--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
+ * <2006> Lutz Mueller <lutz at topfrose dot de>
+ * <2015> Jan Schmidt <jan at centricular dot com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 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).
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * 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
+ * </refsect2>
+ */
+
+/* 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 <unistd.h>
+#endif /* HAVE_UNISTD_H */
+#include <stdlib.h>
+#include <string.h>
+#include <stdio.h>
+#include <stdarg.h>
+
+#include <gst/net/gstnet.h>
+#include <gst/sdp/gstsdpmessage.h>
+#include <gst/sdp/gstmikey.h>
+#include <gst/rtp/rtp.h>
+
+#include "gstrtspclientsink.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_client_sink_debug);
+#define GST_CAT_DEFAULT (rtsp_client_sink_debug)
+
+static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%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,
+ 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 AES_128_KEY_LEN 16
+#define AES_256_KEY_LEN 32
+
+#define HMAC_32_KEY_LEN 4
+#define HMAC_80_KEY_LEN 10
+
+#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
+
+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
+};
+
+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 };
+
+#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));
+
+#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.
+ *
+ */
+ 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::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, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
+ G_TYPE_POINTER, G_TYPE_POINTER);
+
+ /**
+ * 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 | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
+ g_cclosure_marshal_generic, 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 | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
+ g_cclosure_marshal_generic, 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);
+
+ 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_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtptemplate));
+
+ 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 <jan@centricular.com>");
+
+ gstbin_class->handle_message = gst_rtsp_client_sink_handle_message;
+}
+
+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->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;
+
+ sink->internal_bin = (GstBin *) gst_bin_new ("rtspbin");
+ 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->send_lock);
+
+ g_mutex_clear (&rtsp_client_sink->preroll_lock);
+ g_cond_clear (&rtsp_client_sink->preroll_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 (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);
+ 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);
+
+ 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:
+ 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;
+
+ 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++;
+
+ 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);
+
+#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."));
+
+ if (stream)
+ g_object_unref (stream);
+ 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;
+
+ GstElement *payloader;
+ GstPad *sinkpad, *srcpad, *ghostsink;
+
+ context = gst_pad_get_element_private (pad);
+
+ /* Find the payloader. FIXME: Allow user to provide payloader via pad property */
+ 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);
+
+ 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);
+ 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)) {
+ gst_event_unref (event);
+ return FALSE;
+ }
+ }
+ }
+
+ 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) {
+ /* No target yet - return the union of all payloader caps */
+ GstCaps *caps = gst_rtsp_client_sink_get_payloader_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;
+ }
+ }
+
+ 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_ghost_pad_new_no_target_from_template (tmpname, templ);
+ 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);
+
+ (void) gst_rtsp_client_sink_get_factories ();
+
+ 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);
+
+ 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_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.tv_sec = timeout / G_USEC_PER_SEC;
+ rtsp_client_sink->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
+
+ if (timeout != 0)
+ rtsp_client_sink->ptcp_timeout = &rtsp_client_sink->tcp_timeout;
+ else
+ rtsp_client_sink->ptcp_timeout = NULL;
+}
+
+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);
+ } else {
+ 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;
+ 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:
+ {
+ guint64 timeout;
+
+ timeout = rtsp_client_sink->tcp_timeout.tv_sec * G_USEC_PER_SEC +
+ rtsp_client_sink->tcp_timeout.tv_usec;
+ g_value_set_uint64 (value, 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;
+ 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);
+ 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,
+ GstRTSPConnection * conn, GstRTSPMessage * message, GTimeVal * timeout)
+{
+ GstRTSPResult ret;
+
+ if (conn)
+ ret = gst_rtsp_connection_send (conn, message, timeout);
+ else
+ ret = GST_RTSP_ERROR;
+
+ return ret;
+}
+
+static GstRTSPResult
+gst_rtsp_client_sink_connection_receive (GstRTSPClientSink * sink,
+ GstRTSPConnection * conn, GstRTSPMessage * message, GTimeVal * timeout)
+{
+ GstRTSPResult ret;
+
+ if (conn)
+ ret = gst_rtsp_connection_receive (conn, message, timeout);
+ else
+ ret = GST_RTSP_ERROR;
+
+ return ret;
+}
+
+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);
+ }
+
+ 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 (info->connection,
+ sink->ptcp_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);
+ info->connection = NULL;
+ }
+ 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,
+ GstRTSPConnection * conn, 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, conn, &response, NULL);
+ 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.connection,
+ &request, NULL);
+ 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) {
+ GTimeVal tv_timeout;
+
+ /* get the next timeout interval */
+ gst_rtsp_connection_next_timeout (sink->conninfo.connection, &tv_timeout);
+
+ GST_DEBUG_OBJECT (sink, "doing receive with timeout %d seconds",
+ (gint) tv_timeout.tv_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.connection, &message, &tv_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;
+ 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.connection, &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
+
+static const gchar *
+gst_rtsp_client_sink_skip_lws (const gchar * s)
+{
+ while (g_ascii_isspace (*s))
+ s++;
+ return s;
+}
+
+static const gchar *
+gst_rtsp_client_sink_unskip_lws (const gchar * s, const gchar * start)
+{
+ while (s > start && g_ascii_isspace (*(s - 1)))
+ s--;
+ return s;
+}
+
+static const gchar *
+gst_rtsp_client_sink_skip_commas (const gchar * s)
+{
+ /* The grammar allows for multiple commas */
+ while (g_ascii_isspace (*s) || *s == ',')
+ s++;
+ return s;
+}
+
+static const gchar *
+gst_rtsp_client_sink_skip_item (const gchar * s)
+{
+ gboolean quoted = FALSE;
+ const gchar *start = s;
+
+ /* A list item ends at the last non-whitespace character
+ * before a comma which is not inside a quoted-string. Or at
+ * the end of the string.
+ */
+ while (*s) {
+ if (*s == '"')
+ quoted = !quoted;
+ else if (quoted) {
+ if (*s == '\\' && *(s + 1))
+ s++;
+ } else {
+ if (*s == ',')
+ break;
+ }
+ s++;
+ }
+
+ return gst_rtsp_client_sink_unskip_lws (s, start);
+}
+
+static void
+gst_rtsp_decode_quoted_string (gchar * quoted_string)
+{
+ gchar *src, *dst;
+
+ src = quoted_string + 1;
+ dst = quoted_string;
+ while (*src && *src != '"') {
+ if (*src == '\\' && *(src + 1))
+ src++;
+ *dst++ = *src++;
+ }
+ *dst = '\0';
+}
+
+/* Extract the authentication tokens that the server provided for each method
+ * into an array of structures and give those to the connection object.
+ */
+static void
+gst_rtsp_client_sink_parse_digest_challenge (GstRTSPConnection * conn,
+ const gchar * header, gboolean * stale)
+{
+ GSList *list = NULL, *iter;
+ const gchar *end;
+ gchar *item, *eq, *name_end, *value;
+
+ g_return_if_fail (stale != NULL);
+
+ gst_rtsp_connection_clear_auth_params (conn);
+ *stale = FALSE;
+
+ /* Parse a header whose content is described by RFC2616 as
+ * "#something", where "something" does not itself contain commas,
+ * except as part of quoted-strings, into a list of allocated strings.
+ */
+ header = gst_rtsp_client_sink_skip_commas (header);
+ while (*header) {
+ end = gst_rtsp_client_sink_skip_item (header);
+ list = g_slist_prepend (list, g_strndup (header, end - header));
+ header = gst_rtsp_client_sink_skip_commas (end);
+ }
+ if (!list)
+ return;
+
+ list = g_slist_reverse (list);
+ for (iter = list; iter; iter = iter->next) {
+ item = iter->data;
+
+ eq = strchr (item, '=');
+ if (eq) {
+ name_end = (gchar *) gst_rtsp_client_sink_unskip_lws (eq, item);
+ if (name_end == item) {
+ /* That's no good... */
+ g_free (item);
+ continue;
+ }
+
+ *name_end = '\0';
+
+ value = (gchar *) gst_rtsp_client_sink_skip_lws (eq + 1);
+ if (*value == '"')
+ gst_rtsp_decode_quoted_string (value);
+ } else
+ value = NULL;
+
+ if (value && strcmp (item, "stale") == 0 && strcmp (value, "TRUE") == 0)
+ *stale = TRUE;
+ gst_rtsp_connection_set_auth_param (conn, item, value);
+ g_free (item);
+ }
+
+ g_slist_free (list);
+}
+
+/* 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 (gchar * hdr, GstRTSPAuthMethod * methods,
+ GstRTSPConnection * conn, gboolean * stale)
+{
+ gchar *start;
+
+ g_return_if_fail (hdr != NULL);
+ g_return_if_fail (methods != NULL);
+ g_return_if_fail (stale != NULL);
+
+ /* Skip whitespace at the start of the string */
+ for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
+
+ if (g_ascii_strncasecmp (start, "basic", 5) == 0)
+ *methods |= GST_RTSP_AUTH_BASIC;
+ else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
+ *methods |= GST_RTSP_AUTH_DIGEST;
+ gst_rtsp_client_sink_parse_digest_challenge (conn, &start[7], stale);
+ }
+}
+
+/**
+ * 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;
+ gchar *hdr;
+ gboolean stale = FALSE;
+
+ conn = sink->conninfo.connection;
+
+ /* Identify the available auth methods and see if any are supported */
+ if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
+ &hdr, 0) == GST_RTSP_OK) {
+ gst_rtsp_client_sink_parse_auth_hdr (hdr, &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,
+ GstRTSPConnection * conn, GstRTSPMessage * request,
+ GstRTSPMessage * response, GstRTSPStatusCode * code)
+{
+ GstRTSPResult res;
+ GstRTSPStatusCode thecode;
+ gchar *content_base = NULL;
+ gint try = 0;
+
+again:
+ GST_DEBUG_OBJECT (sink, "sending message");
+
+ if (sink->debug)
+ gst_rtsp_message_dump (request);
+
+ g_mutex_lock (&sink->send_lock);
+
+ res =
+ gst_rtsp_client_sink_connection_send (sink, conn, request,
+ sink->ptcp_timeout);
+ if (res < 0) {
+ g_mutex_unlock (&sink->send_lock);
+ goto send_error;
+ }
+
+ gst_rtsp_connection_reset_timeout (conn);
+
+ /* 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, conn, response,
+ sink->ptcp_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, conn, 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, GstRTSPConnection * conn,
+ 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, conn, request, 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));
+
+ 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.connection, &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;
+
+ 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);
+
+ /* Collect all our input streams and create
+ * stream objects before actually returning */
+ gst_rtsp_client_sink_collect_streams (sink);
+
+ 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);
+
+ 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->connection, &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 gboolean
+gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink)
+{
+ GstRTSPStreamContext *context;
+ GList *walk;
+ const gchar *base;
+ gboolean has_slash;
+
+ GST_DEBUG_OBJECT (sink, "Collecting stream information");
+
+ if (!gst_rtsp_client_sink_configure_manager (sink))
+ return FALSE;
+
+ base = get_aggregate_control (sink);
+ /* check if the base ends with / */
+ has_slash = g_str_has_suffix (base, "/");
+
+ 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_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);
+
+ /* concatenate the two strings, insert / when not present */
+ g_free (context->conninfo.location);
+ context->conninfo.location =
+ g_strdup_printf ("%s%sstream=%d", base, has_slash ? "" : "/",
+ context->index);
+
+ 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);
+ }
+
+ 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;
+
+ /* 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.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);
+
+ return TRUE;
+
+join_bin_failed:
+
+ 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 guint8
+enc_key_length_from_cipher_name (const gchar * cipher)
+{
+ if (g_strcmp0 (cipher, "aes-128-icm") == 0)
+ return AES_128_KEY_LEN;
+ else if (g_strcmp0 (cipher, "aes-256-icm") == 0)
+ return AES_256_KEY_LEN;
+ else {
+ GST_ERROR ("encryption algorithm '%s' not supported", cipher);
+ return 0;
+ }
+}
+
+static guint8
+auth_key_length_from_auth_name (const gchar * auth)
+{
+ if (g_strcmp0 (auth, "hmac-sha1-32") == 0)
+ return HMAC_32_KEY_LEN;
+ else if (g_strcmp0 (auth, "hmac-sha1-80") == 0)
+ return HMAC_80_KEY_LEN;
+ else {
+ GST_ERROR ("authentication algorithm '%s' not supported", auth);
+ return 0;
+ }
+}
+
+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)
+{
+ GBytes *bytes;
+ gchar *result, *base64;
+ const guint8 *data;
+ gsize size;
+ GstMIKEYMessage *msg;
+ GstMIKEYPayload *payload, *pkd;
+ guint8 byte;
+ GstStructure *s;
+ GstMapInfo info;
+ GstBuffer *srtpkey;
+ const GValue *val;
+ const gchar *srtcpcipher, *srtcpauth;
+ guint send_ssrc;
+
+ context->srtcpparams = signal_get_srtcp_params (sink, context);
+ if (context->srtcpparams == NULL)
+ context->srtcpparams = gst_rtsp_stream_get_caps (context->stream);
+
+ s = gst_caps_get_structure (context->srtcpparams, 0);
+
+ srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
+ srtcpauth = gst_structure_get_string (s, "srtcp-auth");
+ val = gst_structure_get_value (s, "srtp-key");
+
+ if (srtcpcipher == NULL || srtcpauth == NULL || val == NULL) {
+ GST_ERROR_OBJECT (sink, "could not find the right SRTP parameters in caps");
+ return NULL;
+ }
+
+ srtpkey = gst_value_get_buffer (val);
+
+ gst_rtsp_stream_get_ssrc (context->stream, &send_ssrc);
+ GST_LOG_OBJECT (sink, "Stream %p ssrc %x", context->stream, send_ssrc);
+
+ msg = gst_mikey_message_new ();
+ /* unencrypted MIKEY message, we send this over TLS so this is allowed */
+ gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
+ FALSE, GST_MIKEY_PRF_MIKEY_1, g_random_int (), GST_MIKEY_MAP_TYPE_SRTP);
+ /* add policy '0' for our SSRC */
+ gst_mikey_message_add_cs_srtp (msg, 0, send_ssrc, 0);
+ /* timestamp is now */
+ gst_mikey_message_add_t_now_ntp_utc (msg);
+ /* add some random data */
+ gst_mikey_message_add_rand_len (msg, 16);
+
+ /* the policy '0' is SRTP */
+ payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
+ gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
+
+ /* only AES-CM is supported */
+ byte = 1;
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1, &byte);
+ /* encryption key length */
+ byte = enc_key_length_from_cipher_name (srtcpcipher);
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_KEY_LEN, 1,
+ &byte);
+ /* only HMAC-SHA1 */
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
+ &byte);
+ /* authentication key length */
+ byte = auth_key_length_from_auth_name (srtcpauth);
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_KEY_LEN, 1,
+ &byte);
+ /* we enable encryption on RTP and RTCP */
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
+ &byte);
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
+ &byte);
+ /* we enable authentication on RTP and RTCP */
+ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
+ &byte);
+ gst_mikey_message_add_payload (msg, payload);
+
+ /* make unencrypted KEMAC */
+ payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC);
+ gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL, GST_MIKEY_MAC_NULL);
+ /* add the key in KEMAC */
+ pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA);
+ gst_buffer_map (srtpkey, &info, GST_MAP_READ);
+ gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size,
+ info.data);
+ gst_buffer_unmap (srtpkey, &info);
+ gst_mikey_payload_kemac_add_sub (payload, pkd);
+ gst_mikey_message_add_payload (msg, payload);
+
+ /* now serialize this to bytes */
+ bytes = gst_mikey_message_to_bytes (msg, NULL, NULL);
+ gst_mikey_message_unref (msg);
+ /* and make it into base64 */
+ data = g_bytes_get_data (bytes, &size);
+ base64 = g_base64_encode (data, size);
+ g_bytes_unref (bytes);
+
+ result = g_strdup_printf ("prot=mikey;uri=\"%s\";data=\"%s\"",
+ 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;
+ GstMapInfo map_info;
+ guint8 *data;
+ guint usize;
+
+ gst_rtsp_message_init_data (&message, channel);
+
+ /* FIXME, need some sort of iovec RTSPMessage here */
+ if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
+ return FALSE;
+
+ gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
+
+ res =
+ gst_rtsp_client_sink_try_send (sink, sink->conninfo.connection, &message,
+ NULL, NULL);
+
+ gst_rtsp_message_steal_body (&message, &data, &usize);
+ gst_buffer_unmap (buffer, &map_info);
+
+ gst_rtsp_message_unset (&message);
+
+ 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;
+
+ GstRTSPConnection *conn;
+ 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;
+ }
+ conn = context->conninfo.connection;
+ } else {
+ conn = sink->conninfo.connection;
+ }
+ GST_DEBUG_OBJECT (sink, "doing setup of stream %p with %s", stream,
+ context->conninfo.location);
+
+ conn_socket = gst_rtsp_connection_get_read_socket (conn);
+ 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;
+ }
+
+ /* select transport */
+ gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
+
+ /* 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));
+
+ /* handle the code ourselves */
+ res = gst_rtsp_client_sink_send (sink, conn, &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);
+ }
+
+ /* 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;
+ }
+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, };
+
+ const gchar *proto;
+ gchar *sess_id, *client_ip, *str;
+ GSocketAddress *sa;
+ GInetAddress *ia;
+ GSocket *conn_socket;
+ GList *walk;
+
+ /* Wait for streams to preroll */
+ g_mutex_lock (&sink->preroll_lock);
+ while (sink->in_async) {
+ GST_LOG_OBJECT (sink, "Waiting for ASYNC_DONE preroll");
+ g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
+ }
+ g_mutex_unlock (&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.");
+ goto done;
+ }
+
+ /* 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;
+
+ 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.connection, &request,
+ &response, NULL)) < 0)
+ goto send_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.connection, &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;
+
+ /* 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;
+ }
+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;
+ GstRTSPConnection *conn;
+ 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) {
+ conn = sink->conninfo.connection;
+ } else if (stream->conninfo.connection) {
+ conn = stream->conninfo.connection;
+ } 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, conn, &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 */
+ g_cond_broadcast (&rtsp_client_sink->preroll_cond);
+ }
+ 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);
+ }
+ }
+ 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:
+ gst_rtsp_client_sink_open (sink, TRUE);
+ 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;
+ sink->in_async = TRUE;
+ 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);
+
+ /* 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);
+ 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;
+}
projects / platform / upstream / gstreamer.git / commitdiff