--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005-2009> Wim Taymans <wim.taymans@gmail.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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+/**
+ * SECTION:gstrtspconnection
+ * @short_description: manage RTSP connections
+ * @see_also: gstrtspurl
+ *
+ * This object manages the RTSP connection to the server. It provides function
+ * to receive and send bytes and messages.
+ *
+ * Last reviewed on 2007-07-24 (0.10.14)
+ */
+
+#ifdef HAVE_CONFIG_H
+# include <config.h>
+#endif
+
+#include <stdio.h>
+#include <errno.h>
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+
+#ifdef HAVE_UNISTD_H
+#include <unistd.h>
+#endif
+
+/* we include this here to get the G_OS_* defines */
+#include <glib.h>
+#include <gst/gst.h>
+
+#ifdef G_OS_WIN32
+/* ws2_32.dll has getaddrinfo and freeaddrinfo on Windows XP and later.
+ * minwg32 headers check WINVER before allowing the use of these */
+#ifndef WINVER
+#define WINVER 0x0501
+#endif
+#include <winsock2.h>
+#include <ws2tcpip.h>
+#define EINPROGRESS WSAEINPROGRESS
+#else
+#include <sys/ioctl.h>
+#include <netdb.h>
+#include <sys/socket.h>
+#include <fcntl.h>
+#include <netinet/in.h>
+#endif
+
+#ifdef HAVE_FIONREAD_IN_SYS_FILIO
+#include <sys/filio.h>
+#endif
+
+#include "gstrtspconnection.h"
+#include "gstrtspbase64.h"
+
+union gst_sockaddr
+{
+ struct sockaddr sa;
+ struct sockaddr_in sa_in;
+ struct sockaddr_in6 sa_in6;
+ struct sockaddr_storage sa_stor;
+};
+
+typedef struct
+{
+ gint state;
+ guint save;
+ guchar out[3]; /* the size must be evenly divisible by 3 */
+ guint cout;
+ guint coutl;
+} DecodeCtx;
+
+#ifdef MSG_NOSIGNAL
+#define SEND_FLAGS MSG_NOSIGNAL
+#else
+#define SEND_FLAGS 0
+#endif
+
+#ifdef G_OS_WIN32
+#define READ_SOCKET(fd, buf, len) recv (fd, (char *)buf, len, 0)
+#define WRITE_SOCKET(fd, buf, len) send (fd, (const char *)buf, len, SEND_FLAGS)
+#define SETSOCKOPT(sock, level, name, val, len) setsockopt (sock, level, name, (const char *)val, len)
+#define CLOSE_SOCKET(sock) closesocket (sock)
+#define ERRNO_IS_EAGAIN (WSAGetLastError () == WSAEWOULDBLOCK)
+#define ERRNO_IS_EINTR (WSAGetLastError () == WSAEINTR)
+/* According to Microsoft's connect() documentation this one returns
+ * WSAEWOULDBLOCK and not WSAEINPROGRESS. */
+#define ERRNO_IS_EINPROGRESS (WSAGetLastError () == WSAEWOULDBLOCK)
+#else
+#define READ_SOCKET(fd, buf, len) read (fd, buf, len)
+#define WRITE_SOCKET(fd, buf, len) send (fd, buf, len, SEND_FLAGS)
+#define SETSOCKOPT(sock, level, name, val, len) setsockopt (sock, level, name, val, len)
+#define CLOSE_SOCKET(sock) close (sock)
+#define ERRNO_IS_EAGAIN (errno == EAGAIN)
+#define ERRNO_IS_EINTR (errno == EINTR)
+#define ERRNO_IS_EINPROGRESS (errno == EINPROGRESS)
+#endif
+
+#define ADD_POLLFD(fdset, pfd, fd) \
+G_STMT_START { \
+ (pfd)->fd = fd; \
+ gst_poll_add_fd (fdset, pfd); \
+} G_STMT_END
+
+#define REMOVE_POLLFD(fdset, pfd) \
+G_STMT_START { \
+ if ((pfd)->fd != -1) { \
+ GST_DEBUG ("remove fd %d", (pfd)->fd); \
+ gst_poll_remove_fd (fdset, pfd); \
+ CLOSE_SOCKET ((pfd)->fd); \
+ (pfd)->fd = -1; \
+ } \
+} G_STMT_END
+
+typedef enum
+{
+ TUNNEL_STATE_NONE,
+ TUNNEL_STATE_GET,
+ TUNNEL_STATE_POST,
+ TUNNEL_STATE_COMPLETE
+} GstRTSPTunnelState;
+
+#define TUNNELID_LEN 24
+
+struct _GstRTSPConnection
+{
+ /*< private > */
+ /* URL for the connection */
+ GstRTSPUrl *url;
+
+ /* connection state */
+ GstPollFD fd0;
+ GstPollFD fd1;
+
+ GstPollFD *readfd;
+ GstPollFD *writefd;
+
+ gboolean manual_http;
+
+ gchar tunnelid[TUNNELID_LEN];
+ gboolean tunneled;
+ GstRTSPTunnelState tstate;
+
+ GstPoll *fdset;
+ gchar *ip;
+
+ gint read_ahead;
+
+ gchar *initial_buffer;
+ gsize initial_buffer_offset;
+
+ /* Session state */
+ gint cseq; /* sequence number */
+ gchar session_id[512]; /* session id */
+ gint timeout; /* session timeout in seconds */
+ GTimer *timer; /* timeout timer */
+
+ /* Authentication */
+ GstRTSPAuthMethod auth_method;
+ gchar *username;
+ gchar *passwd;
+ GHashTable *auth_params;
+
+ DecodeCtx ctx;
+ DecodeCtx *ctxp;
+
+ gchar *proxy_host;
+ guint proxy_port;
+};
+
+enum
+{
+ STATE_START = 0,
+ STATE_DATA_HEADER,
+ STATE_DATA_BODY,
+ STATE_READ_LINES,
+ STATE_END,
+ STATE_LAST
+};
+
+enum
+{
+ READ_AHEAD_EOH = -1, /* end of headers */
+ READ_AHEAD_CRLF = -2,
+ READ_AHEAD_CRLFCR = -3
+};
+
+/* a structure for constructing RTSPMessages */
+typedef struct
+{
+ gint state;
+ GstRTSPResult status;
+ guint8 buffer[4096];
+ guint offset;
+
+ guint line;
+ guint8 *body_data;
+ glong body_len;
+} GstRTSPBuilder;
+
+static void
+build_reset (GstRTSPBuilder * builder)
+{
+ g_free (builder->body_data);
+ memset (builder, 0, sizeof (GstRTSPBuilder));
+}
+
+/**
+ * gst_rtsp_connection_create:
+ * @url: a #GstRTSPUrl
+ * @conn: storage for a #GstRTSPConnection
+ *
+ * Create a newly allocated #GstRTSPConnection from @url and store it in @conn.
+ * The connection will not yet attempt to connect to @url, use
+ * gst_rtsp_connection_connect().
+ *
+ * A copy of @url will be made.
+ *
+ * Returns: #GST_RTSP_OK when @conn contains a valid connection.
+ */
+GstRTSPResult
+gst_rtsp_connection_create (const GstRTSPUrl * url, GstRTSPConnection ** conn)
+{
+ GstRTSPConnection *newconn;
+#ifdef G_OS_WIN32
+ WSADATA w;
+ int error;
+#endif
+
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+
+#ifdef G_OS_WIN32
+ error = WSAStartup (0x0202, &w);
+
+ if (error)
+ goto startup_error;
+
+ if (w.wVersion != 0x0202)
+ goto version_error;
+#endif
+
+ newconn = g_new0 (GstRTSPConnection, 1);
+
+ if ((newconn->fdset = gst_poll_new (TRUE)) == NULL)
+ goto no_fdset;
+
+ newconn->url = gst_rtsp_url_copy (url);
+ newconn->fd0.fd = -1;
+ newconn->fd1.fd = -1;
+ newconn->timer = g_timer_new ();
+ newconn->timeout = 60;
+ newconn->cseq = 1;
+
+ newconn->auth_method = GST_RTSP_AUTH_NONE;
+ newconn->username = NULL;
+ newconn->passwd = NULL;
+ newconn->auth_params = NULL;
+
+ *conn = newconn;
+
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+#ifdef G_OS_WIN32
+startup_error:
+ {
+ g_warning ("Error %d on WSAStartup", error);
+ return GST_RTSP_EWSASTART;
+ }
+version_error:
+ {
+ g_warning ("Windows sockets are not version 0x202 (current 0x%x)",
+ w.wVersion);
+ WSACleanup ();
+ return GST_RTSP_EWSAVERSION;
+ }
+#endif
+no_fdset:
+ {
+ g_free (newconn);
+#ifdef G_OS_WIN32
+ WSACleanup ();
+#endif
+ return GST_RTSP_ESYS;
+ }
+}
+
+/**
+ * gst_rtsp_connection_create_from_fd:
+ * @fd: a file descriptor
+ * @ip: the IP address of the other end
+ * @port: the port used by the other end
+ * @initial_buffer: data already read from @fd
+ * @conn: storage for a #GstRTSPConnection
+ *
+ * Create a new #GstRTSPConnection for handling communication on the existing
+ * file descriptor @fd. The @initial_buffer contains any data already read from
+ * @fd which should be used before starting to read new data.
+ *
+ * Returns: #GST_RTSP_OK when @conn contains a valid connection.
+ *
+ * Since: 0.10.25
+ */
+GstRTSPResult
+gst_rtsp_connection_create_from_fd (gint fd, const gchar * ip, guint16 port,
+ const gchar * initial_buffer, GstRTSPConnection ** conn)
+{
+ GstRTSPConnection *newconn = NULL;
+ GstRTSPUrl *url;
+#ifdef G_OS_WIN32
+ gulong flags = 1;
+#endif
+ GstRTSPResult res;
+
+ g_return_val_if_fail (fd >= 0, GST_RTSP_EINVAL);
+ g_return_val_if_fail (ip != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+
+ /* set to non-blocking mode so that we can cancel the communication */
+#ifndef G_OS_WIN32
+ fcntl (fd, F_SETFL, O_NONBLOCK);
+#else
+ ioctlsocket (fd, FIONBIO, &flags);
+#endif /* G_OS_WIN32 */
+
+ /* create a url for the client address */
+ url = g_new0 (GstRTSPUrl, 1);
+ url->host = g_strdup (ip);
+ url->port = port;
+
+ /* now create the connection object */
+ GST_RTSP_CHECK (gst_rtsp_connection_create (url, &newconn), newconn_failed);
+ gst_rtsp_url_free (url);
+
+ ADD_POLLFD (newconn->fdset, &newconn->fd0, fd);
+
+ /* both read and write initially */
+ newconn->readfd = &newconn->fd0;
+ newconn->writefd = &newconn->fd0;
+
+ newconn->ip = g_strdup (ip);
+
+ newconn->initial_buffer = g_strdup (initial_buffer);
+
+ *conn = newconn;
+
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+newconn_failed:
+ {
+ gst_rtsp_url_free (url);
+ return res;
+ }
+}
+
+/**
+ * gst_rtsp_connection_accept:
+ * @sock: a socket
+ * @conn: storage for a #GstRTSPConnection
+ *
+ * Accept a new connection on @sock and create a new #GstRTSPConnection for
+ * handling communication on new socket.
+ *
+ * Returns: #GST_RTSP_OK when @conn contains a valid connection.
+ *
+ * Since: 0.10.23
+ */
+GstRTSPResult
+gst_rtsp_connection_accept (gint sock, GstRTSPConnection ** conn)
+{
+ int fd;
+ union gst_sockaddr sa;
+ socklen_t slen = sizeof (sa);
+ gchar ip[INET6_ADDRSTRLEN];
+ guint16 port;
+
+ g_return_val_if_fail (sock >= 0, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+
+ memset (&sa, 0, slen);
+
+#ifndef G_OS_WIN32
+ fd = accept (sock, &sa.sa, &slen);
+#else
+ fd = accept (sock, &sa.sa, (gint *) & slen);
+#endif /* G_OS_WIN32 */
+ if (fd == -1)
+ goto accept_failed;
+
+ if (getnameinfo (&sa.sa, slen, ip, sizeof (ip), NULL, 0, NI_NUMERICHOST) != 0)
+ goto getnameinfo_failed;
+
+ if (sa.sa.sa_family == AF_INET)
+ port = sa.sa_in.sin_port;
+ else if (sa.sa.sa_family == AF_INET6)
+ port = sa.sa_in6.sin6_port;
+ else
+ goto wrong_family;
+
+ return gst_rtsp_connection_create_from_fd (fd, ip, port, NULL, conn);
+
+ /* ERRORS */
+accept_failed:
+ {
+ return GST_RTSP_ESYS;
+ }
+getnameinfo_failed:
+wrong_family:
+ {
+ CLOSE_SOCKET (fd);
+ return GST_RTSP_ERROR;
+ }
+}
+
+static gchar *
+do_resolve (const gchar * host)
+{
+ static gchar ip[INET6_ADDRSTRLEN];
+ struct addrinfo *aires, hints;
+ struct addrinfo *ai;
+ gint aierr;
+
+ memset (&hints, 0, sizeof (struct addrinfo));
+ hints.ai_family = AF_UNSPEC; /* Allow IPv4 or IPv6 */
+ hints.ai_socktype = SOCK_DGRAM; /* Datagram socket */
+ hints.ai_flags = AI_PASSIVE; /* For wildcard IP address */
+ hints.ai_protocol = 0; /* Any protocol */
+ hints.ai_canonname = NULL;
+ hints.ai_addr = NULL;
+ hints.ai_next = NULL;
+
+ aierr = getaddrinfo (host, NULL, &hints, &aires);
+ if (aierr != 0)
+ goto no_addrinfo;
+
+ for (ai = aires; ai; ai = ai->ai_next) {
+ if (ai->ai_family == AF_INET || ai->ai_family == AF_INET6) {
+ break;
+ }
+ }
+ if (ai == NULL)
+ goto no_family;
+
+ aierr = getnameinfo (ai->ai_addr, ai->ai_addrlen, ip, sizeof (ip), NULL, 0,
+ NI_NUMERICHOST | NI_NUMERICSERV);
+ if (aierr != 0)
+ goto no_address;
+
+ freeaddrinfo (aires);
+
+ return g_strdup (ip);
+
+ /* ERRORS */
+no_addrinfo:
+ {
+ GST_ERROR ("no addrinfo found for %s: %s", host, gai_strerror (aierr));
+ return NULL;
+ }
+no_family:
+ {
+ GST_ERROR ("no family found for %s", host);
+ freeaddrinfo (aires);
+ return NULL;
+ }
+no_address:
+ {
+ GST_ERROR ("no address found for %s: %s", host, gai_strerror (aierr));
+ freeaddrinfo (aires);
+ return NULL;
+ }
+}
+
+static GstRTSPResult
+do_connect (const gchar * ip, guint16 port, GstPollFD * fdout,
+ GstPoll * fdset, GTimeVal * timeout)
+{
+ gint fd;
+ struct addrinfo hints;
+ struct addrinfo *aires;
+ struct addrinfo *ai;
+ gint aierr;
+ gchar service[NI_MAXSERV];
+ gint ret;
+#ifdef G_OS_WIN32
+ unsigned long flags = 1;
+#endif /* G_OS_WIN32 */
+ GstClockTime to;
+ gint retval;
+
+ memset (&hints, 0, sizeof hints);
+ hints.ai_flags = AI_NUMERICHOST;
+ hints.ai_family = AF_UNSPEC;
+ hints.ai_socktype = SOCK_STREAM;
+ g_snprintf (service, sizeof (service) - 1, "%hu", port);
+ service[sizeof (service) - 1] = '\0';
+
+ aierr = getaddrinfo (ip, service, &hints, &aires);
+ if (aierr != 0)
+ goto no_addrinfo;
+
+ for (ai = aires; ai; ai = ai->ai_next) {
+ if (ai->ai_family == AF_INET || ai->ai_family == AF_INET6) {
+ break;
+ }
+ }
+ if (ai == NULL)
+ goto no_family;
+
+ fd = socket (ai->ai_family, SOCK_STREAM, 0);
+ if (fd == -1)
+ goto no_socket;
+
+ /* set to non-blocking mode so that we can cancel the connect */
+#ifndef G_OS_WIN32
+ fcntl (fd, F_SETFL, O_NONBLOCK);
+#else
+ ioctlsocket (fd, FIONBIO, &flags);
+#endif /* G_OS_WIN32 */
+
+ /* add the socket to our fdset */
+ ADD_POLLFD (fdset, fdout, fd);
+
+ /* we are going to connect ASYNC now */
+ ret = connect (fd, ai->ai_addr, ai->ai_addrlen);
+ if (ret == 0)
+ goto done;
+ if (!ERRNO_IS_EINPROGRESS)
+ goto sys_error;
+
+ /* wait for connect to complete up to the specified timeout or until we got
+ * interrupted. */
+ gst_poll_fd_ctl_write (fdset, fdout, TRUE);
+
+ to = timeout ? GST_TIMEVAL_TO_TIME (*timeout) : GST_CLOCK_TIME_NONE;
+
+ do {
+ retval = gst_poll_wait (fdset, to);
+ } while (retval == -1 && (errno == EINTR || errno == EAGAIN));
+
+ if (retval == 0)
+ goto timeout;
+ else if (retval == -1)
+ goto sys_error;
+
+ /* we can still have an error connecting on windows */
+ if (gst_poll_fd_has_error (fdset, fdout)) {
+ socklen_t len = sizeof (errno);
+#ifndef G_OS_WIN32
+ getsockopt (fd, SOL_SOCKET, SO_ERROR, &errno, &len);
+#else
+ getsockopt (fd, SOL_SOCKET, SO_ERROR, (char *) &errno, &len);
+#endif
+ goto sys_error;
+ }
+
+ gst_poll_fd_ignored (fdset, fdout);
+
+done:
+ freeaddrinfo (aires);
+
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+no_addrinfo:
+ {
+ GST_ERROR ("no addrinfo found for %s: %s", ip, gai_strerror (aierr));
+ return GST_RTSP_ERROR;
+ }
+no_family:
+ {
+ GST_ERROR ("no family found for %s", ip);
+ freeaddrinfo (aires);
+ return GST_RTSP_ERROR;
+ }
+no_socket:
+ {
+ GST_ERROR ("no socket %d (%s)", errno, g_strerror (errno));
+ freeaddrinfo (aires);
+ return GST_RTSP_ESYS;
+ }
+sys_error:
+ {
+ GST_ERROR ("system error %d (%s)", errno, g_strerror (errno));
+ REMOVE_POLLFD (fdset, fdout);
+ freeaddrinfo (aires);
+ return GST_RTSP_ESYS;
+ }
+timeout:
+ {
+ GST_ERROR ("timeout");
+ REMOVE_POLLFD (fdset, fdout);
+ freeaddrinfo (aires);
+ return GST_RTSP_ETIMEOUT;
+ }
+}
+
+static GstRTSPResult
+setup_tunneling (GstRTSPConnection * conn, GTimeVal * timeout)
+{
+ gint i;
+ GstRTSPResult res;
+ gchar *ip;
+ gchar *uri;
+ gchar *value;
+ guint16 port, url_port;
+ GstRTSPUrl *url;
+ gchar *hostparam;
+ GstRTSPMessage *msg;
+ GstRTSPMessage response;
+ gboolean old_http;
+
+ memset (&response, 0, sizeof (response));
+ gst_rtsp_message_init (&response);
+
+ /* create a random sessionid */
+ for (i = 0; i < TUNNELID_LEN; i++)
+ conn->tunnelid[i] = g_random_int_range ('a', 'z');
+ conn->tunnelid[TUNNELID_LEN - 1] = '\0';
+
+ url = conn->url;
+ /* get the port from the url */
+ gst_rtsp_url_get_port (url, &url_port);
+
+ if (conn->proxy_host) {
+ uri = g_strdup_printf ("http://%s:%d%s%s%s", url->host, url_port,
+ url->abspath, url->query ? "?" : "", url->query ? url->query : "");
+ hostparam = g_strdup_printf ("%s:%d", url->host, url_port);
+ ip = conn->proxy_host;
+ port = conn->proxy_port;
+ } else {
+ uri = g_strdup_printf ("%s%s%s", url->abspath, url->query ? "?" : "",
+ url->query ? url->query : "");
+ hostparam = NULL;
+ ip = conn->ip;
+ port = url_port;
+ }
+
+ /* create the GET request for the read connection */
+ GST_RTSP_CHECK (gst_rtsp_message_new_request (&msg, GST_RTSP_GET, uri),
+ no_message);
+ msg->type = GST_RTSP_MESSAGE_HTTP_REQUEST;
+
+ if (hostparam != NULL)
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_HOST, hostparam);
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_X_SESSIONCOOKIE,
+ conn->tunnelid);
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_ACCEPT,
+ "application/x-rtsp-tunnelled");
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CACHE_CONTROL, "no-cache");
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_PRAGMA, "no-cache");
+
+ /* we start by writing to this fd */
+ conn->writefd = &conn->fd0;
+
+ /* we need to temporarily set conn->tunneled to FALSE to prevent the HTTP
+ * request from being base64 encoded */
+ conn->tunneled = FALSE;
+ GST_RTSP_CHECK (gst_rtsp_connection_send (conn, msg, timeout), write_failed);
+ gst_rtsp_message_free (msg);
+ conn->tunneled = TRUE;
+
+ /* receive the response to the GET request */
+ /* we need to temporarily set manual_http to TRUE since
+ * gst_rtsp_connection_receive() will treat the HTTP response as a parsing
+ * failure otherwise */
+ old_http = conn->manual_http;
+ conn->manual_http = TRUE;
+ GST_RTSP_CHECK (gst_rtsp_connection_receive (conn, &response, timeout),
+ read_failed);
+ conn->manual_http = old_http;
+
+ if (response.type != GST_RTSP_MESSAGE_HTTP_RESPONSE ||
+ response.type_data.response.code != GST_RTSP_STS_OK)
+ goto wrong_result;
+
+ if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_X_SERVER_IP_ADDRESS,
+ &value, 0) == GST_RTSP_OK) {
+ if (conn->proxy_host) {
+ /* if we use a proxy we need to change the destination url */
+ g_free (url->host);
+ url->host = g_strdup (value);
+ g_free (hostparam);
+ hostparam = g_strdup_printf ("%s:%d", url->host, url_port);
+ } else {
+ /* and resolve the new ip address */
+ if (!(ip = do_resolve (value)))
+ goto not_resolved;
+ g_free (conn->ip);
+ conn->ip = ip;
+ }
+ }
+
+ /* connect to the host/port */
+ res = do_connect (ip, port, &conn->fd1, conn->fdset, timeout);
+ if (res != GST_RTSP_OK)
+ goto connect_failed;
+
+ /* this is now our writing socket */
+ conn->writefd = &conn->fd1;
+
+ /* create the POST request for the write connection */
+ GST_RTSP_CHECK (gst_rtsp_message_new_request (&msg, GST_RTSP_POST, uri),
+ no_message);
+ msg->type = GST_RTSP_MESSAGE_HTTP_REQUEST;
+
+ if (hostparam != NULL)
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_HOST, hostparam);
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_X_SESSIONCOOKIE,
+ conn->tunnelid);
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_ACCEPT,
+ "application/x-rtsp-tunnelled");
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CACHE_CONTROL, "no-cache");
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_PRAGMA, "no-cache");
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_EXPIRES,
+ "Sun, 9 Jan 1972 00:00:00 GMT");
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CONTENT_LENGTH, "32767");
+
+ /* we need to temporarily set conn->tunneled to FALSE to prevent the HTTP
+ * request from being base64 encoded */
+ conn->tunneled = FALSE;
+ GST_RTSP_CHECK (gst_rtsp_connection_send (conn, msg, timeout), write_failed);
+ gst_rtsp_message_free (msg);
+ conn->tunneled = TRUE;
+
+exit:
+ gst_rtsp_message_unset (&response);
+ g_free (hostparam);
+ g_free (uri);
+
+ return res;
+
+ /* ERRORS */
+no_message:
+ {
+ GST_ERROR ("failed to create request (%d)", res);
+ goto exit;
+ }
+write_failed:
+ {
+ GST_ERROR ("write failed (%d)", res);
+ gst_rtsp_message_free (msg);
+ conn->tunneled = TRUE;
+ goto exit;
+ }
+read_failed:
+ {
+ GST_ERROR ("read failed (%d)", res);
+ conn->manual_http = FALSE;
+ goto exit;
+ }
+wrong_result:
+ {
+ GST_ERROR ("got failure response %d %s", response.type_data.response.code,
+ response.type_data.response.reason);
+ res = GST_RTSP_ERROR;
+ goto exit;
+ }
+not_resolved:
+ {
+ GST_ERROR ("could not resolve %s", conn->ip);
+ res = GST_RTSP_ENET;
+ goto exit;
+ }
+connect_failed:
+ {
+ GST_ERROR ("failed to connect");
+ goto exit;
+ }
+}
+
+/**
+ * gst_rtsp_connection_connect:
+ * @conn: a #GstRTSPConnection
+ * @timeout: a #GTimeVal timeout
+ *
+ * Attempt to connect to the url of @conn made with
+ * gst_rtsp_connection_create(). If @timeout is #NULL this function can block
+ * forever. If @timeout contains a valid timeout, this function will return
+ * #GST_RTSP_ETIMEOUT after the timeout expired.
+ *
+ * This function can be cancelled with gst_rtsp_connection_flush().
+ *
+ * Returns: #GST_RTSP_OK when a connection could be made.
+ */
+GstRTSPResult
+gst_rtsp_connection_connect (GstRTSPConnection * conn, GTimeVal * timeout)
+{
+ GstRTSPResult res;
+ gchar *ip;
+ guint16 port;
+ GstRTSPUrl *url;
+
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn->url != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn->fd0.fd < 0, GST_RTSP_EINVAL);
+
+ url = conn->url;
+
+ if (conn->proxy_host && conn->tunneled) {
+ if (!(ip = do_resolve (conn->proxy_host))) {
+ GST_ERROR ("could not resolve %s", conn->proxy_host);
+ goto not_resolved;
+ }
+ port = conn->proxy_port;
+ g_free (conn->proxy_host);
+ conn->proxy_host = ip;
+ } else {
+ if (!(ip = do_resolve (url->host))) {
+ GST_ERROR ("could not resolve %s", url->host);
+ goto not_resolved;
+ }
+ /* get the port from the url */
+ gst_rtsp_url_get_port (url, &port);
+
+ g_free (conn->ip);
+ conn->ip = ip;
+ }
+
+ /* connect to the host/port */
+ res = do_connect (ip, port, &conn->fd0, conn->fdset, timeout);
+ if (res != GST_RTSP_OK)
+ goto connect_failed;
+
+ /* this is our read URL */
+ conn->readfd = &conn->fd0;
+
+ if (conn->tunneled) {
+ res = setup_tunneling (conn, timeout);
+ if (res != GST_RTSP_OK)
+ goto tunneling_failed;
+ } else {
+ conn->writefd = &conn->fd0;
+ }
+
+ return GST_RTSP_OK;
+
+not_resolved:
+ {
+ return GST_RTSP_ENET;
+ }
+connect_failed:
+ {
+ GST_ERROR ("failed to connect");
+ return res;
+ }
+tunneling_failed:
+ {
+ GST_ERROR ("failed to setup tunneling");
+ return res;
+ }
+}
+
+static void
+auth_digest_compute_hex_urp (const gchar * username,
+ const gchar * realm, const gchar * password, gchar hex_urp[33])
+{
+ GChecksum *md5_context = g_checksum_new (G_CHECKSUM_MD5);
+ const gchar *digest_string;
+
+ g_checksum_update (md5_context, (const guchar *) username, strlen (username));
+ g_checksum_update (md5_context, (const guchar *) ":", 1);
+ g_checksum_update (md5_context, (const guchar *) realm, strlen (realm));
+ g_checksum_update (md5_context, (const guchar *) ":", 1);
+ g_checksum_update (md5_context, (const guchar *) password, strlen (password));
+ digest_string = g_checksum_get_string (md5_context);
+
+ memset (hex_urp, 0, 33);
+ memcpy (hex_urp, digest_string, strlen (digest_string));
+
+ g_checksum_free (md5_context);
+}
+
+static void
+auth_digest_compute_response (const gchar * method,
+ const gchar * uri, const gchar * hex_a1, const gchar * nonce,
+ gchar response[33])
+{
+ char hex_a2[33] = { 0, };
+ GChecksum *md5_context = g_checksum_new (G_CHECKSUM_MD5);
+ const gchar *digest_string;
+
+ /* compute A2 */
+ g_checksum_update (md5_context, (const guchar *) method, strlen (method));
+ g_checksum_update (md5_context, (const guchar *) ":", 1);
+ g_checksum_update (md5_context, (const guchar *) uri, strlen (uri));
+ digest_string = g_checksum_get_string (md5_context);
+ memcpy (hex_a2, digest_string, strlen (digest_string));
+
+ /* compute KD */
+ g_checksum_reset (md5_context);
+ g_checksum_update (md5_context, (const guchar *) hex_a1, strlen (hex_a1));
+ g_checksum_update (md5_context, (const guchar *) ":", 1);
+ g_checksum_update (md5_context, (const guchar *) nonce, strlen (nonce));
+ g_checksum_update (md5_context, (const guchar *) ":", 1);
+
+ g_checksum_update (md5_context, (const guchar *) hex_a2, 32);
+ digest_string = g_checksum_get_string (md5_context);
+ memset (response, 0, 33);
+ memcpy (response, digest_string, strlen (digest_string));
+
+ g_checksum_free (md5_context);
+}
+
+static void
+add_auth_header (GstRTSPConnection * conn, GstRTSPMessage * message)
+{
+ switch (conn->auth_method) {
+ case GST_RTSP_AUTH_BASIC:{
+ gchar *user_pass;
+ gchar *user_pass64;
+ gchar *auth_string;
+
+ if (conn->username == NULL || conn->passwd == NULL)
+ break;
+
+ user_pass = g_strdup_printf ("%s:%s", conn->username, conn->passwd);
+ user_pass64 = g_base64_encode ((guchar *) user_pass, strlen (user_pass));
+ auth_string = g_strdup_printf ("Basic %s", user_pass64);
+
+ gst_rtsp_message_take_header (message, GST_RTSP_HDR_AUTHORIZATION,
+ auth_string);
+
+ g_free (user_pass);
+ g_free (user_pass64);
+ break;
+ }
+ case GST_RTSP_AUTH_DIGEST:{
+ gchar response[33], hex_urp[33];
+ gchar *auth_string, *auth_string2;
+ gchar *realm;
+ gchar *nonce;
+ gchar *opaque;
+ const gchar *uri;
+ const gchar *method;
+
+ /* we need to have some params set */
+ if (conn->auth_params == NULL || conn->username == NULL ||
+ conn->passwd == NULL)
+ break;
+
+ /* we need the realm and nonce */
+ realm = (gchar *) g_hash_table_lookup (conn->auth_params, "realm");
+ nonce = (gchar *) g_hash_table_lookup (conn->auth_params, "nonce");
+ if (realm == NULL || nonce == NULL)
+ break;
+
+ auth_digest_compute_hex_urp (conn->username, realm, conn->passwd,
+ hex_urp);
+
+ method = gst_rtsp_method_as_text (message->type_data.request.method);
+ uri = message->type_data.request.uri;
+
+ /* Assume no qop, algorithm=md5, stale=false */
+ /* For algorithm MD5, a1 = urp. */
+ auth_digest_compute_response (method, uri, hex_urp, nonce, response);
+ auth_string = g_strdup_printf ("Digest username=\"%s\", "
+ "realm=\"%s\", nonce=\"%s\", uri=\"%s\", response=\"%s\"",
+ conn->username, realm, nonce, uri, response);
+
+ opaque = (gchar *) g_hash_table_lookup (conn->auth_params, "opaque");
+ if (opaque) {
+ auth_string2 = g_strdup_printf ("%s, opaque=\"%s\"", auth_string,
+ opaque);
+ g_free (auth_string);
+ auth_string = auth_string2;
+ }
+ gst_rtsp_message_take_header (message, GST_RTSP_HDR_AUTHORIZATION,
+ auth_string);
+ break;
+ }
+ default:
+ /* Nothing to do */
+ break;
+ }
+}
+
+static void
+gen_date_string (gchar * date_string, guint len)
+{
+ static const char wkdays[7][4] =
+ { "Sun", "Mon", "Tue", "Wed", "Thu", "Fri", "Sat" };
+ static const char months[12][4] =
+ { "Jan", "Feb", "Mar", "Apr", "May", "Jun", "Jul", "Aug", "Sep", "Oct",
+ "Nov", "Dec"
+ };
+ struct tm tm;
+ time_t t;
+
+ time (&t);
+
+#ifdef HAVE_GMTIME_R
+ gmtime_r (&t, &tm);
+#else
+ tm = *gmtime (&t);
+#endif
+
+ g_snprintf (date_string, len, "%s, %02d %s %04d %02d:%02d:%02d GMT",
+ wkdays[tm.tm_wday], tm.tm_mday, months[tm.tm_mon], tm.tm_year + 1900,
+ tm.tm_hour, tm.tm_min, tm.tm_sec);
+}
+
+static GstRTSPResult
+write_bytes (gint fd, const guint8 * buffer, guint * idx, guint size)
+{
+ guint left;
+
+ if (G_UNLIKELY (*idx > size))
+ return GST_RTSP_ERROR;
+
+ left = size - *idx;
+
+ while (left) {
+ gint r;
+
+ r = WRITE_SOCKET (fd, &buffer[*idx], left);
+ if (G_UNLIKELY (r == 0)) {
+ return GST_RTSP_EINTR;
+ } else if (G_UNLIKELY (r < 0)) {
+ if (ERRNO_IS_EAGAIN)
+ return GST_RTSP_EINTR;
+ if (!ERRNO_IS_EINTR)
+ return GST_RTSP_ESYS;
+ } else {
+ left -= r;
+ *idx += r;
+ }
+ }
+ return GST_RTSP_OK;
+}
+
+static gint
+fill_raw_bytes (GstRTSPConnection * conn, guint8 * buffer, guint size)
+{
+ gint out = 0;
+
+ if (G_UNLIKELY (conn->initial_buffer != NULL)) {
+ gsize left = strlen (&conn->initial_buffer[conn->initial_buffer_offset]);
+
+ out = MIN (left, size);
+ memcpy (buffer, &conn->initial_buffer[conn->initial_buffer_offset], out);
+
+ if (left == (gsize) out) {
+ g_free (conn->initial_buffer);
+ conn->initial_buffer = NULL;
+ conn->initial_buffer_offset = 0;
+ } else
+ conn->initial_buffer_offset += out;
+ }
+
+ if (G_LIKELY (size > (guint) out)) {
+ gint r;
+
+ r = READ_SOCKET (conn->readfd->fd, &buffer[out], size - out);
+ if (r <= 0) {
+ if (out == 0)
+ out = r;
+ } else
+ out += r;
+ }
+
+ return out;
+}
+
+static gint
+fill_bytes (GstRTSPConnection * conn, guint8 * buffer, guint size)
+{
+ DecodeCtx *ctx = conn->ctxp;
+ gint out = 0;
+
+ if (ctx) {
+ while (size > 0) {
+ guint8 in[sizeof (ctx->out) * 4 / 3];
+ gint r;
+
+ while (size > 0 && ctx->cout < ctx->coutl) {
+ /* we have some leftover bytes */
+ *buffer++ = ctx->out[ctx->cout++];
+ size--;
+ out++;
+ }
+
+ /* got what we needed? */
+ if (size == 0)
+ break;
+
+ /* try to read more bytes */
+ r = fill_raw_bytes (conn, in, sizeof (in));
+ if (r <= 0) {
+ if (out == 0)
+ out = r;
+ break;
+ }
+
+ ctx->cout = 0;
+ ctx->coutl =
+ g_base64_decode_step ((gchar *) in, r, ctx->out, &ctx->state,
+ &ctx->save);
+ }
+ } else {
+ out = fill_raw_bytes (conn, buffer, size);
+ }
+
+ return out;
+}
+
+static GstRTSPResult
+read_bytes (GstRTSPConnection * conn, guint8 * buffer, guint * idx, guint size)
+{
+ guint left;
+
+ if (G_UNLIKELY (*idx > size))
+ return GST_RTSP_ERROR;
+
+ left = size - *idx;
+
+ while (left) {
+ gint r;
+
+ r = fill_bytes (conn, &buffer[*idx], left);
+ if (G_UNLIKELY (r == 0)) {
+ return GST_RTSP_EEOF;
+ } else if (G_UNLIKELY (r < 0)) {
+ if (ERRNO_IS_EAGAIN)
+ return GST_RTSP_EINTR;
+ if (!ERRNO_IS_EINTR)
+ return GST_RTSP_ESYS;
+ } else {
+ left -= r;
+ *idx += r;
+ }
+ }
+ return GST_RTSP_OK;
+}
+
+/* The code below tries to handle clients using \r, \n or \r\n to indicate the
+ * end of a line. It even does its best to handle clients which mix them (even
+ * though this is a really stupid idea (tm).) It also handles Line White Space
+ * (LWS), where a line end followed by whitespace is considered LWS. This is
+ * the method used in RTSP (and HTTP) to break long lines.
+ */
+static GstRTSPResult
+read_line (GstRTSPConnection * conn, guint8 * buffer, guint * idx, guint size)
+{
+ while (TRUE) {
+ guint8 c;
+ gint r;
+
+ if (conn->read_ahead == READ_AHEAD_EOH) {
+ /* the last call to read_line() already determined that we have reached
+ * the end of the headers, so convey that information now */
+ conn->read_ahead = 0;
+ break;
+ } else if (conn->read_ahead == READ_AHEAD_CRLF) {
+ /* the last call to read_line() left off after having read \r\n */
+ c = '\n';
+ } else if (conn->read_ahead == READ_AHEAD_CRLFCR) {
+ /* the last call to read_line() left off after having read \r\n\r */
+ c = '\r';
+ } else if (conn->read_ahead != 0) {
+ /* the last call to read_line() left us with a character to start with */
+ c = (guint8) conn->read_ahead;
+ conn->read_ahead = 0;
+ } else {
+ /* read the next character */
+ r = fill_bytes (conn, &c, 1);
+ if (G_UNLIKELY (r == 0)) {
+ return GST_RTSP_EEOF;
+ } else if (G_UNLIKELY (r < 0)) {
+ if (ERRNO_IS_EAGAIN)
+ return GST_RTSP_EINTR;
+ if (!ERRNO_IS_EINTR)
+ return GST_RTSP_ESYS;
+ continue;
+ }
+ }
+
+ /* special treatment of line endings */
+ if (c == '\r' || c == '\n') {
+ guint8 read_ahead;
+
+ retry:
+ /* need to read ahead one more character to know what to do... */
+ r = fill_bytes (conn, &read_ahead, 1);
+ if (G_UNLIKELY (r == 0)) {
+ return GST_RTSP_EEOF;
+ } else if (G_UNLIKELY (r < 0)) {
+ if (ERRNO_IS_EAGAIN) {
+ /* remember the original character we read and try again next time */
+ if (conn->read_ahead == 0)
+ conn->read_ahead = c;
+ return GST_RTSP_EINTR;
+ }
+ if (!ERRNO_IS_EINTR)
+ return GST_RTSP_ESYS;
+ goto retry;
+ }
+
+ if (read_ahead == ' ' || read_ahead == '\t') {
+ if (conn->read_ahead == READ_AHEAD_CRLFCR) {
+ /* got \r\n\r followed by whitespace, treat it as a normal line
+ * followed by one starting with LWS */
+ conn->read_ahead = read_ahead;
+ break;
+ } else {
+ /* got LWS, change the line ending to a space and continue */
+ c = ' ';
+ conn->read_ahead = read_ahead;
+ }
+ } else if (conn->read_ahead == READ_AHEAD_CRLFCR) {
+ if (read_ahead == '\r' || read_ahead == '\n') {
+ /* got \r\n\r\r or \r\n\r\n, treat it as the end of the headers */
+ conn->read_ahead = READ_AHEAD_EOH;
+ break;
+ } else {
+ /* got \r\n\r followed by something else, this is not really
+ * supported since we have probably just eaten the first character
+ * of the body or the next message, so just ignore the second \r
+ * and live with it... */
+ conn->read_ahead = read_ahead;
+ break;
+ }
+ } else if (conn->read_ahead == READ_AHEAD_CRLF) {
+ if (read_ahead == '\r') {
+ /* got \r\n\r so far, need one more character... */
+ conn->read_ahead = READ_AHEAD_CRLFCR;
+ goto retry;
+ } else if (read_ahead == '\n') {
+ /* got \r\n\n, treat it as the end of the headers */
+ conn->read_ahead = READ_AHEAD_EOH;
+ break;
+ } else {
+ /* found the end of a line, keep read_ahead for the next line */
+ conn->read_ahead = read_ahead;
+ break;
+ }
+ } else if (c == read_ahead) {
+ /* got double \r or \n, treat it as the end of the headers */
+ conn->read_ahead = READ_AHEAD_EOH;
+ break;
+ } else if (c == '\r' && read_ahead == '\n') {
+ /* got \r\n so far, still need more to know what to do... */
+ conn->read_ahead = READ_AHEAD_CRLF;
+ goto retry;
+ } else {
+ /* found the end of a line, keep read_ahead for the next line */
+ conn->read_ahead = read_ahead;
+ break;
+ }
+ }
+
+ if (G_LIKELY (*idx < size - 1))
+ buffer[(*idx)++] = c;
+ }
+ buffer[*idx] = '\0';
+
+ return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_connection_write:
+ * @conn: a #GstRTSPConnection
+ * @data: the data to write
+ * @size: the size of @data
+ * @timeout: a timeout value or #NULL
+ *
+ * Attempt to write @size bytes of @data to the connected @conn, blocking up to
+ * the specified @timeout. @timeout can be #NULL, in which case this function
+ * might block forever.
+ *
+ * This function can be cancelled with gst_rtsp_connection_flush().
+ *
+ * Returns: #GST_RTSP_OK on success.
+ */
+GstRTSPResult
+gst_rtsp_connection_write (GstRTSPConnection * conn, const guint8 * data,
+ guint size, GTimeVal * timeout)
+{
+ guint offset;
+ gint retval;
+ GstClockTime to;
+ GstRTSPResult res;
+
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (data != NULL || size == 0, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn->writefd != NULL, GST_RTSP_EINVAL);
+
+ gst_poll_set_controllable (conn->fdset, TRUE);
+ gst_poll_fd_ctl_write (conn->fdset, conn->writefd, TRUE);
+ gst_poll_fd_ctl_read (conn->fdset, conn->readfd, FALSE);
+ /* clear all previous poll results */
+ gst_poll_fd_ignored (conn->fdset, conn->writefd);
+ gst_poll_fd_ignored (conn->fdset, conn->readfd);
+
+ to = timeout ? GST_TIMEVAL_TO_TIME (*timeout) : GST_CLOCK_TIME_NONE;
+
+ offset = 0;
+
+ while (TRUE) {
+ /* try to write */
+ res = write_bytes (conn->writefd->fd, data, &offset, size);
+ if (G_LIKELY (res == GST_RTSP_OK))
+ break;
+ if (G_UNLIKELY (res != GST_RTSP_EINTR))
+ goto write_error;
+
+ /* not all is written, wait until we can write more */
+ do {
+ retval = gst_poll_wait (conn->fdset, to);
+ } while (retval == -1 && (errno == EINTR || errno == EAGAIN));
+
+ if (G_UNLIKELY (retval == 0))
+ goto timeout;
+
+ if (G_UNLIKELY (retval == -1)) {
+ if (errno == EBUSY)
+ goto stopped;
+ else
+ goto select_error;
+ }
+
+ /* could also be an error with read socket */
+ if (gst_poll_fd_has_error (conn->fdset, conn->readfd))
+ goto socket_error;
+ }
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+timeout:
+ {
+ return GST_RTSP_ETIMEOUT;
+ }
+select_error:
+ {
+ return GST_RTSP_ESYS;
+ }
+stopped:
+ {
+ return GST_RTSP_EINTR;
+ }
+socket_error:
+ {
+ return GST_RTSP_ENET;
+ }
+write_error:
+ {
+ return res;
+ }
+}
+
+static GString *
+message_to_string (GstRTSPConnection * conn, GstRTSPMessage * message)
+{
+ GString *str = NULL;
+
+ str = g_string_new ("");
+
+ switch (message->type) {
+ case GST_RTSP_MESSAGE_REQUEST:
+ /* create request string, add CSeq */
+ g_string_append_printf (str, "%s %s RTSP/1.0\r\n"
+ "CSeq: %d\r\n",
+ gst_rtsp_method_as_text (message->type_data.request.method),
+ message->type_data.request.uri, conn->cseq++);
+ /* add session id if we have one */
+ if (conn->session_id[0] != '\0') {
+ gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
+ gst_rtsp_message_add_header (message, GST_RTSP_HDR_SESSION,
+ conn->session_id);
+ }
+ /* add any authentication headers */
+ add_auth_header (conn, message);
+ break;
+ case GST_RTSP_MESSAGE_RESPONSE:
+ /* create response string */
+ g_string_append_printf (str, "RTSP/1.0 %d %s\r\n",
+ message->type_data.response.code, message->type_data.response.reason);
+ break;
+ case GST_RTSP_MESSAGE_HTTP_REQUEST:
+ /* create request string */
+ g_string_append_printf (str, "%s %s HTTP/%s\r\n",
+ gst_rtsp_method_as_text (message->type_data.request.method),
+ message->type_data.request.uri,
+ gst_rtsp_version_as_text (message->type_data.request.version));
+ /* add any authentication headers */
+ add_auth_header (conn, message);
+ break;
+ case GST_RTSP_MESSAGE_HTTP_RESPONSE:
+ /* create response string */
+ g_string_append_printf (str, "HTTP/%s %d %s\r\n",
+ gst_rtsp_version_as_text (message->type_data.request.version),
+ message->type_data.response.code, message->type_data.response.reason);
+ break;
+ case GST_RTSP_MESSAGE_DATA:
+ {
+ guint8 data_header[4];
+
+ /* prepare data header */
+ data_header[0] = '$';
+ data_header[1] = message->type_data.data.channel;
+ data_header[2] = (message->body_size >> 8) & 0xff;
+ data_header[3] = message->body_size & 0xff;
+
+ /* create string with header and data */
+ str = g_string_append_len (str, (gchar *) data_header, 4);
+ str =
+ g_string_append_len (str, (gchar *) message->body,
+ message->body_size);
+ break;
+ }
+ default:
+ g_string_free (str, TRUE);
+ g_return_val_if_reached (NULL);
+ break;
+ }
+
+ /* append headers and body */
+ if (message->type != GST_RTSP_MESSAGE_DATA) {
+ gchar date_string[100];
+
+ gen_date_string (date_string, sizeof (date_string));
+
+ /* add date header */
+ gst_rtsp_message_remove_header (message, GST_RTSP_HDR_DATE, -1);
+ gst_rtsp_message_add_header (message, GST_RTSP_HDR_DATE, date_string);
+
+ /* append headers */
+ gst_rtsp_message_append_headers (message, str);
+
+ /* append Content-Length and body if needed */
+ if (message->body != NULL && message->body_size > 0) {
+ gchar *len;
+
+ len = g_strdup_printf ("%d", message->body_size);
+ g_string_append_printf (str, "%s: %s\r\n",
+ gst_rtsp_header_as_text (GST_RTSP_HDR_CONTENT_LENGTH), len);
+ g_free (len);
+ /* header ends here */
+ g_string_append (str, "\r\n");
+ str =
+ g_string_append_len (str, (gchar *) message->body,
+ message->body_size);
+ } else {
+ /* just end headers */
+ g_string_append (str, "\r\n");
+ }
+ }
+
+ return str;
+}
+
+/**
+ * gst_rtsp_connection_send:
+ * @conn: a #GstRTSPConnection
+ * @message: the message to send
+ * @timeout: a timeout value or #NULL
+ *
+ * Attempt to send @message to the connected @conn, blocking up to
+ * the specified @timeout. @timeout can be #NULL, in which case this function
+ * might block forever.
+ *
+ * This function can be cancelled with gst_rtsp_connection_flush().
+ *
+ * Returns: #GST_RTSP_OK on success.
+ */
+GstRTSPResult
+gst_rtsp_connection_send (GstRTSPConnection * conn, GstRTSPMessage * message,
+ GTimeVal * timeout)
+{
+ GString *string = NULL;
+ GstRTSPResult res;
+ gchar *str;
+ gsize len;
+
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
+
+ if (G_UNLIKELY (!(string = message_to_string (conn, message))))
+ goto no_message;
+
+ if (conn->tunneled) {
+ str = g_base64_encode ((const guchar *) string->str, string->len);
+ g_string_free (string, TRUE);
+ len = strlen (str);
+ } else {
+ str = string->str;
+ len = string->len;
+ g_string_free (string, FALSE);
+ }
+
+ /* write request */
+ res = gst_rtsp_connection_write (conn, (guint8 *) str, len, timeout);
+
+ g_free (str);
+
+ return res;
+
+no_message:
+ {
+ g_warning ("Wrong message");
+ return GST_RTSP_EINVAL;
+ }
+}
+
+static GstRTSPResult
+parse_string (gchar * dest, gint size, gchar ** src)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ gint idx;
+
+ idx = 0;
+ /* skip spaces */
+ while (g_ascii_isspace (**src))
+ (*src)++;
+
+ while (!g_ascii_isspace (**src) && **src != '\0') {
+ if (idx < size - 1)
+ dest[idx++] = **src;
+ else
+ res = GST_RTSP_EPARSE;
+ (*src)++;
+ }
+ if (size > 0)
+ dest[idx] = '\0';
+
+ return res;
+}
+
+static GstRTSPResult
+parse_protocol_version (gchar * protocol, GstRTSPMsgType * type,
+ GstRTSPVersion * version)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ gchar *ver;
+
+ if (G_LIKELY ((ver = strchr (protocol, '/')) != NULL)) {
+ guint major;
+ guint minor;
+ gchar dummychar;
+
+ *ver++ = '\0';
+
+ /* the version number must be formatted as X.Y with nothing following */
+ if (sscanf (ver, "%u.%u%c", &major, &minor, &dummychar) != 2)
+ res = GST_RTSP_EPARSE;
+
+ if (g_ascii_strcasecmp (protocol, "RTSP") == 0) {
+ if (major != 1 || minor != 0) {
+ *version = GST_RTSP_VERSION_INVALID;
+ res = GST_RTSP_ERROR;
+ }
+ } else if (g_ascii_strcasecmp (protocol, "HTTP") == 0) {
+ if (*type == GST_RTSP_MESSAGE_REQUEST)
+ *type = GST_RTSP_MESSAGE_HTTP_REQUEST;
+ else if (*type == GST_RTSP_MESSAGE_RESPONSE)
+ *type = GST_RTSP_MESSAGE_HTTP_RESPONSE;
+
+ if (major == 1 && minor == 1) {
+ *version = GST_RTSP_VERSION_1_1;
+ } else if (major != 1 || minor != 0) {
+ *version = GST_RTSP_VERSION_INVALID;
+ res = GST_RTSP_ERROR;
+ }
+ } else
+ res = GST_RTSP_EPARSE;
+ } else
+ res = GST_RTSP_EPARSE;
+
+ return res;
+}
+
+static GstRTSPResult
+parse_response_status (guint8 * buffer, GstRTSPMessage * msg)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPResult res2;
+ gchar versionstr[20];
+ gchar codestr[4];
+ gint code;
+ gchar *bptr;
+
+ bptr = (gchar *) buffer;
+
+ if (parse_string (versionstr, sizeof (versionstr), &bptr) != GST_RTSP_OK)
+ res = GST_RTSP_EPARSE;
+
+ if (parse_string (codestr, sizeof (codestr), &bptr) != GST_RTSP_OK)
+ res = GST_RTSP_EPARSE;
+ code = atoi (codestr);
+ if (G_UNLIKELY (*codestr == '\0' || code < 0 || code >= 600))
+ res = GST_RTSP_EPARSE;
+
+ while (g_ascii_isspace (*bptr))
+ bptr++;
+
+ if (G_UNLIKELY (gst_rtsp_message_init_response (msg, code, bptr,
+ NULL) != GST_RTSP_OK))
+ res = GST_RTSP_EPARSE;
+
+ res2 = parse_protocol_version (versionstr, &msg->type,
+ &msg->type_data.response.version);
+ if (G_LIKELY (res == GST_RTSP_OK))
+ res = res2;
+
+ return res;
+}
+
+static GstRTSPResult
+parse_request_line (guint8 * buffer, GstRTSPMessage * msg)
+{
+ GstRTSPResult res = GST_RTSP_OK;
+ GstRTSPResult res2;
+ gchar versionstr[20];
+ gchar methodstr[20];
+ gchar urlstr[4096];
+ gchar *bptr;
+ GstRTSPMethod method;
+
+ bptr = (gchar *) buffer;
+
+ if (parse_string (methodstr, sizeof (methodstr), &bptr) != GST_RTSP_OK)
+ res = GST_RTSP_EPARSE;
+ method = gst_rtsp_find_method (methodstr);
+
+ if (parse_string (urlstr, sizeof (urlstr), &bptr) != GST_RTSP_OK)
+ res = GST_RTSP_EPARSE;
+ if (G_UNLIKELY (*urlstr == '\0'))
+ res = GST_RTSP_EPARSE;
+
+ if (parse_string (versionstr, sizeof (versionstr), &bptr) != GST_RTSP_OK)
+ res = GST_RTSP_EPARSE;
+
+ if (G_UNLIKELY (*bptr != '\0'))
+ res = GST_RTSP_EPARSE;
+
+ if (G_UNLIKELY (gst_rtsp_message_init_request (msg, method,
+ urlstr) != GST_RTSP_OK))
+ res = GST_RTSP_EPARSE;
+
+ res2 = parse_protocol_version (versionstr, &msg->type,
+ &msg->type_data.request.version);
+ if (G_LIKELY (res == GST_RTSP_OK))
+ res = res2;
+
+ if (G_LIKELY (msg->type == GST_RTSP_MESSAGE_REQUEST)) {
+ /* GET and POST are not allowed as RTSP methods */
+ if (msg->type_data.request.method == GST_RTSP_GET ||
+ msg->type_data.request.method == GST_RTSP_POST) {
+ msg->type_data.request.method = GST_RTSP_INVALID;
+ if (res == GST_RTSP_OK)
+ res = GST_RTSP_ERROR;
+ }
+ } else if (msg->type == GST_RTSP_MESSAGE_HTTP_REQUEST) {
+ /* only GET and POST are allowed as HTTP methods */
+ if (msg->type_data.request.method != GST_RTSP_GET &&
+ msg->type_data.request.method != GST_RTSP_POST) {
+ msg->type_data.request.method = GST_RTSP_INVALID;
+ if (res == GST_RTSP_OK)
+ res = GST_RTSP_ERROR;
+ }
+ }
+
+ return res;
+}
+
+/* parsing lines means reading a Key: Value pair */
+static GstRTSPResult
+parse_line (guint8 * buffer, GstRTSPMessage * msg)
+{
+ GstRTSPHeaderField field;
+ gchar *line = (gchar *) buffer;
+ gchar *value;
+
+ if ((value = strchr (line, ':')) == NULL || value == line)
+ goto parse_error;
+
+ /* trim space before the colon */
+ if (value[-1] == ' ')
+ value[-1] = '\0';
+
+ /* replace the colon with a NUL */
+ *value++ = '\0';
+
+ /* find the header */
+ field = gst_rtsp_find_header_field (line);
+ if (field == GST_RTSP_HDR_INVALID)
+ goto done;
+
+ /* split up the value in multiple key:value pairs if it contains comma(s) */
+ while (*value != '\0') {
+ gchar *next_value;
+ gchar *comma = NULL;
+ gboolean quoted = FALSE;
+ guint comment = 0;
+
+ /* trim leading space */
+ if (*value == ' ')
+ value++;
+
+ /* for headers which may not appear multiple times, and thus may not
+ * contain multiple values on the same line, we can short-circuit the loop
+ * below and the entire value results in just one key:value pair*/
+ if (!gst_rtsp_header_allow_multiple (field))
+ next_value = value + strlen (value);
+ else
+ next_value = value;
+
+ /* find the next value, taking special care of quotes and comments */
+ while (*next_value != '\0') {
+ if ((quoted || comment != 0) && *next_value == '\\' &&
+ next_value[1] != '\0')
+ next_value++;
+ else if (comment == 0 && *next_value == '"')
+ quoted = !quoted;
+ else if (!quoted && *next_value == '(')
+ comment++;
+ else if (comment != 0 && *next_value == ')')
+ comment--;
+ else if (!quoted && comment == 0) {
+ /* To quote RFC 2068: "User agents MUST take special care in parsing
+ * the WWW-Authenticate field value if it contains more than one
+ * challenge, or if more than one WWW-Authenticate header field is
+ * provided, since the contents of a challenge may itself contain a
+ * comma-separated list of authentication parameters."
+ *
+ * What this means is that we cannot just look for an unquoted comma
+ * when looking for multiple values in Proxy-Authenticate and
+ * WWW-Authenticate headers. Instead we need to look for the sequence
+ * "comma [space] token space token" before we can split after the
+ * comma...
+ */
+ if (field == GST_RTSP_HDR_PROXY_AUTHENTICATE ||
+ field == GST_RTSP_HDR_WWW_AUTHENTICATE) {
+ if (*next_value == ',') {
+ if (next_value[1] == ' ') {
+ /* skip any space following the comma so we do not mistake it for
+ * separating between two tokens */
+ next_value++;
+ }
+ comma = next_value;
+ } else if (*next_value == ' ' && next_value[1] != ',' &&
+ next_value[1] != '=' && comma != NULL) {
+ next_value = comma;
+ comma = NULL;
+ break;
+ }
+ } else if (*next_value == ',')
+ break;
+ }
+
+ next_value++;
+ }
+
+ /* trim space */
+ if (value != next_value && next_value[-1] == ' ')
+ next_value[-1] = '\0';
+
+ if (*next_value != '\0')
+ *next_value++ = '\0';
+
+ /* add the key:value pair */
+ if (*value != '\0')
+ gst_rtsp_message_add_header (msg, field, value);
+
+ value = next_value;
+ }
+
+done:
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+parse_error:
+ {
+ return GST_RTSP_EPARSE;
+ }
+}
+
+/* convert all consecutive whitespace to a single space */
+static void
+normalize_line (guint8 * buffer)
+{
+ while (*buffer) {
+ if (g_ascii_isspace (*buffer)) {
+ guint8 *tmp;
+
+ *buffer++ = ' ';
+ for (tmp = buffer; g_ascii_isspace (*tmp); tmp++) {
+ }
+ if (buffer != tmp)
+ memmove (buffer, tmp, strlen ((gchar *) tmp) + 1);
+ } else {
+ buffer++;
+ }
+ }
+}
+
+/* returns:
+ * GST_RTSP_OK when a complete message was read.
+ * GST_RTSP_EEOF: when the read socket is closed
+ * GST_RTSP_EINTR: when more data is needed.
+ * GST_RTSP_..: some other error occured.
+ */
+static GstRTSPResult
+build_next (GstRTSPBuilder * builder, GstRTSPMessage * message,
+ GstRTSPConnection * conn)
+{
+ GstRTSPResult res;
+
+ while (TRUE) {
+ switch (builder->state) {
+ case STATE_START:
+ {
+ guint8 c;
+
+ builder->offset = 0;
+ res =
+ read_bytes (conn, (guint8 *) builder->buffer, &builder->offset, 1);
+ if (res != GST_RTSP_OK)
+ goto done;
+
+ c = builder->buffer[0];
+
+ /* we have 1 bytes now and we can see if this is a data message or
+ * not */
+ if (c == '$') {
+ /* data message, prepare for the header */
+ builder->state = STATE_DATA_HEADER;
+ } else if (c == '\n' || c == '\r') {
+ /* skip \n and \r */
+ builder->offset = 0;
+ } else {
+ builder->line = 0;
+ builder->state = STATE_READ_LINES;
+ }
+ break;
+ }
+ case STATE_DATA_HEADER:
+ {
+ res =
+ read_bytes (conn, (guint8 *) builder->buffer, &builder->offset, 4);
+ if (res != GST_RTSP_OK)
+ goto done;
+
+ gst_rtsp_message_init_data (message, builder->buffer[1]);
+
+ builder->body_len = (builder->buffer[2] << 8) | builder->buffer[3];
+ builder->body_data = g_malloc (builder->body_len + 1);
+ builder->body_data[builder->body_len] = '\0';
+ builder->offset = 0;
+ builder->state = STATE_DATA_BODY;
+ break;
+ }
+ case STATE_DATA_BODY:
+ {
+ res =
+ read_bytes (conn, builder->body_data, &builder->offset,
+ builder->body_len);
+ if (res != GST_RTSP_OK)
+ goto done;
+
+ /* we have the complete body now, store in the message adjusting the
+ * length to include the traling '\0' */
+ gst_rtsp_message_take_body (message,
+ (guint8 *) builder->body_data, builder->body_len + 1);
+ builder->body_data = NULL;
+ builder->body_len = 0;
+
+ builder->state = STATE_END;
+ break;
+ }
+ case STATE_READ_LINES:
+ {
+ res = read_line (conn, builder->buffer, &builder->offset,
+ sizeof (builder->buffer));
+ if (res != GST_RTSP_OK)
+ goto done;
+
+ /* we have a regular response */
+ if (builder->buffer[0] == '\0') {
+ gchar *hdrval;
+
+ /* empty line, end of message header */
+ /* see if there is a Content-Length header, but ignore it if this
+ * is a POST request with an x-sessioncookie header */
+ if (gst_rtsp_message_get_header (message,
+ GST_RTSP_HDR_CONTENT_LENGTH, &hdrval, 0) == GST_RTSP_OK &&
+ (message->type != GST_RTSP_MESSAGE_HTTP_REQUEST ||
+ message->type_data.request.method != GST_RTSP_POST ||
+ gst_rtsp_message_get_header (message,
+ GST_RTSP_HDR_X_SESSIONCOOKIE, NULL, 0) != GST_RTSP_OK)) {
+ /* there is, prepare to read the body */
+ builder->body_len = atol (hdrval);
+ builder->body_data = g_try_malloc (builder->body_len + 1);
+ /* we can't do much here, we need the length to know how many bytes
+ * we need to read next and when allocation fails, something is
+ * probably wrong with the length. */
+ if (builder->body_data == NULL)
+ goto invalid_body_len;
+
+ builder->body_data[builder->body_len] = '\0';
+ builder->offset = 0;
+ builder->state = STATE_DATA_BODY;
+ } else {
+ builder->state = STATE_END;
+ }
+ break;
+ }
+
+ /* we have a line */
+ normalize_line (builder->buffer);
+ if (builder->line == 0) {
+ /* first line, check for response status */
+ if (memcmp (builder->buffer, "RTSP", 4) == 0 ||
+ memcmp (builder->buffer, "HTTP", 4) == 0) {
+ builder->status = parse_response_status (builder->buffer, message);
+ } else {
+ builder->status = parse_request_line (builder->buffer, message);
+ }
+ } else {
+ /* else just parse the line */
+ res = parse_line (builder->buffer, message);
+ if (res != GST_RTSP_OK)
+ builder->status = res;
+ }
+ builder->line++;
+ builder->offset = 0;
+ break;
+ }
+ case STATE_END:
+ {
+ gchar *session_cookie;
+ gchar *session_id;
+
+ if (message->type == GST_RTSP_MESSAGE_DATA) {
+ /* data messages don't have headers */
+ res = GST_RTSP_OK;
+ goto done;
+ }
+
+ /* save the tunnel session in the connection */
+ if (message->type == GST_RTSP_MESSAGE_HTTP_REQUEST &&
+ !conn->manual_http &&
+ conn->tstate == TUNNEL_STATE_NONE &&
+ gst_rtsp_message_get_header (message, GST_RTSP_HDR_X_SESSIONCOOKIE,
+ &session_cookie, 0) == GST_RTSP_OK) {
+ strncpy (conn->tunnelid, session_cookie, TUNNELID_LEN);
+ conn->tunnelid[TUNNELID_LEN - 1] = '\0';
+ conn->tunneled = TRUE;
+ }
+
+ /* save session id in the connection for further use */
+ if (message->type == GST_RTSP_MESSAGE_RESPONSE &&
+ gst_rtsp_message_get_header (message, GST_RTSP_HDR_SESSION,
+ &session_id, 0) == GST_RTSP_OK) {
+ gint maxlen, i;
+
+ maxlen = sizeof (conn->session_id) - 1;
+ /* the sessionid can have attributes marked with ;
+ * Make sure we strip them */
+ for (i = 0; session_id[i] != '\0'; i++) {
+ if (session_id[i] == ';') {
+ maxlen = i;
+ /* parse timeout */
+ do {
+ i++;
+ } while (g_ascii_isspace (session_id[i]));
+ if (g_str_has_prefix (&session_id[i], "timeout=")) {
+ gint to;
+
+ /* if we parsed something valid, configure */
+ if ((to = atoi (&session_id[i + 8])) > 0)
+ conn->timeout = to;
+ }
+ break;
+ }
+ }
+
+ /* make sure to not overflow */
+ strncpy (conn->session_id, session_id, maxlen);
+ conn->session_id[maxlen] = '\0';
+ }
+ res = builder->status;
+ goto done;
+ }
+ default:
+ res = GST_RTSP_ERROR;
+ break;
+ }
+ }
+done:
+ return res;
+
+ /* ERRORS */
+invalid_body_len:
+ {
+ GST_DEBUG ("could not allocate body");
+ return GST_RTSP_ERROR;
+ }
+}
+
+/**
+ * gst_rtsp_connection_read:
+ * @conn: a #GstRTSPConnection
+ * @data: the data to read
+ * @size: the size of @data
+ * @timeout: a timeout value or #NULL
+ *
+ * Attempt to read @size bytes into @data from the connected @conn, blocking up to
+ * the specified @timeout. @timeout can be #NULL, in which case this function
+ * might block forever.
+ *
+ * This function can be cancelled with gst_rtsp_connection_flush().
+ *
+ * Returns: #GST_RTSP_OK on success.
+ */
+GstRTSPResult
+gst_rtsp_connection_read (GstRTSPConnection * conn, guint8 * data, guint size,
+ GTimeVal * timeout)
+{
+ guint offset;
+ gint retval;
+ GstClockTime to;
+ GstRTSPResult res;
+
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (data != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn->readfd != NULL, GST_RTSP_EINVAL);
+
+ if (G_UNLIKELY (size == 0))
+ return GST_RTSP_OK;
+
+ offset = 0;
+
+ /* configure timeout if any */
+ to = timeout ? GST_TIMEVAL_TO_TIME (*timeout) : GST_CLOCK_TIME_NONE;
+
+ gst_poll_set_controllable (conn->fdset, TRUE);
+ gst_poll_fd_ctl_write (conn->fdset, conn->writefd, FALSE);
+ gst_poll_fd_ctl_read (conn->fdset, conn->readfd, TRUE);
+
+ while (TRUE) {
+ res = read_bytes (conn, data, &offset, size);
+ if (G_UNLIKELY (res == GST_RTSP_EEOF))
+ goto eof;
+ if (G_LIKELY (res == GST_RTSP_OK))
+ break;
+ if (G_UNLIKELY (res != GST_RTSP_EINTR))
+ goto read_error;
+
+ do {
+ retval = gst_poll_wait (conn->fdset, to);
+ } while (retval == -1 && (errno == EINTR || errno == EAGAIN));
+
+ /* check for timeout */
+ if (G_UNLIKELY (retval == 0))
+ goto select_timeout;
+
+ if (G_UNLIKELY (retval == -1)) {
+ if (errno == EBUSY)
+ goto stopped;
+ else
+ goto select_error;
+ }
+
+ /* could also be an error with write socket */
+ if (gst_poll_fd_has_error (conn->fdset, conn->writefd))
+ goto socket_error;
+
+ gst_poll_set_controllable (conn->fdset, FALSE);
+ }
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+select_error:
+ {
+ return GST_RTSP_ESYS;
+ }
+select_timeout:
+ {
+ return GST_RTSP_ETIMEOUT;
+ }
+stopped:
+ {
+ return GST_RTSP_EINTR;
+ }
+eof:
+ {
+ return GST_RTSP_EEOF;
+ }
+socket_error:
+ {
+ res = GST_RTSP_ENET;
+ }
+read_error:
+ {
+ return res;
+ }
+}
+
+static GstRTSPMessage *
+gen_tunnel_reply (GstRTSPConnection * conn, GstRTSPStatusCode code,
+ const GstRTSPMessage * request)
+{
+ GstRTSPMessage *msg;
+ GstRTSPResult res;
+
+ if (gst_rtsp_status_as_text (code) == NULL)
+ code = GST_RTSP_STS_INTERNAL_SERVER_ERROR;
+
+ GST_RTSP_CHECK (gst_rtsp_message_new_response (&msg, code, NULL, request),
+ no_message);
+
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_SERVER,
+ "GStreamer RTSP Server");
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CONNECTION, "close");
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CACHE_CONTROL, "no-store");
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_PRAGMA, "no-cache");
+
+ if (code == GST_RTSP_STS_OK) {
+ if (conn->ip)
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_X_SERVER_IP_ADDRESS,
+ conn->ip);
+ gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/x-rtsp-tunnelled");
+ }
+
+ return msg;
+
+ /* ERRORS */
+no_message:
+ {
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_connection_receive:
+ * @conn: a #GstRTSPConnection
+ * @message: the message to read
+ * @timeout: a timeout value or #NULL
+ *
+ * Attempt to read into @message from the connected @conn, blocking up to
+ * the specified @timeout. @timeout can be #NULL, in which case this function
+ * might block forever.
+ *
+ * This function can be cancelled with gst_rtsp_connection_flush().
+ *
+ * Returns: #GST_RTSP_OK on success.
+ */
+GstRTSPResult
+gst_rtsp_connection_receive (GstRTSPConnection * conn, GstRTSPMessage * message,
+ GTimeVal * timeout)
+{
+ GstRTSPResult res;
+ GstRTSPBuilder builder;
+ gint retval;
+ GstClockTime to;
+
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn->readfd != NULL, GST_RTSP_EINVAL);
+
+ /* configure timeout if any */
+ to = timeout ? GST_TIMEVAL_TO_TIME (*timeout) : GST_CLOCK_TIME_NONE;
+
+ gst_poll_set_controllable (conn->fdset, TRUE);
+ gst_poll_fd_ctl_write (conn->fdset, conn->writefd, FALSE);
+ gst_poll_fd_ctl_read (conn->fdset, conn->readfd, TRUE);
+
+ memset (&builder, 0, sizeof (GstRTSPBuilder));
+ while (TRUE) {
+ res = build_next (&builder, message, conn);
+ if (G_UNLIKELY (res == GST_RTSP_EEOF))
+ goto eof;
+ else if (G_LIKELY (res == GST_RTSP_OK)) {
+ if (!conn->manual_http) {
+ if (message->type == GST_RTSP_MESSAGE_HTTP_REQUEST) {
+ if (conn->tstate == TUNNEL_STATE_NONE &&
+ message->type_data.request.method == GST_RTSP_GET) {
+ GstRTSPMessage *response;
+
+ conn->tstate = TUNNEL_STATE_GET;
+
+ /* tunnel GET request, we can reply now */
+ response = gen_tunnel_reply (conn, GST_RTSP_STS_OK, message);
+ res = gst_rtsp_connection_send (conn, response, timeout);
+ gst_rtsp_message_free (response);
+ if (res == GST_RTSP_OK)
+ res = GST_RTSP_ETGET;
+ goto cleanup;
+ } else if (conn->tstate == TUNNEL_STATE_NONE &&
+ message->type_data.request.method == GST_RTSP_POST) {
+ conn->tstate = TUNNEL_STATE_POST;
+
+ /* tunnel POST request, the caller now has to link the two
+ * connections. */
+ res = GST_RTSP_ETPOST;
+ goto cleanup;
+ } else {
+ res = GST_RTSP_EPARSE;
+ goto cleanup;
+ }
+ } else if (message->type == GST_RTSP_MESSAGE_HTTP_RESPONSE) {
+ res = GST_RTSP_EPARSE;
+ goto cleanup;
+ }
+ }
+
+ break;
+ } else if (G_UNLIKELY (res != GST_RTSP_EINTR))
+ goto read_error;
+
+ do {
+ retval = gst_poll_wait (conn->fdset, to);
+ } while (retval == -1 && (errno == EINTR || errno == EAGAIN));
+
+ /* check for timeout */
+ if (G_UNLIKELY (retval == 0))
+ goto select_timeout;
+
+ if (G_UNLIKELY (retval == -1)) {
+ if (errno == EBUSY)
+ goto stopped;
+ else
+ goto select_error;
+ }
+
+ /* could also be an error with write socket */
+ if (gst_poll_fd_has_error (conn->fdset, conn->writefd))
+ goto socket_error;
+
+ gst_poll_set_controllable (conn->fdset, FALSE);
+ }
+
+ /* we have a message here */
+ build_reset (&builder);
+
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+select_error:
+ {
+ res = GST_RTSP_ESYS;
+ goto cleanup;
+ }
+select_timeout:
+ {
+ res = GST_RTSP_ETIMEOUT;
+ goto cleanup;
+ }
+stopped:
+ {
+ res = GST_RTSP_EINTR;
+ goto cleanup;
+ }
+eof:
+ {
+ res = GST_RTSP_EEOF;
+ goto cleanup;
+ }
+socket_error:
+ {
+ res = GST_RTSP_ENET;
+ goto cleanup;
+ }
+read_error:
+cleanup:
+ {
+ build_reset (&builder);
+ gst_rtsp_message_unset (message);
+ return res;
+ }
+}
+
+/**
+ * gst_rtsp_connection_close:
+ * @conn: a #GstRTSPConnection
+ *
+ * Close the connected @conn. After this call, the connection is in the same
+ * state as when it was first created.
+ *
+ * Returns: #GST_RTSP_OK on success.
+ */
+GstRTSPResult
+gst_rtsp_connection_close (GstRTSPConnection * conn)
+{
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+
+ g_free (conn->ip);
+ conn->ip = NULL;
+
+ conn->read_ahead = 0;
+
+ g_free (conn->initial_buffer);
+ conn->initial_buffer = NULL;
+ conn->initial_buffer_offset = 0;
+
+ REMOVE_POLLFD (conn->fdset, &conn->fd0);
+ REMOVE_POLLFD (conn->fdset, &conn->fd1);
+ conn->writefd = NULL;
+ conn->readfd = NULL;
+ conn->tunneled = FALSE;
+ conn->tstate = TUNNEL_STATE_NONE;
+ conn->ctxp = NULL;
+ g_free (conn->username);
+ conn->username = NULL;
+ g_free (conn->passwd);
+ conn->passwd = NULL;
+ gst_rtsp_connection_clear_auth_params (conn);
+ conn->timeout = 60;
+ conn->cseq = 0;
+ conn->session_id[0] = '\0';
+
+ return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_connection_free:
+ * @conn: a #GstRTSPConnection
+ *
+ * Close and free @conn.
+ *
+ * Returns: #GST_RTSP_OK on success.
+ */
+GstRTSPResult
+gst_rtsp_connection_free (GstRTSPConnection * conn)
+{
+ GstRTSPResult res;
+
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+
+ res = gst_rtsp_connection_close (conn);
+ gst_poll_free (conn->fdset);
+ g_timer_destroy (conn->timer);
+ gst_rtsp_url_free (conn->url);
+ g_free (conn->proxy_host);
+ g_free (conn);
+#ifdef G_OS_WIN32
+ WSACleanup ();
+#endif
+
+ return res;
+}
+
+/**
+ * gst_rtsp_connection_poll:
+ * @conn: a #GstRTSPConnection
+ * @events: a bitmask of #GstRTSPEvent flags to check
+ * @revents: location for result flags
+ * @timeout: a timeout
+ *
+ * Wait up to the specified @timeout for the connection to become available for
+ * at least one of the operations specified in @events. When the function returns
+ * with #GST_RTSP_OK, @revents will contain a bitmask of available operations on
+ * @conn.
+ *
+ * @timeout can be #NULL, in which case this function might block forever.
+ *
+ * This function can be cancelled with gst_rtsp_connection_flush().
+ *
+ * Returns: #GST_RTSP_OK on success.
+ *
+ * Since: 0.10.15
+ */
+GstRTSPResult
+gst_rtsp_connection_poll (GstRTSPConnection * conn, GstRTSPEvent events,
+ GstRTSPEvent * revents, GTimeVal * timeout)
+{
+ GstClockTime to;
+ gint retval;
+
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (events != 0, GST_RTSP_EINVAL);
+ g_return_val_if_fail (revents != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn->readfd != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn->writefd != NULL, GST_RTSP_EINVAL);
+
+ gst_poll_set_controllable (conn->fdset, TRUE);
+
+ /* add fd to writer set when asked to */
+ gst_poll_fd_ctl_write (conn->fdset, conn->writefd,
+ events & GST_RTSP_EV_WRITE);
+
+ /* add fd to reader set when asked to */
+ gst_poll_fd_ctl_read (conn->fdset, conn->readfd, events & GST_RTSP_EV_READ);
+
+ /* configure timeout if any */
+ to = timeout ? GST_TIMEVAL_TO_TIME (*timeout) : GST_CLOCK_TIME_NONE;
+
+ do {
+ retval = gst_poll_wait (conn->fdset, to);
+ } while (retval == -1 && (errno == EINTR || errno == EAGAIN));
+
+ if (G_UNLIKELY (retval == 0))
+ goto select_timeout;
+
+ if (G_UNLIKELY (retval == -1)) {
+ if (errno == EBUSY)
+ goto stopped;
+ else
+ goto select_error;
+ }
+
+ *revents = 0;
+ if (events & GST_RTSP_EV_READ) {
+ if (gst_poll_fd_can_read (conn->fdset, conn->readfd))
+ *revents |= GST_RTSP_EV_READ;
+ }
+ if (events & GST_RTSP_EV_WRITE) {
+ if (gst_poll_fd_can_write (conn->fdset, conn->writefd))
+ *revents |= GST_RTSP_EV_WRITE;
+ }
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+select_timeout:
+ {
+ return GST_RTSP_ETIMEOUT;
+ }
+select_error:
+ {
+ return GST_RTSP_ESYS;
+ }
+stopped:
+ {
+ return GST_RTSP_EINTR;
+ }
+}
+
+/**
+ * gst_rtsp_connection_next_timeout:
+ * @conn: a #GstRTSPConnection
+ * @timeout: a timeout
+ *
+ * Calculate the next timeout for @conn, storing the result in @timeout.
+ *
+ * Returns: #GST_RTSP_OK.
+ */
+GstRTSPResult
+gst_rtsp_connection_next_timeout (GstRTSPConnection * conn, GTimeVal * timeout)
+{
+ gdouble elapsed;
+ glong sec;
+ gulong usec;
+ gint ctimeout;
+
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (timeout != NULL, GST_RTSP_EINVAL);
+
+ ctimeout = conn->timeout;
+ if (ctimeout >= 20) {
+ /* Because we should act before the timeout we timeout 5
+ * seconds in advance. */
+ ctimeout -= 5;
+ } else if (ctimeout >= 5) {
+ /* else timeout 20% earlier */
+ ctimeout -= ctimeout / 5;
+ } else if (ctimeout >= 1) {
+ /* else timeout 1 second earlier */
+ ctimeout -= 1;
+ }
+
+ elapsed = g_timer_elapsed (conn->timer, &usec);
+ if (elapsed >= ctimeout) {
+ sec = 0;
+ usec = 0;
+ } else {
+ sec = ctimeout - elapsed;
+ if (usec <= G_USEC_PER_SEC)
+ usec = G_USEC_PER_SEC - usec;
+ else
+ usec = 0;
+ }
+
+ timeout->tv_sec = sec;
+ timeout->tv_usec = usec;
+
+ return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_connection_reset_timeout:
+ * @conn: a #GstRTSPConnection
+ *
+ * Reset the timeout of @conn.
+ *
+ * Returns: #GST_RTSP_OK.
+ */
+GstRTSPResult
+gst_rtsp_connection_reset_timeout (GstRTSPConnection * conn)
+{
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+
+ g_timer_start (conn->timer);
+
+ return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_connection_flush:
+ * @conn: a #GstRTSPConnection
+ * @flush: start or stop the flush
+ *
+ * Start or stop the flushing action on @conn. When flushing, all current
+ * and future actions on @conn will return #GST_RTSP_EINTR until the connection
+ * is set to non-flushing mode again.
+ *
+ * Returns: #GST_RTSP_OK.
+ */
+GstRTSPResult
+gst_rtsp_connection_flush (GstRTSPConnection * conn, gboolean flush)
+{
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+
+ gst_poll_set_flushing (conn->fdset, flush);
+
+ return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_connection_set_proxy:
+ * @conn: a #GstRTSPConnection
+ * @host: the proxy host
+ * @port: the proxy port
+ *
+ * Set the proxy host and port.
+ *
+ * Returns: #GST_RTSP_OK.
+ *
+ * Since: 0.10.23
+ */
+GstRTSPResult
+gst_rtsp_connection_set_proxy (GstRTSPConnection * conn,
+ const gchar * host, guint port)
+{
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+
+ g_free (conn->proxy_host);
+ conn->proxy_host = g_strdup (host);
+ conn->proxy_port = port;
+
+ return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_connection_set_auth:
+ * @conn: a #GstRTSPConnection
+ * @method: authentication method
+ * @user: the user
+ * @pass: the password
+ *
+ * Configure @conn for authentication mode @method with @user and @pass as the
+ * user and password respectively.
+ *
+ * Returns: #GST_RTSP_OK.
+ */
+GstRTSPResult
+gst_rtsp_connection_set_auth (GstRTSPConnection * conn,
+ GstRTSPAuthMethod method, const gchar * user, const gchar * pass)
+{
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+
+ if (method == GST_RTSP_AUTH_DIGEST && ((user == NULL || pass == NULL)
+ || g_strrstr (user, ":") != NULL))
+ return GST_RTSP_EINVAL;
+
+ /* Make sure the username and passwd are being set for authentication */
+ if (method == GST_RTSP_AUTH_NONE && (user == NULL || pass == NULL))
+ return GST_RTSP_EINVAL;
+
+ /* ":" chars are not allowed in usernames for basic auth */
+ if (method == GST_RTSP_AUTH_BASIC && g_strrstr (user, ":") != NULL)
+ return GST_RTSP_EINVAL;
+
+ g_free (conn->username);
+ g_free (conn->passwd);
+
+ conn->auth_method = method;
+ conn->username = g_strdup (user);
+ conn->passwd = g_strdup (pass);
+
+ return GST_RTSP_OK;
+}
+
+/**
+ * str_case_hash:
+ * @key: ASCII string to hash
+ *
+ * Hashes @key in a case-insensitive manner.
+ *
+ * Returns: the hash code.
+ **/
+static guint
+str_case_hash (gconstpointer key)
+{
+ const char *p = key;
+ guint h = g_ascii_toupper (*p);
+
+ if (h)
+ for (p += 1; *p != '\0'; p++)
+ h = (h << 5) - h + g_ascii_toupper (*p);
+
+ return h;
+}
+
+/**
+ * str_case_equal:
+ * @v1: an ASCII string
+ * @v2: another ASCII string
+ *
+ * Compares @v1 and @v2 in a case-insensitive manner
+ *
+ * Returns: %TRUE if they are equal (modulo case)
+ **/
+static gboolean
+str_case_equal (gconstpointer v1, gconstpointer v2)
+{
+ const char *string1 = v1;
+ const char *string2 = v2;
+
+ return g_ascii_strcasecmp (string1, string2) == 0;
+}
+
+/**
+ * gst_rtsp_connection_set_auth_param:
+ * @conn: a #GstRTSPConnection
+ * @param: authentication directive
+ * @value: value
+ *
+ * Setup @conn with authentication directives. This is not necesary for
+ * methods #GST_RTSP_AUTH_NONE and #GST_RTSP_AUTH_BASIC. For
+ * #GST_RTSP_AUTH_DIGEST, directives should be taken from the digest challenge
+ * in the WWW-Authenticate response header and can include realm, domain,
+ * nonce, opaque, stale, algorithm, qop as per RFC2617.
+ *
+ * Since: 0.10.20
+ */
+void
+gst_rtsp_connection_set_auth_param (GstRTSPConnection * conn,
+ const gchar * param, const gchar * value)
+{
+ g_return_if_fail (conn != NULL);
+ g_return_if_fail (param != NULL);
+
+ if (conn->auth_params == NULL) {
+ conn->auth_params =
+ g_hash_table_new_full (str_case_hash, str_case_equal, g_free, g_free);
+ }
+ g_hash_table_insert (conn->auth_params, g_strdup (param), g_strdup (value));
+}
+
+/**
+ * gst_rtsp_connection_clear_auth_params:
+ * @conn: a #GstRTSPConnection
+ *
+ * Clear the list of authentication directives stored in @conn.
+ *
+ * Since: 0.10.20
+ */
+void
+gst_rtsp_connection_clear_auth_params (GstRTSPConnection * conn)
+{
+ g_return_if_fail (conn != NULL);
+
+ if (conn->auth_params != NULL) {
+ g_hash_table_destroy (conn->auth_params);
+ conn->auth_params = NULL;
+ }
+}
+
+static GstRTSPResult
+set_qos_dscp (gint fd, guint qos_dscp)
+{
+ union gst_sockaddr sa;
+ socklen_t slen = sizeof (sa);
+ gint af;
+ gint tos;
+
+ if (fd == -1)
+ return GST_RTSP_OK;
+
+ if (getsockname (fd, &sa.sa, &slen) < 0)
+ goto no_getsockname;
+
+ af = sa.sa.sa_family;
+
+ /* if this is an IPv4-mapped address then do IPv4 QoS */
+ if (af == AF_INET6) {
+ if (IN6_IS_ADDR_V4MAPPED (&sa.sa_in6.sin6_addr))
+ af = AF_INET;
+ }
+
+ /* extract and shift 6 bits of the DSCP */
+ tos = (qos_dscp & 0x3f) << 2;
+
+ switch (af) {
+ case AF_INET:
+ if (SETSOCKOPT (fd, IPPROTO_IP, IP_TOS, &tos, sizeof (tos)) < 0)
+ goto no_setsockopt;
+ break;
+ case AF_INET6:
+#ifdef IPV6_TCLASS
+ if (SETSOCKOPT (fd, IPPROTO_IPV6, IPV6_TCLASS, &tos, sizeof (tos)) < 0)
+ goto no_setsockopt;
+ break;
+#endif
+ default:
+ goto wrong_family;
+ }
+
+ return GST_RTSP_OK;
+
+ /* ERRORS */
+no_getsockname:
+no_setsockopt:
+ {
+ return GST_RTSP_ESYS;
+ }
+
+wrong_family:
+ {
+ return GST_RTSP_ERROR;
+ }
+}
+
+/**
+ * gst_rtsp_connection_set_qos_dscp:
+ * @conn: a #GstRTSPConnection
+ * @qos_dscp: DSCP value
+ *
+ * Configure @conn to use the specified DSCP value.
+ *
+ * Returns: #GST_RTSP_OK on success.
+ *
+ * Since: 0.10.20
+ */
+GstRTSPResult
+gst_rtsp_connection_set_qos_dscp (GstRTSPConnection * conn, guint qos_dscp)
+{
+ GstRTSPResult res;
+
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn->readfd != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn->writefd != NULL, GST_RTSP_EINVAL);
+
+ res = set_qos_dscp (conn->fd0.fd, qos_dscp);
+ if (res == GST_RTSP_OK)
+ res = set_qos_dscp (conn->fd1.fd, qos_dscp);
+
+ return res;
+}
+
+
+/**
+ * gst_rtsp_connection_get_url:
+ * @conn: a #GstRTSPConnection
+ *
+ * Retrieve the URL of the other end of @conn.
+ *
+ * Returns: The URL. This value remains valid until the
+ * connection is freed.
+ *
+ * Since: 0.10.23
+ */
+GstRTSPUrl *
+gst_rtsp_connection_get_url (const GstRTSPConnection * conn)
+{
+ g_return_val_if_fail (conn != NULL, NULL);
+
+ return conn->url;
+}
+
+/**
+ * gst_rtsp_connection_get_ip:
+ * @conn: a #GstRTSPConnection
+ *
+ * Retrieve the IP address of the other end of @conn.
+ *
+ * Returns: The IP address as a string. this value remains valid until the
+ * connection is closed.
+ *
+ * Since: 0.10.20
+ */
+const gchar *
+gst_rtsp_connection_get_ip (const GstRTSPConnection * conn)
+{
+ g_return_val_if_fail (conn != NULL, NULL);
+
+ return conn->ip;
+}
+
+/**
+ * gst_rtsp_connection_set_ip:
+ * @conn: a #GstRTSPConnection
+ * @ip: an ip address
+ *
+ * Set the IP address of the server.
+ *
+ * Since: 0.10.23
+ */
+void
+gst_rtsp_connection_set_ip (GstRTSPConnection * conn, const gchar * ip)
+{
+ g_return_if_fail (conn != NULL);
+
+ g_free (conn->ip);
+ conn->ip = g_strdup (ip);
+}
+
+/**
+ * gst_rtsp_connection_get_readfd:
+ * @conn: a #GstRTSPConnection
+ *
+ * Get the file descriptor for reading.
+ *
+ * Returns: the file descriptor used for reading or -1 on error. The file
+ * descriptor remains valid until the connection is closed.
+ *
+ * Since: 0.10.23
+ */
+gint
+gst_rtsp_connection_get_readfd (const GstRTSPConnection * conn)
+{
+ g_return_val_if_fail (conn != NULL, -1);
+ g_return_val_if_fail (conn->readfd != NULL, -1);
+
+ return conn->readfd->fd;
+}
+
+/**
+ * gst_rtsp_connection_get_writefd:
+ * @conn: a #GstRTSPConnection
+ *
+ * Get the file descriptor for writing.
+ *
+ * Returns: the file descriptor used for writing or -1 on error. The file
+ * descriptor remains valid until the connection is closed.
+ *
+ * Since: 0.10.23
+ */
+gint
+gst_rtsp_connection_get_writefd (const GstRTSPConnection * conn)
+{
+ g_return_val_if_fail (conn != NULL, -1);
+ g_return_val_if_fail (conn->writefd != NULL, -1);
+
+ return conn->writefd->fd;
+}
+
+/**
+ * gst_rtsp_connection_set_http_mode:
+ * @conn: a #GstRTSPConnection
+ * @enable: %TRUE to enable manual HTTP mode
+ *
+ * By setting the HTTP mode to %TRUE the message parsing will support HTTP
+ * messages in addition to the RTSP messages. It will also disable the
+ * automatic handling of setting up an HTTP tunnel.
+ *
+ * Since: 0.10.25
+ */
+void
+gst_rtsp_connection_set_http_mode (GstRTSPConnection * conn, gboolean enable)
+{
+ g_return_if_fail (conn != NULL);
+
+ conn->manual_http = enable;
+}
+
+/**
+ * gst_rtsp_connection_set_tunneled:
+ * @conn: a #GstRTSPConnection
+ * @tunneled: the new state
+ *
+ * Set the HTTP tunneling state of the connection. This must be configured before
+ * the @conn is connected.
+ *
+ * Since: 0.10.23
+ */
+void
+gst_rtsp_connection_set_tunneled (GstRTSPConnection * conn, gboolean tunneled)
+{
+ g_return_if_fail (conn != NULL);
+ g_return_if_fail (conn->readfd == NULL);
+ g_return_if_fail (conn->writefd == NULL);
+
+ conn->tunneled = tunneled;
+}
+
+/**
+ * gst_rtsp_connection_is_tunneled:
+ * @conn: a #GstRTSPConnection
+ *
+ * Get the tunneling state of the connection.
+ *
+ * Returns: if @conn is using HTTP tunneling.
+ *
+ * Since: 0.10.23
+ */
+gboolean
+gst_rtsp_connection_is_tunneled (const GstRTSPConnection * conn)
+{
+ g_return_val_if_fail (conn != NULL, FALSE);
+
+ return conn->tunneled;
+}
+
+/**
+ * gst_rtsp_connection_get_tunnelid:
+ * @conn: a #GstRTSPConnection
+ *
+ * Get the tunnel session id the connection.
+ *
+ * Returns: returns a non-empty string if @conn is being tunneled over HTTP.
+ *
+ * Since: 0.10.23
+ */
+const gchar *
+gst_rtsp_connection_get_tunnelid (const GstRTSPConnection * conn)
+{
+ g_return_val_if_fail (conn != NULL, NULL);
+
+ if (!conn->tunneled)
+ return NULL;
+
+ return conn->tunnelid;
+}
+
+/**
+ * gst_rtsp_connection_do_tunnel:
+ * @conn: a #GstRTSPConnection
+ * @conn2: a #GstRTSPConnection or %NULL
+ *
+ * If @conn received the first tunnel connection and @conn2 received
+ * the second tunnel connection, link the two connections together so that
+ * @conn manages the tunneled connection.
+ *
+ * After this call, @conn2 cannot be used anymore and must be freed with
+ * gst_rtsp_connection_free().
+ *
+ * If @conn2 is %NULL then only the base64 decoding context will be setup for
+ * @conn.
+ *
+ * Returns: return GST_RTSP_OK on success.
+ *
+ * Since: 0.10.23
+ */
+GstRTSPResult
+gst_rtsp_connection_do_tunnel (GstRTSPConnection * conn,
+ GstRTSPConnection * conn2)
+{
+ g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
+
+ if (conn2 != NULL) {
+ g_return_val_if_fail (conn->tstate == TUNNEL_STATE_GET, GST_RTSP_EINVAL);
+ g_return_val_if_fail (conn2->tstate == TUNNEL_STATE_POST, GST_RTSP_EINVAL);
+ g_return_val_if_fail (!memcmp (conn2->tunnelid, conn->tunnelid,
+ TUNNELID_LEN), GST_RTSP_EINVAL);
+
+ /* both connections have fd0 as the read/write socket. start by taking the
+ * socket from conn2 and set it as the socket in conn */
+ conn->fd1 = conn2->fd0;
+
+ /* clean up some of the state of conn2 */
+ gst_poll_remove_fd (conn2->fdset, &conn2->fd0);
+ conn2->fd0.fd = -1;
+ conn2->readfd = conn2->writefd = NULL;
+
+ /* We make fd0 the write socket and fd1 the read socket. */
+ conn->writefd = &conn->fd0;
+ conn->readfd = &conn->fd1;
+
+ conn->tstate = TUNNEL_STATE_COMPLETE;
+ }
+
+ /* we need base64 decoding for the readfd */
+ conn->ctx.state = 0;
+ conn->ctx.save = 0;
+ conn->ctx.cout = 0;
+ conn->ctx.coutl = 0;
+ conn->ctxp = &conn->ctx;
+
+ return GST_RTSP_OK;
+}
+
+#define READ_ERR (G_IO_HUP | G_IO_ERR | G_IO_NVAL)
+#define READ_COND (G_IO_IN | READ_ERR)
+#define WRITE_ERR (G_IO_HUP | G_IO_ERR | G_IO_NVAL)
+#define WRITE_COND (G_IO_OUT | WRITE_ERR)
+
+typedef struct
+{
+ guint8 *data;
+ guint size;
+ guint id;
+} GstRTSPRec;
+
+/* async functions */
+struct _GstRTSPWatch
+{
+ GSource source;
+
+ GstRTSPConnection *conn;
+
+ GstRTSPBuilder builder;
+ GstRTSPMessage message;
+
+ GPollFD readfd;
+ GPollFD writefd;
+
+ /* queued message for transmission */
+ guint id;
+ GMutex *mutex;
+ GQueue *messages;
+ guint8 *write_data;
+ guint write_off;
+ guint write_size;
+ guint write_id;
+
+ GstRTSPWatchFuncs funcs;
+
+ gpointer user_data;
+ GDestroyNotify notify;
+};
+
+static gboolean
+gst_rtsp_source_prepare (GSource * source, gint * timeout)
+{
+ GstRTSPWatch *watch = (GstRTSPWatch *) source;
+
+ if (watch->conn->initial_buffer != NULL)
+ return TRUE;
+
+ *timeout = (watch->conn->timeout * 1000);
+
+ return FALSE;
+}
+
+static gboolean
+gst_rtsp_source_check (GSource * source)
+{
+ GstRTSPWatch *watch = (GstRTSPWatch *) source;
+
+ if (watch->readfd.revents & READ_COND)
+ return TRUE;
+
+ if (watch->writefd.revents & WRITE_COND)
+ return TRUE;
+
+ return FALSE;
+}
+
+static gboolean
+gst_rtsp_source_dispatch (GSource * source, GSourceFunc callback G_GNUC_UNUSED,
+ gpointer user_data G_GNUC_UNUSED)
+{
+ GstRTSPWatch *watch = (GstRTSPWatch *) source;
+ GstRTSPResult res = GST_RTSP_ERROR;
+ gboolean keep_running = TRUE;
+
+ /* first read as much as we can */
+ if (watch->readfd.revents & READ_COND || watch->conn->initial_buffer != NULL) {
+ do {
+ if (watch->readfd.revents & READ_ERR)
+ goto read_error;
+
+ res = build_next (&watch->builder, &watch->message, watch->conn);
+ if (res == GST_RTSP_EINTR)
+ break;
+ else if (G_UNLIKELY (res == GST_RTSP_EEOF)) {
+ watch->readfd.events = 0;
+ watch->readfd.revents = 0;
+ g_source_remove_poll ((GSource *) watch, &watch->readfd);
+ /* When we are in tunnelled mode, the read socket can be closed and we
+ * should be prepared for a new POST method to reopen it */
+ if (watch->conn->tstate == TUNNEL_STATE_COMPLETE) {
+ /* remove the read connection for the tunnel */
+ /* we accept a new POST request */
+ watch->conn->tstate = TUNNEL_STATE_GET;
+ /* and signal that we lost our tunnel */
+ if (watch->funcs.tunnel_lost)
+ res = watch->funcs.tunnel_lost (watch, watch->user_data);
+ goto read_done;
+ } else
+ goto eof;
+ } else if (G_LIKELY (res == GST_RTSP_OK)) {
+ if (!watch->conn->manual_http &&
+ watch->message.type == GST_RTSP_MESSAGE_HTTP_REQUEST) {
+ if (watch->conn->tstate == TUNNEL_STATE_NONE &&
+ watch->message.type_data.request.method == GST_RTSP_GET) {
+ GstRTSPMessage *response;
+ GstRTSPStatusCode code;
+
+ watch->conn->tstate = TUNNEL_STATE_GET;
+
+ if (watch->funcs.tunnel_start)
+ code = watch->funcs.tunnel_start (watch, watch->user_data);
+ else
+ code = GST_RTSP_STS_OK;
+
+ /* queue the response */
+ response = gen_tunnel_reply (watch->conn, code, &watch->message);
+ gst_rtsp_watch_send_message (watch, response, NULL);
+ gst_rtsp_message_free (response);
+ goto read_done;
+ } else if (watch->conn->tstate == TUNNEL_STATE_NONE &&
+ watch->message.type_data.request.method == GST_RTSP_POST) {
+ watch->conn->tstate = TUNNEL_STATE_POST;
+
+ /* in the callback the connection should be tunneled with the
+ * GET connection */
+ if (watch->funcs.tunnel_complete)
+ watch->funcs.tunnel_complete (watch, watch->user_data);
+ goto read_done;
+ }
+ }
+ }
+
+ if (!watch->conn->manual_http) {
+ /* if manual HTTP support is not enabled, then restore the message to
+ * what it would have looked like without the support for parsing HTTP
+ * messages being present */
+ if (watch->message.type == GST_RTSP_MESSAGE_HTTP_REQUEST) {
+ watch->message.type = GST_RTSP_MESSAGE_REQUEST;
+ watch->message.type_data.request.method = GST_RTSP_INVALID;
+ if (watch->message.type_data.request.version != GST_RTSP_VERSION_1_0)
+ watch->message.type_data.request.version = GST_RTSP_VERSION_INVALID;
+ res = GST_RTSP_EPARSE;
+ } else if (watch->message.type == GST_RTSP_MESSAGE_HTTP_RESPONSE) {
+ watch->message.type = GST_RTSP_MESSAGE_RESPONSE;
+ if (watch->message.type_data.response.version != GST_RTSP_VERSION_1_0)
+ watch->message.type_data.response.version =
+ GST_RTSP_VERSION_INVALID;
+ res = GST_RTSP_EPARSE;
+ }
+ }
+
+ if (G_LIKELY (res == GST_RTSP_OK)) {
+ if (watch->funcs.message_received)
+ watch->funcs.message_received (watch, &watch->message,
+ watch->user_data);
+ } else {
+ goto read_error;
+ }
+
+ read_done:
+ gst_rtsp_message_unset (&watch->message);
+ build_reset (&watch->builder);
+ } while (FALSE);
+ }
+
+ if (watch->writefd.revents & WRITE_COND) {
+ if (watch->writefd.revents & WRITE_ERR)
+ goto write_error;
+
+ g_mutex_lock (watch->mutex);
+ do {
+ if (watch->write_data == NULL) {
+ GstRTSPRec *rec;
+
+ /* get a new message from the queue */
+ rec = g_queue_pop_tail (watch->messages);
+ if (rec == NULL)
+ break;
+
+ watch->write_off = 0;
+ watch->write_data = rec->data;
+ watch->write_size = rec->size;
+ watch->write_id = rec->id;
+
+ g_slice_free (GstRTSPRec, rec);
+ }
+
+ res = write_bytes (watch->writefd.fd, watch->write_data,
+ &watch->write_off, watch->write_size);
+ g_mutex_unlock (watch->mutex);
+
+ if (res == GST_RTSP_EINTR)
+ goto write_blocked;
+ else if (G_LIKELY (res == GST_RTSP_OK)) {
+ if (watch->funcs.message_sent)
+ watch->funcs.message_sent (watch, watch->write_id, watch->user_data);
+ } else {
+ goto write_error;
+ }
+ g_mutex_lock (watch->mutex);
+
+ g_free (watch->write_data);
+ watch->write_data = NULL;
+ } while (TRUE);
+
+ watch->writefd.events = WRITE_ERR;
+
+ g_mutex_unlock (watch->mutex);
+ }
+
+write_blocked:
+ return keep_running;
+
+ /* ERRORS */
+eof:
+ {
+ if (watch->funcs.closed)
+ watch->funcs.closed (watch, watch->user_data);
+
+ /* always stop when the readfd returns EOF in non-tunneled mode */
+ return FALSE;
+ }
+read_error:
+ {
+ watch->readfd.events = 0;
+ watch->readfd.revents = 0;
+ g_source_remove_poll ((GSource *) watch, &watch->readfd);
+ keep_running = (watch->writefd.events != 0);
+
+ if (keep_running) {
+ if (watch->funcs.error_full)
+ GST_RTSP_CHECK (watch->funcs.error_full (watch, res, &watch->message,
+ 0, watch->user_data), error);
+ else
+ goto error;
+ } else
+ goto eof;
+ }
+write_error:
+ {
+ watch->writefd.events = 0;
+ watch->writefd.revents = 0;
+ g_source_remove_poll ((GSource *) watch, &watch->writefd);
+ keep_running = (watch->readfd.events != 0);
+
+ if (keep_running) {
+ if (watch->funcs.error_full)
+ GST_RTSP_CHECK (watch->funcs.error_full (watch, res, NULL,
+ watch->write_id, watch->user_data), error);
+ else
+ goto error;
+ } else
+ goto eof;
+ }
+error:
+ {
+ if (watch->funcs.error)
+ watch->funcs.error (watch, res, watch->user_data);
+
+ return keep_running;
+ }
+}
+
+static void
+gst_rtsp_rec_free (gpointer data)
+{
+ GstRTSPRec *rec = data;
+
+ g_free (rec->data);
+ g_slice_free (GstRTSPRec, rec);
+}
+
+static void
+gst_rtsp_source_finalize (GSource * source)
+{
+ GstRTSPWatch *watch = (GstRTSPWatch *) source;
+
+ build_reset (&watch->builder);
+ gst_rtsp_message_unset (&watch->message);
+
+ g_queue_foreach (watch->messages, (GFunc) gst_rtsp_rec_free, NULL);
+ g_queue_free (watch->messages);
+ watch->messages = NULL;
+ g_free (watch->write_data);
+
+ g_mutex_free (watch->mutex);
+
+ if (watch->notify)
+ watch->notify (watch->user_data);
+}
+
+static GSourceFuncs gst_rtsp_source_funcs = {
+ gst_rtsp_source_prepare,
+ gst_rtsp_source_check,
+ gst_rtsp_source_dispatch,
+ gst_rtsp_source_finalize,
+ NULL,
+ NULL
+};
+
+/**
+ * gst_rtsp_watch_new:
+ * @conn: a #GstRTSPConnection
+ * @funcs: watch functions
+ * @user_data: user data to pass to @funcs
+ * @notify: notify when @user_data is not referenced anymore
+ *
+ * Create a watch object for @conn. The functions provided in @funcs will be
+ * called with @user_data when activity happened on the watch.
+ *
+ * The new watch is usually created so that it can be attached to a
+ * maincontext with gst_rtsp_watch_attach().
+ *
+ * @conn must exist for the entire lifetime of the watch.
+ *
+ * Returns: a #GstRTSPWatch that can be used for asynchronous RTSP
+ * communication. Free with gst_rtsp_watch_unref () after usage.
+ *
+ * Since: 0.10.23
+ */
+GstRTSPWatch *
+gst_rtsp_watch_new (GstRTSPConnection * conn,
+ GstRTSPWatchFuncs * funcs, gpointer user_data, GDestroyNotify notify)
+{
+ GstRTSPWatch *result;
+
+ g_return_val_if_fail (conn != NULL, NULL);
+ g_return_val_if_fail (funcs != NULL, NULL);
+ g_return_val_if_fail (conn->readfd != NULL, NULL);
+ g_return_val_if_fail (conn->writefd != NULL, NULL);
+
+ result = (GstRTSPWatch *) g_source_new (&gst_rtsp_source_funcs,
+ sizeof (GstRTSPWatch));
+
+ result->conn = conn;
+ result->builder.state = STATE_START;
+
+ result->mutex = g_mutex_new ();
+ result->messages = g_queue_new ();
+
+ result->readfd.fd = -1;
+ result->writefd.fd = -1;
+
+ gst_rtsp_watch_reset (result);
+
+ result->funcs = *funcs;
+ result->user_data = user_data;
+ result->notify = notify;
+
+ return result;
+}
+
+/**
+ * gst_rtsp_watch_reset:
+ * @watch: a #GstRTSPWatch
+ *
+ * Reset @watch, this is usually called after gst_rtsp_connection_do_tunnel()
+ * when the file descriptors of the connection might have changed.
+ *
+ * Since: 0.10.23
+ */
+void
+gst_rtsp_watch_reset (GstRTSPWatch * watch)
+{
+ if (watch->readfd.fd != -1)
+ g_source_remove_poll ((GSource *) watch, &watch->readfd);
+ if (watch->writefd.fd != -1)
+ g_source_remove_poll ((GSource *) watch, &watch->writefd);
+
+ watch->readfd.fd = watch->conn->readfd->fd;
+ watch->readfd.events = READ_COND;
+ watch->readfd.revents = 0;
+
+ watch->writefd.fd = watch->conn->writefd->fd;
+ watch->writefd.events = WRITE_ERR;
+ watch->writefd.revents = 0;
+
+ if (watch->readfd.fd != -1)
+ g_source_add_poll ((GSource *) watch, &watch->readfd);
+ if (watch->writefd.fd != -1)
+ g_source_add_poll ((GSource *) watch, &watch->writefd);
+}
+
+/**
+ * gst_rtsp_watch_attach:
+ * @watch: a #GstRTSPWatch
+ * @context: a GMainContext (if NULL, the default context will be used)
+ *
+ * Adds a #GstRTSPWatch to a context so that it will be executed within that context.
+ *
+ * Returns: the ID (greater than 0) for the watch within the GMainContext.
+ *
+ * Since: 0.10.23
+ */
+guint
+gst_rtsp_watch_attach (GstRTSPWatch * watch, GMainContext * context)
+{
+ g_return_val_if_fail (watch != NULL, 0);
+
+ return g_source_attach ((GSource *) watch, context);
+}
+
+/**
+ * gst_rtsp_watch_unref:
+ * @watch: a #GstRTSPWatch
+ *
+ * Decreases the reference count of @watch by one. If the resulting reference
+ * count is zero the watch and associated memory will be destroyed.
+ *
+ * Since: 0.10.23
+ */
+void
+gst_rtsp_watch_unref (GstRTSPWatch * watch)
+{
+ g_return_if_fail (watch != NULL);
+
+ g_source_unref ((GSource *) watch);
+}
+
+/**
+ * gst_rtsp_watch_write_data:
+ * @watch: a #GstRTSPWatch
+ * @data: the data to queue
+ * @size: the size of @data
+ * @id: location for a message ID or %NULL
+ *
+ * Write @data using the connection of the @watch. If it cannot be sent
+ * immediately, it will be queued for transmission in @watch. The contents of
+ * @message will then be serialized and transmitted when the connection of the
+ * @watch becomes writable. In case the @message is queued, the ID returned in
+ * @id will be non-zero and used as the ID argument in the message_sent
+ * callback.
+ *
+ * This function will take ownership of @data and g_free() it after use.
+ *
+ * Returns: #GST_RTSP_OK on success.
+ *
+ * Since: 0.10.25
+ */
+GstRTSPResult
+gst_rtsp_watch_write_data (GstRTSPWatch * watch, const guint8 * data,
+ guint size, guint * id)
+{
+ GstRTSPResult res;
+ GstRTSPRec *rec;
+ guint off = 0;
+ GMainContext *context = NULL;
+
+ g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (data != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (size != 0, GST_RTSP_EINVAL);
+
+ g_mutex_lock (watch->mutex);
+
+ /* try to send the message synchronously first */
+ if (watch->messages->length == 0) {
+ res = write_bytes (watch->writefd.fd, data, &off, size);
+ if (res != GST_RTSP_EINTR) {
+ if (id != NULL)
+ *id = 0;
+ g_free ((gpointer) data);
+ goto done;
+ }
+ }
+
+ /* make a record with the data and id for sending async */
+ rec = g_slice_new (GstRTSPRec);
+ if (off == 0) {
+ rec->data = (guint8 *) data;
+ rec->size = size;
+ } else {
+ rec->data = g_memdup (data + off, size - off);
+ rec->size = size - off;
+ g_free ((gpointer) data);
+ }
+
+ do {
+ /* make sure rec->id is never 0 */
+ rec->id = ++watch->id;
+ } while (G_UNLIKELY (rec->id == 0));
+
+ /* add the record to a queue. FIXME we would like to have an upper limit here */
+ g_queue_push_head (watch->messages, rec);
+
+ /* make sure the main context will now also check for writability on the
+ * socket */
+ if (watch->writefd.events != WRITE_COND) {
+ watch->writefd.events = WRITE_COND;
+ context = ((GSource *) watch)->context;
+ }
+
+ if (id != NULL)
+ *id = rec->id;
+ res = GST_RTSP_OK;
+
+done:
+ g_mutex_unlock (watch->mutex);
+
+ if (context)
+ g_main_context_wakeup (context);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_watch_send_message:
+ * @watch: a #GstRTSPWatch
+ * @message: a #GstRTSPMessage
+ * @id: location for a message ID or %NULL
+ *
+ * Send a @message using the connection of the @watch. If it cannot be sent
+ * immediately, it will be queued for transmission in @watch. The contents of
+ * @message will then be serialized and transmitted when the connection of the
+ * @watch becomes writable. In case the @message is queued, the ID returned in
+ * @id will be non-zero and used as the ID argument in the message_sent
+ * callback.
+ *
+ * Returns: #GST_RTSP_OK on success.
+ *
+ * Since: 0.10.25
+ */
+GstRTSPResult
+gst_rtsp_watch_send_message (GstRTSPWatch * watch, GstRTSPMessage * message,
+ guint * id)
+{
+ GString *str;
+ guint size;
+
+ g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
+
+ /* make a record with the message as a string and id */
+ str = message_to_string (watch->conn, message);
+ size = str->len;
+ return gst_rtsp_watch_write_data (watch,
+ (guint8 *) g_string_free (str, FALSE), size, id);
+}
+
+/**
+ * gst_rtsp_watch_queue_data:
+ * @watch: a #GstRTSPWatch
+ * @data: the data to queue
+ * @size: the size of @data
+ *
+ * Queue @data for transmission in @watch. It will be transmitted when the
+ * connection of the @watch becomes writable.
+ *
+ * This function will take ownership of @data and g_free() it after use.
+ *
+ * The return value of this function will be used as the id argument in the
+ * message_sent callback.
+ *
+ * Deprecated: Use gst_rtsp_watch_write_data()
+ *
+ * Returns: an id.
+ *
+ * Since: 0.10.24
+ */
+#ifndef GST_REMOVE_DEPRECATED
+#ifdef GST_DISABLE_DEPRECATED
+guint
+gst_rtsp_watch_queue_data (GstRTSPWatch * watch, const guint8 * data,
+ guint size);
+#endif
+guint
+gst_rtsp_watch_queue_data (GstRTSPWatch * watch, const guint8 * data,
+ guint size)
+{
+ GstRTSPRec *rec;
+ GMainContext *context = NULL;
+
+ g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (data != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (size != 0, GST_RTSP_EINVAL);
+
+ g_mutex_lock (watch->mutex);
+
+ /* make a record with the data and id */
+ rec = g_slice_new (GstRTSPRec);
+ rec->data = (guint8 *) data;
+ rec->size = size;
+ do {
+ /* make sure rec->id is never 0 */
+ rec->id = ++watch->id;
+ } while (G_UNLIKELY (rec->id == 0));
+
+ /* add the record to a queue. FIXME we would like to have an upper limit here */
+ g_queue_push_head (watch->messages, rec);
+
+ /* make sure the main context will now also check for writability on the
+ * socket */
+ if (watch->writefd.events != WRITE_COND) {
+ watch->writefd.events = WRITE_COND;
+ context = ((GSource *) watch)->context;
+ }
+ g_mutex_unlock (watch->mutex);
+
+ if (context)
+ g_main_context_wakeup (context);
+
+ return rec->id;
+}
+#endif /* GST_REMOVE_DEPRECATED */
+
+/**
+ * gst_rtsp_watch_queue_message:
+ * @watch: a #GstRTSPWatch
+ * @message: a #GstRTSPMessage
+ *
+ * Queue a @message for transmission in @watch. The contents of this
+ * message will be serialized and transmitted when the connection of the
+ * @watch becomes writable.
+ *
+ * The return value of this function will be used as the id argument in the
+ * message_sent callback.
+ *
+ * Deprecated: Use gst_rtsp_watch_send_message()
+ *
+ * Returns: an id.
+ *
+ * Since: 0.10.23
+ */
+#ifndef GST_REMOVE_DEPRECATED
+#ifdef GST_DISABLE_DEPRECATED
+guint
+gst_rtsp_watch_queue_message (GstRTSPWatch * watch, GstRTSPMessage * message);
+#endif
+guint
+gst_rtsp_watch_queue_message (GstRTSPWatch * watch, GstRTSPMessage * message)
+{
+ GString *str;
+ guint size;
+
+ g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
+ g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
+
+ /* make a record with the message as a string and id */
+ str = message_to_string (watch->conn, message);
+ size = str->len;
+ return gst_rtsp_watch_queue_data (watch,
+ (guint8 *) g_string_free (str, FALSE), size);
+}
+#endif /* GST_REMOVE_DEPRECATED */
projects / mafwsubrenderer / blobdiff