as we stop sending. Is there * a way to know from the receiver that it's got all data? Some session timeout? * - Implement extension support for Real / WMS if they support RECORD? * - Add support for network clock synchronised streaming? * - Fix crypto key nego so SAVP/SAVPF profiles work. * - Test (&fix?) HTTP tunnel support * - Add an address pool object for GstRTSPStreams to use for multicast * - Test multicast UDP transport */ #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 "gstrtspclientsink.h" typedef struct _GstRtspClientSinkPad GstRtspClientSinkPad; typedef GstGhostPadClass GstRtspClientSinkPadClass; struct _GstRtspClientSinkPad { GstGhostPad parent; GstElement *custom_payloader; guint ulpfec_percentage; }; enum { PROP_PAD_0, PROP_PAD_PAYLOADER, PROP_PAD_ULPFEC_PERCENTAGE }; #define DEFAULT_PAD_ULPFEC_PERCENTAGE 0 static GType gst_rtsp_client_sink_pad_get_type (void); G_DEFINE_TYPE (GstRtspClientSinkPad, gst_rtsp_client_sink_pad, GST_TYPE_GHOST_PAD); #define GST_TYPE_RTSP_CLIENT_SINK_PAD (gst_rtsp_client_sink_pad_get_type ()) #define GST_RTSP_CLIENT_SINK_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSP_CLIENT_SINK_PAD,GstRtspClientSinkPad)) static void gst_rtsp_client_sink_pad_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtspClientSinkPad *pad; pad = GST_RTSP_CLIENT_SINK_PAD (object); switch (prop_id) { case PROP_PAD_PAYLOADER: GST_OBJECT_LOCK (pad); if (pad->custom_payloader) gst_object_unref (pad->custom_payloader); pad->custom_payloader = g_value_get_object (value); gst_object_ref_sink (pad->custom_payloader); GST_OBJECT_UNLOCK (pad); break; case PROP_PAD_ULPFEC_PERCENTAGE: GST_OBJECT_LOCK (pad); pad->ulpfec_percentage = g_value_get_uint (value); GST_OBJECT_UNLOCK (pad); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtsp_client_sink_pad_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtspClientSinkPad *pad; pad = GST_RTSP_CLIENT_SINK_PAD (object); switch (prop_id) { case PROP_PAD_PAYLOADER: GST_OBJECT_LOCK (pad); g_value_set_object (value, pad->custom_payloader); GST_OBJECT_UNLOCK (pad); break; case PROP_PAD_ULPFEC_PERCENTAGE: GST_OBJECT_LOCK (pad); g_value_set_uint (value, pad->ulpfec_percentage); GST_OBJECT_UNLOCK (pad); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtsp_client_sink_pad_dispose (GObject * object) { GstRtspClientSinkPad *pad = GST_RTSP_CLIENT_SINK_PAD (object); if (pad->custom_payloader) gst_object_unref (pad->custom_payloader); G_OBJECT_CLASS (gst_rtsp_client_sink_pad_parent_class)->dispose (object); } static void gst_rtsp_client_sink_pad_class_init (GstRtspClientSinkPadClass * klass) { GObjectClass *gobject_klass; gobject_klass = (GObjectClass *) klass; gobject_klass->set_property = gst_rtsp_client_sink_pad_set_property; gobject_klass->get_property = gst_rtsp_client_sink_pad_get_property; gobject_klass->dispose = gst_rtsp_client_sink_pad_dispose; g_object_class_install_property (gobject_klass, PROP_PAD_PAYLOADER, g_param_spec_object ("payloader", "Payloader", "The payloader element to use (NULL = default automatically selected)", GST_TYPE_ELEMENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_klass, PROP_PAD_ULPFEC_PERCENTAGE, g_param_spec_uint ("ulpfec-percentage", "ULPFEC percentage", "The percentage of ULP redundancy to apply", 0, 100, DEFAULT_PAD_ULPFEC_PERCENTAGE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); } static void gst_rtsp_client_sink_pad_init (GstRtspClientSinkPad * pad) { } static GstPad * gst_rtsp_client_sink_pad_new (const GstPadTemplate * pad_tmpl, const gchar * name) { GstRtspClientSinkPad *ret; ret = g_object_new (GST_TYPE_RTSP_CLIENT_SINK_PAD, "direction", GST_PAD_SINK, "template", pad_tmpl, "name", name, NULL); return GST_PAD (ret); } GST_DEBUG_CATEGORY_STATIC (rtsp_client_sink_debug); #define GST_CAT_DEFAULT (rtsp_client_sink_debug) static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("sink_%u", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS_ANY); /* Actual caps come from available set of payloaders */ enum { SIGNAL_HANDLE_REQUEST, SIGNAL_NEW_MANAGER, SIGNAL_NEW_PAYLOADER, SIGNAL_REQUEST_RTCP_KEY, SIGNAL_ACCEPT_CERTIFICATE, SIGNAL_UPDATE_SDP, LAST_SIGNAL }; enum _GstRTSPClientSinkNtpTimeSource { NTP_TIME_SOURCE_NTP, NTP_TIME_SOURCE_UNIX, NTP_TIME_SOURCE_RUNNING_TIME, NTP_TIME_SOURCE_CLOCK_TIME }; #define GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE (gst_rtsp_client_sink_ntp_time_source_get_type()) static GType gst_rtsp_client_sink_ntp_time_source_get_type (void) { static GType ntp_time_source_type = 0; static const GEnumValue ntp_time_source_values[] = { {NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"}, {NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"}, {NTP_TIME_SOURCE_RUNNING_TIME, "Running time based on pipeline clock", "running-time"}, {NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"}, {0, NULL, NULL}, }; if (!ntp_time_source_type) { ntp_time_source_type = g_enum_register_static ("GstRTSPClientSinkNtpTimeSource", ntp_time_source_values); } return ntp_time_source_type; } #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_UDP_BUFFER_SIZE 0x80000 #define DEFAULT_TCP_TIMEOUT 20000000 #define DEFAULT_LATENCY_MS 2000 #define DEFAULT_DO_RTSP_KEEP_ALIVE TRUE #define DEFAULT_PROXY NULL #define DEFAULT_RTP_BLOCKSIZE 0 #define DEFAULT_USER_ID NULL #define DEFAULT_USER_PW NULL #define DEFAULT_PORT_RANGE NULL #define DEFAULT_UDP_RECONNECT TRUE #define DEFAULT_MULTICAST_IFACE NULL #define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL #define DEFAULT_TLS_DATABASE NULL #define DEFAULT_TLS_INTERACTION NULL #define DEFAULT_NTP_TIME_SOURCE NTP_TIME_SOURCE_NTP #define DEFAULT_USER_AGENT "GStreamer/" PACKAGE_VERSION #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP #define DEFAULT_RTX_TIME_MS 500 #define DEFAULT_PUBLISH_CLOCK_MODE GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK enum { PROP_0, PROP_LOCATION, PROP_PROTOCOLS, PROP_DEBUG, PROP_RETRY, PROP_TIMEOUT, PROP_TCP_TIMEOUT, PROP_LATENCY, PROP_RTX_TIME, PROP_DO_RTSP_KEEP_ALIVE, PROP_PROXY, PROP_PROXY_ID, PROP_PROXY_PW, PROP_RTP_BLOCKSIZE, PROP_USER_ID, PROP_USER_PW, PROP_PORT_RANGE, PROP_UDP_BUFFER_SIZE, PROP_UDP_RECONNECT, PROP_MULTICAST_IFACE, PROP_SDES, PROP_TLS_VALIDATION_FLAGS, PROP_TLS_DATABASE, PROP_TLS_INTERACTION, PROP_NTP_TIME_SOURCE, PROP_USER_AGENT, PROP_PROFILES, PROP_PUBLISH_CLOCK_MODE, }; static void gst_rtsp_client_sink_finalize (GObject * object); static void gst_rtsp_client_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtsp_client_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstClock *gst_rtsp_client_sink_provide_clock (GstElement * element); static void gst_rtsp_client_sink_uri_handler_init (gpointer g_iface, gpointer iface_data); static gboolean gst_rtsp_client_sink_set_proxy (GstRTSPClientSink * rtsp, const gchar * proxy); static void gst_rtsp_client_sink_set_tcp_timeout (GstRTSPClientSink * rtsp_client_sink, guint64 timeout); static GstStateChangeReturn gst_rtsp_client_sink_change_state (GstElement * element, GstStateChange transition); static void gst_rtsp_client_sink_handle_message (GstBin * bin, GstMessage * message); static gboolean gst_rtsp_client_sink_setup_auth (GstRTSPClientSink * sink, GstRTSPMessage * response); static gboolean gst_rtsp_client_sink_loop_send_cmd (GstRTSPClientSink * sink, gint cmd, gint mask); static GstRTSPResult gst_rtsp_client_sink_open (GstRTSPClientSink * sink, gboolean async); static GstRTSPResult gst_rtsp_client_sink_record (GstRTSPClientSink * sink, gboolean async); static GstRTSPResult gst_rtsp_client_sink_pause (GstRTSPClientSink * sink, gboolean async); static GstRTSPResult gst_rtsp_client_sink_close (GstRTSPClientSink * sink, gboolean async, gboolean only_close); static gboolean gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink); static gboolean gst_rtsp_client_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri, GError ** error); static gchar *gst_rtsp_client_sink_uri_get_uri (GstURIHandler * handler); static gboolean gst_rtsp_client_sink_loop (GstRTSPClientSink * sink); static void gst_rtsp_client_sink_connection_flush (GstRTSPClientSink * sink, gboolean flush); static GstPad *gst_rtsp_client_sink_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name, const GstCaps * caps); static void gst_rtsp_client_sink_release_pad (GstElement * element, GstPad * pad); /* commands we send to out loop to notify it of events */ #define CMD_OPEN (1 << 0) #define CMD_RECORD (1 << 1) #define CMD_PAUSE (1 << 2) #define CMD_CLOSE (1 << 3) #define CMD_WAIT (1 << 4) #define CMD_RECONNECT (1 << 5) #define CMD_LOOP (1 << 6) /* mask for all commands */ #define CMD_ALL ((CMD_LOOP << 1) - 1) #define GST_ELEMENT_PROGRESS(el, type, code, text) \ G_STMT_START { \ gchar *__txt = _gst_element_error_printf text; \ gst_element_post_message (GST_ELEMENT_CAST (el), \ gst_message_new_progress (GST_OBJECT_CAST (el), \ GST_PROGRESS_TYPE_ ##type, code, __txt)); \ g_free (__txt); \ } G_STMT_END static guint gst_rtsp_client_sink_signals[LAST_SIGNAL] = { 0 }; /********************************* * GstChildProxy implementation * *********************************/ static GObject * gst_rtsp_client_sink_child_proxy_get_child_by_index (GstChildProxy * child_proxy, guint index) { GObject *obj; GstRTSPClientSink *cs = GST_RTSP_CLIENT_SINK (child_proxy); GST_OBJECT_LOCK (cs); if ((obj = g_list_nth_data (GST_ELEMENT (cs)->sinkpads, index))) g_object_ref (obj); GST_OBJECT_UNLOCK (cs); return obj; } static guint gst_rtsp_client_sink_child_proxy_get_children_count (GstChildProxy * child_proxy) { guint count = 0; GST_OBJECT_LOCK (child_proxy); count = GST_ELEMENT (child_proxy)->numsinkpads; GST_OBJECT_UNLOCK (child_proxy); GST_INFO_OBJECT (child_proxy, "Children Count: %d", count); return count; } static void gst_rtsp_client_sink_child_proxy_init (gpointer g_iface, gpointer iface_data) { GstChildProxyInterface *iface = g_iface; GST_INFO ("intializing child proxy interface"); iface->get_child_by_index = gst_rtsp_client_sink_child_proxy_get_child_by_index; iface->get_children_count = gst_rtsp_client_sink_child_proxy_get_children_count; } #define gst_rtsp_client_sink_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstRTSPClientSink, gst_rtsp_client_sink, GST_TYPE_BIN, G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtsp_client_sink_uri_handler_init); G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY, gst_rtsp_client_sink_child_proxy_init); ); #ifndef GST_DISABLE_GST_DEBUG static inline const gchar * cmd_to_string (guint cmd) { switch (cmd) { case CMD_OPEN: return "OPEN"; case CMD_RECORD: return "RECORD"; case CMD_PAUSE: return "PAUSE"; case CMD_CLOSE: return "CLOSE"; case CMD_WAIT: return "WAIT"; case CMD_RECONNECT: return "RECONNECT"; case CMD_LOOP: return "LOOP"; } return "unknown"; } #endif static void gst_rtsp_client_sink_class_init (GstRTSPClientSinkClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBinClass *gstbin_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbin_class = (GstBinClass *) klass; GST_DEBUG_CATEGORY_INIT (rtsp_client_sink_debug, "rtspclientsink", 0, "RTSP sink element"); gobject_class->set_property = gst_rtsp_client_sink_set_property; gobject_class->get_property = gst_rtsp_client_sink_get_property; gobject_class->finalize = gst_rtsp_client_sink_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_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PROTOCOLS, g_param_spec_flags ("protocols", "Protocols", "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS, DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PROFILES, g_param_spec_flags ("profiles", "Profiles", "Allowed RTSP profiles", GST_TYPE_RTSP_PROFILE, DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); 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_STATIC_STRINGS)); 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_STATIC_STRINGS)); 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_STATIC_STRINGS)); 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_STATIC_STRINGS)); 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_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_RTX_TIME, g_param_spec_uint ("rtx-time", "Retransmission buffer in ms", "Amount of ms to buffer for retransmission. 0 disables retransmission", 0, G_MAXUINT, DEFAULT_RTX_TIME_MS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:do-rtsp-keep-alive: * * Enable RTSP keep alive support. Some old server don't like RTSP * keep alive and then this property needs to be set to FALSE. */ g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE, g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive", "Send RTSP keep alive packets, disable for old incompatible server.", DEFAULT_DO_RTSP_KEEP_ALIVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:proxy: * * Set the proxy parameters. This has to be a string of the format * [http://][user:passwd@]host[:port]. */ g_object_class_install_property (gobject_class, PROP_PROXY, g_param_spec_string ("proxy", "Proxy", "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]", DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:proxy-id: * * Sets the proxy URI user id for authentication. If the URI set via the * "proxy" property contains a user-id already, that will take precedence. * */ g_object_class_install_property (gobject_class, PROP_PROXY_ID, g_param_spec_string ("proxy-id", "proxy-id", "HTTP proxy URI user id for authentication", "", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:proxy-pw: * * Sets the proxy URI password for authentication. If the URI set via the * "proxy" property contains a password already, that will take precedence. * */ g_object_class_install_property (gobject_class, PROP_PROXY_PW, g_param_spec_string ("proxy-pw", "proxy-pw", "HTTP proxy URI user password for authentication", "", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:rtp-blocksize: * * RTP package size to suggest to server. */ g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE, g_param_spec_uint ("rtp-blocksize", "RTP Blocksize", "RTP package size to suggest to server (0 = disabled)", 0, 65536, DEFAULT_RTP_BLOCKSIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_USER_ID, g_param_spec_string ("user-id", "user-id", "RTSP location URI user id for authentication", DEFAULT_USER_ID, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_USER_PW, g_param_spec_string ("user-pw", "user-pw", "RTSP location URI user password for authentication", DEFAULT_USER_PW, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:port-range: * * Configure the client port numbers that can be used to receive * RTCP. */ g_object_class_install_property (gobject_class, PROP_PORT_RANGE, g_param_spec_string ("port-range", "Port range", "Client port range that can be used to receive RTCP data, " "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:udp-buffer-size: * * Size of the kernel UDP receive buffer in bytes. */ g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE, g_param_spec_int ("udp-buffer-size", "UDP Buffer Size", "Size of the kernel UDP receive buffer in bytes, 0=default", 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT, g_param_spec_boolean ("udp-reconnect", "Reconnect to the server", "Reconnect to the server if RTSP connection is closed when doing UDP", DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE, g_param_spec_string ("multicast-iface", "Multicast Interface", "The network interface on which to join the multicast group", DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_SDES, g_param_spec_boxed ("sdes", "SDES", "The SDES items of this session", GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:tls-validation-flags: * * TLS certificate validation flags used to validate server * certificate. * * GLib guarantees that if certificate verification fails, at least one * error will be set, but it does not guarantee that all possible errors * will be set. Accordingly, you may not safely decide to ignore any * particular type of error. * * For example, it would be incorrect to mask %G_TLS_CERTIFICATE_EXPIRED if * you want to allow expired certificates, because this could potentially be * the only error flag set even if other problems exist with the * certificate. * */ g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS, g_param_spec_flags ("tls-validation-flags", "TLS validation flags", "TLS certificate validation flags used to validate the server certificate", G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:tls-database: * * TLS database with anchor certificate authorities used to validate * the server certificate. * */ g_object_class_install_property (gobject_class, PROP_TLS_DATABASE, g_param_spec_object ("tls-database", "TLS database", "TLS database with anchor certificate authorities used to validate the server certificate", G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:tls-interaction: * * A #GTlsInteraction object to be used when the connection or certificate * database need to interact with the user. This will be used to prompt the * user for passwords where necessary. * */ g_object_class_install_property (gobject_class, PROP_TLS_INTERACTION, g_param_spec_object ("tls-interaction", "TLS interaction", "A GTlsInteraction object to prompt the user for password or certificate", G_TYPE_TLS_INTERACTION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:ntp-time-source: * * allows to select the time source that should be used * for the NTP time in outgoing packets * */ g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE, g_param_spec_enum ("ntp-time-source", "NTP Time Source", "NTP time source for RTCP packets", GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE, DEFAULT_NTP_TIME_SOURCE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:user-agent: * * The string to set in the User-Agent header. * */ g_object_class_install_property (gobject_class, PROP_USER_AGENT, g_param_spec_string ("user-agent", "User Agent", "The User-Agent string to send to the server", DEFAULT_USER_AGENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink:publish-clock-mode: * * Sets if and how the media clock should be published according to RFC7273. * * Since: 1.22 * */ g_object_class_install_property (gobject_class, PROP_PUBLISH_CLOCK_MODE, g_param_spec_enum ("publish-clock-mode", "Publish Clock Mode", "Clock publishing mode according to RFC7273", GST_TYPE_RTSP_PUBLISH_CLOCK_MODE, DEFAULT_PUBLISH_CLOCK_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTSPClientSink::handle-request: * @rtsp_client_sink: a #GstRTSPClientSink * @request: a #GstRTSPMessage * @response: a #GstRTSPMessage * * Handle a server request in @request and prepare @response. * * This signal is called from the streaming thread, you should therefore not * do any state changes on @rtsp_client_sink because this might deadlock. If you want * to modify the state as a result of this signal, post a * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread * in some other way. * */ gst_rtsp_client_sink_signals[SIGNAL_HANDLE_REQUEST] = g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0, 0, NULL, NULL, NULL, G_TYPE_NONE, 2, GST_TYPE_RTSP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE, GST_TYPE_RTSP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE); /** * GstRTSPClientSink::new-manager: * @rtsp_client_sink: a #GstRTSPClientSink * @manager: a #GstElement * * Emitted after a new manager (like rtpbin) was created and the default * properties were configured. * */ gst_rtsp_client_sink_signals[SIGNAL_NEW_MANAGER] = g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_FIRST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_ELEMENT); /** * GstRTSPClientSink::new-payloader: * @rtsp_client_sink: a #GstRTSPClientSink * @payloader: a #GstElement * * Emitted after a new RTP payloader was created and the default * properties were configured. * */ gst_rtsp_client_sink_signals[SIGNAL_NEW_PAYLOADER] = g_signal_new_class_handler ("new-payloader", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_FIRST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_ELEMENT); /** * GstRTSPClientSink::request-rtcp-key: * @rtsp_client_sink: a #GstRTSPClientSink * @num: the stream number * * Signal emitted to get the crypto parameters relevant to the RTCP * stream. User should provide the key and the RTCP encryption ciphers * and authentication, and return them wrapped in a GstCaps. * */ gst_rtsp_client_sink_signals[SIGNAL_REQUEST_RTCP_KEY] = g_signal_new ("request-rtcp-key", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT); /** * GstRTSPClientSink::accept-certificate: * @rtsp_client_sink: a #GstRTSPClientSink * @peer_cert: the peer's #GTlsCertificate * @errors: the problems with @peer_cert * @user_data: user data set when the signal handler was connected. * * This will directly map to #GTlsConnection 's "accept-certificate" * signal and be performed after the default checks of #GstRTSPConnection * (checking against the #GTlsDatabase with the given #GTlsCertificateFlags) * have failed. If no #GTlsDatabase is set on this connection, only this * signal will be emitted. * * Since: 1.14 */ gst_rtsp_client_sink_signals[SIGNAL_ACCEPT_CERTIFICATE] = g_signal_new ("accept-certificate", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, g_signal_accumulator_true_handled, NULL, NULL, G_TYPE_BOOLEAN, 3, G_TYPE_TLS_CONNECTION, G_TYPE_TLS_CERTIFICATE, G_TYPE_TLS_CERTIFICATE_FLAGS); /** * GstRTSPClientSink::update-sdp: * @rtsp_client_sink: a #GstRTSPClientSink * @sdp: a #GstSDPMessage * * Emitted right before the ANNOUNCE request is sent to the server with the * generated SDP. The SDP can be updated from signal handlers but the order * and number of medias must not be changed. * * Since: 1.20 */ gst_rtsp_client_sink_signals[SIGNAL_UPDATE_SDP] = g_signal_new_class_handler ("update-sdp", G_TYPE_FROM_CLASS (klass), 0, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE); gstelement_class->provide_clock = gst_rtsp_client_sink_provide_clock; gstelement_class->change_state = gst_rtsp_client_sink_change_state; gstelement_class->request_new_pad = GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_request_new_pad); gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_release_pad); gst_element_class_add_static_pad_template_with_gtype (gstelement_class, &rtptemplate, GST_TYPE_RTSP_CLIENT_SINK_PAD); gst_element_class_set_static_metadata (gstelement_class, "RTSP RECORD client", "Sink/Network", "Send data over the network via RTSP RECORD(RFC 2326)", "Jan Schmidt "); gstbin_class->handle_message = gst_rtsp_client_sink_handle_message; gst_type_mark_as_plugin_api (GST_TYPE_RTSP_CLIENT_SINK_PAD, 0); gst_type_mark_as_plugin_api (GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE, 0); } static void gst_rtsp_client_sink_init (GstRTSPClientSink * sink) { sink->conninfo.location = g_strdup (DEFAULT_LOCATION); sink->protocols = DEFAULT_PROTOCOLS; sink->debug = DEFAULT_DEBUG; sink->retry = DEFAULT_RETRY; sink->udp_timeout = DEFAULT_TIMEOUT; gst_rtsp_client_sink_set_tcp_timeout (sink, DEFAULT_TCP_TIMEOUT); sink->latency = DEFAULT_LATENCY_MS; sink->rtx_time = DEFAULT_RTX_TIME_MS; sink->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE; gst_rtsp_client_sink_set_proxy (sink, DEFAULT_PROXY); sink->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE; sink->user_id = g_strdup (DEFAULT_USER_ID); sink->user_pw = g_strdup (DEFAULT_USER_PW); sink->client_port_range.min = 0; sink->client_port_range.max = 0; sink->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE; sink->udp_reconnect = DEFAULT_UDP_RECONNECT; sink->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE); sink->sdes = NULL; sink->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS; sink->tls_database = DEFAULT_TLS_DATABASE; sink->tls_interaction = DEFAULT_TLS_INTERACTION; sink->ntp_time_source = DEFAULT_NTP_TIME_SOURCE; sink->user_agent = g_strdup (DEFAULT_USER_AGENT); sink->publish_clock_mode = DEFAULT_PUBLISH_CLOCK_MODE; sink->profiles = DEFAULT_PROFILES; /* protects the streaming thread in interleaved mode or the polling * thread in UDP mode. */ g_rec_mutex_init (&sink->stream_rec_lock); /* protects our state changes from multiple invocations */ g_rec_mutex_init (&sink->state_rec_lock); g_mutex_init (&sink->send_lock); g_mutex_init (&sink->preroll_lock); g_cond_init (&sink->preroll_cond); sink->state = GST_RTSP_STATE_INVALID; g_mutex_init (&sink->conninfo.send_lock); g_mutex_init (&sink->conninfo.recv_lock); g_mutex_init (&sink->block_streams_lock); g_cond_init (&sink->block_streams_cond); g_mutex_init (&sink->open_conn_lock); g_cond_init (&sink->open_conn_cond); sink->internal_bin = (GstBin *) gst_bin_new ("rtspbin"); g_object_set (sink->internal_bin, "async-handling", TRUE, NULL); gst_element_set_locked_state (GST_ELEMENT_CAST (sink->internal_bin), TRUE); gst_bin_add (GST_BIN (sink), GST_ELEMENT_CAST (sink->internal_bin)); sink->next_dyn_pt = 96; gst_sdp_message_init (&sink->cursdp); GST_OBJECT_FLAG_SET (sink, GST_ELEMENT_FLAG_SINK); } static void gst_rtsp_client_sink_finalize (GObject * object) { GstRTSPClientSink *rtsp_client_sink; rtsp_client_sink = GST_RTSP_CLIENT_SINK (object); gst_sdp_message_uninit (&rtsp_client_sink->cursdp); g_free (rtsp_client_sink->conninfo.location); gst_rtsp_url_free (rtsp_client_sink->conninfo.url); g_free (rtsp_client_sink->conninfo.url_str); g_free (rtsp_client_sink->user_id); g_free (rtsp_client_sink->user_pw); g_free (rtsp_client_sink->multi_iface); g_free (rtsp_client_sink->user_agent); if (rtsp_client_sink->uri_sdp) { gst_sdp_message_free (rtsp_client_sink->uri_sdp); rtsp_client_sink->uri_sdp = NULL; } if (rtsp_client_sink->provided_clock) gst_object_unref (rtsp_client_sink->provided_clock); if (rtsp_client_sink->sdes) gst_structure_free (rtsp_client_sink->sdes); if (rtsp_client_sink->tls_database) g_object_unref (rtsp_client_sink->tls_database); if (rtsp_client_sink->tls_interaction) g_object_unref (rtsp_client_sink->tls_interaction); /* free locks */ g_rec_mutex_clear (&rtsp_client_sink->stream_rec_lock); g_rec_mutex_clear (&rtsp_client_sink->state_rec_lock); g_mutex_clear (&rtsp_client_sink->conninfo.send_lock); g_mutex_clear (&rtsp_client_sink->conninfo.recv_lock); g_mutex_clear (&rtsp_client_sink->send_lock); g_mutex_clear (&rtsp_client_sink->preroll_lock); g_cond_clear (&rtsp_client_sink->preroll_cond); g_mutex_clear (&rtsp_client_sink->block_streams_lock); g_cond_clear (&rtsp_client_sink->block_streams_cond); g_mutex_clear (&rtsp_client_sink->open_conn_lock); g_cond_clear (&rtsp_client_sink->open_conn_cond); G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_rtp_payloader_filter_func (GstPluginFeature * feature, gpointer user_data) { GstElementFactory *factory = NULL; const gchar *klass; if (!GST_IS_ELEMENT_FACTORY (feature)) return FALSE; factory = GST_ELEMENT_FACTORY (feature); if (gst_plugin_feature_get_rank (feature) == GST_RANK_NONE) return FALSE; if (!gst_element_factory_list_is_type (factory, GST_ELEMENT_FACTORY_TYPE_PAYLOADER)) return FALSE; klass = gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_KLASS); if (strstr (klass, "Codec") == NULL) return FALSE; if (strstr (klass, "RTP") == NULL) return FALSE; return TRUE; } static gint compare_ranks (GstPluginFeature * f1, GstPluginFeature * f2) { gint diff; const gchar *rname1, *rname2; GstRank rank1, rank2; rname1 = gst_plugin_feature_get_name (f1); rname2 = gst_plugin_feature_get_name (f2); rank1 = gst_plugin_feature_get_rank (f1); rank2 = gst_plugin_feature_get_rank (f2); /* HACK: Prefer rtpmp4apay over rtpmp4gpay */ if (g_str_equal (rname1, "rtpmp4apay")) rank1 = GST_RANK_SECONDARY + 1; if (g_str_equal (rname2, "rtpmp4apay")) rank2 = GST_RANK_SECONDARY + 1; diff = rank2 - rank1; if (diff != 0) return diff; diff = strcmp (rname2, rname1); return diff; } static GList * gst_rtsp_client_sink_get_factories (void) { static GList *payloader_factories = NULL; if (g_once_init_enter (&payloader_factories)) { GList *all_factories; all_factories = gst_registry_feature_filter (gst_registry_get (), gst_rtp_payloader_filter_func, FALSE, NULL); all_factories = g_list_sort (all_factories, (GCompareFunc) compare_ranks); g_once_init_leave (&payloader_factories, all_factories); } return payloader_factories; } static GstCaps * gst_rtsp_client_sink_get_payloader_caps (GstElementFactory * factory) { const GList *tmp; GstCaps *caps = gst_caps_new_empty (); for (tmp = gst_element_factory_get_static_pad_templates (factory); tmp; tmp = g_list_next (tmp)) { GstStaticPadTemplate *template = tmp->data; if (template->direction == GST_PAD_SINK) { GstCaps *static_caps = gst_static_pad_template_get_caps (template); GST_LOG ("Found pad template %s on factory %s", template->name_template, gst_plugin_feature_get_name (factory)); if (static_caps) caps = gst_caps_merge (caps, static_caps); /* Early out, any is absorbing */ if (gst_caps_is_any (caps)) goto out; } } out: return caps; } static GstCaps * gst_rtsp_client_sink_get_all_payloaders_caps (void) { /* Cached caps result */ static GstCaps *ret; if (g_once_init_enter (&ret)) { GList *factories, *cur; GstCaps *caps = gst_caps_new_empty (); factories = gst_rtsp_client_sink_get_factories (); for (cur = factories; cur != NULL; cur = g_list_next (cur)) { GstElementFactory *factory = GST_ELEMENT_FACTORY (cur->data); GstCaps *payloader_caps = gst_rtsp_client_sink_get_payloader_caps (factory); caps = gst_caps_merge (caps, payloader_caps); /* Early out, any is absorbing */ if (gst_caps_is_any (caps)) goto out; } GST_MINI_OBJECT_FLAG_SET (caps, GST_MINI_OBJECT_FLAG_MAY_BE_LEAKED); out: g_once_init_leave (&ret, caps); } /* Return cached result */ return gst_caps_ref (ret); } static GstElement * gst_rtsp_client_sink_make_payloader (GstCaps * caps) { GList *factories, *cur; factories = gst_rtsp_client_sink_get_factories (); for (cur = factories; cur != NULL; cur = g_list_next (cur)) { GstElementFactory *factory = GST_ELEMENT_FACTORY (cur->data); const GList *tmp; for (tmp = gst_element_factory_get_static_pad_templates (factory); tmp; tmp = g_list_next (tmp)) { GstStaticPadTemplate *template = tmp->data; if (template->direction == GST_PAD_SINK) { GstCaps *static_caps = gst_static_pad_template_get_caps (template); GstElement *payloader = NULL; if (gst_caps_can_intersect (static_caps, caps)) { GST_DEBUG ("caps %" GST_PTR_FORMAT " intersects with template %" GST_PTR_FORMAT " for payloader %s", caps, static_caps, gst_plugin_feature_get_name (factory)); payloader = gst_element_factory_create (factory, NULL); } gst_caps_unref (static_caps); if (payloader) return payloader; } } } return NULL; } static GstRTSPStream * gst_rtsp_client_sink_create_stream (GstRTSPClientSink * sink, GstRTSPStreamContext * context, GstElement * payloader, GstPad * pad) { GstRTSPStream *stream = NULL; guint pt, aux_pt, ulpfec_pt; GST_OBJECT_LOCK (sink); g_object_get (G_OBJECT (payloader), "pt", &pt, NULL); if (pt >= 96 && pt <= sink->next_dyn_pt) { /* Payloader has a dynamic PT, but one that's already used */ /* FIXME: Create a caps->ptmap instead? */ pt = sink->next_dyn_pt; if (pt > 127) goto no_free_pt; GST_DEBUG_OBJECT (sink, "Assigning pt %u to stream %d", pt, context->index); sink->next_dyn_pt++; } else { GST_DEBUG_OBJECT (sink, "Keeping existing pt %u for stream %d", pt, context->index); } aux_pt = sink->next_dyn_pt; if (aux_pt > 127) goto no_free_pt; sink->next_dyn_pt++; ulpfec_pt = sink->next_dyn_pt; if (ulpfec_pt > 127) goto no_free_pt; sink->next_dyn_pt++; GST_OBJECT_UNLOCK (sink); g_object_set (G_OBJECT (payloader), "pt", pt, NULL); stream = gst_rtsp_stream_new (context->index, payloader, pad); gst_rtsp_stream_set_client_side (stream, TRUE); gst_rtsp_stream_set_retransmission_time (stream, (GstClockTime) (sink->rtx_time) * GST_MSECOND); gst_rtsp_stream_set_protocols (stream, sink->protocols); gst_rtsp_stream_set_profiles (stream, sink->profiles); gst_rtsp_stream_set_retransmission_pt (stream, aux_pt); gst_rtsp_stream_set_buffer_size (stream, sink->udp_buffer_size); if (sink->rtp_blocksize > 0) gst_rtsp_stream_set_mtu (stream, sink->rtp_blocksize); gst_rtsp_stream_set_multicast_iface (stream, sink->multi_iface); gst_rtsp_stream_set_ulpfec_pt (stream, ulpfec_pt); gst_rtsp_stream_set_ulpfec_percentage (stream, context->ulpfec_percentage); gst_rtsp_stream_set_publish_clock_mode (stream, sink->publish_clock_mode); #if 0 if (priv->pool) gst_rtsp_stream_set_address_pool (stream, priv->pool); #endif return stream; no_free_pt: GST_OBJECT_UNLOCK (sink); GST_ELEMENT_ERROR (sink, RESOURCE, NO_SPACE_LEFT, (NULL), ("Ran out of dynamic payload types.")); return NULL; } static GstPadProbeReturn handle_payloader_block (GstPad * pad, GstPadProbeInfo * info, GstRTSPStreamContext * context) { GstRTSPClientSink *sink = context->parent; GST_INFO_OBJECT (sink, "Block on pad %" GST_PTR_FORMAT, pad); g_mutex_lock (&sink->preroll_lock); context->prerolled = TRUE; g_cond_broadcast (&sink->preroll_cond); g_mutex_unlock (&sink->preroll_lock); GST_INFO_OBJECT (sink, "Announced preroll on pad %" GST_PTR_FORMAT, pad); return GST_PAD_PROBE_OK; } static gboolean gst_rtsp_client_sink_setup_payloader (GstRTSPClientSink * sink, GstPad * pad, GstCaps * caps) { GstRTSPStreamContext *context; GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad); GstElement *payloader; GstPad *sinkpad, *srcpad, *ghostsink; context = gst_pad_get_element_private (pad); if (cspad->custom_payloader) { payloader = cspad->custom_payloader; } else { /* Find the payloader. */ payloader = gst_rtsp_client_sink_make_payloader (caps); } if (payloader == NULL) return FALSE; GST_DEBUG_OBJECT (sink, "Configuring payloader %" GST_PTR_FORMAT " for pad %" GST_PTR_FORMAT, payloader, pad); sinkpad = gst_element_get_static_pad (payloader, "sink"); if (sinkpad == NULL) goto no_sinkpad; srcpad = gst_element_get_static_pad (payloader, "src"); if (srcpad == NULL) goto no_srcpad; gst_bin_add (GST_BIN (sink->internal_bin), payloader); ghostsink = gst_ghost_pad_new (NULL, sinkpad); gst_pad_set_active (ghostsink, TRUE); gst_element_add_pad (GST_ELEMENT (sink->internal_bin), ghostsink); g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_NEW_PAYLOADER], 0, payloader); GST_RTSP_STATE_LOCK (sink); context->payloader_block_id = gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, (GstPadProbeCallback) handle_payloader_block, context, NULL); context->payloader = payloader; payloader = gst_object_ref (payloader); gst_ghost_pad_set_target (GST_GHOST_PAD (pad), ghostsink); gst_object_unref (GST_OBJECT (sinkpad)); GST_RTSP_STATE_UNLOCK (sink); context->ulpfec_percentage = cspad->ulpfec_percentage; gst_element_sync_state_with_parent (payloader); gst_object_unref (payloader); gst_object_unref (GST_OBJECT (srcpad)); return TRUE; no_sinkpad: GST_ERROR_OBJECT (sink, "Could not find sink pad on payloader %" GST_PTR_FORMAT, payloader); if (!cspad->custom_payloader) gst_object_unref (payloader); return FALSE; no_srcpad: GST_ERROR_OBJECT (sink, "Could not find src pad on payloader %" GST_PTR_FORMAT, payloader); gst_object_unref (GST_OBJECT (sinkpad)); gst_object_unref (payloader); return TRUE; } static gboolean gst_rtsp_client_sink_sinkpad_event (GstPad * pad, GstObject * parent, GstEvent * event) { if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) { GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad)); if (target == NULL) { GstCaps *caps; /* No target yet - choose a payloader and configure it */ gst_event_parse_caps (event, &caps); GST_DEBUG_OBJECT (parent, "Have set caps event on pad %" GST_PTR_FORMAT " caps %" GST_PTR_FORMAT, pad, caps); if (!gst_rtsp_client_sink_setup_payloader (GST_RTSP_CLIENT_SINK (parent), pad, caps)) { GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad); GST_ELEMENT_ERROR (parent, CORE, NEGOTIATION, ("Could not create payloader"), ("Custom payloader: %p, caps: %" GST_PTR_FORMAT, cspad->custom_payloader, caps)); gst_event_unref (event); return FALSE; } } else { gst_object_unref (target); } } return gst_pad_event_default (pad, parent, event); } static gboolean gst_rtsp_client_sink_sinkpad_query (GstPad * pad, GstObject * parent, GstQuery * query) { if (GST_QUERY_TYPE (query) == GST_QUERY_CAPS) { GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad)); if (target == NULL) { GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad); GstCaps *caps; if (cspad->custom_payloader) { GstPad *sinkpad = gst_element_get_static_pad (cspad->custom_payloader, "sink"); if (sinkpad) { caps = gst_pad_query_caps (sinkpad, NULL); gst_object_unref (sinkpad); } else { GST_ELEMENT_ERROR (parent, CORE, NEGOTIATION, (NULL), ("Custom payloaders are expected to expose a sink pad named 'sink'")); return FALSE; } } else { /* No target yet - return the union of all payloader caps */ caps = gst_rtsp_client_sink_get_all_payloaders_caps (); } GST_TRACE_OBJECT (parent, "Returning payloader caps %" GST_PTR_FORMAT, caps); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); return TRUE; } gst_object_unref (target); } return gst_pad_query_default (pad, parent, query); } static GstPad * gst_rtsp_client_sink_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name, const GstCaps * caps) { GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element); GstPad *pad; GstRTSPStreamContext *context; guint idx = (guint) - 1; gchar *tmpname; g_mutex_lock (&sink->preroll_lock); if (sink->streams_collected) { GST_WARNING_OBJECT (element, "Can't add streams to a running session"); g_mutex_unlock (&sink->preroll_lock); return NULL; } g_mutex_unlock (&sink->preroll_lock); GST_OBJECT_LOCK (sink); if (name) { if (!sscanf (name, "sink_%u", &idx)) { GST_OBJECT_UNLOCK (sink); GST_ERROR_OBJECT (element, "Invalid sink pad name %s", name); return NULL; } if (idx >= sink->next_pad_id) sink->next_pad_id = idx + 1; } if (idx == (guint) - 1) { idx = sink->next_pad_id; sink->next_pad_id++; } GST_OBJECT_UNLOCK (sink); tmpname = g_strdup_printf ("sink_%u", idx); pad = gst_rtsp_client_sink_pad_new (templ, tmpname); g_free (tmpname); GST_DEBUG_OBJECT (element, "Creating request pad %" GST_PTR_FORMAT, pad); gst_pad_set_event_function (pad, GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_sinkpad_event)); gst_pad_set_query_function (pad, GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_sinkpad_query)); context = g_new0 (GstRTSPStreamContext, 1); context->parent = sink; context->index = idx; gst_pad_set_element_private (pad, context); /* The rest of the context is configured on a caps set */ gst_pad_set_active (pad, TRUE); gst_element_add_pad (element, pad); gst_child_proxy_child_added (GST_CHILD_PROXY (element), G_OBJECT (pad), GST_PAD_NAME (pad)); (void) gst_rtsp_client_sink_get_factories (); g_mutex_init (&context->conninfo.send_lock); g_mutex_init (&context->conninfo.recv_lock); GST_RTSP_STATE_LOCK (sink); sink->contexts = g_list_prepend (sink->contexts, context); GST_RTSP_STATE_UNLOCK (sink); return pad; } static void gst_rtsp_client_sink_release_pad (GstElement * element, GstPad * pad) { GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element); GstRTSPStreamContext *context; context = gst_pad_get_element_private (pad); /* FIXME: we may need to change our blocking state waiting for * GstRTSPStreamBlocking messages */ GST_RTSP_STATE_LOCK (sink); sink->contexts = g_list_remove (sink->contexts, context); GST_RTSP_STATE_UNLOCK (sink); /* FIXME: Shut down and clean up streaming on this pad, * do teardown if needed */ GST_LOG_OBJECT (sink, "Cleaning up payloader and stream for released pad %" GST_PTR_FORMAT, pad); if (context->stream_transport) { gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE); gst_object_unref (context->stream_transport); context->stream_transport = NULL; } if (context->stream) { if (context->joined) { gst_rtsp_stream_leave_bin (context->stream, GST_BIN (sink->internal_bin), sink->rtpbin); context->joined = FALSE; } gst_object_unref (context->stream); context->stream = NULL; } if (context->srtcpparams) gst_caps_unref (context->srtcpparams); g_free (context->conninfo.location); context->conninfo.location = NULL; g_mutex_clear (&context->conninfo.send_lock); g_mutex_clear (&context->conninfo.recv_lock); g_free (context); gst_element_remove_pad (element, pad); } static GstClock * gst_rtsp_client_sink_provide_clock (GstElement * element) { GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element); GstClock *clock; if ((clock = sink->provided_clock) != NULL) gst_object_ref (clock); return clock; } /* a proxy string of the format [user:passwd@]host[:port] */ static gboolean gst_rtsp_client_sink_set_proxy (GstRTSPClientSink * rtsp, const gchar * proxy) { gchar *p, *at, *col; g_free (rtsp->proxy_user); rtsp->proxy_user = NULL; g_free (rtsp->proxy_passwd); rtsp->proxy_passwd = NULL; g_free (rtsp->proxy_host); rtsp->proxy_host = NULL; rtsp->proxy_port = 0; p = (gchar *) proxy; if (p == NULL) return TRUE; /* we allow http:// in front but ignore it */ if (g_str_has_prefix (p, "http://")) p += 7; at = strchr (p, '@'); if (at) { /* look for user:passwd */ col = strchr (proxy, ':'); if (col == NULL || col > at) return FALSE; rtsp->proxy_user = g_strndup (p, col - p); col++; rtsp->proxy_passwd = g_strndup (col, at - col); /* move to host */ p = at + 1; } else { if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0') rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id); if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0') rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw); if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) { GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s", GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd)); } } col = strchr (p, ':'); if (col) { /* everything before the colon is the hostname */ rtsp->proxy_host = g_strndup (p, col - p); p = col + 1; rtsp->proxy_port = strtoul (p, (char **) &p, 10); } else { rtsp->proxy_host = g_strdup (p); rtsp->proxy_port = 8080; } return TRUE; } static void gst_rtsp_client_sink_set_tcp_timeout (GstRTSPClientSink * rtsp_client_sink, guint64 timeout) { rtsp_client_sink->tcp_timeout = timeout; } static void gst_rtsp_client_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRTSPClientSink *rtsp_client_sink; rtsp_client_sink = GST_RTSP_CLIENT_SINK (object); switch (prop_id) { case PROP_LOCATION: gst_rtsp_client_sink_uri_set_uri (GST_URI_HANDLER (rtsp_client_sink), g_value_get_string (value), NULL); break; case PROP_PROTOCOLS: rtsp_client_sink->protocols = g_value_get_flags (value); break; case PROP_PROFILES: rtsp_client_sink->profiles = g_value_get_flags (value); break; case PROP_DEBUG: rtsp_client_sink->debug = g_value_get_boolean (value); break; case PROP_RETRY: rtsp_client_sink->retry = g_value_get_uint (value); break; case PROP_TIMEOUT: rtsp_client_sink->udp_timeout = g_value_get_uint64 (value); break; case PROP_TCP_TIMEOUT: gst_rtsp_client_sink_set_tcp_timeout (rtsp_client_sink, g_value_get_uint64 (value)); break; case PROP_LATENCY: rtsp_client_sink->latency = g_value_get_uint (value); break; case PROP_RTX_TIME: rtsp_client_sink->rtx_time = g_value_get_uint (value); break; case PROP_DO_RTSP_KEEP_ALIVE: rtsp_client_sink->do_rtsp_keep_alive = g_value_get_boolean (value); break; case PROP_PROXY: gst_rtsp_client_sink_set_proxy (rtsp_client_sink, g_value_get_string (value)); break; case PROP_PROXY_ID: if (rtsp_client_sink->prop_proxy_id) g_free (rtsp_client_sink->prop_proxy_id); rtsp_client_sink->prop_proxy_id = g_value_dup_string (value); break; case PROP_PROXY_PW: if (rtsp_client_sink->prop_proxy_pw) g_free (rtsp_client_sink->prop_proxy_pw); rtsp_client_sink->prop_proxy_pw = g_value_dup_string (value); break; case PROP_RTP_BLOCKSIZE: rtsp_client_sink->rtp_blocksize = g_value_get_uint (value); break; case PROP_USER_ID: if (rtsp_client_sink->user_id) g_free (rtsp_client_sink->user_id); rtsp_client_sink->user_id = g_value_dup_string (value); break; case PROP_USER_PW: if (rtsp_client_sink->user_pw) g_free (rtsp_client_sink->user_pw); rtsp_client_sink->user_pw = g_value_dup_string (value); break; case PROP_PORT_RANGE: { const gchar *str; str = g_value_get_string (value); if (!str || !sscanf (str, "%u-%u", &rtsp_client_sink->client_port_range.min, &rtsp_client_sink->client_port_range.max)) { rtsp_client_sink->client_port_range.min = 0; rtsp_client_sink->client_port_range.max = 0; } break; } case PROP_UDP_BUFFER_SIZE: rtsp_client_sink->udp_buffer_size = g_value_get_int (value); break; case PROP_UDP_RECONNECT: rtsp_client_sink->udp_reconnect = g_value_get_boolean (value); break; case PROP_MULTICAST_IFACE: g_free (rtsp_client_sink->multi_iface); if (g_value_get_string (value) == NULL) rtsp_client_sink->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE); else rtsp_client_sink->multi_iface = g_value_dup_string (value); break; case PROP_SDES: rtsp_client_sink->sdes = g_value_dup_boxed (value); break; case PROP_TLS_VALIDATION_FLAGS: rtsp_client_sink->tls_validation_flags = g_value_get_flags (value); break; case PROP_TLS_DATABASE: g_clear_object (&rtsp_client_sink->tls_database); rtsp_client_sink->tls_database = g_value_dup_object (value); break; case PROP_TLS_INTERACTION: g_clear_object (&rtsp_client_sink->tls_interaction); rtsp_client_sink->tls_interaction = g_value_dup_object (value); break; case PROP_NTP_TIME_SOURCE: rtsp_client_sink->ntp_time_source = g_value_get_enum (value); break; case PROP_USER_AGENT: g_free (rtsp_client_sink->user_agent); rtsp_client_sink->user_agent = g_value_dup_string (value); break; case PROP_PUBLISH_CLOCK_MODE: rtsp_client_sink->publish_clock_mode = g_value_get_enum (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtsp_client_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRTSPClientSink *rtsp_client_sink; rtsp_client_sink = GST_RTSP_CLIENT_SINK (object); switch (prop_id) { case PROP_LOCATION: g_value_set_string (value, rtsp_client_sink->conninfo.location); break; case PROP_PROTOCOLS: g_value_set_flags (value, rtsp_client_sink->protocols); break; case PROP_PROFILES: g_value_set_flags (value, rtsp_client_sink->profiles); break; case PROP_DEBUG: g_value_set_boolean (value, rtsp_client_sink->debug); break; case PROP_RETRY: g_value_set_uint (value, rtsp_client_sink->retry); break; case PROP_TIMEOUT: g_value_set_uint64 (value, rtsp_client_sink->udp_timeout); break; case PROP_TCP_TIMEOUT: g_value_set_uint64 (value, rtsp_client_sink->tcp_timeout); break; case PROP_LATENCY: g_value_set_uint (value, rtsp_client_sink->latency); break; case PROP_RTX_TIME: g_value_set_uint (value, rtsp_client_sink->rtx_time); break; case PROP_DO_RTSP_KEEP_ALIVE: g_value_set_boolean (value, rtsp_client_sink->do_rtsp_keep_alive); break; case PROP_PROXY: { gchar *str; if (rtsp_client_sink->proxy_host) { str = g_strdup_printf ("%s:%d", rtsp_client_sink->proxy_host, rtsp_client_sink->proxy_port); } else { str = NULL; } g_value_take_string (value, str); break; } case PROP_PROXY_ID: g_value_set_string (value, rtsp_client_sink->prop_proxy_id); break; case PROP_PROXY_PW: g_value_set_string (value, rtsp_client_sink->prop_proxy_pw); break; case PROP_RTP_BLOCKSIZE: g_value_set_uint (value, rtsp_client_sink->rtp_blocksize); break; case PROP_USER_ID: g_value_set_string (value, rtsp_client_sink->user_id); break; case PROP_USER_PW: g_value_set_string (value, rtsp_client_sink->user_pw); break; case PROP_PORT_RANGE: { gchar *str; if (rtsp_client_sink->client_port_range.min != 0) { str = g_strdup_printf ("%u-%u", rtsp_client_sink->client_port_range.min, rtsp_client_sink->client_port_range.max); } else { str = NULL; } g_value_take_string (value, str); break; } case PROP_UDP_BUFFER_SIZE: g_value_set_int (value, rtsp_client_sink->udp_buffer_size); break; case PROP_UDP_RECONNECT: g_value_set_boolean (value, rtsp_client_sink->udp_reconnect); break; case PROP_MULTICAST_IFACE: g_value_set_string (value, rtsp_client_sink->multi_iface); break; case PROP_SDES: g_value_set_boxed (value, rtsp_client_sink->sdes); break; case PROP_TLS_VALIDATION_FLAGS: g_value_set_flags (value, rtsp_client_sink->tls_validation_flags); break; case PROP_TLS_DATABASE: g_value_set_object (value, rtsp_client_sink->tls_database); break; case PROP_TLS_INTERACTION: g_value_set_object (value, rtsp_client_sink->tls_interaction); break; case PROP_NTP_TIME_SOURCE: g_value_set_enum (value, rtsp_client_sink->ntp_time_source); break; case PROP_USER_AGENT: g_value_set_string (value, rtsp_client_sink->user_agent); break; case PROP_PUBLISH_CLOCK_MODE: g_value_set_enum (value, rtsp_client_sink->publish_clock_mode); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static const gchar * get_aggregate_control (GstRTSPClientSink * sink) { const gchar *base; if (sink->control) base = sink->control; else if (sink->content_base) base = sink->content_base; else if (sink->conninfo.url_str) base = sink->conninfo.url_str; else base = "/"; return base; } static void gst_rtsp_client_sink_cleanup (GstRTSPClientSink * sink) { GList *walk; GST_DEBUG_OBJECT (sink, "cleanup"); gst_element_set_state (GST_ELEMENT (sink->internal_bin), GST_STATE_NULL); /* Clean up any left over stream objects */ for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstRTSPStreamContext *context = (GstRTSPStreamContext *) (walk->data); if (context->stream_transport) { gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE); gst_object_unref (context->stream_transport); context->stream_transport = NULL; } if (context->stream) { if (context->joined) { gst_rtsp_stream_leave_bin (context->stream, GST_BIN (sink->internal_bin), sink->rtpbin); context->joined = FALSE; } gst_object_unref (context->stream); context->stream = NULL; } if (context->srtcpparams) { gst_caps_unref (context->srtcpparams); context->srtcpparams = NULL; } g_free (context->conninfo.location); context->conninfo.location = NULL; } if (sink->rtpbin) { gst_element_set_state (sink->rtpbin, GST_STATE_NULL); gst_bin_remove (GST_BIN_CAST (sink->internal_bin), sink->rtpbin); sink->rtpbin = NULL; } g_free (sink->content_base); sink->content_base = NULL; g_free (sink->control); sink->control = NULL; if (sink->range) gst_rtsp_range_free (sink->range); sink->range = NULL; /* don't clear the SDP when it was used in the url */ if (sink->uri_sdp && !sink->from_sdp) { gst_sdp_message_free (sink->uri_sdp); sink->uri_sdp = NULL; } if (sink->provided_clock) { gst_object_unref (sink->provided_clock); sink->provided_clock = NULL; } g_free (sink->server_ip); sink->server_ip = NULL; sink->next_pad_id = 0; sink->next_dyn_pt = 96; } static GstRTSPResult gst_rtsp_client_sink_connection_send (GstRTSPClientSink * sink, GstRTSPConnInfo * conninfo, GstRTSPMessage * message, gint64 timeout) { GstRTSPResult ret; if (conninfo->connection) { g_mutex_lock (&conninfo->send_lock); ret = gst_rtsp_connection_send_usec (conninfo->connection, message, timeout); g_mutex_unlock (&conninfo->send_lock); } else { ret = GST_RTSP_ERROR; } return ret; } static GstRTSPResult gst_rtsp_client_sink_connection_send_messages (GstRTSPClientSink * sink, GstRTSPConnInfo * conninfo, GstRTSPMessage * messages, guint n_messages, gint64 timeout) { GstRTSPResult ret; if (conninfo->connection) { g_mutex_lock (&conninfo->send_lock); ret = gst_rtsp_connection_send_messages_usec (conninfo->connection, messages, n_messages, timeout); g_mutex_unlock (&conninfo->send_lock); } else { ret = GST_RTSP_ERROR; } return ret; } static GstRTSPResult gst_rtsp_client_sink_connection_receive (GstRTSPClientSink * sink, GstRTSPConnInfo * conninfo, GstRTSPMessage * message, gint64 timeout) { GstRTSPResult ret; if (conninfo->connection) { g_mutex_lock (&conninfo->recv_lock); ret = gst_rtsp_connection_receive_usec (conninfo->connection, message, timeout); g_mutex_unlock (&conninfo->recv_lock); } else { ret = GST_RTSP_ERROR; } return ret; } static gboolean accept_certificate_cb (GTlsConnection * conn, GTlsCertificate * peer_cert, GTlsCertificateFlags errors, gpointer user_data) { GstRTSPClientSink *sink = user_data; gboolean accept = FALSE; g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_ACCEPT_CERTIFICATE], 0, conn, peer_cert, errors, &accept); return accept; } static GstRTSPResult gst_rtsp_conninfo_connect (GstRTSPClientSink * sink, GstRTSPConnInfo * info, gboolean async) { GstRTSPResult res; if (info->connection == NULL) { if (info->url == NULL) { GST_DEBUG_OBJECT (sink, "parsing uri (%s)...", info->location); if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0) goto parse_error; } /* create connection */ GST_DEBUG_OBJECT (sink, "creating connection (%s)...", info->location); if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0) goto could_not_create; if (info->url_str) g_free (info->url_str); info->url_str = gst_rtsp_url_get_request_uri (info->url); GST_DEBUG_OBJECT (sink, "sanitized uri %s", info->url_str); if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) { if (!gst_rtsp_connection_set_tls_validation_flags (info->connection, sink->tls_validation_flags)) GST_WARNING_OBJECT (sink, "Unable to set TLS validation flags"); if (sink->tls_database) gst_rtsp_connection_set_tls_database (info->connection, sink->tls_database); if (sink->tls_interaction) gst_rtsp_connection_set_tls_interaction (info->connection, sink->tls_interaction); gst_rtsp_connection_set_accept_certificate_func (info->connection, accept_certificate_cb, sink, NULL); } if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP) gst_rtsp_connection_set_tunneled (info->connection, TRUE); if (sink->proxy_host) { GST_DEBUG_OBJECT (sink, "setting proxy %s:%d", sink->proxy_host, sink->proxy_port); gst_rtsp_connection_set_proxy (info->connection, sink->proxy_host, sink->proxy_port); } } if (!info->connected) { /* connect */ if (async) GST_ELEMENT_PROGRESS (sink, CONTINUE, "connect", ("Connecting to %s", info->location)); GST_DEBUG_OBJECT (sink, "connecting (%s)...", info->location); if ((res = gst_rtsp_connection_connect_usec (info->connection, sink->tcp_timeout)) < 0) goto could_not_connect; info->connected = TRUE; } return GST_RTSP_OK; /* ERRORS */ parse_error: { GST_ERROR_OBJECT (sink, "No valid RTSP URL was provided"); return res; } could_not_create: { gchar *str = gst_rtsp_strresult (res); GST_ERROR_OBJECT (sink, "Could not create connection. (%s)", str); g_free (str); return res; } could_not_connect: { gchar *str = gst_rtsp_strresult (res); GST_ERROR_OBJECT (sink, "Could not connect to server. (%s)", str); g_free (str); return res; } } static GstRTSPResult gst_rtsp_conninfo_close (GstRTSPClientSink * sink, GstRTSPConnInfo * info, gboolean free) { GST_RTSP_STATE_LOCK (sink); if (info->connected) { GST_DEBUG_OBJECT (sink, "closing connection..."); gst_rtsp_connection_close (info->connection); info->connected = FALSE; } if (free && info->connection) { /* free connection */ GST_DEBUG_OBJECT (sink, "freeing connection..."); gst_rtsp_connection_free (info->connection); g_mutex_lock (&sink->preroll_lock); info->connection = NULL; g_cond_broadcast (&sink->preroll_cond); g_mutex_unlock (&sink->preroll_lock); } GST_RTSP_STATE_UNLOCK (sink); return GST_RTSP_OK; } static GstRTSPResult gst_rtsp_conninfo_reconnect (GstRTSPClientSink * sink, GstRTSPConnInfo * info, gboolean async) { GstRTSPResult res; GST_DEBUG_OBJECT (sink, "reconnecting connection..."); gst_rtsp_conninfo_close (sink, info, FALSE); res = gst_rtsp_conninfo_connect (sink, info, async); return res; } static void gst_rtsp_client_sink_connection_flush (GstRTSPClientSink * sink, gboolean flush) { GList *walk; GST_DEBUG_OBJECT (sink, "set flushing %d", flush); g_mutex_lock (&sink->preroll_lock); if (sink->conninfo.connection && sink->conninfo.flushing != flush) { GST_DEBUG_OBJECT (sink, "connection flush"); gst_rtsp_connection_flush (sink->conninfo.connection, flush); sink->conninfo.flushing = flush; } for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data; if (stream->conninfo.connection && stream->conninfo.flushing != flush) { GST_DEBUG_OBJECT (sink, "stream %p flush", stream); gst_rtsp_connection_flush (stream->conninfo.connection, flush); stream->conninfo.flushing = flush; } } g_cond_broadcast (&sink->preroll_cond); g_mutex_unlock (&sink->preroll_lock); } static GstRTSPResult gst_rtsp_client_sink_init_request (GstRTSPClientSink * sink, GstRTSPMessage * msg, GstRTSPMethod method, const gchar * uri) { GstRTSPResult res; res = gst_rtsp_message_init_request (msg, method, uri); if (res < 0) return res; /* set user-agent */ if (sink->user_agent) gst_rtsp_message_add_header (msg, GST_RTSP_HDR_USER_AGENT, sink->user_agent); return res; } /* FIXME, handle server request, reply with OK, for now */ static GstRTSPResult gst_rtsp_client_sink_handle_request (GstRTSPClientSink * sink, GstRTSPConnInfo * conninfo, GstRTSPMessage * request) { GstRTSPMessage response = { 0 }; GstRTSPResult res; GST_DEBUG_OBJECT (sink, "got server request message"); if (sink->debug) gst_rtsp_message_dump (request); /* default implementation, send OK */ GST_DEBUG_OBJECT (sink, "prepare OK reply"); res = gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK", request); if (res < 0) goto send_error; /* let app parse and reply */ g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_HANDLE_REQUEST], 0, request, &response); if (sink->debug) gst_rtsp_message_dump (&response); res = gst_rtsp_client_sink_connection_send (sink, conninfo, &response, 0); if (res < 0) goto send_error; gst_rtsp_message_unset (&response); return GST_RTSP_OK; /* ERRORS */ send_error: { gst_rtsp_message_unset (&response); return res; } } /* send server keep-alive */ static GstRTSPResult gst_rtsp_client_sink_send_keep_alive (GstRTSPClientSink * sink) { GstRTSPMessage request = { 0 }; GstRTSPResult res; GstRTSPMethod method; const gchar *control; if (sink->do_rtsp_keep_alive == FALSE) { GST_DEBUG_OBJECT (sink, "do-rtsp-keep-alive is FALSE, not sending."); gst_rtsp_connection_reset_timeout (sink->conninfo.connection); return GST_RTSP_OK; } GST_DEBUG_OBJECT (sink, "creating server keep-alive"); /* find a method to use for keep-alive */ if (sink->methods & GST_RTSP_GET_PARAMETER) method = GST_RTSP_GET_PARAMETER; else method = GST_RTSP_OPTIONS; control = get_aggregate_control (sink); if (control == NULL) goto no_control; res = gst_rtsp_client_sink_init_request (sink, &request, method, control); if (res < 0) goto send_error; if (sink->debug) gst_rtsp_message_dump (&request); res = gst_rtsp_client_sink_connection_send (sink, &sink->conninfo, &request, 0); if (res < 0) goto send_error; gst_rtsp_connection_reset_timeout (sink->conninfo.connection); gst_rtsp_message_unset (&request); return GST_RTSP_OK; /* ERRORS */ no_control: { GST_WARNING_OBJECT (sink, "no control url to send keepalive"); return GST_RTSP_OK; } send_error: { gchar *str = gst_rtsp_strresult (res); gst_rtsp_message_unset (&request); GST_ELEMENT_WARNING (sink, RESOURCE, WRITE, (NULL), ("Could not send keep-alive. (%s)", str)); g_free (str); return res; } } static GstFlowReturn gst_rtsp_client_sink_loop_rx (GstRTSPClientSink * sink) { GstRTSPResult res; GstRTSPMessage message = { 0 }; gint retry = 0; while (TRUE) { gint64 timeout; /* get the next timeout interval */ timeout = gst_rtsp_connection_next_timeout_usec (sink->conninfo.connection); GST_DEBUG_OBJECT (sink, "doing receive with timeout %d seconds", (gint) timeout / G_USEC_PER_SEC); gst_rtsp_message_unset (&message); /* 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_rtsp_client_sink_connection_receive (sink, &sink->conninfo, &message, timeout); switch (res) { case GST_RTSP_OK: GST_DEBUG_OBJECT (sink, "we received a server message"); break; case GST_RTSP_EINTR: /* we got interrupted, see what we have to do */ goto interrupt; case GST_RTSP_ETIMEOUT: /* send keep-alive, ignore the result, a warning will be posted. */ GST_DEBUG_OBJECT (sink, "timeout, sending keep-alive"); if ((res = gst_rtsp_client_sink_send_keep_alive (sink)) == GST_RTSP_EINTR) goto interrupt; continue; case GST_RTSP_EEOF: /* server closed the connection. not very fatal for UDP, reconnect and * see what happens. */ GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL), ("The server closed the connection.")); if (sink->udp_reconnect) { if ((res = gst_rtsp_conninfo_reconnect (sink, &sink->conninfo, FALSE)) < 0) goto connect_error; } else { goto server_eof; } continue; break; case GST_RTSP_ENET: GST_DEBUG_OBJECT (sink, "An ethernet problem occured."); default: GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL), ("Unhandled return value %d.", res)); goto receive_error; } switch (message.type) { case GST_RTSP_MESSAGE_REQUEST: /* server sends us a request message, handle it */ res = gst_rtsp_client_sink_handle_request (sink, &sink->conninfo, &message); if (res == GST_RTSP_EEOF) goto server_eof; else if (res < 0) goto handle_request_failed; break; case GST_RTSP_MESSAGE_RESPONSE: /* we ignore response and data messages */ GST_DEBUG_OBJECT (sink, "ignoring response message"); if (sink->debug) gst_rtsp_message_dump (&message); if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) { GST_DEBUG_OBJECT (sink, "but is Unauthorized response ..."); if (gst_rtsp_client_sink_setup_auth (sink, &message) && !(retry++)) { GST_DEBUG_OBJECT (sink, "so retrying keep-alive"); if ((res = gst_rtsp_client_sink_send_keep_alive (sink)) == GST_RTSP_EINTR) goto interrupt; } } else { retry = 0; } break; case GST_RTSP_MESSAGE_DATA: /* we ignore response and data messages */ GST_DEBUG_OBJECT (sink, "ignoring data message"); break; default: GST_WARNING_OBJECT (sink, "ignoring unknown message type %d", message.type); break; } } g_assert_not_reached (); /* we get here when the connection got interrupted */ interrupt: { gst_rtsp_message_unset (&message); GST_DEBUG_OBJECT (sink, "got interrupted"); return GST_FLOW_FLUSHING; } connect_error: { gchar *str = gst_rtsp_strresult (res); GstFlowReturn ret; sink->conninfo.connected = FALSE; if (res != GST_RTSP_EINTR) { GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ_WRITE, (NULL), ("Could not connect to server. (%s)", str)); g_free (str); ret = GST_FLOW_ERROR; } else { ret = GST_FLOW_FLUSHING; } return ret; } receive_error: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (sink, 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); GstFlowReturn ret; gst_rtsp_message_unset (&message); if (res != GST_RTSP_EINTR) { GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Could not handle server message. (%s)", str)); g_free (str); ret = GST_FLOW_ERROR; } else { ret = GST_FLOW_FLUSHING; } return ret; } server_eof: { GST_DEBUG_OBJECT (sink, "we got an eof from the server"); GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL), ("The server closed the connection.")); sink->conninfo.connected = FALSE; gst_rtsp_message_unset (&message); return GST_FLOW_EOS; } } static GstRTSPResult gst_rtsp_client_sink_reconnect (GstRTSPClientSink * sink, gboolean async) { GstRTSPResult res = GST_RTSP_OK; gboolean restart = FALSE; GST_DEBUG_OBJECT (sink, "doing reconnect"); GST_FIXME_OBJECT (sink, "Reconnection is not yet implemented"); /* no need to restart, we're done */ if (!restart) goto done; /* we can try only TCP now */ sink->cur_protocols = GST_RTSP_LOWER_TRANS_TCP; /* close and cleanup our state */ if ((res = gst_rtsp_client_sink_close (sink, async, FALSE)) < 0) goto done; /* see if we have TCP left to try. Also don't try TCP when we were configured * with an SDP. */ if (!(sink->protocols & GST_RTSP_LOWER_TRANS_TCP) || sink->from_sdp) goto no_protocols; /* We post a warning message now to inform the user * that nothing happened. It's most likely a firewall thing. */ GST_ELEMENT_WARNING (sink, 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 (sink->udp_timeout / 1000000.0))); /* open new connection using tcp */ if (gst_rtsp_client_sink_open (sink, async) < 0) goto open_failed; /* start recording */ if (gst_rtsp_client_sink_record (sink, async) < 0) goto play_failed; done: return res; /* ERRORS */ no_protocols: { sink->cur_protocols = 0; /* no transport possible, post an error and stop */ GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL), ("Could not receive any UDP packets for %.4f seconds, maybe your " "firewall is blocking it. No other protocols to try.", gst_guint64_to_gdouble (sink->udp_timeout / 1000000.0))); return GST_RTSP_ERROR; } open_failed: { GST_DEBUG_OBJECT (sink, "open failed"); return GST_RTSP_OK; } play_failed: { GST_DEBUG_OBJECT (sink, "play failed"); return GST_RTSP_OK; } } static void gst_rtsp_client_sink_loop_start_cmd (GstRTSPClientSink * sink, gint cmd) { switch (cmd) { case CMD_OPEN: GST_ELEMENT_PROGRESS (sink, START, "open", ("Opening Stream")); break; case CMD_RECORD: GST_ELEMENT_PROGRESS (sink, START, "request", ("Sending RECORD request")); break; case CMD_PAUSE: GST_ELEMENT_PROGRESS (sink, START, "request", ("Sending PAUSE request")); break; case CMD_CLOSE: GST_ELEMENT_PROGRESS (sink, START, "close", ("Closing Stream")); break; default: break; } } static void gst_rtsp_client_sink_loop_complete_cmd (GstRTSPClientSink * sink, gint cmd) { switch (cmd) { case CMD_OPEN: GST_ELEMENT_PROGRESS (sink, COMPLETE, "open", ("Opened Stream")); break; case CMD_RECORD: GST_ELEMENT_PROGRESS (sink, COMPLETE, "request", ("Sent RECORD request")); break; case CMD_PAUSE: GST_ELEMENT_PROGRESS (sink, COMPLETE, "request", ("Sent PAUSE request")); break; case CMD_CLOSE: GST_ELEMENT_PROGRESS (sink, COMPLETE, "close", ("Closed Stream")); break; default: break; } } static void gst_rtsp_client_sink_loop_cancel_cmd (GstRTSPClientSink * sink, gint cmd) { switch (cmd) { case CMD_OPEN: GST_ELEMENT_PROGRESS (sink, CANCELED, "open", ("Open canceled")); break; case CMD_RECORD: GST_ELEMENT_PROGRESS (sink, CANCELED, "request", ("RECORD canceled")); break; case CMD_PAUSE: GST_ELEMENT_PROGRESS (sink, CANCELED, "request", ("PAUSE canceled")); break; case CMD_CLOSE: GST_ELEMENT_PROGRESS (sink, CANCELED, "close", ("Close canceled")); break; default: break; } } static void gst_rtsp_client_sink_loop_error_cmd (GstRTSPClientSink * sink, gint cmd) { switch (cmd) { case CMD_OPEN: GST_ELEMENT_PROGRESS (sink, ERROR, "open", ("Open failed")); break; case CMD_RECORD: GST_ELEMENT_PROGRESS (sink, ERROR, "request", ("RECORD failed")); break; case CMD_PAUSE: GST_ELEMENT_PROGRESS (sink, ERROR, "request", ("PAUSE failed")); break; case CMD_CLOSE: GST_ELEMENT_PROGRESS (sink, ERROR, "close", ("Close failed")); break; default: break; } } static void gst_rtsp_client_sink_loop_end_cmd (GstRTSPClientSink * sink, gint cmd, GstRTSPResult ret) { if (ret == GST_RTSP_OK) gst_rtsp_client_sink_loop_complete_cmd (sink, cmd); else if (ret == GST_RTSP_EINTR) gst_rtsp_client_sink_loop_cancel_cmd (sink, cmd); else gst_rtsp_client_sink_loop_error_cmd (sink, cmd); } static gboolean gst_rtsp_client_sink_loop_send_cmd (GstRTSPClientSink * sink, gint cmd, gint mask) { gint old; gboolean flushed = FALSE; /* start new request */ gst_rtsp_client_sink_loop_start_cmd (sink, cmd); GST_DEBUG_OBJECT (sink, "sending cmd %s", cmd_to_string (cmd)); GST_OBJECT_LOCK (sink); old = sink->pending_cmd; if (old == CMD_RECONNECT) { GST_DEBUG_OBJECT (sink, "ignore, we were reconnecting"); cmd = CMD_RECONNECT; } if (old != CMD_WAIT) { sink->pending_cmd = CMD_WAIT; GST_OBJECT_UNLOCK (sink); /* cancel previous request */ GST_DEBUG_OBJECT (sink, "cancel previous request %s", cmd_to_string (old)); gst_rtsp_client_sink_loop_cancel_cmd (sink, old); GST_OBJECT_LOCK (sink); } sink->pending_cmd = cmd; /* interrupt if allowed */ if (sink->busy_cmd & mask) { GST_DEBUG_OBJECT (sink, "connection flush busy %s", cmd_to_string (sink->busy_cmd)); gst_rtsp_client_sink_connection_flush (sink, TRUE); flushed = TRUE; } else { GST_DEBUG_OBJECT (sink, "not interrupting busy cmd %s", cmd_to_string (sink->busy_cmd)); } if (sink->task) gst_task_start (sink->task); GST_OBJECT_UNLOCK (sink); return flushed; } static gboolean gst_rtsp_client_sink_loop (GstRTSPClientSink * sink) { GstFlowReturn ret; if (!sink->conninfo.connection || !sink->conninfo.connected) goto no_connection; ret = gst_rtsp_client_sink_loop_rx (sink); if (ret != GST_FLOW_OK) goto pause; return TRUE; /* ERRORS */ no_connection: { GST_WARNING_OBJECT (sink, "we are not connected"); ret = GST_FLOW_FLUSHING; goto pause; } pause: { const gchar *reason = gst_flow_get_name (ret); GST_DEBUG_OBJECT (sink, "pausing task, reason %s", reason); gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT, CMD_LOOP); return FALSE; } } #ifndef GST_DISABLE_GST_DEBUG static 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 * * This code should also cope with the fact that each WWW-Authenticate * header can contain multiple challenge methods + tokens * * At the moment, for Basic auth, we just do a minimal check and don't * even parse out the realm */ static void gst_rtsp_client_sink_parse_auth_hdr (GstRTSPMessage * response, GstRTSPAuthMethod * methods, GstRTSPConnection * conn, gboolean * stale) { GstRTSPAuthCredential **credentials, **credential; g_return_if_fail (response != NULL); g_return_if_fail (methods != NULL); g_return_if_fail (stale != NULL); credentials = gst_rtsp_message_parse_auth_credentials (response, GST_RTSP_HDR_WWW_AUTHENTICATE); if (!credentials) return; credential = credentials; while (*credential) { if ((*credential)->scheme == GST_RTSP_AUTH_BASIC) { *methods |= GST_RTSP_AUTH_BASIC; } else if ((*credential)->scheme == GST_RTSP_AUTH_DIGEST) { GstRTSPAuthParam **param = (*credential)->params; *methods |= GST_RTSP_AUTH_DIGEST; gst_rtsp_connection_clear_auth_params (conn); *stale = FALSE; while (*param) { if (strcmp ((*param)->name, "stale") == 0 && g_ascii_strcasecmp ((*param)->value, "TRUE") == 0) *stale = TRUE; gst_rtsp_connection_set_auth_param (conn, (*param)->name, (*param)->value); param++; } } credential++; } gst_rtsp_auth_credentials_free (credentials); } /** * gst_rtsp_client_sink_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_rtsp_client_sink_setup_auth (GstRTSPClientSink * sink, GstRTSPMessage * response) { gchar *user = NULL; gchar *pass = NULL; GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE; GstRTSPAuthMethod method; GstRTSPResult auth_result; GstRTSPUrl *url; GstRTSPConnection *conn; gboolean stale = FALSE; conn = sink->conninfo.connection; /* Identify the available auth methods and see if any are supported */ gst_rtsp_client_sink_parse_auth_hdr (response, &avail_methods, conn, &stale); if (avail_methods == GST_RTSP_AUTH_NONE) goto no_auth_available; /* 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 */ if (stale) sink->tried_url_auth = FALSE; url = gst_rtsp_connection_get_url (conn); /* Do we have username and password available? */ if (url != NULL && !sink->tried_url_auth && url->user != NULL && url->passwd != NULL) { user = url->user; pass = url->passwd; sink->tried_url_auth = TRUE; GST_DEBUG_OBJECT (sink, "Attempting authentication using credentials from the URL"); } else { user = sink->user_id; pass = sink->user_pw; GST_DEBUG_OBJECT (sink, "Attempting authentication using credentials from the properties"); } /* 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 (conn, 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 (sink, "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 (sink, 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_rtsp_client_sink_try_send (GstRTSPClientSink * sink, GstRTSPConnInfo * conninfo, GstRTSPMessage * requests, guint n_requests, GstRTSPMessage * response, GstRTSPStatusCode * code) { GstRTSPResult res; GstRTSPStatusCode thecode; gchar *content_base = NULL; gint try = 0; g_assert (n_requests == 1 || response == NULL); again: GST_DEBUG_OBJECT (sink, "sending message"); if (sink->debug && n_requests == 1) gst_rtsp_message_dump (&requests[0]); g_mutex_lock (&sink->send_lock); res = gst_rtsp_client_sink_connection_send_messages (sink, conninfo, requests, n_requests, sink->tcp_timeout); if (res < 0) { g_mutex_unlock (&sink->send_lock); goto send_error; } gst_rtsp_connection_reset_timeout (conninfo->connection); /* See if we should handle the response */ if (response == NULL) { g_mutex_unlock (&sink->send_lock); return GST_RTSP_OK; } next: res = gst_rtsp_client_sink_connection_receive (sink, conninfo, response, sink->tcp_timeout); g_mutex_unlock (&sink->send_lock); if (res < 0) goto receive_error; if (sink->debug) gst_rtsp_message_dump (response); switch (response->type) { case GST_RTSP_MESSAGE_REQUEST: res = gst_rtsp_client_sink_handle_request (sink, conninfo, response); if (res == GST_RTSP_EEOF) goto server_eof; else if (res < 0) goto handle_request_failed; g_mutex_lock (&sink->send_lock); goto next; case GST_RTSP_MESSAGE_RESPONSE: /* ok, a response is good */ GST_DEBUG_OBJECT (sink, "received response message"); break; case GST_RTSP_MESSAGE_DATA: /* we ignore data messages */ GST_DEBUG_OBJECT (sink, "ignoring data message"); g_mutex_lock (&sink->send_lock); goto next; default: GST_WARNING_OBJECT (sink, "ignoring unknown message type %d", response->type); g_mutex_lock (&sink->send_lock); goto next; } thecode = response->type_data.response.code; GST_DEBUG_OBJECT (sink, "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 (sink->content_base); sink->content_base = g_strdup (content_base); } return GST_RTSP_OK; /* ERRORS */ send_error: { gchar *str = gst_rtsp_strresult (res); if (res != GST_RTSP_EINTR) { GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Could not send message. (%s)", str)); } else { GST_WARNING_OBJECT (sink, "send interrupted"); } g_free (str); return res; } receive_error: { switch (res) { case GST_RTSP_EEOF: GST_WARNING_OBJECT (sink, "server closed connection"); if ((try == 0) && !sink->interleaved && sink->udp_reconnect) { try++; /* if reconnect succeeds, try again */ if ((res = gst_rtsp_conninfo_reconnect (sink, &sink->conninfo, FALSE)) == 0) goto again; } /* only try once after reconnect, then fallthrough and error out */ default: { gchar *str = gst_rtsp_strresult (res); if (res != GST_RTSP_EINTR) { GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL), ("Could not receive message. (%s)", str)); } else { GST_WARNING_OBJECT (sink, "receive interrupted"); } g_free (str); break; } } return res; } handle_request_failed: { /* ERROR was posted */ gst_rtsp_message_unset (response); return res; } server_eof: { GST_DEBUG_OBJECT (sink, "we got an eof from the server"); GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL), ("The server closed the connection.")); gst_rtsp_message_unset (response); return res; } } static void gst_rtsp_client_sink_set_state (GstRTSPClientSink * sink, GstState state) { GST_DEBUG_OBJECT (sink, "Setting internal state to %s", gst_element_state_get_name (state)); gst_element_set_state (GST_ELEMENT (sink->internal_bin), state); } /** * gst_rtsp_client_sink_send: * @src: the rtsp source * @conn: the connection to send on * @request: must point to a valid request * @response: must point to an empty #GstRTSPMessage * @code: an optional code result * * 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_rtsp_client_sink_setup_auth and retry * the request. * * Returns: #GST_RTSP_OK if the processing was successful. */ static GstRTSPResult gst_rtsp_client_sink_send (GstRTSPClientSink * sink, GstRTSPConnInfo * conninfo, 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_rtsp_client_sink_try_send (sink, conninfo, request, 1, response, &int_code)) < 0) goto error; switch (int_code) { case GST_RTSP_STS_UNAUTHORIZED: if (gst_rtsp_client_sink_setup_auth (sink, 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 (sink, "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 (sink, RESOURCE, NOT_FOUND, (NULL), ("%s", response->type_data.response.reason)); break; case GST_RTSP_STS_UNAUTHORIZED: GST_ELEMENT_ERROR (sink, RESOURCE, NOT_AUTHORIZED, (NULL), ("%s", response->type_data.response.reason)); break; case GST_RTSP_STS_MOVED_PERMANENTLY: case GST_RTSP_STS_MOVE_TEMPORARILY: { gchar *new_location; GstRTSPLowerTrans transports; GST_DEBUG_OBJECT (sink, "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 (sink, "redirection to %s", new_location); /* save current transports */ if (sink->conninfo.url) transports = sink->conninfo.url->transports; else transports = GST_RTSP_LOWER_TRANS_UNKNOWN; gst_rtsp_client_sink_uri_set_uri (GST_URI_HANDLER (sink), new_location, NULL); /* set old transports */ if (sink->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN) sink->conninfo.url->transports = transports; sink->need_redirect = TRUE; sink->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 (sink, "got NOT IMPLEMENTED, disable method %s", gst_rtsp_method_as_text (method)); sink->methods &= ~method; res = GST_RTSP_OK; break; default: GST_ELEMENT_ERROR (sink, 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; } } /* 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_rtsp_client_sink_parse_methods (GstRTSPClientSink * sink, GstRTSPMessage * response) { GstRTSPHeaderField field; gchar *respoptions; gint indx = 0; /* reset supported methods */ sink->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; sink->methods |= gst_rtsp_options_from_text (respoptions); indx++; } if (sink->methods == 0) { /* neither Allow nor Public are required, assume the server supports * at least SETUP. */ GST_DEBUG_OBJECT (sink, "could not get OPTIONS"); sink->methods = GST_RTSP_SETUP; } /* Even if the server replied, and didn't say it supports * RECORD|ANNOUNCE, try anyway by assuming it does */ sink->methods |= GST_RTSP_ANNOUNCE | GST_RTSP_RECORD; if (!(sink->methods & GST_RTSP_SETUP)) goto no_setup; return TRUE; /* ERRORS */ no_setup: { GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ, (NULL), ("Server does not support SETUP.")); return FALSE; } } static GstRTSPResult gst_rtsp_client_sink_connect_to_server (GstRTSPClientSink * sink, gboolean async) { GstRTSPResult res; GstRTSPMessage request = { 0 }; GstRTSPMessage response = { 0 }; GSocket *conn_socket; GSocketAddress *sa; GInetAddress *ia; sink->need_redirect = FALSE; /* can't continue without a valid url */ if (G_UNLIKELY (sink->conninfo.url == NULL)) { res = GST_RTSP_EINVAL; goto no_url; } sink->tried_url_auth = FALSE; if ((res = gst_rtsp_conninfo_connect (sink, &sink->conninfo, async)) < 0) goto connect_failed; conn_socket = gst_rtsp_connection_get_read_socket (sink->conninfo.connection); sa = g_socket_get_remote_address (conn_socket, NULL); ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa)); g_free (sink->server_ip); sink->server_ip = g_inet_address_to_string (ia); g_object_unref (sa); /* create OPTIONS */ GST_DEBUG_OBJECT (sink, "create options..."); res = gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_OPTIONS, sink->conninfo.url_str); if (res < 0) goto create_request_failed; /* send OPTIONS */ GST_DEBUG_OBJECT (sink, "send options..."); if (async) GST_ELEMENT_PROGRESS (sink, CONTINUE, "open", ("Retrieving server options")); if ((res = gst_rtsp_client_sink_send (sink, &sink->conninfo, &request, &response, NULL)) < 0) goto send_error; /* parse OPTIONS */ if (!gst_rtsp_client_sink_parse_methods (sink, &response)) goto methods_error; /* FIXME: Do we need to handle REDIRECT responses for OPTIONS? */ /* clean up any messages */ gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); return res; /* ERRORS */ no_url: { GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, (NULL), ("No valid RTSP URL was provided")); goto cleanup_error; } connect_failed: { gchar *str = gst_rtsp_strresult (res); if (res != GST_RTSP_EINTR) { GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ_WRITE, (NULL), ("Failed to connect. (%s)", str)); } else { GST_WARNING_OBJECT (sink, "connect interrupted"); } g_free (str); goto cleanup_error; } create_request_failed: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (sink, 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 */ res = GST_RTSP_ERROR; goto cleanup_error; } cleanup_error: { if (sink->conninfo.connection) { GST_DEBUG_OBJECT (sink, "free connection"); gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE); } gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); return res; } } static GstRTSPResult gst_rtsp_client_sink_open (GstRTSPClientSink * sink, gboolean async) { GstRTSPResult ret; sink->methods = GST_RTSP_SETUP | GST_RTSP_RECORD | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN; g_mutex_lock (&sink->open_conn_lock); sink->open_conn_start = TRUE; g_cond_broadcast (&sink->open_conn_cond); GST_DEBUG_OBJECT (sink, "connection to server started"); g_mutex_unlock (&sink->open_conn_lock); if ((ret = gst_rtsp_client_sink_connect_to_server (sink, async)) < 0) goto open_failed; if (async) gst_rtsp_client_sink_loop_end_cmd (sink, CMD_OPEN, ret); return ret; /* ERRORS */ open_failed: { GST_WARNING_OBJECT (sink, "Failed to connect to server"); sink->open_error = TRUE; if (async) gst_rtsp_client_sink_loop_end_cmd (sink, CMD_OPEN, ret); return ret; } } static GstRTSPResult gst_rtsp_client_sink_close (GstRTSPClientSink * sink, gboolean async, gboolean only_close) { GstRTSPMessage request = { 0 }; GstRTSPMessage response = { 0 }; GstRTSPResult res = GST_RTSP_OK; GList *walk; const gchar *control; GST_DEBUG_OBJECT (sink, "TEARDOWN..."); gst_rtsp_client_sink_set_state (sink, GST_STATE_NULL); if (sink->state < GST_RTSP_STATE_READY) { GST_DEBUG_OBJECT (sink, "not ready, doing cleanup"); goto close; } if (only_close) goto close; /* construct a control url */ control = get_aggregate_control (sink); if (!(sink->methods & (GST_RTSP_RECORD | GST_RTSP_TEARDOWN))) goto not_supported; /* stop streaming */ for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data; if (context->stream_transport) { gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE); gst_object_unref (context->stream_transport); context->stream_transport = NULL; } if (context->joined) { gst_rtsp_stream_leave_bin (context->stream, GST_BIN (sink->internal_bin), sink->rtpbin); context->joined = FALSE; } } for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data; const gchar *setup_url; GstRTSPConnInfo *info; GST_DEBUG_OBJECT (sink, "Looking at stream %p for teardown", context->stream); /* try aggregate control first but do non-aggregate control otherwise */ if (control) setup_url = control; else if ((setup_url = context->conninfo.location) == NULL) { GST_DEBUG_OBJECT (sink, "Skipping TEARDOWN stream %p - no setup URL", context->stream); continue; } if (sink->conninfo.connection) { info = &sink->conninfo; } else if (context->conninfo.connection) { info = &context->conninfo; } else { continue; } if (!info->connected) goto next; /* do TEARDOWN */ GST_DEBUG_OBJECT (sink, "Sending teardown for stream %p at URL %s", context->stream, setup_url); res = gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_TEARDOWN, setup_url); if (res < 0) goto create_request_failed; if (async) GST_ELEMENT_PROGRESS (sink, CONTINUE, "close", ("Closing stream")); if ((res = gst_rtsp_client_sink_send (sink, info, &request, &response, NULL)) < 0) goto send_error; /* FIXME, parse result? */ gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); next: /* early exit when we did aggregate control */ if (control) break; } close: /* close connections */ GST_DEBUG_OBJECT (sink, "closing connection..."); gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE); for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data; gst_rtsp_conninfo_close (sink, &stream->conninfo, TRUE); } /* cleanup */ gst_rtsp_client_sink_cleanup (sink); sink->state = GST_RTSP_STATE_INVALID; if (async) gst_rtsp_client_sink_loop_end_cmd (sink, CMD_CLOSE, res); return res; /* ERRORS */ create_request_failed: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL), ("Could not create request. (%s)", str)); g_free (str); goto close; } send_error: { gchar *str = gst_rtsp_strresult (res); gst_rtsp_message_unset (&request); if (res != GST_RTSP_EINTR) { GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Could not send message. (%s)", str)); } else { GST_WARNING_OBJECT (sink, "TEARDOWN interrupted"); } g_free (str); goto close; } not_supported: { GST_DEBUG_OBJECT (sink, "TEARDOWN and PLAY not supported, can't do TEARDOWN"); goto close; } } static gboolean gst_rtsp_client_sink_configure_manager (GstRTSPClientSink * sink) { GstElement *rtpbin; GstStateChangeReturn ret; rtpbin = sink->rtpbin; if (rtpbin == NULL) { GObjectClass *klass; rtpbin = gst_element_factory_make ("rtpbin", NULL); if (rtpbin == NULL) goto no_rtpbin; gst_bin_add (GST_BIN_CAST (sink->internal_bin), rtpbin); sink->rtpbin = rtpbin; /* Any more settings we should configure on rtpbin here? */ g_object_set (sink->rtpbin, "latency", sink->latency, NULL); klass = G_OBJECT_GET_CLASS (G_OBJECT (rtpbin)); if (g_object_class_find_property (klass, "ntp-time-source")) { g_object_set (sink->rtpbin, "ntp-time-source", sink->ntp_time_source, NULL); } if (sink->sdes && g_object_class_find_property (klass, "sdes")) { g_object_set (sink->rtpbin, "sdes", sink->sdes, NULL); } g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_NEW_MANAGER], 0, sink->rtpbin); } ret = gst_element_set_state (rtpbin, GST_STATE_PAUSED); if (ret == GST_STATE_CHANGE_FAILURE) goto start_manager_failure; return TRUE; no_rtpbin: { GST_WARNING ("no rtpbin element"); g_warning ("failed to create element 'rtpbin', check your installation"); return FALSE; } start_manager_failure: { GST_DEBUG_OBJECT (sink, "could not start session manager"); gst_bin_remove (GST_BIN_CAST (sink->internal_bin), rtpbin); return FALSE; } } static GstElement * request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPClientSink * sink) { GstRTSPStream *stream = NULL; GstElement *ret = NULL; GList *walk; GST_RTSP_STATE_LOCK (sink); for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data; if (sessid == gst_rtsp_stream_get_index (context->stream)) { stream = context->stream; break; } } if (stream != NULL) { GST_DEBUG_OBJECT (sink, "Creating aux sender for stream %u", sessid); ret = gst_rtsp_stream_request_aux_sender (stream, sessid); } GST_RTSP_STATE_UNLOCK (sink); return ret; } static GstElement * request_fec_encoder (GstElement * rtpbin, guint sessid, GstRTSPClientSink * sink) { GstRTSPStream *stream = NULL; GstElement *ret = NULL; GList *walk; GST_RTSP_STATE_LOCK (sink); for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data; if (sessid == gst_rtsp_stream_get_index (context->stream)) { stream = context->stream; break; } } if (stream != NULL) { ret = gst_rtsp_stream_request_ulpfec_encoder (stream, sessid); } GST_RTSP_STATE_UNLOCK (sink); return ret; } static gboolean gst_rtsp_client_sink_is_stopping (GstRTSPClientSink * sink) { gboolean is_stopping; GST_OBJECT_LOCK (sink); is_stopping = sink->task == NULL; GST_OBJECT_UNLOCK (sink); return is_stopping; } static gboolean gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink) { GstRTSPStreamContext *context; GList *walk; const gchar *base; gchar *stream_path; GstUri *base_uri, *uri; GST_DEBUG_OBJECT (sink, "Collecting stream information"); if (!gst_rtsp_client_sink_configure_manager (sink)) return FALSE; base = get_aggregate_control (sink); base_uri = gst_uri_from_string (base); if (!base_uri) { GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, (NULL), ("Could not parse uri %s", base)); return FALSE; } g_mutex_lock (&sink->preroll_lock); while (sink->contexts == NULL && !sink->conninfo.flushing) { g_cond_wait (&sink->preroll_cond, &sink->preroll_lock); } g_mutex_unlock (&sink->preroll_lock); /* FIXME: Need different locking - need to protect against pad releases * and potential state changes ruining things here */ for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstPad *srcpad; context = (GstRTSPStreamContext *) walk->data; if (context->stream) continue; g_mutex_lock (&sink->preroll_lock); while (!context->prerolled && !sink->conninfo.flushing && !gst_rtsp_client_sink_is_stopping (sink)) { GST_DEBUG_OBJECT (sink, "Waiting for caps on stream %d", context->index); g_cond_wait (&sink->preroll_cond, &sink->preroll_lock); } if (sink->conninfo.flushing) { g_mutex_unlock (&sink->preroll_lock); break; } g_mutex_unlock (&sink->preroll_lock); if (context->payloader == NULL) continue; srcpad = gst_element_get_static_pad (context->payloader, "src"); GST_DEBUG_OBJECT (sink, "Creating stream object for stream %d", context->index); context->stream = gst_rtsp_client_sink_create_stream (sink, context, context->payloader, srcpad); /* append stream index to uri path */ g_free (context->conninfo.location); stream_path = g_strdup_printf ("stream=%d", context->index); uri = gst_uri_copy (base_uri); gst_uri_append_path (uri, stream_path); context->conninfo.location = gst_uri_to_string (uri); gst_uri_unref (uri); g_free (stream_path); if (sink->rtx_time > 0) { /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */ g_signal_connect (sink->rtpbin, "request-aux-sender", (GCallback) request_aux_sender, sink); } g_signal_connect (sink->rtpbin, "request-fec-encoder", (GCallback) request_fec_encoder, sink); if (!gst_rtsp_stream_join_bin (context->stream, GST_BIN (sink->internal_bin), sink->rtpbin, GST_STATE_PAUSED)) { goto join_bin_failed; } context->joined = TRUE; /* Block the stream, as it does not have any transport parts yet */ gst_rtsp_stream_set_blocked (context->stream, TRUE); /* Let the stream object receive data */ gst_pad_remove_probe (srcpad, context->payloader_block_id); gst_object_unref (srcpad); } /* Now wait for the preroll of the rtp bin */ g_mutex_lock (&sink->preroll_lock); while (!sink->prerolled && sink->conninfo.connection && !sink->conninfo.flushing) { GST_LOG_OBJECT (sink, "Waiting for preroll before continuing"); g_cond_wait (&sink->preroll_cond, &sink->preroll_lock); } GST_LOG_OBJECT (sink, "Marking streams as collected"); sink->streams_collected = TRUE; g_mutex_unlock (&sink->preroll_lock); gst_uri_unref (base_uri); return TRUE; join_bin_failed: gst_uri_unref (base_uri); GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL), ("Could not start stream %d", context->index)); return FALSE; } static GstRTSPResult gst_rtsp_client_sink_create_transports_string (GstRTSPClientSink * sink, GstRTSPStreamContext * context, GSocketFamily family, GstRTSPLowerTrans protocols, GstRTSPProfile profiles, gchar ** transports) { GString *result; GstRTSPStream *stream = context->stream; gboolean first = TRUE; /* the default RTSP transports */ result = g_string_new ("RTP"); while (profiles != 0) { if (!first) g_string_append (result, ",RTP"); if (profiles & GST_RTSP_PROFILE_SAVPF) { g_string_append (result, "/SAVPF"); profiles &= ~GST_RTSP_PROFILE_SAVPF; } else if (profiles & GST_RTSP_PROFILE_SAVP) { g_string_append (result, "/SAVP"); profiles &= ~GST_RTSP_PROFILE_SAVP; } else if (profiles & GST_RTSP_PROFILE_AVPF) { g_string_append (result, "/AVPF"); profiles &= ~GST_RTSP_PROFILE_AVPF; } else if (profiles & GST_RTSP_PROFILE_AVP) { g_string_append (result, "/AVP"); profiles &= ~GST_RTSP_PROFILE_AVP; } else { GST_WARNING_OBJECT (sink, "Unimplemented profile(s) 0x%x", profiles); break; } if (protocols & GST_RTSP_LOWER_TRANS_UDP) { GstRTSPRange ports; GST_DEBUG_OBJECT (sink, "adding UDP unicast"); gst_rtsp_stream_get_server_port (stream, &ports, family); g_string_append_printf (result, "/UDP;unicast;client_port=%d-%d", ports.min, ports.max); } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) { GstRTSPAddress *addr = gst_rtsp_stream_get_multicast_address (stream, family); if (addr) { GST_DEBUG_OBJECT (sink, "adding UDP multicast"); g_string_append_printf (result, "/UDP;multicast;client_port=%d-%d", addr->port, addr->port + addr->n_ports - 1); gst_rtsp_address_free (addr); } } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) { GST_DEBUG_OBJECT (sink, "adding TCP"); g_string_append_printf (result, "/TCP;unicast;interleaved=%d-%d", sink->free_channel, sink->free_channel + 1); } g_string_append (result, ";mode=RECORD"); /* FIXME: Support appending too: if (sink->append) g_string_append (result, ";append"); */ first = FALSE; } if (first) { /* No valid transport could be constructed */ GST_ERROR_OBJECT (sink, "No supported profiles configured"); goto fail; } *transports = g_string_free (result, FALSE); GST_DEBUG_OBJECT (sink, "prepared transports %s", GST_STR_NULL (*transports)); return GST_RTSP_OK; fail: g_string_free (result, TRUE); return GST_RTSP_ERROR; } static GstCaps * signal_get_srtcp_params (GstRTSPClientSink * sink, GstRTSPStreamContext * context) { GstCaps *caps = NULL; g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_REQUEST_RTCP_KEY], 0, context->index, &caps); if (caps != NULL) GST_DEBUG_OBJECT (sink, "SRTP parameters received"); return caps; } static gchar * gst_rtsp_client_sink_stream_make_keymgmt (GstRTSPClientSink * sink, GstRTSPStreamContext * context) { gchar *base64, *result = NULL; GstMIKEYMessage *mikey_msg; context->srtcpparams = signal_get_srtcp_params (sink, context); if (context->srtcpparams == NULL) context->srtcpparams = gst_rtsp_stream_get_caps (context->stream); mikey_msg = gst_mikey_message_new_from_caps (context->srtcpparams); if (mikey_msg) { guint send_ssrc, send_rtx_ssrc; const GstStructure *s = gst_caps_get_structure (context->srtcpparams, 0); /* add policy '0' for our SSRC */ gst_rtsp_stream_get_ssrc (context->stream, &send_ssrc); GST_LOG_OBJECT (sink, "Stream %p ssrc %x", context->stream, send_ssrc); gst_mikey_message_add_cs_srtp (mikey_msg, 0, send_ssrc, 0); if (gst_structure_get_uint (s, "rtx-ssrc", &send_rtx_ssrc)) gst_mikey_message_add_cs_srtp (mikey_msg, 0, send_rtx_ssrc, 0); base64 = gst_mikey_message_base64_encode (mikey_msg); gst_mikey_message_unref (mikey_msg); if (base64) { result = gst_sdp_make_keymgmt (context->conninfo.location, base64); g_free (base64); } } return result; } /* masks to be kept in sync with the hardcoded protocol order of preference * in code below */ static const guint protocol_masks[] = { GST_RTSP_LOWER_TRANS_UDP, GST_RTSP_LOWER_TRANS_UDP_MCAST, GST_RTSP_LOWER_TRANS_TCP, 0 }; /* Same for profile_masks */ static const guint profile_masks[] = { GST_RTSP_PROFILE_SAVPF, GST_RTSP_PROFILE_SAVP, GST_RTSP_PROFILE_AVPF, GST_RTSP_PROFILE_AVP, 0 }; static gboolean do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPStreamContext * context) { GstRTSPClientSink *sink = context->parent; GstRTSPMessage message = { 0 }; GstRTSPResult res = GST_RTSP_OK; gst_rtsp_message_init_data (&message, channel); gst_rtsp_message_set_body_buffer (&message, buffer); res = gst_rtsp_client_sink_try_send (sink, &sink->conninfo, &message, 1, NULL, NULL); gst_rtsp_message_unset (&message); gst_rtsp_stream_transport_message_sent (context->stream_transport); return res == GST_RTSP_OK; } static gboolean do_send_data_list (GstBufferList * buffer_list, guint8 channel, GstRTSPStreamContext * context) { GstRTSPClientSink *sink = context->parent; GstRTSPResult res = GST_RTSP_OK; guint i, n = gst_buffer_list_length (buffer_list); GstRTSPMessage *messages = g_newa (GstRTSPMessage, n); memset (messages, 0, n * sizeof (GstRTSPMessage)); for (i = 0; i < n; i++) { GstBuffer *buffer = gst_buffer_list_get (buffer_list, i); gst_rtsp_message_init_data (&messages[i], channel); gst_rtsp_message_set_body_buffer (&messages[i], buffer); } res = gst_rtsp_client_sink_try_send (sink, &sink->conninfo, messages, n, NULL, NULL); for (i = 0; i < n; i++) { gst_rtsp_message_unset (&messages[i]); gst_rtsp_stream_transport_message_sent (context->stream_transport); } return res == GST_RTSP_OK; } static GstRTSPResult gst_rtsp_client_sink_setup_streams (GstRTSPClientSink * sink, gboolean async) { GstRTSPResult res = GST_RTSP_ERROR; GstRTSPMessage request = { 0 }; GstRTSPMessage response = { 0 }; GstRTSPLowerTrans protocols; GstRTSPStatusCode code; GSocketFamily family; GSocketAddress *sa; GSocket *conn_socket; GstRTSPUrl *url; GList *walk; gchar *hval; if (sink->conninfo.connection) { url = gst_rtsp_connection_get_url (sink->conninfo.connection); /* we initially allow all configured lower transports. based on the URL * transports and the replies from the server we narrow them down. */ protocols = url->transports & sink->cur_protocols; } else { url = NULL; protocols = sink->cur_protocols; } if (protocols == 0) goto no_protocols; GST_RTSP_STATE_LOCK (sink); if (G_UNLIKELY (sink->contexts == NULL)) goto no_streams; for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data; GstRTSPStream *stream; GstRTSPConnInfo *info; GstRTSPProfile profiles; GstRTSPProfile cur_profile; gchar *transports; gint retry = 0; guint profile_mask = 0; guint mask = 0; GstCaps *caps; const GstSDPMedia *media; stream = context->stream; profiles = gst_rtsp_stream_get_profiles (stream); caps = gst_rtsp_stream_get_caps (stream); if (caps == NULL) { GST_DEBUG_OBJECT (sink, "skipping stream %p, no caps", stream); continue; } gst_caps_unref (caps); media = gst_sdp_message_get_media (&sink->cursdp, context->sdp_index); if (media == NULL) { GST_DEBUG_OBJECT (sink, "skipping stream %p, no SDP info", stream); continue; } /* skip setup if we have no URL for it */ if (context->conninfo.location == NULL) { GST_DEBUG_OBJECT (sink, "skipping stream %p, no setup", stream); continue; } if (sink->conninfo.connection == NULL) { if (!gst_rtsp_conninfo_connect (sink, &context->conninfo, async)) { GST_DEBUG_OBJECT (sink, "skipping stream %p, failed to connect", stream); continue; } info = &context->conninfo; } else { info = &sink->conninfo; } GST_DEBUG_OBJECT (sink, "doing setup of stream %p with %s", stream, context->conninfo.location); conn_socket = gst_rtsp_connection_get_read_socket (info->connection); sa = g_socket_get_local_address (conn_socket, NULL); family = g_socket_address_get_family (sa); g_object_unref (sa); next_protocol: /* first selectable profile */ while (profile_masks[profile_mask] && !(profiles & profile_masks[profile_mask])) profile_mask++; if (!profile_masks[profile_mask]) goto no_profiles; /* first selectable protocol */ while (protocol_masks[mask] && !(protocols & protocol_masks[mask])) mask++; if (!protocol_masks[mask]) goto no_protocols; retry: GST_DEBUG_OBJECT (sink, "protocols = 0x%x, protocol mask = 0x%x", protocols, protocol_masks[mask]); /* create a string with first transport in line */ transports = NULL; cur_profile = profiles & profile_masks[profile_mask]; res = gst_rtsp_client_sink_create_transports_string (sink, context, family, protocols & protocol_masks[mask], cur_profile, &transports); if (res < 0 || transports == NULL) goto setup_transport_failed; if (strlen (transports) == 0) { g_free (transports); GST_DEBUG_OBJECT (sink, "no transports found"); mask++; profile_mask = 0; goto next_protocol; } GST_DEBUG_OBJECT (sink, "transport is %s", GST_STR_NULL (transports)); /* create SETUP request */ res = gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_SETUP, context->conninfo.location); if (res < 0) { g_free (transports); goto create_request_failed; } /* set up keys */ if (cur_profile == GST_RTSP_PROFILE_SAVP || cur_profile == GST_RTSP_PROFILE_SAVPF) { hval = gst_rtsp_client_sink_stream_make_keymgmt (sink, context); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_KEYMGMT, hval); } /* if the user wants a non default RTP packet size we add the blocksize * parameter */ if (sink->rtp_blocksize > 0) { hval = g_strdup_printf ("%d", sink->rtp_blocksize); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval); } if (async) GST_ELEMENT_PROGRESS (sink, CONTINUE, "request", ("SETUP stream %d", context->index)); { GstRTSPTransport *transport; gst_rtsp_transport_new (&transport); if (gst_rtsp_transport_parse (transports, transport) != GST_RTSP_OK) goto parse_transport_failed; if (transport->lower_transport != GST_RTSP_LOWER_TRANS_TCP) { if (!gst_rtsp_stream_allocate_udp_sockets (stream, family, transport, FALSE)) { gst_rtsp_transport_free (transport); goto allocate_udp_ports_failed; } } if (!gst_rtsp_stream_complete_stream (stream, transport)) { gst_rtsp_transport_free (transport); goto complete_stream_failed; } gst_rtsp_transport_free (transport); gst_rtsp_stream_set_blocked (stream, FALSE); } /* FIXME: * the creation of the transports string depends on * calling stream_get_server_port, which only starts returning * something meaningful after a call to stream_allocate_udp_sockets * has been made, this function expects a transport that we parse * from the transport string ... * * Significant refactoring is in order, but does not look entirely * trivial, for now we put a band aid on and create a second transport * string after the stream has been completed, to pass it in * the request headers instead of the previous, incomplete one. */ g_free (transports); transports = NULL; res = gst_rtsp_client_sink_create_transports_string (sink, context, family, protocols & protocol_masks[mask], cur_profile, &transports); if (res < 0 || transports == NULL) goto setup_transport_failed; /* select transport */ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports); /* handle the code ourselves */ res = gst_rtsp_client_sink_send (sink, info, &request, &response, &code); if (res < 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); /* Try another profile. If no more, move to the next protocol */ profile_mask++; while (profile_masks[profile_mask] && !(profiles & profile_masks[profile_mask])) profile_mask++; if (profile_masks[profile_mask]) goto retry; /* select next available protocol, give up on this stream if none */ /* Reset profiles to try: */ profile_mask = 0; mask++; while (protocol_masks[mask] && !(protocols & protocol_masks[mask])) mask++; if (!protocol_masks[mask]) continue; else goto retry; default: goto response_error; } /* parse response transport */ { gchar *resptrans = NULL; GstRTSPTransport *transport; gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT, &resptrans, 0); if (!resptrans) { goto no_transport; } gst_rtsp_transport_new (&transport); /* parse transport, go to next stream on parse error */ if (gst_rtsp_transport_parse (resptrans, transport) != GST_RTSP_OK) { GST_WARNING_OBJECT (sink, "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 (sink, "stream %p as TCP interleaved", stream); protocols = GST_RTSP_LOWER_TRANS_TCP; sink->interleaved = TRUE; /* update free channels */ sink->free_channel = MAX (transport->interleaved.min, sink->free_channel); sink->free_channel = MAX (transport->interleaved.max, sink->free_channel); sink->free_channel++; break; case GST_RTSP_LOWER_TRANS_UDP_MCAST: /* only allow multicast for other streams */ GST_DEBUG_OBJECT (sink, "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 (sink, "stream %p as UDP unicast", stream); protocols = GST_RTSP_LOWER_TRANS_UDP; /* Update transport with server destination if not provided by the server */ if (transport->destination == NULL) { transport->destination = g_strdup (sink->server_ip); } break; default: GST_DEBUG_OBJECT (sink, "stream %p unknown transport %d", stream, transport->lower_transport); break; } if (!retry) { GST_DEBUG ("Configuring the stream transport for stream %d", context->index); if (context->stream_transport == NULL) context->stream_transport = gst_rtsp_stream_transport_new (stream, transport); else gst_rtsp_stream_transport_set_transport (context->stream_transport, transport); if (transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP) { /* our callbacks to send data on this TCP connection */ gst_rtsp_stream_transport_set_callbacks (context->stream_transport, (GstRTSPSendFunc) do_send_data, (GstRTSPSendFunc) do_send_data, context, NULL); gst_rtsp_stream_transport_set_list_callbacks (context->stream_transport, (GstRTSPSendListFunc) do_send_data_list, (GstRTSPSendListFunc) do_send_data_list, context, NULL); } /* The stream_transport now owns the transport */ transport = NULL; gst_rtsp_stream_transport_set_active (context->stream_transport, TRUE); } next: if (transport) gst_rtsp_transport_free (transport); /* clean up used RTSP messages */ gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); } } GST_RTSP_STATE_UNLOCK (sink); /* store the transport protocol that was configured */ sink->cur_protocols = protocols; return res; no_streams: { GST_RTSP_STATE_UNLOCK (sink); GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), ("SDP contains no streams")); return GST_RTSP_ERROR; } setup_transport_failed: { GST_RTSP_STATE_UNLOCK (sink); GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), ("Could not setup transport.")); res = GST_RTSP_ERROR; goto cleanup_error; } no_profiles: { GST_RTSP_STATE_UNLOCK (sink); /* no transport possible, post an error and stop */ GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL), ("Could not connect to server, no profiles left")); return GST_RTSP_ERROR; } no_protocols: { GST_RTSP_STATE_UNLOCK (sink); /* no transport possible, post an error and stop */ GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL), ("Could not connect to server, no protocols left")); return GST_RTSP_ERROR; } no_transport: { GST_RTSP_STATE_UNLOCK (sink); GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), ("Server did not select transport.")); res = GST_RTSP_ERROR; goto cleanup_error; } create_request_failed: { gchar *str = gst_rtsp_strresult (res); GST_RTSP_STATE_UNLOCK (sink); GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL), ("Could not create request. (%s)", str)); g_free (str); goto cleanup_error; } parse_transport_failed: { GST_RTSP_STATE_UNLOCK (sink); GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), ("Could not parse transport.")); res = GST_RTSP_ERROR; goto cleanup_error; } allocate_udp_ports_failed: { GST_RTSP_STATE_UNLOCK (sink); GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), ("Could not parse transport.")); res = GST_RTSP_ERROR; goto cleanup_error; } complete_stream_failed: { GST_RTSP_STATE_UNLOCK (sink); GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), ("Could not parse transport.")); res = GST_RTSP_ERROR; goto cleanup_error; } send_error: { gchar *str = gst_rtsp_strresult (res); GST_RTSP_STATE_UNLOCK (sink); if (res != GST_RTSP_EINTR) { GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Could not send message. (%s)", str)); } else { GST_WARNING_OBJECT (sink, "send interrupted"); } g_free (str); goto cleanup_error; } response_error: { const gchar *str = gst_rtsp_status_as_text (code); GST_RTSP_STATE_UNLOCK (sink); GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Error (%d): %s", code, GST_STR_NULL (str))); res = GST_RTSP_ERROR; goto cleanup_error; } cleanup_error: { gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); return res; } } static GstRTSPResult gst_rtsp_client_sink_ensure_open (GstRTSPClientSink * sink, gboolean async) { GstRTSPResult res = GST_RTSP_OK; if (sink->state < GST_RTSP_STATE_READY) { res = GST_RTSP_ERROR; if (sink->open_error) { GST_DEBUG_OBJECT (sink, "the stream was in error"); goto done; } if (async) gst_rtsp_client_sink_loop_start_cmd (sink, CMD_OPEN); if ((res = gst_rtsp_client_sink_open (sink, async)) < 0) { GST_DEBUG_OBJECT (sink, "failed to open stream"); goto done; } } done: return res; } static GstRTSPResult gst_rtsp_client_sink_record (GstRTSPClientSink * sink, gboolean async) { GstRTSPMessage request = { 0 }; GstRTSPMessage response = { 0 }; GstRTSPResult res = GST_RTSP_OK; GstSDPMessage *sdp; guint sdp_index = 0; GstSDPInfo info = { 0, }; gchar *keymgmt; guint i; const gchar *proto; gchar *sess_id, *client_ip, *str; GSocketAddress *sa; GInetAddress *ia; GSocket *conn_socket; GList *walk; g_mutex_lock (&sink->preroll_lock); if (sink->state == GST_RTSP_STATE_PLAYING) { /* Already recording, don't send another request */ GST_LOG_OBJECT (sink, "Already in RECORD. Skipping duplicate request."); g_mutex_unlock (&sink->preroll_lock); goto done; } g_mutex_unlock (&sink->preroll_lock); /* Collect all our input streams and create * stream objects before actually returning. * The streams are blocked at this point as we do not have any transport * parts yet. */ gst_rtsp_client_sink_collect_streams (sink); if (gst_rtsp_client_sink_is_stopping (sink)) { GST_INFO_OBJECT (sink, "task stopped, shutting down"); return GST_RTSP_EINTR; } g_mutex_lock (&sink->block_streams_lock); /* Wait for streams to be blocked */ while (sink->n_streams_blocked < g_list_length (sink->contexts) && !gst_rtsp_client_sink_is_stopping (sink)) { GST_DEBUG_OBJECT (sink, "waiting for streams to be blocked"); g_cond_wait (&sink->block_streams_cond, &sink->block_streams_lock); } g_mutex_unlock (&sink->block_streams_lock); if (gst_rtsp_client_sink_is_stopping (sink)) { GST_INFO_OBJECT (sink, "task stopped, shutting down"); return GST_RTSP_EINTR; } /* Send announce, then setup for all streams */ gst_sdp_message_init (&sink->cursdp); sdp = &sink->cursdp; /* some standard things first */ gst_sdp_message_set_version (sdp, "0"); /* session ID doesn't have to be super-unique in this case */ sess_id = g_strdup_printf ("%u", g_random_int ()); if (sink->conninfo.connection == NULL) return GST_RTSP_ERROR; conn_socket = gst_rtsp_connection_get_read_socket (sink->conninfo.connection); sa = g_socket_get_local_address (conn_socket, NULL); ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa)); client_ip = g_inet_address_to_string (ia); if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV6) { info.is_ipv6 = TRUE; proto = "IP6"; } else if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV4) proto = "IP4"; else g_assert_not_reached (); g_object_unref (sa); /* FIXME: Should this actually be the server's IP or ours? */ info.server_ip = sink->server_ip; gst_sdp_message_set_origin (sdp, "-", sess_id, "1", "IN", proto, client_ip); gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer"); gst_sdp_message_set_information (sdp, "rtspclientsink"); gst_sdp_message_add_time (sdp, "0", "0", NULL); gst_sdp_message_add_attribute (sdp, "tool", "GStreamer"); /* add stream */ for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data; gst_rtsp_sdp_from_stream (sdp, &info, context->stream); context->sdp_index = sdp_index++; } g_free (sess_id); g_free (client_ip); /* send ANNOUNCE request */ GST_DEBUG_OBJECT (sink, "create ANNOUNCE request..."); res = gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_ANNOUNCE, sink->conninfo.url_str); if (res < 0) goto create_request_failed; g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_UPDATE_SDP], 0, sdp); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CONTENT_TYPE, "application/sdp"); /* add SDP to the request body */ str = gst_sdp_message_as_text (sdp); gst_rtsp_message_take_body (&request, (guint8 *) str, strlen (str)); /* send ANNOUNCE */ GST_DEBUG_OBJECT (sink, "sending announce..."); if (async) GST_ELEMENT_PROGRESS (sink, CONTINUE, "record", ("Sending server stream info")); if ((res = gst_rtsp_client_sink_send (sink, &sink->conninfo, &request, &response, NULL)) < 0) goto send_error; /* parse the keymgmt */ i = 0; walk = sink->contexts; while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_KEYMGMT, &keymgmt, i++) == GST_RTSP_OK) { GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data; walk = g_list_next (walk); if (!gst_rtsp_stream_handle_keymgmt (context->stream, keymgmt)) goto keymgmt_error; } /* send setup for all streams */ if ((res = gst_rtsp_client_sink_setup_streams (sink, async)) < 0) goto setup_failed; res = gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_RECORD, sink->conninfo.url_str); if (res < 0) goto create_request_failed; #if 0 /* FIXME: Configure a range based on input segments? */ if (src->need_range) { hval = gen_range_header (src, segment); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval); } if (segment->rate != 1.0) { gchar hval[G_ASCII_DTOSTR_BUF_SIZE]; g_ascii_dtostr (hval, sizeof (hval), segment->rate); if (src->skip) gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval); else gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval); } #endif if (async) GST_ELEMENT_PROGRESS (sink, CONTINUE, "record", ("Starting recording")); if ((res = gst_rtsp_client_sink_send (sink, &sink->conninfo, &request, &response, NULL)) < 0) goto send_error; #if 0 /* FIXME: Check if servers return these for record: */ /* 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. */ hval_idx = 0; while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO, &hval, hval_idx++) == GST_RTSP_OK) gst_rtspsrc_parse_rtpinfo (src, hval); /* some servers indicate RTCP parameters in PLAY response, * rather than properly in SDP */ if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL, &hval, 0) == GST_RTSP_OK) gst_rtspsrc_handle_rtcp_interval (src, hval); #endif gst_rtsp_client_sink_set_state (sink, GST_STATE_PLAYING); sink->state = GST_RTSP_STATE_PLAYING; for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data; gst_rtsp_stream_unblock_rtcp (context->stream); } /* clean up any messages */ gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); done: return res; create_request_failed: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (sink, 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; } keymgmt_error: { GST_ELEMENT_ERROR (sink, STREAM, DECRYPT_NOKEY, (NULL), ("Could not handle KeyMgmt")); } setup_failed: { GST_ERROR_OBJECT (sink, "setup failed"); goto cleanup_error; } cleanup_error: { if (sink->conninfo.connection) { GST_DEBUG_OBJECT (sink, "free connection"); gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE); } gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); return res; } } static GstRTSPResult gst_rtsp_client_sink_pause (GstRTSPClientSink * sink, gboolean async) { GstRTSPResult res = GST_RTSP_OK; GstRTSPMessage request = { 0 }; GstRTSPMessage response = { 0 }; GList *walk; const gchar *control; GST_DEBUG_OBJECT (sink, "PAUSE..."); if ((res = gst_rtsp_client_sink_ensure_open (sink, async)) < 0) goto open_failed; if (!(sink->methods & GST_RTSP_PAUSE)) goto not_supported; if (sink->state == GST_RTSP_STATE_READY) goto was_paused; if (!sink->conninfo.connection || !sink->conninfo.connected) goto no_connection; /* construct a control url */ control = get_aggregate_control (sink); /* loop over the streams. We might exit the loop early when we could do an * aggregate control */ for (walk = sink->contexts; walk; walk = g_list_next (walk)) { GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data; GstRTSPConnInfo *info; const gchar *setup_url; /* try aggregate control first but do non-aggregate control otherwise */ if (control) setup_url = control; else if ((setup_url = stream->conninfo.location) == NULL) continue; if (sink->conninfo.connection) { info = &sink->conninfo; } else if (stream->conninfo.connection) { info = &stream->conninfo; } else { continue; } if (async) GST_ELEMENT_PROGRESS (sink, CONTINUE, "request", ("Sending PAUSE request")); if ((res = gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_PAUSE, setup_url)) < 0) goto create_request_failed; if ((res = gst_rtsp_client_sink_send (sink, info, &request, &response, NULL)) < 0) goto send_error; gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); /* exit early when we did agregate control */ if (control) break; } /* change element states now */ gst_rtsp_client_sink_set_state (sink, GST_STATE_PAUSED); no_connection: sink->state = GST_RTSP_STATE_READY; done: if (async) gst_rtsp_client_sink_loop_end_cmd (sink, CMD_PAUSE, res); return res; /* ERRORS */ open_failed: { GST_DEBUG_OBJECT (sink, "failed to open stream"); goto done; } not_supported: { GST_DEBUG_OBJECT (sink, "PAUSE is not supported"); goto done; } was_paused: { GST_DEBUG_OBJECT (sink, "we were already PAUSED"); goto done; } create_request_failed: { gchar *str = gst_rtsp_strresult (res); GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL), ("Could not create request. (%s)", str)); g_free (str); goto done; } send_error: { gchar *str = gst_rtsp_strresult (res); gst_rtsp_message_unset (&request); if (res != GST_RTSP_EINTR) { GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Could not send message. (%s)", str)); } else { GST_WARNING_OBJECT (sink, "PAUSE interrupted"); } g_free (str); goto done; } } static void gst_rtsp_client_sink_handle_message (GstBin * bin, GstMessage * message) { GstRTSPClientSink *rtsp_client_sink; rtsp_client_sink = GST_RTSP_CLIENT_SINK (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 (rtsp_client_sink); ignore_timeout = rtsp_client_sink->ignore_timeout; rtsp_client_sink->ignore_timeout = TRUE; GST_OBJECT_UNLOCK (rtsp_client_sink); /* we only act on the first udp timeout message, others are irrelevant * and can be ignored. */ if (!ignore_timeout) gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_RECONNECT, CMD_LOOP); /* eat and free */ gst_message_unref (message); return; } else if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) { /* An RTSPStream has prerolled */ GST_DEBUG_OBJECT (rtsp_client_sink, "received GstRTSPStreamBlocking"); g_mutex_lock (&rtsp_client_sink->block_streams_lock); rtsp_client_sink->n_streams_blocked++; g_cond_broadcast (&rtsp_client_sink->block_streams_cond); g_mutex_unlock (&rtsp_client_sink->block_streams_lock); } GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } case GST_MESSAGE_ASYNC_START:{ GstObject *sender; sender = GST_MESSAGE_SRC (message); GST_LOG_OBJECT (rtsp_client_sink, "Have async-start from %" GST_PTR_FORMAT, sender); if (sender == GST_OBJECT (rtsp_client_sink->internal_bin)) { GST_LOG_OBJECT (rtsp_client_sink, "child bin is now ASYNC"); } GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } case GST_MESSAGE_ASYNC_DONE: { GstObject *sender; gboolean need_async_done; sender = GST_MESSAGE_SRC (message); GST_LOG_OBJECT (rtsp_client_sink, "Have async-done from %" GST_PTR_FORMAT, sender); g_mutex_lock (&rtsp_client_sink->preroll_lock); if (sender == GST_OBJECT_CAST (rtsp_client_sink->internal_bin)) { GST_LOG_OBJECT (rtsp_client_sink, "child bin is no longer ASYNC"); } need_async_done = rtsp_client_sink->in_async; if (rtsp_client_sink->in_async) { rtsp_client_sink->in_async = FALSE; g_cond_broadcast (&rtsp_client_sink->preroll_cond); } g_mutex_unlock (&rtsp_client_sink->preroll_lock); GST_BIN_CLASS (parent_class)->handle_message (bin, message); if (need_async_done) { GST_DEBUG_OBJECT (rtsp_client_sink, "Posting ASYNC-DONE"); gst_element_post_message (GST_ELEMENT_CAST (rtsp_client_sink), gst_message_new_async_done (GST_OBJECT_CAST (rtsp_client_sink), GST_CLOCK_TIME_NONE)); } break; } case GST_MESSAGE_ERROR: { GstObject *sender; sender = GST_MESSAGE_SRC (message); GST_DEBUG_OBJECT (rtsp_client_sink, "got error from %s", GST_ELEMENT_NAME (sender)); /* FIXME: Ignore errors on RTCP? */ /* fatal but not our message, forward */ GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } case GST_MESSAGE_STATE_CHANGED: { if (GST_MESSAGE_SRC (message) == (GstObject *) rtsp_client_sink->internal_bin) { GstState newstate, pending; gst_message_parse_state_changed (message, NULL, &newstate, &pending); g_mutex_lock (&rtsp_client_sink->preroll_lock); rtsp_client_sink->prerolled = (newstate >= GST_STATE_PAUSED) && pending == GST_STATE_VOID_PENDING; g_cond_broadcast (&rtsp_client_sink->preroll_cond); g_mutex_unlock (&rtsp_client_sink->preroll_lock); GST_DEBUG_OBJECT (bin, "Internal bin changed state to %s (pending %s). Prerolled now %d", gst_element_state_get_name (newstate), gst_element_state_get_name (pending), rtsp_client_sink->prerolled); } /* fallthrough */ } default: { GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } } } /* the thread where everything happens */ static void gst_rtsp_client_sink_thread (GstRTSPClientSink * sink) { gint cmd; GST_OBJECT_LOCK (sink); cmd = sink->pending_cmd; if (cmd == CMD_RECONNECT || cmd == CMD_RECORD || cmd == CMD_PAUSE || cmd == CMD_LOOP || cmd == CMD_OPEN) sink->pending_cmd = CMD_LOOP; else sink->pending_cmd = CMD_WAIT; GST_DEBUG_OBJECT (sink, "got command %s", cmd_to_string (cmd)); /* we got the message command, so ensure communication is possible again */ gst_rtsp_client_sink_connection_flush (sink, FALSE); sink->busy_cmd = cmd; GST_OBJECT_UNLOCK (sink); switch (cmd) { case CMD_OPEN: if (gst_rtsp_client_sink_open (sink, TRUE) == GST_RTSP_ERROR) gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT, CMD_ALL & ~CMD_CLOSE); break; case CMD_RECORD: gst_rtsp_client_sink_record (sink, TRUE); break; case CMD_PAUSE: gst_rtsp_client_sink_pause (sink, TRUE); break; case CMD_CLOSE: gst_rtsp_client_sink_close (sink, TRUE, FALSE); break; case CMD_LOOP: gst_rtsp_client_sink_loop (sink); break; case CMD_RECONNECT: gst_rtsp_client_sink_reconnect (sink, FALSE); break; default: break; } GST_OBJECT_LOCK (sink); /* and go back to sleep */ if (sink->pending_cmd == CMD_WAIT) { if (sink->task) gst_task_pause (sink->task); } /* reset waiting */ sink->busy_cmd = CMD_WAIT; GST_OBJECT_UNLOCK (sink); } static gboolean gst_rtsp_client_sink_start (GstRTSPClientSink * sink) { GST_DEBUG_OBJECT (sink, "starting"); sink->streams_collected = FALSE; gst_element_set_locked_state (GST_ELEMENT (sink->internal_bin), TRUE); gst_rtsp_client_sink_set_state (sink, GST_STATE_READY); GST_OBJECT_LOCK (sink); sink->pending_cmd = CMD_WAIT; if (sink->task == NULL) { sink->task = gst_task_new ((GstTaskFunction) gst_rtsp_client_sink_thread, sink, NULL); if (sink->task == NULL) goto task_error; gst_task_set_lock (sink->task, GST_RTSP_STREAM_GET_LOCK (sink)); } GST_OBJECT_UNLOCK (sink); return TRUE; /* ERRORS */ task_error: { GST_OBJECT_UNLOCK (sink); GST_ERROR_OBJECT (sink, "failed to create task"); return FALSE; } } static gboolean gst_rtsp_client_sink_stop (GstRTSPClientSink * sink) { GstTask *task; GST_DEBUG_OBJECT (sink, "stopping"); /* also cancels pending task */ gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT, CMD_ALL & ~CMD_CLOSE); GST_OBJECT_LOCK (sink); if ((task = sink->task)) { sink->task = NULL; GST_OBJECT_UNLOCK (sink); gst_task_stop (task); g_mutex_lock (&sink->block_streams_lock); g_cond_broadcast (&sink->block_streams_cond); g_mutex_unlock (&sink->block_streams_lock); g_mutex_lock (&sink->preroll_lock); g_cond_broadcast (&sink->preroll_cond); g_mutex_unlock (&sink->preroll_lock); /* make sure it is not running */ GST_RTSP_STREAM_LOCK (sink); GST_RTSP_STREAM_UNLOCK (sink); /* now wait for the task to finish */ gst_task_join (task); /* and free the task */ gst_object_unref (GST_OBJECT (task)); GST_OBJECT_LOCK (sink); } GST_OBJECT_UNLOCK (sink); /* ensure synchronously all is closed and clean */ gst_rtsp_client_sink_close (sink, FALSE, TRUE); return TRUE; } static GstStateChangeReturn gst_rtsp_client_sink_change_state (GstElement * element, GstStateChange transition) { GstRTSPClientSink *rtsp_client_sink; GstStateChangeReturn ret; rtsp_client_sink = GST_RTSP_CLIENT_SINK (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: if (!gst_rtsp_client_sink_start (rtsp_client_sink)) goto start_failed; break; case GST_STATE_CHANGE_READY_TO_PAUSED: /* init some state */ rtsp_client_sink->cur_protocols = rtsp_client_sink->protocols; /* first attempt, don't ignore timeouts */ rtsp_client_sink->ignore_timeout = FALSE; rtsp_client_sink->open_error = FALSE; gst_rtsp_client_sink_set_state (rtsp_client_sink, GST_STATE_PAUSED); g_mutex_lock (&rtsp_client_sink->preroll_lock); if (rtsp_client_sink->in_async) { GST_DEBUG_OBJECT (rtsp_client_sink, "Posting ASYNC-START"); gst_element_post_message (GST_ELEMENT_CAST (rtsp_client_sink), gst_message_new_async_start (GST_OBJECT_CAST (rtsp_client_sink))); } g_mutex_unlock (&rtsp_client_sink->preroll_lock); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: /* fall-through */ case GST_STATE_CHANGE_PLAYING_TO_PAUSED: /* unblock the tcp tasks and make the loop waiting */ if (gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_WAIT, CMD_LOOP)) { /* make sure it is waiting before we send PLAY below */ GST_RTSP_STREAM_LOCK (rtsp_client_sink); GST_RTSP_STREAM_UNLOCK (rtsp_client_sink); } break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_rtsp_client_sink_set_state (rtsp_client_sink, GST_STATE_READY); 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_NULL_TO_READY: ret = GST_STATE_CHANGE_SUCCESS; break; case GST_STATE_CHANGE_READY_TO_PAUSED: /* Return ASYNC and preroll input streams */ g_mutex_lock (&rtsp_client_sink->preroll_lock); if (rtsp_client_sink->in_async) ret = GST_STATE_CHANGE_ASYNC; g_mutex_unlock (&rtsp_client_sink->preroll_lock); gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_OPEN, 0); /* CMD_OPEN has been scheduled. Wait until the sink thread starts * opening connection to the server */ g_mutex_lock (&rtsp_client_sink->open_conn_lock); while (!rtsp_client_sink->open_conn_start) { GST_DEBUG_OBJECT (rtsp_client_sink, "wait for connection to be started"); g_cond_wait (&rtsp_client_sink->open_conn_cond, &rtsp_client_sink->open_conn_lock); } rtsp_client_sink->open_conn_start = FALSE; g_mutex_unlock (&rtsp_client_sink->open_conn_lock); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{ GST_DEBUG_OBJECT (rtsp_client_sink, "Switching to playing -sending RECORD"); gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_RECORD, 0); ret = GST_STATE_CHANGE_SUCCESS; break; } case GST_STATE_CHANGE_PLAYING_TO_PAUSED: /* send pause request and keep the idle task around */ gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_PAUSE, CMD_LOOP); ret = GST_STATE_CHANGE_NO_PREROLL; break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_CLOSE, CMD_PAUSE); ret = GST_STATE_CHANGE_SUCCESS; break; case GST_STATE_CHANGE_READY_TO_NULL: gst_rtsp_client_sink_stop (rtsp_client_sink); ret = GST_STATE_CHANGE_SUCCESS; break; default: break; } done: return ret; start_failed: { GST_DEBUG_OBJECT (rtsp_client_sink, "start failed"); return GST_STATE_CHANGE_FAILURE; } } /*** GSTURIHANDLER INTERFACE *************************************************/ static GstURIType gst_rtsp_client_sink_uri_get_type (GType type) { return GST_URI_SINK; } static const gchar *const * gst_rtsp_client_sink_uri_get_protocols (GType type) { static const gchar *protocols[] = { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp", "rtsps", "rtspsu", "rtspst", "rtspsh", NULL }; return protocols; } static gchar * gst_rtsp_client_sink_uri_get_uri (GstURIHandler * handler) { GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (handler); /* FIXME: make thread-safe */ return g_strdup (sink->conninfo.location); } static gboolean gst_rtsp_client_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri, GError ** error) { GstRTSPClientSink *sink; GstRTSPResult res; GstSDPResult sres; GstRTSPUrl *newurl = NULL; GstSDPMessage *sdp = NULL; sink = GST_RTSP_CLIENT_SINK (handler); /* same URI, we're fine */ if (sink->conninfo.location && uri && !strcmp (uri, sink->conninfo.location)) goto was_ok; if (g_str_has_prefix (uri, "rtsp-sdp://")) { sres = gst_sdp_message_new (&sdp); if (sres < 0) goto sdp_failed; GST_DEBUG_OBJECT (sink, "parsing SDP message"); sres = gst_sdp_message_parse_uri (uri, sdp); if (sres < 0) goto invalid_sdp; } else { /* try to parse */ GST_DEBUG_OBJECT (sink, "parsing URI"); 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_DEBUG_OBJECT (sink, "configuring URI"); g_free (sink->conninfo.location); sink->conninfo.location = g_strdup (uri); gst_rtsp_url_free (sink->conninfo.url); sink->conninfo.url = newurl; g_free (sink->conninfo.url_str); if (newurl) sink->conninfo.url_str = gst_rtsp_url_get_request_uri (sink->conninfo.url); else sink->conninfo.url_str = NULL; if (sink->uri_sdp) gst_sdp_message_free (sink->uri_sdp); sink->uri_sdp = sdp; sink->from_sdp = sdp != NULL; GST_DEBUG_OBJECT (sink, "set uri: %s", GST_STR_NULL (uri)); GST_DEBUG_OBJECT (sink, "request uri is: %s", GST_STR_NULL (sink->conninfo.url_str)); return TRUE; /* Special cases */ was_ok: { GST_DEBUG_OBJECT (sink, "URI was ok: '%s'", GST_STR_NULL (uri)); return TRUE; } sdp_failed: { GST_ERROR_OBJECT (sink, "Could not create new SDP (%d)", sres); g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI, "Could not create SDP"); return FALSE; } invalid_sdp: { GST_ERROR_OBJECT (sink, "Not a valid SDP (%d) '%s'", sres, GST_STR_NULL (uri)); gst_sdp_message_free (sdp); g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI, "Invalid SDP"); return FALSE; } parse_error: { GST_ERROR_OBJECT (sink, "Not a valid RTSP url '%s' (%d)", GST_STR_NULL (uri), res); g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI, "Invalid RTSP URI"); return FALSE; } } static void gst_rtsp_client_sink_uri_handler_init (gpointer g_iface, gpointer iface_data) { GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface; iface->get_type = gst_rtsp_client_sink_uri_get_type; iface->get_protocols = gst_rtsp_client_sink_uri_get_protocols; iface->get_uri = gst_rtsp_client_sink_uri_get_uri; iface->set_uri = gst_rtsp_client_sink_uri_set_uri; }