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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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3654 lines
96 KiB
C
3654 lines
96 KiB
C
/* GStreamer
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* Copyright (C) <2005-2009> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/*
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* Unless otherwise indicated, Source Code is licensed under MIT license.
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* See further explanation attached in License Statement (distributed in the file
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* LICENSE).
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
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* of the Software, and to permit persons to whom the Software is furnished to do
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* so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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/**
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* SECTION:gstrtspconnection
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* @short_description: manage RTSP connections
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* @see_also: gstrtspurl
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*
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* This object manages the RTSP connection to the server. It provides function
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* to receive and send bytes and messages.
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*
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* Last reviewed on 2007-07-24 (0.10.14)
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*/
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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#include <stdio.h>
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#include <errno.h>
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#include <stdlib.h>
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#include <string.h>
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#include <time.h>
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#ifdef HAVE_UNISTD_H
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#include <unistd.h>
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#endif
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/* we include this here to get the G_OS_* defines */
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#include <glib.h>
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#include <gst/gst.h>
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#ifdef G_OS_WIN32
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/* ws2_32.dll has getaddrinfo and freeaddrinfo on Windows XP and later.
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* minwg32 headers check WINVER before allowing the use of these */
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#ifndef WINVER
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#define WINVER 0x0501
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#endif
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#include <winsock2.h>
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#include <ws2tcpip.h>
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#define EINPROGRESS WSAEINPROGRESS
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#else
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#include <sys/ioctl.h>
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#include <netdb.h>
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#include <sys/socket.h>
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#include <fcntl.h>
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#include <netinet/in.h>
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#endif
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#ifdef HAVE_FIONREAD_IN_SYS_FILIO
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#include <sys/filio.h>
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#endif
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#include "gstrtspconnection.h"
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#include "gstrtspbase64.h"
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union gst_sockaddr
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{
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struct sockaddr sa;
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struct sockaddr_in sa_in;
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struct sockaddr_in6 sa_in6;
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struct sockaddr_storage sa_stor;
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};
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typedef struct
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{
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gint state;
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guint save;
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guchar out[3]; /* the size must be evenly divisible by 3 */
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guint cout;
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guint coutl;
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} DecodeCtx;
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#ifdef MSG_NOSIGNAL
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#define SEND_FLAGS MSG_NOSIGNAL
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#else
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#define SEND_FLAGS 0
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#endif
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#ifdef G_OS_WIN32
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#define READ_SOCKET(fd, buf, len) recv (fd, (char *)buf, len, 0)
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#define WRITE_SOCKET(fd, buf, len) send (fd, (const char *)buf, len, SEND_FLAGS)
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#define SETSOCKOPT(sock, level, name, val, len) setsockopt (sock, level, name, (const char *)val, len)
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#define CLOSE_SOCKET(sock) closesocket (sock)
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#define ERRNO_IS_EAGAIN (WSAGetLastError () == WSAEWOULDBLOCK)
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#define ERRNO_IS_EINTR (WSAGetLastError () == WSAEINTR)
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/* According to Microsoft's connect() documentation this one returns
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* WSAEWOULDBLOCK and not WSAEINPROGRESS. */
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#define ERRNO_IS_EINPROGRESS (WSAGetLastError () == WSAEWOULDBLOCK)
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#else
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#define READ_SOCKET(fd, buf, len) read (fd, buf, len)
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#define WRITE_SOCKET(fd, buf, len) send (fd, buf, len, SEND_FLAGS)
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#define SETSOCKOPT(sock, level, name, val, len) setsockopt (sock, level, name, val, len)
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#define CLOSE_SOCKET(sock) close (sock)
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#define ERRNO_IS_EAGAIN (errno == EAGAIN)
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#define ERRNO_IS_EINTR (errno == EINTR)
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#define ERRNO_IS_EINPROGRESS (errno == EINPROGRESS)
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#endif
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#define ADD_POLLFD(fdset, pfd, fd) \
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G_STMT_START { \
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(pfd)->fd = fd; \
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gst_poll_add_fd (fdset, pfd); \
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} G_STMT_END
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#define REMOVE_POLLFD(fdset, pfd) \
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G_STMT_START { \
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if ((pfd)->fd != -1) { \
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GST_DEBUG ("remove fd %d", (pfd)->fd); \
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gst_poll_remove_fd (fdset, pfd); \
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CLOSE_SOCKET ((pfd)->fd); \
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(pfd)->fd = -1; \
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} \
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} G_STMT_END
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typedef enum
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{
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TUNNEL_STATE_NONE,
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TUNNEL_STATE_GET,
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TUNNEL_STATE_POST,
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TUNNEL_STATE_COMPLETE
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} GstRTSPTunnelState;
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#define TUNNELID_LEN 24
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struct _GstRTSPConnection
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{
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/*< private > */
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/* URL for the connection */
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GstRTSPUrl *url;
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/* connection state */
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GstPollFD fd0;
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GstPollFD fd1;
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GstPollFD *readfd;
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GstPollFD *writefd;
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gboolean manual_http;
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gchar tunnelid[TUNNELID_LEN];
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gboolean tunneled;
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GstRTSPTunnelState tstate;
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GstPoll *fdset;
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gchar *ip;
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gint read_ahead;
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gchar *initial_buffer;
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gsize initial_buffer_offset;
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/* Session state */
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gint cseq; /* sequence number */
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gchar session_id[512]; /* session id */
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gint timeout; /* session timeout in seconds */
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GTimer *timer; /* timeout timer */
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/* Authentication */
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GstRTSPAuthMethod auth_method;
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gchar *username;
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gchar *passwd;
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GHashTable *auth_params;
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DecodeCtx ctx;
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DecodeCtx *ctxp;
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gchar *proxy_host;
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guint proxy_port;
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};
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enum
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{
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STATE_START = 0,
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STATE_DATA_HEADER,
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STATE_DATA_BODY,
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STATE_READ_LINES,
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STATE_END,
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STATE_LAST
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};
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enum
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{
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READ_AHEAD_EOH = -1, /* end of headers */
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READ_AHEAD_CRLF = -2,
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READ_AHEAD_CRLFCR = -3
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};
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/* a structure for constructing RTSPMessages */
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typedef struct
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{
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gint state;
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GstRTSPResult status;
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guint8 buffer[4096];
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guint offset;
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guint line;
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guint8 *body_data;
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glong body_len;
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} GstRTSPBuilder;
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static void
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build_reset (GstRTSPBuilder * builder)
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{
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g_free (builder->body_data);
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memset (builder, 0, sizeof (GstRTSPBuilder));
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}
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/**
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* gst_rtsp_connection_create:
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* @url: a #GstRTSPUrl
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* @conn: storage for a #GstRTSPConnection
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*
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* Create a newly allocated #GstRTSPConnection from @url and store it in @conn.
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* The connection will not yet attempt to connect to @url, use
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* gst_rtsp_connection_connect().
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*
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* A copy of @url will be made.
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*
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* Returns: #GST_RTSP_OK when @conn contains a valid connection.
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*/
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GstRTSPResult
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gst_rtsp_connection_create (const GstRTSPUrl * url, GstRTSPConnection ** conn)
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{
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GstRTSPConnection *newconn;
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#ifdef G_OS_WIN32
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WSADATA w;
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int error;
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#endif
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g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
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#ifdef G_OS_WIN32
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error = WSAStartup (0x0202, &w);
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if (error)
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goto startup_error;
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if (w.wVersion != 0x0202)
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goto version_error;
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#endif
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newconn = g_new0 (GstRTSPConnection, 1);
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if ((newconn->fdset = gst_poll_new (TRUE)) == NULL)
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goto no_fdset;
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newconn->url = gst_rtsp_url_copy (url);
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newconn->fd0.fd = -1;
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newconn->fd1.fd = -1;
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newconn->timer = g_timer_new ();
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newconn->timeout = 60;
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newconn->cseq = 1;
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newconn->auth_method = GST_RTSP_AUTH_NONE;
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newconn->username = NULL;
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newconn->passwd = NULL;
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newconn->auth_params = NULL;
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*conn = newconn;
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return GST_RTSP_OK;
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/* ERRORS */
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#ifdef G_OS_WIN32
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startup_error:
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{
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g_warning ("Error %d on WSAStartup", error);
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return GST_RTSP_EWSASTART;
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}
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version_error:
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{
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g_warning ("Windows sockets are not version 0x202 (current 0x%x)",
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w.wVersion);
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WSACleanup ();
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return GST_RTSP_EWSAVERSION;
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}
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#endif
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no_fdset:
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{
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g_free (newconn);
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#ifdef G_OS_WIN32
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WSACleanup ();
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#endif
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return GST_RTSP_ESYS;
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}
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}
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/**
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* gst_rtsp_connection_create_from_fd:
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* @fd: a file descriptor
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* @ip: the IP address of the other end
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* @port: the port used by the other end
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* @initial_buffer: data already read from @fd
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* @conn: storage for a #GstRTSPConnection
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*
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* Create a new #GstRTSPConnection for handling communication on the existing
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* file descriptor @fd. The @initial_buffer contains any data already read from
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* @fd which should be used before starting to read new data.
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*
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* Returns: #GST_RTSP_OK when @conn contains a valid connection.
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*
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* Since: 0.10.25
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*/
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GstRTSPResult
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gst_rtsp_connection_create_from_fd (gint fd, const gchar * ip, guint16 port,
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const gchar * initial_buffer, GstRTSPConnection ** conn)
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{
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GstRTSPConnection *newconn = NULL;
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GstRTSPUrl *url;
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#ifdef G_OS_WIN32
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gulong flags = 1;
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#endif
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GstRTSPResult res;
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g_return_val_if_fail (fd >= 0, GST_RTSP_EINVAL);
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g_return_val_if_fail (ip != NULL, GST_RTSP_EINVAL);
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g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
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/* set to non-blocking mode so that we can cancel the communication */
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#ifndef G_OS_WIN32
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fcntl (fd, F_SETFL, O_NONBLOCK);
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#else
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ioctlsocket (fd, FIONBIO, &flags);
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#endif /* G_OS_WIN32 */
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/* create a url for the client address */
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url = g_new0 (GstRTSPUrl, 1);
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url->host = g_strdup (ip);
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url->port = port;
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/* now create the connection object */
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GST_RTSP_CHECK (gst_rtsp_connection_create (url, &newconn), newconn_failed);
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gst_rtsp_url_free (url);
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ADD_POLLFD (newconn->fdset, &newconn->fd0, fd);
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/* both read and write initially */
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newconn->readfd = &newconn->fd0;
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newconn->writefd = &newconn->fd0;
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newconn->ip = g_strdup (ip);
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newconn->initial_buffer = g_strdup (initial_buffer);
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*conn = newconn;
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return GST_RTSP_OK;
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/* ERRORS */
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newconn_failed:
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{
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gst_rtsp_url_free (url);
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return res;
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}
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}
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/**
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* gst_rtsp_connection_accept:
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* @sock: a socket
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* @conn: storage for a #GstRTSPConnection
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*
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* Accept a new connection on @sock and create a new #GstRTSPConnection for
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* handling communication on new socket.
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*
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* Returns: #GST_RTSP_OK when @conn contains a valid connection.
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*
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* Since: 0.10.23
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*/
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GstRTSPResult
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gst_rtsp_connection_accept (gint sock, GstRTSPConnection ** conn)
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{
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int fd;
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union gst_sockaddr sa;
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socklen_t slen = sizeof (sa);
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gchar ip[INET6_ADDRSTRLEN];
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guint16 port;
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g_return_val_if_fail (sock >= 0, GST_RTSP_EINVAL);
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g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
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memset (&sa, 0, slen);
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#ifndef G_OS_WIN32
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fd = accept (sock, &sa.sa, &slen);
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#else
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fd = accept (sock, &sa.sa, (gint *) & slen);
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#endif /* G_OS_WIN32 */
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if (fd == -1)
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goto accept_failed;
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|
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if (getnameinfo (&sa.sa, slen, ip, sizeof (ip), NULL, 0, NI_NUMERICHOST) != 0)
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goto getnameinfo_failed;
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|
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if (sa.sa.sa_family == AF_INET)
|
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port = sa.sa_in.sin_port;
|
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else if (sa.sa.sa_family == AF_INET6)
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port = sa.sa_in6.sin6_port;
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else
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goto wrong_family;
|
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|
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return gst_rtsp_connection_create_from_fd (fd, ip, port, NULL, conn);
|
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|
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/* ERRORS */
|
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accept_failed:
|
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{
|
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return GST_RTSP_ESYS;
|
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}
|
|
getnameinfo_failed:
|
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wrong_family:
|
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{
|
|
CLOSE_SOCKET (fd);
|
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return GST_RTSP_ERROR;
|
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}
|
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}
|
|
|
|
static gchar *
|
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do_resolve (const gchar * host)
|
|
{
|
|
static gchar ip[INET6_ADDRSTRLEN];
|
|
struct addrinfo *aires, hints;
|
|
struct addrinfo *ai;
|
|
gint aierr;
|
|
|
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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)
|
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goto no_addrinfo;
|
|
|
|
for (ai = aires; ai; ai = ai->ai_next) {
|
|
if (ai->ai_family == AF_INET || ai->ai_family == AF_INET6) {
|
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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)
|
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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:
|
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{
|
|
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;
|
|
|
|
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (timeout != NULL, GST_RTSP_EINVAL);
|
|
|
|
elapsed = g_timer_elapsed (conn->timer, &usec);
|
|
if (elapsed >= conn->timeout) {
|
|
sec = 0;
|
|
usec = 0;
|
|
} else {
|
|
sec = conn->timeout - 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()
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*
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|
* Returns: an id.
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|
*
|
|
* Since: 0.10.23
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*/
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|
#ifndef GST_REMOVE_DEPRECATED
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|
#ifdef GST_DISABLE_DEPRECATED
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|
guint
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|
gst_rtsp_watch_queue_message (GstRTSPWatch * watch, GstRTSPMessage * message);
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#endif
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|
guint
|
|
gst_rtsp_watch_queue_message (GstRTSPWatch * watch, GstRTSPMessage * message)
|
|
{
|
|
GString *str;
|
|
guint size;
|
|
|
|
g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
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|
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
|
|
|
|
/* make a record with the message as a string and id */
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|
str = message_to_string (watch->conn, message);
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|
size = str->len;
|
|
return gst_rtsp_watch_queue_data (watch,
|
|
(guint8 *) g_string_free (str, FALSE), size);
|
|
}
|
|
#endif /* GST_REMOVE_DEPRECATED */
|