mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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5da9f62313
With the 2.72 release, glib-networking developers have decided that TLS certificate validation cannot be implemented correctly by them, so they've deprecated it. In a nutshell: a cert can have several validation errors, but there are no guarantees that the TLS backend will return all those errors, and things are made even more complicated by the fact that the list of errors might refer to certs that are added for backwards-compat and won't actually be used by the TLS library. Our best option is to ignore the deprecation and pass the warning onto users so they can make an appropriate security decision regarding this. We can't deprecate the tls-validation-flags property because it is very useful when connecting to RTSP cameras that will never get updates to fix certificate errors. Relevant upstream merge requests / issues: https://gitlab.gnome.org/GNOME/glib/-/merge_requests/2214 https://gitlab.gnome.org/GNOME/glib-networking/-/issues/179 https://gitlab.gnome.org/GNOME/glib-networking/-/merge_requests/193 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2494>
5263 lines
148 KiB
C
5263 lines
148 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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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* @title: 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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#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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/* 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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#include <gst/base/base.h>
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/* necessary for IP_TOS define */
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#include <gio/gnetworking.h>
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#include "gstrtspconnection.h"
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#ifdef IP_TOS
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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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#endif
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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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typedef struct
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{
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/* If %TRUE we only own data and none of the
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* other fields
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*/
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gboolean borrowed;
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/* Header or full message */
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guint8 *data;
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guint data_size;
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gboolean data_is_data_header;
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/* Payload following data, if any */
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guint8 *body_data;
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guint body_data_size;
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/* or */
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GstBuffer *body_buffer;
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/* DATA packet header statically allocated for above */
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guint8 data_header[4];
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/* all below only for async writing */
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guint data_offset; /* == data_size when done */
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guint body_offset; /* into body_data or the buffer */
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/* ID of the message for notification */
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guint id;
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} GstRTSPSerializedMessage;
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static void
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gst_rtsp_serialized_message_clear (GstRTSPSerializedMessage * msg)
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{
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if (!msg->borrowed) {
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g_free (msg->body_data);
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gst_buffer_replace (&msg->body_buffer, NULL);
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}
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g_free (msg->data);
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}
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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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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 remote connection */
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GstRTSPUrl *url;
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GstRTSPVersion version;
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gboolean server;
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GSocketClient *client;
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GIOStream *stream0;
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GIOStream *stream1;
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GInputStream *input_stream;
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GOutputStream *output_stream;
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/* this is a read source we add on the write socket in tunneled mode to be
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* able to detect when client disconnects the GET channel */
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GInputStream *control_stream;
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/* connection state */
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GSocket *read_socket;
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GSocket *write_socket;
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GSocket *socket0, *socket1;
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gboolean read_socket_used;
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gboolean write_socket_used;
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GMutex socket_use_mutex;
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gboolean manual_http;
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gboolean may_cancel;
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GMutex cancellable_mutex;
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GCancellable *cancellable; /* protected by cancellable_mutex */
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gchar tunnelid[TUNNELID_LEN];
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gboolean tunneled;
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gboolean ignore_x_server_reply;
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GstRTSPTunnelState tstate;
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/* the remote and local ip */
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gchar *remote_ip;
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gchar *local_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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gboolean remember_session_id; /* remember the session id or not */
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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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guint content_length_limit;
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/* TLS */
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GTlsDatabase *tls_database;
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GTlsInteraction *tls_interaction;
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GstRTSPConnectionAcceptCertificateFunc accept_certificate_func;
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GDestroyNotify accept_certificate_destroy_notify;
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gpointer accept_certificate_user_data;
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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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guint body_len;
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} GstRTSPBuilder;
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/* function prototypes */
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static void add_auth_header (GstRTSPConnection * conn,
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GstRTSPMessage * message);
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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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static GstRTSPResult
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gst_rtsp_result_from_g_io_error (GError * error, GstRTSPResult default_res)
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{
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if (error == NULL)
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return GST_RTSP_OK;
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if (error->domain != G_IO_ERROR)
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return default_res;
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switch (error->code) {
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case G_IO_ERROR_TIMED_OUT:
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return GST_RTSP_ETIMEOUT;
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case G_IO_ERROR_INVALID_ARGUMENT:
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return GST_RTSP_EINVAL;
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case G_IO_ERROR_CANCELLED:
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case G_IO_ERROR_WOULD_BLOCK:
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return GST_RTSP_EINTR;
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default:
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return default_res;
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}
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}
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static gboolean
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tls_accept_certificate (GTlsConnection * conn, GTlsCertificate * peer_cert,
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GTlsCertificateFlags errors, GstRTSPConnection * rtspconn)
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{
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GError *error = NULL;
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gboolean accept = FALSE;
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if (rtspconn->tls_database) {
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GSocketConnectable *peer_identity;
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GTlsCertificateFlags validation_flags;
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GST_DEBUG ("TLS peer certificate not accepted, checking user database...");
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peer_identity =
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g_tls_client_connection_get_server_identity (G_TLS_CLIENT_CONNECTION
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(conn));
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errors =
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g_tls_database_verify_chain (rtspconn->tls_database, peer_cert,
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G_TLS_DATABASE_PURPOSE_AUTHENTICATE_SERVER, peer_identity,
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g_tls_connection_get_interaction (conn), G_TLS_DATABASE_VERIFY_NONE,
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NULL, &error);
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if (error)
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goto verify_error;
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validation_flags = gst_rtsp_connection_get_tls_validation_flags (rtspconn);
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accept = ((errors & validation_flags) == 0);
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if (accept)
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GST_DEBUG ("Peer certificate accepted");
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else
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GST_DEBUG ("Peer certificate not accepted (errors: 0x%08X)", errors);
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}
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if (!accept && rtspconn->accept_certificate_func) {
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accept =
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rtspconn->accept_certificate_func (conn, peer_cert, errors,
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rtspconn->accept_certificate_user_data);
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GST_DEBUG ("Peer certificate %saccepted by accept-certificate function",
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accept ? "" : "not ");
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}
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return accept;
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/* ERRORS */
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verify_error:
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{
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GST_ERROR ("An error occurred while verifying the peer certificate: %s",
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error->message);
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g_clear_error (&error);
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return FALSE;
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}
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}
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static void
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socket_client_event (GSocketClient * client, GSocketClientEvent event,
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GSocketConnectable * connectable, GTlsConnection * connection,
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GstRTSPConnection * rtspconn)
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{
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if (event == G_SOCKET_CLIENT_TLS_HANDSHAKING) {
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GST_DEBUG ("TLS handshaking about to start...");
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g_signal_connect (connection, "accept-certificate",
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(GCallback) tls_accept_certificate, rtspconn);
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g_tls_connection_set_interaction (connection, rtspconn->tls_interaction);
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}
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}
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/* transfer full */
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static GCancellable *
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get_cancellable (GstRTSPConnection * conn)
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{
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GCancellable *cancellable = NULL;
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g_mutex_lock (&conn->cancellable_mutex);
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if (conn->cancellable)
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cancellable = g_object_ref (conn->cancellable);
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g_mutex_unlock (&conn->cancellable_mutex);
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return cancellable;
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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: (out) (transfer full): 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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g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
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g_return_val_if_fail (url != NULL, GST_RTSP_EINVAL);
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newconn = g_new0 (GstRTSPConnection, 1);
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newconn->may_cancel = TRUE;
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newconn->cancellable = g_cancellable_new ();
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g_mutex_init (&newconn->cancellable_mutex);
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newconn->client = g_socket_client_new ();
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if (url->transports & GST_RTSP_LOWER_TRANS_TLS)
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g_socket_client_set_tls (newconn->client, TRUE);
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g_signal_connect (newconn->client, "event", (GCallback) socket_client_event,
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newconn);
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newconn->url = gst_rtsp_url_copy (url);
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newconn->timer = g_timer_new ();
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newconn->timeout = 60;
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newconn->cseq = 1; /* RFC 7826: "it is RECOMMENDED to start at 0.",
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but some servers don't copy values <1 due to bugs. */
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newconn->remember_session_id = TRUE;
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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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newconn->version = 0;
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newconn->content_length_limit = G_MAXUINT;
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*conn = newconn;
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return GST_RTSP_OK;
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}
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static gboolean
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collect_addresses (GSocket * socket, gchar ** ip, guint16 * port,
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gboolean remote, GError ** error)
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{
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GSocketAddress *addr;
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if (remote)
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addr = g_socket_get_remote_address (socket, error);
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else
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addr = g_socket_get_local_address (socket, error);
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if (!addr)
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return FALSE;
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if (ip)
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*ip = g_inet_address_to_string (g_inet_socket_address_get_address
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(G_INET_SOCKET_ADDRESS (addr)));
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if (port)
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*port = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
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g_object_unref (addr);
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return TRUE;
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}
|
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|
|
|
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/**
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* gst_rtsp_connection_create_from_socket:
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* @socket: a #GSocket
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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: (out) (transfer full): 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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* socket @socket. The @initial_buffer contains zero terminated data already
|
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* read from @socket 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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/* FIXME 2.0 We don't need the ip and port since they can be got from the
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* GSocket */
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GstRTSPResult
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gst_rtsp_connection_create_from_socket (GSocket * socket, const gchar * ip,
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guint16 port, 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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GstRTSPResult res;
|
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GError *err = NULL;
|
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gchar *local_ip;
|
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GIOStream *stream;
|
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|
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g_return_val_if_fail (G_IS_SOCKET (socket), 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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|
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if (!collect_addresses (socket, &local_ip, NULL, FALSE, &err))
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goto getnameinfo_failed;
|
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|
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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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|
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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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|
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stream = G_IO_STREAM (g_socket_connection_factory_create_connection (socket));
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|
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/* both read and write initially */
|
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newconn->server = TRUE;
|
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newconn->socket0 = socket;
|
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newconn->stream0 = stream;
|
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newconn->write_socket = newconn->read_socket = newconn->socket0;
|
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newconn->read_socket_used = FALSE;
|
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newconn->write_socket_used = FALSE;
|
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g_mutex_init (&newconn->socket_use_mutex);
|
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newconn->input_stream = g_io_stream_get_input_stream (stream);
|
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newconn->output_stream = g_io_stream_get_output_stream (stream);
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newconn->control_stream = NULL;
|
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newconn->remote_ip = g_strdup (ip);
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|
newconn->local_ip = local_ip;
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newconn->initial_buffer = g_strdup (initial_buffer);
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|
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*conn = newconn;
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|
|
return GST_RTSP_OK;
|
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|
|
/* ERRORS */
|
|
getnameinfo_failed:
|
|
{
|
|
GST_ERROR ("failed to get local address: %s", err->message);
|
|
res = gst_rtsp_result_from_g_io_error (err, GST_RTSP_ERROR);
|
|
g_clear_error (&err);
|
|
return res;
|
|
}
|
|
newconn_failed:
|
|
{
|
|
GST_ERROR ("failed to make connection");
|
|
g_free (local_ip);
|
|
gst_rtsp_url_free (url);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_accept:
|
|
* @socket: a socket
|
|
* @conn: (out) (transfer full): storage for a #GstRTSPConnection
|
|
* @cancellable: a #GCancellable to cancel the operation
|
|
*
|
|
* Accept a new connection on @socket and create a new #GstRTSPConnection for
|
|
* handling communication on new socket.
|
|
*
|
|
* Returns: #GST_RTSP_OK when @conn contains a valid connection.
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_accept (GSocket * socket, GstRTSPConnection ** conn,
|
|
GCancellable * cancellable)
|
|
{
|
|
GError *err = NULL;
|
|
gchar *ip;
|
|
guint16 port;
|
|
GSocket *client_sock;
|
|
GstRTSPResult ret;
|
|
|
|
g_return_val_if_fail (G_IS_SOCKET (socket), GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
|
|
|
|
client_sock = g_socket_accept (socket, cancellable, &err);
|
|
if (!client_sock)
|
|
goto accept_failed;
|
|
|
|
/* get the remote ip address and port */
|
|
if (!collect_addresses (client_sock, &ip, &port, TRUE, &err))
|
|
goto getnameinfo_failed;
|
|
|
|
ret =
|
|
gst_rtsp_connection_create_from_socket (client_sock, ip, port, NULL,
|
|
conn);
|
|
g_object_unref (client_sock);
|
|
g_free (ip);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
accept_failed:
|
|
{
|
|
GST_DEBUG ("Accepting client failed: %s", err->message);
|
|
ret = gst_rtsp_result_from_g_io_error (err, GST_RTSP_ESYS);
|
|
g_clear_error (&err);
|
|
return ret;
|
|
}
|
|
getnameinfo_failed:
|
|
{
|
|
GST_DEBUG ("getnameinfo failed: %s", err->message);
|
|
ret = gst_rtsp_result_from_g_io_error (err, GST_RTSP_ERROR);
|
|
g_clear_error (&err);
|
|
if (!g_socket_close (client_sock, &err)) {
|
|
GST_DEBUG ("Closing socket failed: %s", err->message);
|
|
g_clear_error (&err);
|
|
}
|
|
g_object_unref (client_sock);
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_get_tls:
|
|
* @conn: a #GstRTSPConnection
|
|
* @error: #GError for error reporting, or NULL to ignore.
|
|
*
|
|
* Get the TLS connection of @conn.
|
|
*
|
|
* For client side this will return the #GTlsClientConnection when connected
|
|
* over TLS.
|
|
*
|
|
* For server side connections, this function will create a GTlsServerConnection
|
|
* when called the first time and will return that same connection on subsequent
|
|
* calls. The server is then responsible for configuring the TLS connection.
|
|
*
|
|
* Returns: (transfer none): the TLS connection for @conn.
|
|
*
|
|
* Since: 1.2
|
|
*/
|
|
GTlsConnection *
|
|
gst_rtsp_connection_get_tls (GstRTSPConnection * conn, GError ** error)
|
|
{
|
|
GTlsConnection *result;
|
|
|
|
if (G_IS_TLS_CONNECTION (conn->stream0)) {
|
|
/* we already had one, return it */
|
|
result = G_TLS_CONNECTION (conn->stream0);
|
|
} else if (conn->server) {
|
|
/* no TLS connection but we are server, make one */
|
|
result = (GTlsConnection *)
|
|
g_tls_server_connection_new (conn->stream0, NULL, error);
|
|
if (result) {
|
|
g_object_unref (conn->stream0);
|
|
conn->stream0 = G_IO_STREAM (result);
|
|
conn->input_stream = g_io_stream_get_input_stream (conn->stream0);
|
|
conn->output_stream = g_io_stream_get_output_stream (conn->stream0);
|
|
}
|
|
} else {
|
|
/* client */
|
|
result = NULL;
|
|
g_set_error (error, GST_LIBRARY_ERROR, GST_LIBRARY_ERROR_FAILED,
|
|
"client not connected with TLS");
|
|
}
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_set_tls_validation_flags:
|
|
* @conn: a #GstRTSPConnection
|
|
* @flags: the validation flags.
|
|
*
|
|
* Sets the TLS validation flags to be used to verify the peer
|
|
* certificate when a TLS connection is established.
|
|
*
|
|
* GLib guarantees that if certificate verification fails, at least one error
|
|
* will be set, but it does not guarantee that all possible errors will be
|
|
* set. Accordingly, you may not safely decide to ignore any particular type
|
|
* of error.
|
|
*
|
|
* For example, it would be incorrect to mask %G_TLS_CERTIFICATE_EXPIRED if
|
|
* you want to allow expired certificates, because this could potentially be
|
|
* the only error flag set even if other problems exist with the certificate.
|
|
*
|
|
* Returns: TRUE if the validation flags are set correctly, or FALSE if
|
|
* @conn is NULL or is not a TLS connection.
|
|
*
|
|
* Since: 1.2.1
|
|
*/
|
|
gboolean
|
|
gst_rtsp_connection_set_tls_validation_flags (GstRTSPConnection * conn,
|
|
GTlsCertificateFlags flags)
|
|
{
|
|
gboolean res = FALSE;
|
|
|
|
g_return_val_if_fail (conn != NULL, FALSE);
|
|
|
|
res = g_socket_client_get_tls (conn->client);
|
|
G_GNUC_BEGIN_IGNORE_DEPRECATIONS;
|
|
if (res)
|
|
g_socket_client_set_tls_validation_flags (conn->client, flags);
|
|
G_GNUC_END_IGNORE_DEPRECATIONS;
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_get_tls_validation_flags:
|
|
* @conn: a #GstRTSPConnection
|
|
*
|
|
* Gets the TLS validation flags used to verify the peer certificate
|
|
* when a TLS connection is established.
|
|
*
|
|
* GLib guarantees that if certificate verification fails, at least one error
|
|
* will be set, but it does not guarantee that all possible errors will be
|
|
* set. Accordingly, you may not safely decide to ignore any particular type
|
|
* of error.
|
|
*
|
|
* For example, it would be incorrect to ignore %G_TLS_CERTIFICATE_EXPIRED if
|
|
* you want to allow expired certificates, because this could potentially be
|
|
* the only error flag set even if other problems exist with the certificate.
|
|
*
|
|
* Returns: the validation flags.
|
|
*
|
|
* Since: 1.2.1
|
|
*/
|
|
GTlsCertificateFlags
|
|
gst_rtsp_connection_get_tls_validation_flags (GstRTSPConnection * conn)
|
|
{
|
|
g_return_val_if_fail (conn != NULL, 0);
|
|
|
|
G_GNUC_BEGIN_IGNORE_DEPRECATIONS;
|
|
return g_socket_client_get_tls_validation_flags (conn->client);
|
|
G_GNUC_END_IGNORE_DEPRECATIONS;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_set_tls_database:
|
|
* @conn: a #GstRTSPConnection
|
|
* @database: a #GTlsDatabase
|
|
*
|
|
* Sets the anchor certificate authorities database. This certificate
|
|
* database will be used to verify the server's certificate in case it
|
|
* can't be verified with the default certificate database first.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
void
|
|
gst_rtsp_connection_set_tls_database (GstRTSPConnection * conn,
|
|
GTlsDatabase * database)
|
|
{
|
|
GTlsDatabase *old_db;
|
|
|
|
g_return_if_fail (conn != NULL);
|
|
|
|
if (database)
|
|
g_object_ref (database);
|
|
|
|
old_db = conn->tls_database;
|
|
conn->tls_database = database;
|
|
|
|
if (old_db)
|
|
g_object_unref (old_db);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_get_tls_database:
|
|
* @conn: a #GstRTSPConnection
|
|
*
|
|
* Gets the anchor certificate authorities database that will be used
|
|
* after a server certificate can't be verified with the default
|
|
* certificate database.
|
|
*
|
|
* Returns: (transfer full): the anchor certificate authorities database, or NULL if no
|
|
* database has been previously set. Use g_object_unref() to release the
|
|
* certificate database.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
GTlsDatabase *
|
|
gst_rtsp_connection_get_tls_database (GstRTSPConnection * conn)
|
|
{
|
|
GTlsDatabase *result;
|
|
|
|
g_return_val_if_fail (conn != NULL, NULL);
|
|
|
|
if ((result = conn->tls_database))
|
|
g_object_ref (result);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_set_tls_interaction:
|
|
* @conn: a #GstRTSPConnection
|
|
* @interaction: a #GTlsInteraction
|
|
*
|
|
* Sets a #GTlsInteraction object to be used when the connection or certificate
|
|
* database need to interact with the user. This will be used to prompt the
|
|
* user for passwords where necessary.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
void
|
|
gst_rtsp_connection_set_tls_interaction (GstRTSPConnection * conn,
|
|
GTlsInteraction * interaction)
|
|
{
|
|
GTlsInteraction *old_interaction;
|
|
|
|
g_return_if_fail (conn != NULL);
|
|
|
|
if (interaction)
|
|
g_object_ref (interaction);
|
|
|
|
old_interaction = conn->tls_interaction;
|
|
conn->tls_interaction = interaction;
|
|
|
|
if (old_interaction)
|
|
g_object_unref (old_interaction);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_get_tls_interaction:
|
|
* @conn: a #GstRTSPConnection
|
|
*
|
|
* Gets a #GTlsInteraction object to be used when the connection or certificate
|
|
* database need to interact with the user. This will be used to prompt the
|
|
* user for passwords where necessary.
|
|
*
|
|
* Returns: (transfer full): a reference on the #GTlsInteraction. Use
|
|
* g_object_unref() to release.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
GTlsInteraction *
|
|
gst_rtsp_connection_get_tls_interaction (GstRTSPConnection * conn)
|
|
{
|
|
GTlsInteraction *result;
|
|
|
|
g_return_val_if_fail (conn != NULL, NULL);
|
|
|
|
if ((result = conn->tls_interaction))
|
|
g_object_ref (result);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_set_accept_certificate_func:
|
|
* @conn: a #GstRTSPConnection
|
|
* @func: a #GstRTSPConnectionAcceptCertificateFunc to check certificates
|
|
* @destroy_notify: #GDestroyNotify for @user_data
|
|
* @user_data: User data passed to @func
|
|
*
|
|
* Sets a custom accept-certificate function for checking certificates for
|
|
* validity. This will directly map to #GTlsConnection 's "accept-certificate"
|
|
* signal and be performed after the default checks of #GstRTSPConnection
|
|
* (checking against the #GTlsDatabase with the given #GTlsCertificateFlags)
|
|
* have failed. If no #GTlsDatabase is set on this connection, only @func will
|
|
* be called.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
void
|
|
gst_rtsp_connection_set_accept_certificate_func (GstRTSPConnection * conn,
|
|
GstRTSPConnectionAcceptCertificateFunc func,
|
|
gpointer user_data, GDestroyNotify destroy_notify)
|
|
{
|
|
if (conn->accept_certificate_destroy_notify)
|
|
conn->
|
|
accept_certificate_destroy_notify (conn->accept_certificate_user_data);
|
|
conn->accept_certificate_func = func;
|
|
conn->accept_certificate_user_data = user_data;
|
|
conn->accept_certificate_destroy_notify = destroy_notify;
|
|
}
|
|
|
|
static gchar *
|
|
get_tunneled_connection_uri_strdup (GstRTSPUrl * url, guint16 port)
|
|
{
|
|
const gchar *pre_host = "";
|
|
const gchar *post_host = "";
|
|
|
|
if (url->family == GST_RTSP_FAM_INET6) {
|
|
pre_host = "[";
|
|
post_host = "]";
|
|
}
|
|
|
|
return g_strdup_printf ("http://%s%s%s:%d%s%s%s", pre_host, url->host,
|
|
post_host, port, url->abspath, url->query ? "?" : "",
|
|
url->query ? url->query : "");
|
|
}
|
|
|
|
static GstRTSPResult
|
|
setup_tunneling (GstRTSPConnection * conn, gint64 timeout, gchar * uri,
|
|
GstRTSPMessage * response)
|
|
{
|
|
gint i;
|
|
GstRTSPResult res;
|
|
gchar *value;
|
|
guint16 url_port;
|
|
GstRTSPMessage *msg;
|
|
gboolean old_http;
|
|
GstRTSPUrl *url;
|
|
GError *error = NULL;
|
|
GSocketConnection *connection;
|
|
GSocket *socket;
|
|
gchar *connection_uri = NULL;
|
|
gchar *request_uri = NULL;
|
|
gchar *host = NULL;
|
|
GCancellable *cancellable;
|
|
|
|
url = conn->url;
|
|
|
|
gst_rtsp_url_get_port (url, &url_port);
|
|
host = g_strdup_printf ("%s:%d", url->host, url_port);
|
|
|
|
/* 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';
|
|
|
|
/* 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;
|
|
|
|
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_HOST, host);
|
|
|
|
/* 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_usec (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_usec (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 (!conn->ignore_x_server_reply &&
|
|
gst_rtsp_message_get_header (response, GST_RTSP_HDR_X_SERVER_IP_ADDRESS,
|
|
&value, 0) == GST_RTSP_OK) {
|
|
g_free (url->host);
|
|
url->host = g_strdup (value);
|
|
g_free (conn->remote_ip);
|
|
conn->remote_ip = g_strdup (value);
|
|
}
|
|
|
|
connection_uri = get_tunneled_connection_uri_strdup (url, url_port);
|
|
|
|
cancellable = get_cancellable (conn);
|
|
|
|
/* connect to the host/port */
|
|
if (conn->proxy_host) {
|
|
connection = g_socket_client_connect_to_host (conn->client,
|
|
conn->proxy_host, conn->proxy_port, cancellable, &error);
|
|
request_uri = g_strdup (connection_uri);
|
|
} else {
|
|
connection = g_socket_client_connect_to_uri (conn->client,
|
|
connection_uri, 0, cancellable, &error);
|
|
request_uri =
|
|
g_strdup_printf ("%s%s%s", url->abspath,
|
|
url->query ? "?" : "", url->query ? url->query : "");
|
|
}
|
|
|
|
g_clear_object (&cancellable);
|
|
|
|
if (connection == NULL)
|
|
goto connect_failed;
|
|
|
|
socket = g_socket_connection_get_socket (connection);
|
|
|
|
/* get remote address */
|
|
g_free (conn->remote_ip);
|
|
conn->remote_ip = NULL;
|
|
|
|
if (!collect_addresses (socket, &conn->remote_ip, NULL, TRUE, &error))
|
|
goto remote_address_failed;
|
|
|
|
/* this is now our writing socket */
|
|
conn->stream1 = G_IO_STREAM (connection);
|
|
conn->socket1 = socket;
|
|
conn->write_socket = conn->socket1;
|
|
conn->output_stream = g_io_stream_get_output_stream (conn->stream1);
|
|
conn->control_stream = NULL;
|
|
|
|
/* create the POST request for the write connection */
|
|
GST_RTSP_CHECK (gst_rtsp_message_new_request (&msg, GST_RTSP_POST,
|
|
request_uri), no_message);
|
|
msg->type = GST_RTSP_MESSAGE_HTTP_REQUEST;
|
|
|
|
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_CONTENT_TYPE,
|
|
"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");
|
|
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_HOST, host);
|
|
|
|
/* 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_usec (conn, msg, timeout),
|
|
write_failed);
|
|
gst_rtsp_message_free (msg);
|
|
conn->tunneled = TRUE;
|
|
|
|
exit:
|
|
g_free (connection_uri);
|
|
g_free (request_uri);
|
|
g_free (host);
|
|
|
|
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;
|
|
}
|
|
connect_failed:
|
|
{
|
|
GST_ERROR ("failed to connect: %s", error->message);
|
|
res = gst_rtsp_result_from_g_io_error (error, GST_RTSP_ERROR);
|
|
g_clear_error (&error);
|
|
goto exit;
|
|
}
|
|
remote_address_failed:
|
|
{
|
|
GST_ERROR ("failed to resolve address: %s", error->message);
|
|
res = gst_rtsp_result_from_g_io_error (error, GST_RTSP_ERROR);
|
|
g_object_unref (connection);
|
|
g_clear_error (&error);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_connect_with_response_usec:
|
|
* @conn: a #GstRTSPConnection
|
|
* @timeout: a timeout in microseconds
|
|
* @response: a #GstRTSPMessage
|
|
*
|
|
* Attempt to connect to the url of @conn made with
|
|
* gst_rtsp_connection_create(). If @timeout is 0 this function can block
|
|
* forever. If @timeout contains a valid timeout, this function will return
|
|
* #GST_RTSP_ETIMEOUT after the timeout expired. If @conn is set to tunneled,
|
|
* @response will contain a response to the tunneling request messages.
|
|
*
|
|
* This function can be cancelled with gst_rtsp_connection_flush().
|
|
*
|
|
* Returns: #GST_RTSP_OK when a connection could be made.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_connect_with_response_usec (GstRTSPConnection * conn,
|
|
gint64 timeout, GstRTSPMessage * response)
|
|
{
|
|
GstRTSPResult res;
|
|
GSocketConnection *connection;
|
|
GSocket *socket;
|
|
GError *error = NULL;
|
|
gchar *connection_uri, *request_uri, *remote_ip;
|
|
GstClockTime to;
|
|
guint16 url_port;
|
|
GstRTSPUrl *url;
|
|
GCancellable *cancellable;
|
|
|
|
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->stream0 == NULL, GST_RTSP_EINVAL);
|
|
|
|
to = timeout * 1000;
|
|
g_socket_client_set_timeout (conn->client,
|
|
(to + GST_SECOND - 1) / GST_SECOND);
|
|
|
|
url = conn->url;
|
|
|
|
gst_rtsp_url_get_port (url, &url_port);
|
|
|
|
if (conn->tunneled) {
|
|
connection_uri = get_tunneled_connection_uri_strdup (url, url_port);
|
|
} else {
|
|
connection_uri = gst_rtsp_url_get_request_uri (url);
|
|
}
|
|
|
|
cancellable = get_cancellable (conn);
|
|
|
|
if (conn->proxy_host) {
|
|
connection = g_socket_client_connect_to_host (conn->client,
|
|
conn->proxy_host, conn->proxy_port, cancellable, &error);
|
|
request_uri = g_strdup (connection_uri);
|
|
} else {
|
|
connection = g_socket_client_connect_to_uri (conn->client,
|
|
connection_uri, url_port, cancellable, &error);
|
|
|
|
/* use the relative component of the uri for non-proxy connections */
|
|
request_uri = g_strdup_printf ("%s%s%s", url->abspath,
|
|
url->query ? "?" : "", url->query ? url->query : "");
|
|
}
|
|
|
|
g_clear_object (&cancellable);
|
|
|
|
if (connection == NULL)
|
|
goto connect_failed;
|
|
|
|
/* get remote address */
|
|
socket = g_socket_connection_get_socket (connection);
|
|
|
|
if (!collect_addresses (socket, &remote_ip, NULL, TRUE, &error))
|
|
goto remote_address_failed;
|
|
|
|
g_free (conn->remote_ip);
|
|
conn->remote_ip = remote_ip;
|
|
conn->stream0 = G_IO_STREAM (connection);
|
|
conn->socket0 = socket;
|
|
/* this is our read socket */
|
|
conn->read_socket = conn->socket0;
|
|
conn->write_socket = conn->socket0;
|
|
conn->read_socket_used = FALSE;
|
|
conn->write_socket_used = FALSE;
|
|
conn->input_stream = g_io_stream_get_input_stream (conn->stream0);
|
|
conn->output_stream = g_io_stream_get_output_stream (conn->stream0);
|
|
conn->control_stream = NULL;
|
|
|
|
if (conn->tunneled) {
|
|
res = setup_tunneling (conn, timeout, request_uri, response);
|
|
if (res != GST_RTSP_OK)
|
|
goto tunneling_failed;
|
|
}
|
|
g_free (connection_uri);
|
|
g_free (request_uri);
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
connect_failed:
|
|
{
|
|
GST_ERROR ("failed to connect: %s", error->message);
|
|
res = gst_rtsp_result_from_g_io_error (error, GST_RTSP_ERROR);
|
|
g_clear_error (&error);
|
|
g_free (connection_uri);
|
|
g_free (request_uri);
|
|
return res;
|
|
}
|
|
remote_address_failed:
|
|
{
|
|
GST_ERROR ("failed to connect: %s", error->message);
|
|
res = gst_rtsp_result_from_g_io_error (error, GST_RTSP_ERROR);
|
|
g_object_unref (connection);
|
|
g_clear_error (&error);
|
|
g_free (connection_uri);
|
|
g_free (request_uri);
|
|
return res;
|
|
}
|
|
tunneling_failed:
|
|
{
|
|
GST_ERROR ("failed to setup tunneling");
|
|
g_free (connection_uri);
|
|
g_free (request_uri);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
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;
|
|
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;
|
|
|
|
method = gst_rtsp_method_as_text (message->type_data.request.method);
|
|
uri = message->type_data.request.uri;
|
|
|
|
response =
|
|
gst_rtsp_generate_digest_auth_response (NULL, method, realm,
|
|
conn->username, conn->passwd, uri, nonce);
|
|
auth_string =
|
|
g_strdup_printf ("Digest username=\"%s\", "
|
|
"realm=\"%s\", nonce=\"%s\", uri=\"%s\", response=\"%s\"",
|
|
conn->username, realm, nonce, uri, response);
|
|
g_free (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;
|
|
}
|
|
/* Do not keep any old Authorization headers */
|
|
gst_rtsp_message_remove_header (message, GST_RTSP_HDR_AUTHORIZATION, -1);
|
|
gst_rtsp_message_take_header (message, GST_RTSP_HDR_AUTHORIZATION,
|
|
auth_string);
|
|
break;
|
|
}
|
|
default:
|
|
/* Nothing to do */
|
|
break;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_connect_usec:
|
|
* @conn: a #GstRTSPConnection
|
|
* @timeout: a timeout in microseconds
|
|
*
|
|
* Attempt to connect to the url of @conn made with
|
|
* gst_rtsp_connection_create(). If @timeout is 0 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.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_connect_usec (GstRTSPConnection * conn, gint64 timeout)
|
|
{
|
|
GstRTSPResult result;
|
|
GstRTSPMessage response;
|
|
|
|
memset (&response, 0, sizeof (response));
|
|
gst_rtsp_message_init (&response);
|
|
|
|
result = gst_rtsp_connection_connect_with_response_usec (conn, timeout,
|
|
&response);
|
|
|
|
gst_rtsp_message_unset (&response);
|
|
|
|
return result;
|
|
}
|
|
|
|
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 (GOutputStream * stream, const guint8 * buffer, guint * idx,
|
|
guint size, gboolean block, GCancellable * cancellable)
|
|
{
|
|
guint left;
|
|
gssize r;
|
|
GstRTSPResult res;
|
|
GError *err = NULL;
|
|
|
|
if (G_UNLIKELY (*idx > size))
|
|
return GST_RTSP_ERROR;
|
|
|
|
left = size - *idx;
|
|
|
|
while (left) {
|
|
if (block)
|
|
r = g_output_stream_write (stream, (gchar *) & buffer[*idx], left,
|
|
cancellable, &err);
|
|
else
|
|
r = g_pollable_output_stream_write_nonblocking (G_POLLABLE_OUTPUT_STREAM
|
|
(stream), (gchar *) & buffer[*idx], left, cancellable, &err);
|
|
if (G_UNLIKELY (r < 0))
|
|
goto error;
|
|
|
|
left -= r;
|
|
*idx += r;
|
|
}
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
g_object_unref (cancellable);
|
|
|
|
if (G_UNLIKELY (r == 0))
|
|
return GST_RTSP_EEOF;
|
|
|
|
if (!g_error_matches (err, G_IO_ERROR, G_IO_ERROR_WOULD_BLOCK))
|
|
GST_WARNING ("%s", err->message);
|
|
else
|
|
GST_DEBUG ("%s", err->message);
|
|
|
|
res = gst_rtsp_result_from_g_io_error (err, GST_RTSP_ESYS);
|
|
g_clear_error (&err);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
/* NOTE: This changes the values of vectors if multiple iterations are needed! */
|
|
static GstRTSPResult
|
|
writev_bytes (GOutputStream * stream, GOutputVector * vectors, gint n_vectors,
|
|
gsize * bytes_written, gboolean block, GCancellable * cancellable)
|
|
{
|
|
gsize _bytes_written = 0;
|
|
gsize written;
|
|
GstRTSPResult ret;
|
|
GError *err = NULL;
|
|
GPollableReturn res = G_POLLABLE_RETURN_OK;
|
|
|
|
while (n_vectors > 0) {
|
|
if (block) {
|
|
if (G_UNLIKELY (!g_output_stream_writev (stream, vectors, n_vectors,
|
|
&written, cancellable, &err))) {
|
|
/* This will never return G_IO_ERROR_WOULD_BLOCK */
|
|
res = G_POLLABLE_RETURN_FAILED;
|
|
goto error;
|
|
}
|
|
} else {
|
|
res =
|
|
g_pollable_output_stream_writev_nonblocking (G_POLLABLE_OUTPUT_STREAM
|
|
(stream), vectors, n_vectors, &written, cancellable, &err);
|
|
|
|
if (res != G_POLLABLE_RETURN_OK) {
|
|
g_assert (written == 0);
|
|
goto error;
|
|
}
|
|
}
|
|
_bytes_written += written;
|
|
|
|
/* skip vectors that have been written in full */
|
|
while (written > 0 && written >= vectors[0].size) {
|
|
written -= vectors[0].size;
|
|
++vectors;
|
|
--n_vectors;
|
|
}
|
|
|
|
/* skip partially written vector data */
|
|
if (written > 0) {
|
|
vectors[0].size -= written;
|
|
vectors[0].buffer = ((guint8 *) vectors[0].buffer) + written;
|
|
}
|
|
}
|
|
|
|
*bytes_written = _bytes_written;
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
*bytes_written = _bytes_written;
|
|
|
|
if (err)
|
|
GST_WARNING ("%s", err->message);
|
|
if (res == G_POLLABLE_RETURN_WOULD_BLOCK) {
|
|
g_assert (!err);
|
|
return GST_RTSP_EINTR;
|
|
} else if (G_UNLIKELY (written == 0)) {
|
|
g_clear_error (&err);
|
|
return GST_RTSP_EEOF;
|
|
}
|
|
|
|
ret = gst_rtsp_result_from_g_io_error (err, GST_RTSP_ESYS);
|
|
g_clear_error (&err);
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
static gint
|
|
fill_raw_bytes (GstRTSPConnection * conn, guint8 * buffer, guint size,
|
|
gboolean block, GError ** err)
|
|
{
|
|
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)) {
|
|
gssize r;
|
|
gsize count = size - out;
|
|
GCancellable *cancellable;
|
|
|
|
cancellable = conn->may_cancel ? get_cancellable (conn) : NULL;
|
|
|
|
if (block)
|
|
r = g_input_stream_read (conn->input_stream, (gchar *) & buffer[out],
|
|
count, cancellable, err);
|
|
else
|
|
r = g_pollable_input_stream_read_nonblocking (G_POLLABLE_INPUT_STREAM
|
|
(conn->input_stream), (gchar *) & buffer[out], count,
|
|
cancellable, err);
|
|
|
|
g_clear_object (&cancellable);
|
|
|
|
if (G_UNLIKELY (r < 0)) {
|
|
if (out == 0) {
|
|
/* propagate the error */
|
|
out = r;
|
|
} else {
|
|
/* we have some data ignore error */
|
|
g_clear_error (err);
|
|
}
|
|
} else
|
|
out += r;
|
|
}
|
|
|
|
return out;
|
|
}
|
|
|
|
static gint
|
|
fill_bytes (GstRTSPConnection * conn, guint8 * buffer, guint size,
|
|
gboolean block, GError ** err)
|
|
{
|
|
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), block, err);
|
|
if (r <= 0) {
|
|
if (out == 0) {
|
|
out = r;
|
|
} else {
|
|
/* we have some data ignore error */
|
|
g_clear_error (err);
|
|
}
|
|
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, block, err);
|
|
}
|
|
|
|
return out;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
read_bytes (GstRTSPConnection * conn, guint8 * buffer, guint * idx, guint size,
|
|
gboolean block)
|
|
{
|
|
guint left;
|
|
gint r;
|
|
GstRTSPResult res;
|
|
GError *err = NULL;
|
|
|
|
if (G_UNLIKELY (*idx > size))
|
|
return GST_RTSP_ERROR;
|
|
|
|
left = size - *idx;
|
|
|
|
while (left) {
|
|
r = fill_bytes (conn, &buffer[*idx], left, block, &err);
|
|
if (G_UNLIKELY (r <= 0))
|
|
goto error;
|
|
|
|
left -= r;
|
|
*idx += r;
|
|
}
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
if (G_UNLIKELY (r == 0))
|
|
return GST_RTSP_EEOF;
|
|
|
|
GST_DEBUG ("%s", err->message);
|
|
res = gst_rtsp_result_from_g_io_error (err, GST_RTSP_ESYS);
|
|
g_clear_error (&err);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
/* 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,
|
|
gboolean block)
|
|
{
|
|
GstRTSPResult res;
|
|
|
|
while (TRUE) {
|
|
guint8 c;
|
|
guint i;
|
|
|
|
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 */
|
|
i = 0;
|
|
res = read_bytes (conn, &c, &i, 1, block);
|
|
if (G_UNLIKELY (res != GST_RTSP_OK))
|
|
return res;
|
|
}
|
|
|
|
/* 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... */
|
|
i = 0;
|
|
res = read_bytes (conn, &read_ahead, &i, 1, block);
|
|
if (G_UNLIKELY (res != GST_RTSP_OK))
|
|
return res;
|
|
|
|
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;
|
|
}
|
|
|
|
static void
|
|
set_read_socket_timeout (GstRTSPConnection * conn, gint64 timeout)
|
|
{
|
|
GstClockTime to_nsecs;
|
|
guint to_secs;
|
|
|
|
g_mutex_lock (&conn->socket_use_mutex);
|
|
|
|
g_assert (!conn->read_socket_used);
|
|
conn->read_socket_used = TRUE;
|
|
|
|
to_nsecs = timeout * 1000;
|
|
to_secs = (to_nsecs + GST_SECOND - 1) / GST_SECOND;
|
|
|
|
if (to_secs > g_socket_get_timeout (conn->read_socket)) {
|
|
g_socket_set_timeout (conn->read_socket, to_secs);
|
|
}
|
|
|
|
g_mutex_unlock (&conn->socket_use_mutex);
|
|
}
|
|
|
|
static void
|
|
set_write_socket_timeout (GstRTSPConnection * conn, gint64 timeout)
|
|
{
|
|
GstClockTime to_nsecs;
|
|
guint to_secs;
|
|
|
|
g_mutex_lock (&conn->socket_use_mutex);
|
|
|
|
g_assert (!conn->write_socket_used);
|
|
conn->write_socket_used = TRUE;
|
|
|
|
to_nsecs = timeout * 1000;
|
|
to_secs = (to_nsecs + GST_SECOND - 1) / GST_SECOND;
|
|
|
|
if (to_secs > g_socket_get_timeout (conn->write_socket)) {
|
|
g_socket_set_timeout (conn->write_socket, to_secs);
|
|
}
|
|
|
|
g_mutex_unlock (&conn->socket_use_mutex);
|
|
}
|
|
|
|
static void
|
|
clear_read_socket_timeout (GstRTSPConnection * conn)
|
|
{
|
|
g_mutex_lock (&conn->socket_use_mutex);
|
|
|
|
conn->read_socket_used = FALSE;
|
|
if (conn->read_socket != conn->write_socket || !conn->write_socket_used) {
|
|
g_socket_set_timeout (conn->read_socket, 0);
|
|
}
|
|
|
|
g_mutex_unlock (&conn->socket_use_mutex);
|
|
}
|
|
|
|
static void
|
|
clear_write_socket_timeout (GstRTSPConnection * conn)
|
|
{
|
|
g_mutex_lock (&conn->socket_use_mutex);
|
|
|
|
conn->write_socket_used = FALSE;
|
|
if (conn->write_socket != conn->read_socket || !conn->read_socket_used) {
|
|
g_socket_set_timeout (conn->write_socket, 0);
|
|
}
|
|
|
|
g_mutex_unlock (&conn->socket_use_mutex);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_write_usec:
|
|
* @conn: a #GstRTSPConnection
|
|
* @data: the data to write
|
|
* @size: the size of @data
|
|
* @timeout: a timeout value or 0
|
|
*
|
|
* Attempt to write @size bytes of @data to the connected @conn, blocking up to
|
|
* the specified @timeout. @timeout can be 0, 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: 1.18
|
|
*/
|
|
/* FIXME 2.0: This should've been static! */
|
|
GstRTSPResult
|
|
gst_rtsp_connection_write_usec (GstRTSPConnection * conn, const guint8 * data,
|
|
guint size, gint64 timeout)
|
|
{
|
|
guint offset;
|
|
GstRTSPResult res;
|
|
GCancellable *cancellable;
|
|
|
|
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->output_stream != NULL, GST_RTSP_EINVAL);
|
|
|
|
offset = 0;
|
|
|
|
set_write_socket_timeout (conn, timeout);
|
|
|
|
cancellable = get_cancellable (conn);
|
|
res =
|
|
write_bytes (conn->output_stream, data, &offset, size, TRUE, cancellable);
|
|
g_clear_object (&cancellable);
|
|
|
|
clear_write_socket_timeout (conn);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
serialize_message (GstRTSPConnection * conn, GstRTSPMessage * message,
|
|
GstRTSPSerializedMessage * serialized_message)
|
|
{
|
|
GString *str = NULL;
|
|
|
|
memset (serialized_message, 0, sizeof (*serialized_message));
|
|
|
|
/* Initially we borrow the body_data / body_buffer fields from
|
|
* the message */
|
|
serialized_message->borrowed = TRUE;
|
|
|
|
switch (message->type) {
|
|
case GST_RTSP_MESSAGE_REQUEST:
|
|
str = g_string_new ("");
|
|
|
|
/* create request string, add CSeq */
|
|
g_string_append_printf (str, "%s %s RTSP/%s\r\n"
|
|
"CSeq: %d\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),
|
|
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:
|
|
str = g_string_new ("");
|
|
|
|
/* create response string */
|
|
g_string_append_printf (str, "RTSP/%s %d %s\r\n",
|
|
gst_rtsp_version_as_text (message->type_data.response.version),
|
|
message->type_data.response.code, message->type_data.response.reason);
|
|
break;
|
|
case GST_RTSP_MESSAGE_HTTP_REQUEST:
|
|
str = g_string_new ("");
|
|
|
|
/* 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:
|
|
str = g_string_new ("");
|
|
|
|
/* 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 = serialized_message->data_header;
|
|
|
|
/* 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 serialized message with header and data */
|
|
serialized_message->data_is_data_header = TRUE;
|
|
serialized_message->data_size = 4;
|
|
|
|
if (message->body) {
|
|
serialized_message->body_data = message->body;
|
|
serialized_message->body_data_size = message->body_size;
|
|
} else {
|
|
g_assert (message->body_buffer != NULL);
|
|
serialized_message->body_buffer = message->body_buffer;
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
g_string_free (str, TRUE);
|
|
g_return_val_if_reached (FALSE);
|
|
break;
|
|
}
|
|
|
|
/* append headers and body */
|
|
if (message->type != GST_RTSP_MESSAGE_DATA) {
|
|
gchar date_string[100];
|
|
|
|
g_assert (str != NULL);
|
|
|
|
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_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");
|
|
|
|
if (message->body) {
|
|
serialized_message->body_data = message->body;
|
|
serialized_message->body_data_size = message->body_size;
|
|
} else {
|
|
g_assert (message->body_buffer != NULL);
|
|
serialized_message->body_buffer = message->body_buffer;
|
|
}
|
|
} else {
|
|
/* just end headers */
|
|
g_string_append (str, "\r\n");
|
|
}
|
|
|
|
serialized_message->data_size = str->len;
|
|
serialized_message->data = (guint8 *) g_string_free (str, FALSE);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_send_usec:
|
|
* @conn: a #GstRTSPConnection
|
|
* @message: the message to send
|
|
* @timeout: a timeout value in microseconds
|
|
*
|
|
* Attempt to send @message to the connected @conn, blocking up to
|
|
* the specified @timeout. @timeout can be 0, 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: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_send_usec (GstRTSPConnection * conn,
|
|
GstRTSPMessage * message, gint64 timeout)
|
|
{
|
|
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
|
|
|
|
return gst_rtsp_connection_send_messages_usec (conn, message, 1, timeout);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_send_messages_usec:
|
|
* @conn: a #GstRTSPConnection
|
|
* @messages: (array length=n_messages): the messages to send
|
|
* @n_messages: the number of messages to send
|
|
* @timeout: a timeout value in microseconds
|
|
*
|
|
* Attempt to send @messages to the connected @conn, blocking up to
|
|
* the specified @timeout. @timeout can be 0, in which case this function
|
|
* might block forever.
|
|
*
|
|
* This function can be cancelled with gst_rtsp_connection_flush().
|
|
*
|
|
* Returns: #GST_RTSP_OK on Since.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_send_messages_usec (GstRTSPConnection * conn,
|
|
GstRTSPMessage * messages, guint n_messages, gint64 timeout)
|
|
{
|
|
GstRTSPResult res;
|
|
GstRTSPSerializedMessage *serialized_messages;
|
|
GOutputVector *vectors;
|
|
GstMapInfo *map_infos;
|
|
guint n_vectors, n_memories;
|
|
gint i, j, k;
|
|
gsize bytes_to_write, bytes_written;
|
|
GCancellable *cancellable;
|
|
|
|
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (messages != NULL || n_messages == 0, GST_RTSP_EINVAL);
|
|
|
|
serialized_messages = g_newa (GstRTSPSerializedMessage, n_messages);
|
|
memset (serialized_messages, 0,
|
|
sizeof (GstRTSPSerializedMessage) * n_messages);
|
|
|
|
for (i = 0, n_vectors = 0, n_memories = 0, bytes_to_write = 0; i < n_messages;
|
|
i++) {
|
|
if (G_UNLIKELY (!serialize_message (conn, &messages[i],
|
|
&serialized_messages[i])))
|
|
goto no_message;
|
|
|
|
if (conn->tunneled) {
|
|
gint state = 0, save = 0;
|
|
gchar *base64_buffer, *out_buffer;
|
|
gsize written = 0;
|
|
gsize in_length;
|
|
|
|
in_length = serialized_messages[i].data_size;
|
|
if (serialized_messages[i].body_data)
|
|
in_length += serialized_messages[i].body_data_size;
|
|
else if (serialized_messages[i].body_buffer)
|
|
in_length += gst_buffer_get_size (serialized_messages[i].body_buffer);
|
|
|
|
in_length = (in_length / 3 + 1) * 4 + 4 + 1;
|
|
base64_buffer = out_buffer = g_malloc0 (in_length);
|
|
|
|
written =
|
|
g_base64_encode_step (serialized_messages[i].data_is_data_header ?
|
|
serialized_messages[i].data_header : serialized_messages[i].data,
|
|
serialized_messages[i].data_size, FALSE, out_buffer, &state, &save);
|
|
out_buffer += written;
|
|
|
|
if (serialized_messages[i].body_data) {
|
|
written =
|
|
g_base64_encode_step (serialized_messages[i].body_data,
|
|
serialized_messages[i].body_data_size, FALSE, out_buffer, &state,
|
|
&save);
|
|
out_buffer += written;
|
|
} else if (serialized_messages[i].body_buffer) {
|
|
guint j, n = gst_buffer_n_memory (serialized_messages[i].body_buffer);
|
|
|
|
for (j = 0; j < n; j++) {
|
|
GstMemory *mem =
|
|
gst_buffer_peek_memory (serialized_messages[i].body_buffer, j);
|
|
GstMapInfo map;
|
|
|
|
gst_memory_map (mem, &map, GST_MAP_READ);
|
|
|
|
written = g_base64_encode_step (map.data, map.size,
|
|
FALSE, out_buffer, &state, &save);
|
|
out_buffer += written;
|
|
|
|
gst_memory_unmap (mem, &map);
|
|
}
|
|
}
|
|
|
|
written = g_base64_encode_close (FALSE, out_buffer, &state, &save);
|
|
out_buffer += written;
|
|
|
|
gst_rtsp_serialized_message_clear (&serialized_messages[i]);
|
|
memset (&serialized_messages[i], 0, sizeof (serialized_messages[i]));
|
|
|
|
serialized_messages[i].data = (guint8 *) base64_buffer;
|
|
serialized_messages[i].data_size = (out_buffer - base64_buffer);
|
|
n_vectors++;
|
|
} else {
|
|
n_vectors++;
|
|
if (serialized_messages[i].body_data) {
|
|
n_vectors++;
|
|
} else if (serialized_messages[i].body_buffer) {
|
|
n_vectors += gst_buffer_n_memory (serialized_messages[i].body_buffer);
|
|
n_memories += gst_buffer_n_memory (serialized_messages[i].body_buffer);
|
|
}
|
|
}
|
|
}
|
|
|
|
vectors = g_newa (GOutputVector, n_vectors);
|
|
map_infos = n_memories ? g_newa (GstMapInfo, n_memories) : NULL;
|
|
|
|
for (i = 0, j = 0, k = 0; i < n_messages; i++) {
|
|
vectors[j].buffer = serialized_messages[i].data_is_data_header ?
|
|
serialized_messages[i].data_header : serialized_messages[i].data;
|
|
vectors[j].size = serialized_messages[i].data_size;
|
|
bytes_to_write += vectors[j].size;
|
|
j++;
|
|
|
|
if (serialized_messages[i].body_data) {
|
|
vectors[j].buffer = serialized_messages[i].body_data;
|
|
vectors[j].size = serialized_messages[i].body_data_size;
|
|
bytes_to_write += vectors[j].size;
|
|
j++;
|
|
} else if (serialized_messages[i].body_buffer) {
|
|
gint l, n;
|
|
|
|
n = gst_buffer_n_memory (serialized_messages[i].body_buffer);
|
|
for (l = 0; l < n; l++) {
|
|
GstMemory *mem =
|
|
gst_buffer_peek_memory (serialized_messages[i].body_buffer, l);
|
|
|
|
gst_memory_map (mem, &map_infos[k], GST_MAP_READ);
|
|
vectors[j].buffer = map_infos[k].data;
|
|
vectors[j].size = map_infos[k].size;
|
|
bytes_to_write += vectors[j].size;
|
|
|
|
k++;
|
|
j++;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* write request: this is synchronous */
|
|
set_write_socket_timeout (conn, timeout);
|
|
|
|
cancellable = get_cancellable (conn);
|
|
res =
|
|
writev_bytes (conn->output_stream, vectors, n_vectors, &bytes_written,
|
|
TRUE, cancellable);
|
|
g_clear_object (&cancellable);
|
|
|
|
clear_write_socket_timeout (conn);
|
|
|
|
g_assert (bytes_written == bytes_to_write || res != GST_RTSP_OK);
|
|
|
|
/* free everything */
|
|
for (i = 0, k = 0; i < n_messages; i++) {
|
|
if (serialized_messages[i].body_buffer) {
|
|
gint l, n;
|
|
|
|
n = gst_buffer_n_memory (serialized_messages[i].body_buffer);
|
|
for (l = 0; l < n; l++) {
|
|
GstMemory *mem =
|
|
gst_buffer_peek_memory (serialized_messages[i].body_buffer, l);
|
|
|
|
gst_memory_unmap (mem, &map_infos[k]);
|
|
k++;
|
|
}
|
|
}
|
|
|
|
g_free (serialized_messages[i].data);
|
|
}
|
|
|
|
return res;
|
|
|
|
no_message:
|
|
{
|
|
for (i = 0; i < n_messages; i++) {
|
|
gst_rtsp_serialized_message_clear (&serialized_messages[i]);
|
|
}
|
|
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)
|
|
{
|
|
GstRTSPVersion rversion;
|
|
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;
|
|
|
|
rversion = major * 0x10 + minor;
|
|
if (g_ascii_strcasecmp (protocol, "RTSP") == 0) {
|
|
|
|
if (rversion != GST_RTSP_VERSION_1_0 && rversion != GST_RTSP_VERSION_2_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 (rversion != GST_RTSP_VERSION_1_0 &&
|
|
rversion != GST_RTSP_VERSION_1_1 && rversion != GST_RTSP_VERSION_2_0)
|
|
res = GST_RTSP_ERROR;
|
|
} else
|
|
res = GST_RTSP_EPARSE;
|
|
} else
|
|
res = GST_RTSP_EPARSE;
|
|
|
|
if (res == GST_RTSP_OK)
|
|
*version = rversion;
|
|
|
|
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 *field_name = NULL;
|
|
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);
|
|
/* custom header not present in the list of pre-defined headers */
|
|
if (field == GST_RTSP_HDR_INVALID)
|
|
field_name = line;
|
|
|
|
/* 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++;
|
|
}
|
|
|
|
if (msg->type == GST_RTSP_MESSAGE_REQUEST && field == GST_RTSP_HDR_SESSION) {
|
|
/* The timeout parameter is only allowed in a session response header
|
|
* but some clients send it as part of the session request header.
|
|
* Ignore everything from the semicolon to the end of the line. */
|
|
next_value = value;
|
|
while (*next_value != '\0') {
|
|
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') {
|
|
if (field != GST_RTSP_HDR_INVALID)
|
|
gst_rtsp_message_add_header (msg, field, value);
|
|
else
|
|
gst_rtsp_message_add_header_by_name (msg, field_name, value);
|
|
}
|
|
|
|
value = next_value;
|
|
}
|
|
|
|
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++;
|
|
}
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
cseq_validation (GstRTSPConnection * conn, GstRTSPMessage * message)
|
|
{
|
|
gchar *cseq_header;
|
|
gint64 cseq = 0;
|
|
GstRTSPResult res;
|
|
|
|
if (message->type == GST_RTSP_MESSAGE_RESPONSE ||
|
|
message->type == GST_RTSP_MESSAGE_REQUEST) {
|
|
if ((res = gst_rtsp_message_get_header (message, GST_RTSP_HDR_CSEQ,
|
|
&cseq_header, 0)) != GST_RTSP_OK) {
|
|
/* rfc2326 This field MUST be present in all RTSP req and resp */
|
|
goto invalid_format;
|
|
}
|
|
|
|
errno = 0;
|
|
cseq = g_ascii_strtoll (cseq_header, NULL, 10);
|
|
if (errno != 0 || cseq < 0) {
|
|
/* CSeq has no valid value */
|
|
goto invalid_format;
|
|
}
|
|
|
|
if (message->type == GST_RTSP_MESSAGE_RESPONSE &&
|
|
(conn->cseq == 0 || conn->cseq < cseq)) {
|
|
/* Response CSeq can't be higher than the number of outgoing requests
|
|
* neither is a response valid if no request has been made */
|
|
goto invalid_format;
|
|
}
|
|
}
|
|
return GST_RTSP_OK;
|
|
|
|
invalid_format:
|
|
{
|
|
return GST_RTSP_EPARSE;
|
|
}
|
|
}
|
|
|
|
/* 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 occurred.
|
|
*/
|
|
static GstRTSPResult
|
|
build_next (GstRTSPBuilder * builder, GstRTSPMessage * message,
|
|
GstRTSPConnection * conn, gboolean block)
|
|
{
|
|
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,
|
|
block);
|
|
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;
|
|
conn->may_cancel = FALSE;
|
|
} else if (c == '\n' || c == '\r') {
|
|
/* skip \n and \r */
|
|
builder->offset = 0;
|
|
} else {
|
|
builder->line = 0;
|
|
builder->state = STATE_READ_LINES;
|
|
conn->may_cancel = FALSE;
|
|
}
|
|
break;
|
|
}
|
|
case STATE_DATA_HEADER:
|
|
{
|
|
res =
|
|
read_bytes (conn, (guint8 *) builder->buffer, &builder->offset, 4,
|
|
block);
|
|
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, block);
|
|
if (res != GST_RTSP_OK)
|
|
goto done;
|
|
|
|
/* we have the complete body now, store in the message adjusting the
|
|
* length to include the trailing '\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), block);
|
|
if (res != GST_RTSP_OK)
|
|
goto done;
|
|
|
|
/* we have a regular response */
|
|
if (builder->buffer[0] == '\0') {
|
|
gchar *hdrval;
|
|
gint64 content_length_parsed = 0;
|
|
|
|
/* 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 */
|
|
errno = 0;
|
|
content_length_parsed = g_ascii_strtoll (hdrval, NULL, 10);
|
|
if (errno != 0 || content_length_parsed < 0) {
|
|
res = GST_RTSP_EPARSE;
|
|
goto invalid_body_len;
|
|
} else if (content_length_parsed > conn->content_length_limit) {
|
|
res = GST_RTSP_ENOMEM;
|
|
goto invalid_body_len;
|
|
}
|
|
builder->body_len = content_length_parsed;
|
|
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, we can't read the payload. */
|
|
if (builder->body_data == NULL) {
|
|
res = GST_RTSP_ENOMEM;
|
|
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;
|
|
}
|
|
if (builder->status != GST_RTSP_OK) {
|
|
res = builder->status;
|
|
goto invalid_format;
|
|
}
|
|
|
|
builder->line++;
|
|
builder->offset = 0;
|
|
break;
|
|
}
|
|
case STATE_END:
|
|
{
|
|
gchar *session_cookie;
|
|
gchar *session_id;
|
|
|
|
conn->may_cancel = TRUE;
|
|
|
|
if ((res = cseq_validation (conn, message)) != GST_RTSP_OK) {
|
|
/* message don't comply with rfc2326 regarding CSeq */
|
|
goto invalid_format;
|
|
}
|
|
|
|
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; i < maxlen && 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 */
|
|
if (conn->remember_session_id) {
|
|
strncpy (conn->session_id, session_id, maxlen);
|
|
conn->session_id[maxlen] = '\0';
|
|
}
|
|
}
|
|
res = builder->status;
|
|
goto done;
|
|
}
|
|
default:
|
|
res = GST_RTSP_ERROR;
|
|
goto done;
|
|
}
|
|
}
|
|
done:
|
|
conn->may_cancel = TRUE;
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
invalid_body_len:
|
|
{
|
|
conn->may_cancel = TRUE;
|
|
GST_DEBUG ("could not allocate body");
|
|
return res;
|
|
}
|
|
invalid_format:
|
|
{
|
|
conn->may_cancel = TRUE;
|
|
GST_DEBUG ("could not parse");
|
|
return res;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_read_usec:
|
|
* @conn: a #GstRTSPConnection
|
|
* @data: the data to read
|
|
* @size: the size of @data
|
|
* @timeout: a timeout value in microseconds
|
|
*
|
|
* Attempt to read @size bytes into @data from the connected @conn, blocking up to
|
|
* the specified @timeout. @timeout can be 0, 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: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_read_usec (GstRTSPConnection * conn, guint8 * data,
|
|
guint size, gint64 timeout)
|
|
{
|
|
guint offset;
|
|
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->read_socket != NULL, GST_RTSP_EINVAL);
|
|
|
|
if (G_UNLIKELY (size == 0))
|
|
return GST_RTSP_OK;
|
|
|
|
offset = 0;
|
|
|
|
/* configure timeout if any */
|
|
set_read_socket_timeout (conn, timeout);
|
|
|
|
res = read_bytes (conn, data, &offset, size, TRUE);
|
|
|
|
clear_read_socket_timeout (conn);
|
|
|
|
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) {
|
|
/* add the local ip address to the tunnel reply, this is where the client
|
|
* should send the POST request to */
|
|
if (conn->local_ip)
|
|
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_X_SERVER_IP_ADDRESS,
|
|
conn->local_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_usec:
|
|
* @conn: a #GstRTSPConnection
|
|
* @message: the message to read
|
|
* @timeout: a timeout value or 0
|
|
*
|
|
* Attempt to read into @message from the connected @conn, blocking up to
|
|
* the specified @timeout. @timeout can be 0, 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: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_receive_usec (GstRTSPConnection * conn,
|
|
GstRTSPMessage * message, gint64 timeout)
|
|
{
|
|
GstRTSPResult res;
|
|
GstRTSPBuilder builder;
|
|
|
|
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->read_socket != NULL, GST_RTSP_EINVAL);
|
|
|
|
/* configure timeout if any */
|
|
set_read_socket_timeout (conn, timeout);
|
|
|
|
memset (&builder, 0, sizeof (GstRTSPBuilder));
|
|
res = build_next (&builder, message, conn, TRUE);
|
|
|
|
clear_read_socket_timeout (conn);
|
|
|
|
if (G_UNLIKELY (res != GST_RTSP_OK))
|
|
goto read_error;
|
|
|
|
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_usec (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;
|
|
}
|
|
}
|
|
|
|
/* we have a message here */
|
|
build_reset (&builder);
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
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);
|
|
|
|
/* last unref closes the connection we don't want to explicitly close here
|
|
* because these sockets might have been provided at construction */
|
|
if (conn->stream0) {
|
|
g_object_unref (conn->stream0);
|
|
conn->stream0 = NULL;
|
|
conn->socket0 = NULL;
|
|
}
|
|
if (conn->stream1) {
|
|
g_object_unref (conn->stream1);
|
|
conn->stream1 = NULL;
|
|
conn->socket1 = NULL;
|
|
}
|
|
|
|
/* these were owned by the stream */
|
|
conn->input_stream = NULL;
|
|
conn->output_stream = NULL;
|
|
conn->control_stream = NULL;
|
|
|
|
g_free (conn->remote_ip);
|
|
conn->remote_ip = NULL;
|
|
g_free (conn->local_ip);
|
|
conn->local_ip = NULL;
|
|
|
|
conn->read_ahead = 0;
|
|
|
|
g_free (conn->initial_buffer);
|
|
conn->initial_buffer = NULL;
|
|
conn->initial_buffer_offset = 0;
|
|
|
|
conn->write_socket = NULL;
|
|
conn->read_socket = NULL;
|
|
conn->write_socket_used = FALSE;
|
|
conn->read_socket_used = FALSE;
|
|
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);
|
|
|
|
g_mutex_lock (&conn->cancellable_mutex);
|
|
g_clear_object (&conn->cancellable);
|
|
g_mutex_unlock (&conn->cancellable_mutex);
|
|
g_mutex_clear (&conn->cancellable_mutex);
|
|
if (conn->client)
|
|
g_object_unref (conn->client);
|
|
if (conn->tls_database)
|
|
g_object_unref (conn->tls_database);
|
|
if (conn->tls_interaction)
|
|
g_object_unref (conn->tls_interaction);
|
|
if (conn->accept_certificate_destroy_notify)
|
|
conn->
|
|
accept_certificate_destroy_notify (conn->accept_certificate_user_data);
|
|
|
|
g_timer_destroy (conn->timer);
|
|
gst_rtsp_url_free (conn->url);
|
|
g_free (conn->proxy_host);
|
|
g_free (conn);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_poll_usec:
|
|
* @conn: a #GstRTSPConnection
|
|
* @events: a bitmask of #GstRTSPEvent flags to check
|
|
* @revents: location for result flags
|
|
* @timeout: a timeout in microseconds
|
|
*
|
|
* 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 0, 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: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_poll_usec (GstRTSPConnection * conn, GstRTSPEvent events,
|
|
GstRTSPEvent * revents, gint64 timeout)
|
|
{
|
|
GMainContext *ctx;
|
|
GSource *rs, *ws, *ts;
|
|
GIOCondition condition;
|
|
GCancellable *cancellable;
|
|
|
|
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->read_socket != NULL, GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (conn->write_socket != NULL, GST_RTSP_EINVAL);
|
|
|
|
ctx = g_main_context_new ();
|
|
|
|
/* configure timeout if any */
|
|
if (timeout) {
|
|
ts = g_timeout_source_new (timeout / 1000);
|
|
g_source_set_dummy_callback (ts);
|
|
g_source_attach (ts, ctx);
|
|
g_source_unref (ts);
|
|
}
|
|
|
|
cancellable = get_cancellable (conn);
|
|
if (events & GST_RTSP_EV_READ) {
|
|
rs = g_socket_create_source (conn->read_socket, G_IO_IN | G_IO_PRI,
|
|
cancellable);
|
|
g_source_set_dummy_callback (rs);
|
|
g_source_attach (rs, ctx);
|
|
g_source_unref (rs);
|
|
}
|
|
|
|
if (events & GST_RTSP_EV_WRITE) {
|
|
ws = g_socket_create_source (conn->write_socket, G_IO_OUT, cancellable);
|
|
g_source_set_dummy_callback (ws);
|
|
g_source_attach (ws, ctx);
|
|
g_source_unref (ws);
|
|
}
|
|
g_clear_object (&cancellable);
|
|
|
|
/* Returns after handling all pending events */
|
|
while (!g_main_context_iteration (ctx, TRUE));
|
|
|
|
g_main_context_unref (ctx);
|
|
|
|
*revents = 0;
|
|
if (events & GST_RTSP_EV_READ) {
|
|
condition = g_socket_condition_check (conn->read_socket,
|
|
G_IO_IN | G_IO_PRI);
|
|
if ((condition & G_IO_IN) || (condition & G_IO_PRI))
|
|
*revents |= GST_RTSP_EV_READ;
|
|
}
|
|
if (events & GST_RTSP_EV_WRITE) {
|
|
condition = g_socket_condition_check (conn->write_socket, G_IO_OUT);
|
|
if ((condition & G_IO_OUT))
|
|
*revents |= GST_RTSP_EV_WRITE;
|
|
}
|
|
|
|
if (*revents == 0)
|
|
return GST_RTSP_ETIMEOUT;
|
|
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_next_timeout_usec:
|
|
* @conn: a #GstRTSPConnection
|
|
*
|
|
* Calculate the next timeout for @conn
|
|
*
|
|
* Returns: #the next timeout in microseconds
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
gint64
|
|
gst_rtsp_connection_next_timeout_usec (GstRTSPConnection * conn)
|
|
{
|
|
gdouble elapsed;
|
|
gulong usec;
|
|
gint ctimeout;
|
|
gint64 timeout = 0;
|
|
|
|
g_return_val_if_fail (conn != NULL, 1);
|
|
|
|
ctimeout = conn->timeout;
|
|
if (ctimeout >= 20) {
|
|
/* Because we should act before the timeout we timeout 5
|
|
* seconds in advance. */
|
|
ctimeout -= 5;
|
|
} else if (ctimeout >= 5) {
|
|
/* else timeout 20% earlier */
|
|
ctimeout -= ctimeout / 5;
|
|
} else if (ctimeout >= 1) {
|
|
/* else timeout 1 second earlier */
|
|
ctimeout -= 1;
|
|
}
|
|
|
|
elapsed = g_timer_elapsed (conn->timer, &usec);
|
|
if (elapsed >= ctimeout) {
|
|
timeout = 0;
|
|
} else {
|
|
gint64 sec = ctimeout - elapsed;
|
|
if (usec <= G_USEC_PER_SEC)
|
|
usec = G_USEC_PER_SEC - usec;
|
|
else
|
|
usec = 0;
|
|
timeout = usec + sec * G_USEC_PER_SEC;
|
|
}
|
|
|
|
return timeout;
|
|
}
|
|
|
|
/**
|
|
* 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);
|
|
|
|
if (flush) {
|
|
GCancellable *cancellable = get_cancellable (conn);
|
|
g_cancellable_cancel (cancellable);
|
|
g_clear_object (&cancellable);
|
|
} else {
|
|
g_mutex_lock (&conn->cancellable_mutex);
|
|
g_object_unref (conn->cancellable);
|
|
conn->cancellable = g_cancellable_new ();
|
|
g_mutex_unlock (&conn->cancellable_mutex);
|
|
}
|
|
|
|
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.
|
|
*/
|
|
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 necessary 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.
|
|
*/
|
|
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.
|
|
*/
|
|
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 (GSocket * socket, guint qos_dscp)
|
|
{
|
|
#ifndef IP_TOS
|
|
GST_FIXME ("IP_TOS socket option is not defined, not setting dscp");
|
|
return GST_RTSP_OK;
|
|
#else
|
|
gint fd;
|
|
union gst_sockaddr sa;
|
|
socklen_t slen = sizeof (sa);
|
|
gint af;
|
|
gint tos;
|
|
|
|
if (!socket)
|
|
return GST_RTSP_OK;
|
|
|
|
fd = g_socket_get_fd (socket);
|
|
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;
|
|
|
|
#ifdef G_OS_WIN32
|
|
# define SETSOCKOPT_ARG4_TYPE const char *
|
|
#else
|
|
# define SETSOCKOPT_ARG4_TYPE const void *
|
|
#endif
|
|
|
|
switch (af) {
|
|
case AF_INET:
|
|
if (setsockopt (fd, IPPROTO_IP, IP_TOS, (SETSOCKOPT_ARG4_TYPE) & tos,
|
|
sizeof (tos)) < 0)
|
|
goto no_setsockopt;
|
|
break;
|
|
case AF_INET6:
|
|
#ifdef IPV6_TCLASS
|
|
if (setsockopt (fd, IPPROTO_IPV6, IPV6_TCLASS,
|
|
(SETSOCKOPT_ARG4_TYPE) & 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;
|
|
}
|
|
#endif
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*/
|
|
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->read_socket != NULL, GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (conn->write_socket != NULL, GST_RTSP_EINVAL);
|
|
|
|
res = set_qos_dscp (conn->socket0, qos_dscp);
|
|
if (res == GST_RTSP_OK)
|
|
res = set_qos_dscp (conn->socket1, qos_dscp);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_set_content_length_limit:
|
|
* @conn: a #GstRTSPConnection
|
|
* @limit: Content-Length limit
|
|
*
|
|
* Configure @conn to use the specified Content-Length limit.
|
|
* Both requests and responses are validated. If content-length is
|
|
* exceeded, ENOMEM error will be returned.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
void
|
|
gst_rtsp_connection_set_content_length_limit (GstRTSPConnection * conn,
|
|
guint limit)
|
|
{
|
|
g_return_if_fail (conn != NULL);
|
|
|
|
conn->content_length_limit = limit;
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*/
|
|
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.
|
|
*/
|
|
const gchar *
|
|
gst_rtsp_connection_get_ip (const GstRTSPConnection * conn)
|
|
{
|
|
g_return_val_if_fail (conn != NULL, NULL);
|
|
|
|
return conn->remote_ip;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_set_ip:
|
|
* @conn: a #GstRTSPConnection
|
|
* @ip: an ip address
|
|
*
|
|
* Set the IP address of the server.
|
|
*/
|
|
void
|
|
gst_rtsp_connection_set_ip (GstRTSPConnection * conn, const gchar * ip)
|
|
{
|
|
g_return_if_fail (conn != NULL);
|
|
|
|
g_free (conn->remote_ip);
|
|
conn->remote_ip = g_strdup (ip);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_get_read_socket:
|
|
* @conn: a #GstRTSPConnection
|
|
*
|
|
* Get the file descriptor for reading.
|
|
*
|
|
* Returns: (transfer none): the file descriptor used for reading or %NULL on
|
|
* error. The file descriptor remains valid until the connection is closed.
|
|
*/
|
|
GSocket *
|
|
gst_rtsp_connection_get_read_socket (const GstRTSPConnection * conn)
|
|
{
|
|
g_return_val_if_fail (conn != NULL, NULL);
|
|
g_return_val_if_fail (conn->read_socket != NULL, NULL);
|
|
|
|
return conn->read_socket;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_get_write_socket:
|
|
* @conn: a #GstRTSPConnection
|
|
*
|
|
* Get the file descriptor for writing.
|
|
*
|
|
* Returns: (transfer none): the file descriptor used for writing or NULL on
|
|
* error. The file descriptor remains valid until the connection is closed.
|
|
*/
|
|
GSocket *
|
|
gst_rtsp_connection_get_write_socket (const GstRTSPConnection * conn)
|
|
{
|
|
g_return_val_if_fail (conn != NULL, NULL);
|
|
g_return_val_if_fail (conn->write_socket != NULL, NULL);
|
|
|
|
return conn->write_socket;
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*/
|
|
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.
|
|
*/
|
|
void
|
|
gst_rtsp_connection_set_tunneled (GstRTSPConnection * conn, gboolean tunneled)
|
|
{
|
|
g_return_if_fail (conn != NULL);
|
|
g_return_if_fail (conn->read_socket == NULL);
|
|
g_return_if_fail (conn->write_socket == 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.
|
|
*/
|
|
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.
|
|
*/
|
|
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_set_ignore_x_server_reply:
|
|
* @conn: a #GstRTSPConnection
|
|
* @ignore: %TRUE to ignore the x-server-ip-address header reply or %FALSE to
|
|
* comply with it (%FALSE is the default).
|
|
*
|
|
* Set whether to ignore the x-server-ip-address header reply or not. If the
|
|
* header is ignored, the original address will be used instead.
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
void
|
|
gst_rtsp_connection_set_ignore_x_server_reply (GstRTSPConnection * conn,
|
|
gboolean ignore)
|
|
{
|
|
g_return_if_fail (conn != NULL);
|
|
|
|
conn->ignore_x_server_reply = ignore;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_get_ignore_x_server_reply:
|
|
* @conn: a #GstRTSPConnection
|
|
*
|
|
* Get the ignore_x_server_reply value.
|
|
*
|
|
* Returns: returns %TRUE if the x-server-ip-address header reply will be
|
|
* ignored, else returns %FALSE
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
gboolean
|
|
gst_rtsp_connection_get_ignore_x_server_reply (const GstRTSPConnection * conn)
|
|
{
|
|
g_return_val_if_fail (conn != NULL, FALSE);
|
|
|
|
return conn->ignore_x_server_reply;
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_do_tunnel (GstRTSPConnection * conn,
|
|
GstRTSPConnection * conn2)
|
|
{
|
|
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
|
|
|
|
if (conn2 != NULL) {
|
|
GstRTSPTunnelState ts1 = conn->tstate;
|
|
GstRTSPTunnelState ts2 = conn2->tstate;
|
|
|
|
g_return_val_if_fail ((ts1 == TUNNEL_STATE_GET && ts2 == TUNNEL_STATE_POST)
|
|
|| (ts1 == TUNNEL_STATE_POST && ts2 == TUNNEL_STATE_GET),
|
|
GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (!memcmp (conn2->tunnelid, conn->tunnelid,
|
|
TUNNELID_LEN), GST_RTSP_EINVAL);
|
|
|
|
/* both connections have socket0 as the read/write socket */
|
|
if (ts1 == TUNNEL_STATE_GET) {
|
|
/* conn2 is the HTTP POST channel. take its socket and set it as read
|
|
* socket in conn */
|
|
conn->socket1 = conn2->socket0;
|
|
conn->stream1 = conn2->stream0;
|
|
conn->input_stream = conn2->input_stream;
|
|
conn->control_stream = g_io_stream_get_input_stream (conn->stream0);
|
|
conn2->output_stream = NULL;
|
|
} else {
|
|
/* conn2 is the HTTP GET channel. take its socket and set it as write
|
|
* socket in conn */
|
|
conn->socket1 = conn->socket0;
|
|
conn->stream1 = conn->stream0;
|
|
conn->socket0 = conn2->socket0;
|
|
conn->stream0 = conn2->stream0;
|
|
conn->output_stream = conn2->output_stream;
|
|
conn->control_stream = g_io_stream_get_input_stream (conn->stream0);
|
|
}
|
|
|
|
/* clean up some of the state of conn2 */
|
|
g_mutex_lock (&conn2->cancellable_mutex);
|
|
g_cancellable_cancel (conn2->cancellable);
|
|
conn2->write_socket = conn2->read_socket = NULL;
|
|
conn2->socket0 = NULL;
|
|
conn2->stream0 = NULL;
|
|
conn2->socket1 = NULL;
|
|
conn2->stream1 = NULL;
|
|
conn2->input_stream = NULL;
|
|
conn2->control_stream = NULL;
|
|
g_object_unref (conn2->cancellable);
|
|
conn2->cancellable = NULL;
|
|
g_mutex_unlock (&conn2->cancellable_mutex);
|
|
|
|
/* We make socket0 the write socket and socket1 the read socket. */
|
|
conn->write_socket = conn->socket0;
|
|
conn->read_socket = conn->socket1;
|
|
|
|
conn->tstate = TUNNEL_STATE_COMPLETE;
|
|
|
|
g_free (conn->initial_buffer);
|
|
conn->initial_buffer = conn2->initial_buffer;
|
|
conn2->initial_buffer = NULL;
|
|
conn->initial_buffer_offset = conn2->initial_buffer_offset;
|
|
}
|
|
|
|
/* 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;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_set_remember_session_id:
|
|
* @conn: a #GstRTSPConnection
|
|
* @remember: %TRUE if the connection should remember the session id
|
|
*
|
|
* Sets if the #GstRTSPConnection should remember the session id from the last
|
|
* response received and force it onto any further requests.
|
|
*
|
|
* The default value is %TRUE
|
|
*/
|
|
|
|
void
|
|
gst_rtsp_connection_set_remember_session_id (GstRTSPConnection * conn,
|
|
gboolean remember)
|
|
{
|
|
conn->remember_session_id = remember;
|
|
if (!remember)
|
|
conn->session_id[0] = '\0';
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_get_remember_session_id:
|
|
* @conn: a #GstRTSPConnection
|
|
*
|
|
* Returns: %TRUE if the #GstRTSPConnection remembers the session id in the
|
|
* last response to set it on any further request.
|
|
*/
|
|
|
|
gboolean
|
|
gst_rtsp_connection_get_remember_session_id (GstRTSPConnection * conn)
|
|
{
|
|
return conn->remember_session_id;
|
|
}
|
|
|
|
|
|
#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)
|
|
|
|
/* async functions */
|
|
struct _GstRTSPWatch
|
|
{
|
|
GSource source;
|
|
|
|
GstRTSPConnection *conn;
|
|
|
|
GstRTSPBuilder builder;
|
|
GstRTSPMessage message;
|
|
|
|
GSource *readsrc;
|
|
GSource *writesrc;
|
|
GSource *controlsrc;
|
|
|
|
gboolean keep_running;
|
|
|
|
/* queued message for transmission */
|
|
guint id;
|
|
GMutex mutex;
|
|
GstQueueArray *messages;
|
|
gsize messages_bytes;
|
|
guint messages_count;
|
|
|
|
gsize max_bytes;
|
|
guint max_messages;
|
|
GCond queue_not_full;
|
|
gboolean flushing;
|
|
|
|
GstRTSPWatchFuncs funcs;
|
|
|
|
gpointer user_data;
|
|
GDestroyNotify notify;
|
|
};
|
|
|
|
#define IS_BACKLOG_FULL(w) (((w)->max_bytes != 0 && (w)->messages_bytes >= (w)->max_bytes) || \
|
|
((w)->max_messages != 0 && (w)->messages_count >= (w)->max_messages))
|
|
|
|
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)
|
|
{
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_source_dispatch_read_get_channel (GPollableInputStream * stream,
|
|
GstRTSPWatch * watch)
|
|
{
|
|
gssize count;
|
|
guint8 buffer[1024];
|
|
GError *error = NULL;
|
|
|
|
/* try to read in order to be able to detect errors, we read 1k in case some
|
|
* client actually decides to send data on the GET channel */
|
|
count = g_pollable_input_stream_read_nonblocking (stream, buffer, 1024, NULL,
|
|
&error);
|
|
if (count == 0) {
|
|
/* other end closed the socket */
|
|
goto eof;
|
|
}
|
|
|
|
if (count < 0) {
|
|
GST_DEBUG ("%s", error->message);
|
|
if (g_error_matches (error, G_IO_ERROR, G_IO_ERROR_WOULD_BLOCK) ||
|
|
g_error_matches (error, G_IO_ERROR, G_IO_ERROR_TIMED_OUT)) {
|
|
g_clear_error (&error);
|
|
goto done;
|
|
}
|
|
g_clear_error (&error);
|
|
goto read_error;
|
|
}
|
|
|
|
/* client sent data on the GET channel, ignore it */
|
|
|
|
done:
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
eof:
|
|
{
|
|
if (watch->funcs.closed)
|
|
watch->funcs.closed (watch, watch->user_data);
|
|
|
|
/* the read connection was closed, stop the watch now */
|
|
watch->keep_running = FALSE;
|
|
|
|
return FALSE;
|
|
}
|
|
read_error:
|
|
{
|
|
if (watch->funcs.error_full)
|
|
watch->funcs.error_full (watch, GST_RTSP_ESYS, &watch->message,
|
|
0, watch->user_data);
|
|
else if (watch->funcs.error)
|
|
watch->funcs.error (watch, GST_RTSP_ESYS, watch->user_data);
|
|
|
|
goto eof;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_source_dispatch_read (GPollableInputStream * stream,
|
|
GstRTSPWatch * watch)
|
|
{
|
|
GstRTSPResult res = GST_RTSP_ERROR;
|
|
GstRTSPConnection *conn = watch->conn;
|
|
|
|
/* if this connection was already closed, stop now */
|
|
if (G_POLLABLE_INPUT_STREAM (conn->input_stream) != stream)
|
|
goto eof;
|
|
|
|
res = build_next (&watch->builder, &watch->message, conn, FALSE);
|
|
if (res == GST_RTSP_EINTR)
|
|
goto done;
|
|
else if (G_UNLIKELY (res == GST_RTSP_EEOF)) {
|
|
g_mutex_lock (&watch->mutex);
|
|
if (watch->readsrc) {
|
|
if (!g_source_is_destroyed ((GSource *) watch))
|
|
g_source_remove_child_source ((GSource *) watch, watch->readsrc);
|
|
g_source_unref (watch->readsrc);
|
|
watch->readsrc = NULL;
|
|
}
|
|
|
|
if (conn->stream1) {
|
|
g_object_unref (conn->stream1);
|
|
conn->stream1 = NULL;
|
|
conn->socket1 = NULL;
|
|
conn->input_stream = NULL;
|
|
}
|
|
g_mutex_unlock (&watch->mutex);
|
|
|
|
/* 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 (conn->tstate == TUNNEL_STATE_COMPLETE) {
|
|
/* remove the read connection for the tunnel */
|
|
/* we accept a new POST request */
|
|
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);
|
|
/* we add read source on the write socket able to detect when client closes get channel in tunneled mode */
|
|
g_mutex_lock (&watch->mutex);
|
|
if (watch->conn->control_stream && !watch->controlsrc) {
|
|
watch->controlsrc =
|
|
g_pollable_input_stream_create_source (G_POLLABLE_INPUT_STREAM
|
|
(watch->conn->control_stream), NULL);
|
|
g_source_set_callback (watch->controlsrc,
|
|
(GSourceFunc) gst_rtsp_source_dispatch_read_get_channel, watch,
|
|
NULL);
|
|
g_source_add_child_source ((GSource *) watch, watch->controlsrc);
|
|
}
|
|
g_mutex_unlock (&watch->mutex);
|
|
goto read_done;
|
|
} else
|
|
goto eof;
|
|
} else if (G_LIKELY (res == GST_RTSP_OK)) {
|
|
if (!conn->manual_http &&
|
|
watch->message.type == GST_RTSP_MESSAGE_HTTP_REQUEST) {
|
|
if (conn->tstate == TUNNEL_STATE_NONE &&
|
|
watch->message.type_data.request.method == GST_RTSP_GET) {
|
|
GstRTSPMessage *response;
|
|
GstRTSPStatusCode code;
|
|
|
|
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 (conn, code, &watch->message);
|
|
if (watch->funcs.tunnel_http_response)
|
|
watch->funcs.tunnel_http_response (watch, &watch->message, response,
|
|
watch->user_data);
|
|
gst_rtsp_watch_send_message (watch, response, NULL);
|
|
gst_rtsp_message_free (response);
|
|
goto read_done;
|
|
} else if (conn->tstate == TUNNEL_STATE_NONE &&
|
|
watch->message.type_data.request.method == GST_RTSP_POST) {
|
|
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;
|
|
}
|
|
}
|
|
} else
|
|
goto read_error;
|
|
|
|
if (!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))
|
|
goto read_error;
|
|
|
|
if (watch->funcs.message_received)
|
|
watch->funcs.message_received (watch, &watch->message, watch->user_data);
|
|
|
|
read_done:
|
|
gst_rtsp_message_unset (&watch->message);
|
|
build_reset (&watch->builder);
|
|
|
|
done:
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
eof:
|
|
{
|
|
if (watch->funcs.closed)
|
|
watch->funcs.closed (watch, watch->user_data);
|
|
|
|
/* we closed the read connection, stop the watch now */
|
|
watch->keep_running = FALSE;
|
|
|
|
/* always stop when the input returns EOF in non-tunneled mode */
|
|
return FALSE;
|
|
}
|
|
read_error:
|
|
{
|
|
if (watch->funcs.error_full)
|
|
watch->funcs.error_full (watch, res, &watch->message,
|
|
0, watch->user_data);
|
|
else if (watch->funcs.error)
|
|
watch->funcs.error (watch, res, watch->user_data);
|
|
|
|
goto eof;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_source_dispatch (GSource * source, GSourceFunc callback G_GNUC_UNUSED,
|
|
gpointer user_data G_GNUC_UNUSED)
|
|
{
|
|
GstRTSPWatch *watch = (GstRTSPWatch *) source;
|
|
GstRTSPConnection *conn = watch->conn;
|
|
|
|
if (conn->initial_buffer != NULL) {
|
|
gst_rtsp_source_dispatch_read (G_POLLABLE_INPUT_STREAM (conn->input_stream),
|
|
watch);
|
|
}
|
|
return watch->keep_running;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_source_dispatch_write (GPollableOutputStream * stream,
|
|
GstRTSPWatch * watch)
|
|
{
|
|
GstRTSPResult res = GST_RTSP_ERROR;
|
|
GstRTSPConnection *conn = watch->conn;
|
|
|
|
/* if this connection was already closed, stop now */
|
|
if (G_POLLABLE_OUTPUT_STREAM (conn->output_stream) != stream ||
|
|
!watch->messages)
|
|
goto eof;
|
|
|
|
g_mutex_lock (&watch->mutex);
|
|
do {
|
|
guint n_messages = gst_queue_array_get_length (watch->messages);
|
|
GOutputVector *vectors;
|
|
GstMapInfo *map_infos;
|
|
guint *ids;
|
|
gsize bytes_to_write, bytes_written;
|
|
guint n_vectors, n_memories, n_ids, drop_messages;
|
|
gint i, j, l, n_mmap;
|
|
GstRTSPSerializedMessage *msg;
|
|
GCancellable *cancellable;
|
|
|
|
/* if this connection was already closed, stop now */
|
|
if (G_POLLABLE_OUTPUT_STREAM (conn->output_stream) != stream ||
|
|
!watch->messages) {
|
|
g_mutex_unlock (&watch->mutex);
|
|
goto eof;
|
|
}
|
|
|
|
if (n_messages == 0) {
|
|
if (watch->writesrc) {
|
|
if (!g_source_is_destroyed ((GSource *) watch))
|
|
g_source_remove_child_source ((GSource *) watch, watch->writesrc);
|
|
g_source_unref (watch->writesrc);
|
|
watch->writesrc = NULL;
|
|
/* we create and add the write source again when we actually have
|
|
* something to write */
|
|
|
|
/* since write source is now removed we add read source on the write
|
|
* socket instead to be able to detect when client closes get channel
|
|
* in tunneled mode */
|
|
if (watch->conn->control_stream) {
|
|
watch->controlsrc =
|
|
g_pollable_input_stream_create_source (G_POLLABLE_INPUT_STREAM
|
|
(watch->conn->control_stream), NULL);
|
|
g_source_set_callback (watch->controlsrc,
|
|
(GSourceFunc) gst_rtsp_source_dispatch_read_get_channel, watch,
|
|
NULL);
|
|
g_source_add_child_source ((GSource *) watch, watch->controlsrc);
|
|
} else {
|
|
watch->controlsrc = NULL;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
|
|
for (i = 0, n_vectors = 0, n_memories = 0, n_ids = 0; i < n_messages; i++) {
|
|
msg = gst_queue_array_peek_nth_struct (watch->messages, i);
|
|
if (msg->id != 0)
|
|
n_ids++;
|
|
|
|
if (msg->data_offset < msg->data_size)
|
|
n_vectors++;
|
|
|
|
if (msg->body_data && msg->body_offset < msg->body_data_size) {
|
|
n_vectors++;
|
|
} else if (msg->body_buffer) {
|
|
guint m, n;
|
|
guint offset = 0;
|
|
|
|
n = gst_buffer_n_memory (msg->body_buffer);
|
|
for (m = 0; m < n; m++) {
|
|
GstMemory *mem = gst_buffer_peek_memory (msg->body_buffer, m);
|
|
|
|
/* Skip all memories we already wrote */
|
|
if (offset + mem->size <= msg->body_offset) {
|
|
offset += mem->size;
|
|
continue;
|
|
}
|
|
offset += mem->size;
|
|
|
|
n_memories++;
|
|
n_vectors++;
|
|
}
|
|
}
|
|
}
|
|
|
|
vectors = g_newa (GOutputVector, n_vectors);
|
|
map_infos = n_memories ? g_newa (GstMapInfo, n_memories) : NULL;
|
|
ids = n_ids ? g_newa (guint, n_ids + 1) : NULL;
|
|
if (ids)
|
|
memset (ids, 0, sizeof (guint) * (n_ids + 1));
|
|
|
|
for (i = 0, j = 0, n_mmap = 0, l = 0, bytes_to_write = 0; i < n_messages;
|
|
i++) {
|
|
msg = gst_queue_array_peek_nth_struct (watch->messages, i);
|
|
|
|
if (msg->data_offset < msg->data_size) {
|
|
vectors[j].buffer = (msg->data_is_data_header ?
|
|
msg->data_header : msg->data) + msg->data_offset;
|
|
vectors[j].size = msg->data_size - msg->data_offset;
|
|
bytes_to_write += vectors[j].size;
|
|
j++;
|
|
}
|
|
|
|
if (msg->body_data) {
|
|
if (msg->body_offset < msg->body_data_size) {
|
|
vectors[j].buffer = msg->body_data + msg->body_offset;
|
|
vectors[j].size = msg->body_data_size - msg->body_offset;
|
|
bytes_to_write += vectors[j].size;
|
|
j++;
|
|
}
|
|
} else if (msg->body_buffer) {
|
|
guint m, n;
|
|
guint offset = 0;
|
|
n = gst_buffer_n_memory (msg->body_buffer);
|
|
for (m = 0; m < n; m++) {
|
|
GstMemory *mem = gst_buffer_peek_memory (msg->body_buffer, m);
|
|
guint off;
|
|
|
|
/* Skip all memories we already wrote */
|
|
if (offset + mem->size <= msg->body_offset) {
|
|
offset += mem->size;
|
|
continue;
|
|
}
|
|
|
|
if (offset < msg->body_offset)
|
|
off = msg->body_offset - offset;
|
|
else
|
|
off = 0;
|
|
|
|
offset += mem->size;
|
|
|
|
g_assert (off < mem->size);
|
|
|
|
gst_memory_map (mem, &map_infos[n_mmap], GST_MAP_READ);
|
|
vectors[j].buffer = map_infos[n_mmap].data + off;
|
|
vectors[j].size = map_infos[n_mmap].size - off;
|
|
bytes_to_write += vectors[j].size;
|
|
|
|
n_mmap++;
|
|
j++;
|
|
}
|
|
}
|
|
}
|
|
|
|
cancellable = get_cancellable (watch->conn);
|
|
|
|
res =
|
|
writev_bytes (watch->conn->output_stream, vectors, n_vectors,
|
|
&bytes_written, FALSE, cancellable);
|
|
g_assert (bytes_written == bytes_to_write || res != GST_RTSP_OK);
|
|
|
|
g_clear_object (&cancellable);
|
|
|
|
/* First unmap all memories here, this simplifies the code below
|
|
* as we don't have to skip all memories that were already written
|
|
* before */
|
|
for (i = 0; i < n_mmap; i++) {
|
|
gst_memory_unmap (map_infos[i].memory, &map_infos[i]);
|
|
}
|
|
|
|
if (bytes_written == bytes_to_write) {
|
|
/* fast path, just unmap all memories, free memory, drop all messages and notify them */
|
|
l = 0;
|
|
while ((msg = gst_queue_array_pop_head_struct (watch->messages))) {
|
|
if (msg->id) {
|
|
ids[l] = msg->id;
|
|
l++;
|
|
}
|
|
|
|
gst_rtsp_serialized_message_clear (msg);
|
|
}
|
|
|
|
g_assert (watch->messages_bytes >= bytes_written);
|
|
watch->messages_bytes -= bytes_written;
|
|
} else if (bytes_written > 0) {
|
|
/* not done, let's skip all messages that were sent already and free them */
|
|
for (i = 0, drop_messages = 0; i < n_messages; i++) {
|
|
msg = gst_queue_array_peek_nth_struct (watch->messages, i);
|
|
|
|
if (bytes_written >= msg->data_size - msg->data_offset) {
|
|
guint body_size;
|
|
|
|
/* all data of this message is sent, check body and otherwise
|
|
* skip the whole message for next time */
|
|
bytes_written -= (msg->data_size - msg->data_offset);
|
|
watch->messages_bytes -= (msg->data_size - msg->data_offset);
|
|
msg->data_offset = msg->data_size;
|
|
|
|
if (msg->body_data) {
|
|
body_size = msg->body_data_size;
|
|
} else if (msg->body_buffer) {
|
|
body_size = gst_buffer_get_size (msg->body_buffer);
|
|
} else {
|
|
body_size = 0;
|
|
}
|
|
|
|
if (bytes_written + msg->body_offset >= body_size) {
|
|
/* body written, drop this message */
|
|
bytes_written -= body_size - msg->body_offset;
|
|
watch->messages_bytes -= body_size - msg->body_offset;
|
|
msg->body_offset = body_size;
|
|
drop_messages++;
|
|
|
|
if (msg->id) {
|
|
ids[l] = msg->id;
|
|
l++;
|
|
}
|
|
|
|
gst_rtsp_serialized_message_clear (msg);
|
|
} else {
|
|
msg->body_offset += bytes_written;
|
|
watch->messages_bytes -= bytes_written;
|
|
bytes_written = 0;
|
|
}
|
|
} else {
|
|
/* Need to continue sending from the data of this message */
|
|
msg->data_offset += bytes_written;
|
|
watch->messages_bytes -= bytes_written;
|
|
bytes_written = 0;
|
|
}
|
|
}
|
|
|
|
while (drop_messages > 0) {
|
|
msg = gst_queue_array_pop_head_struct (watch->messages);
|
|
g_assert (msg);
|
|
drop_messages--;
|
|
}
|
|
|
|
g_assert (watch->messages_bytes >= bytes_written);
|
|
watch->messages_bytes -= bytes_written;
|
|
}
|
|
|
|
if (!IS_BACKLOG_FULL (watch))
|
|
g_cond_signal (&watch->queue_not_full);
|
|
g_mutex_unlock (&watch->mutex);
|
|
|
|
/* notify all messages that were successfully written */
|
|
if (ids) {
|
|
while (*ids) {
|
|
/* only decrease the counter for messages that have an id. Only
|
|
* the last message of a messages chunk is counted */
|
|
watch->messages_count--;
|
|
|
|
if (watch->funcs.message_sent)
|
|
watch->funcs.message_sent (watch, *ids, watch->user_data);
|
|
ids++;
|
|
}
|
|
}
|
|
|
|
if (res == GST_RTSP_EINTR) {
|
|
goto write_blocked;
|
|
} else if (G_UNLIKELY (res != GST_RTSP_OK)) {
|
|
goto write_error;
|
|
}
|
|
g_mutex_lock (&watch->mutex);
|
|
} while (TRUE);
|
|
g_mutex_unlock (&watch->mutex);
|
|
|
|
write_blocked:
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
eof:
|
|
{
|
|
return FALSE;
|
|
}
|
|
write_error:
|
|
{
|
|
if (watch->funcs.error_full) {
|
|
guint i, n_messages;
|
|
|
|
n_messages = gst_queue_array_get_length (watch->messages);
|
|
for (i = 0; i < n_messages; i++) {
|
|
GstRTSPSerializedMessage *msg =
|
|
gst_queue_array_peek_nth_struct (watch->messages, i);
|
|
if (msg->id)
|
|
watch->funcs.error_full (watch, res, NULL, msg->id, watch->user_data);
|
|
}
|
|
} else if (watch->funcs.error) {
|
|
watch->funcs.error (watch, res, watch->user_data);
|
|
}
|
|
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_source_finalize (GSource * source)
|
|
{
|
|
GstRTSPWatch *watch = (GstRTSPWatch *) source;
|
|
GstRTSPSerializedMessage *msg;
|
|
|
|
if (watch->notify)
|
|
watch->notify (watch->user_data);
|
|
|
|
build_reset (&watch->builder);
|
|
gst_rtsp_message_unset (&watch->message);
|
|
|
|
while ((msg = gst_queue_array_pop_head_struct (watch->messages))) {
|
|
gst_rtsp_serialized_message_clear (msg);
|
|
}
|
|
gst_queue_array_free (watch->messages);
|
|
watch->messages = NULL;
|
|
watch->messages_bytes = 0;
|
|
watch->messages_count = 0;
|
|
|
|
g_cond_clear (&watch->queue_not_full);
|
|
|
|
if (watch->readsrc)
|
|
g_source_unref (watch->readsrc);
|
|
if (watch->writesrc)
|
|
g_source_unref (watch->writesrc);
|
|
if (watch->controlsrc)
|
|
g_source_unref (watch->controlsrc);
|
|
|
|
g_mutex_clear (&watch->mutex);
|
|
}
|
|
|
|
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: (skip)
|
|
* @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.
|
|
*/
|
|
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->read_socket != NULL, NULL);
|
|
g_return_val_if_fail (conn->write_socket != NULL, NULL);
|
|
|
|
result = (GstRTSPWatch *) g_source_new (&gst_rtsp_source_funcs,
|
|
sizeof (GstRTSPWatch));
|
|
|
|
result->conn = conn;
|
|
result->builder.state = STATE_START;
|
|
|
|
g_mutex_init (&result->mutex);
|
|
result->messages =
|
|
gst_queue_array_new_for_struct (sizeof (GstRTSPSerializedMessage), 10);
|
|
g_cond_init (&result->queue_not_full);
|
|
|
|
gst_rtsp_watch_reset (result);
|
|
result->keep_running = TRUE;
|
|
result->flushing = FALSE;
|
|
|
|
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.
|
|
*/
|
|
void
|
|
gst_rtsp_watch_reset (GstRTSPWatch * watch)
|
|
{
|
|
g_mutex_lock (&watch->mutex);
|
|
if (watch->readsrc) {
|
|
g_source_remove_child_source ((GSource *) watch, watch->readsrc);
|
|
g_source_unref (watch->readsrc);
|
|
}
|
|
if (watch->writesrc) {
|
|
g_source_remove_child_source ((GSource *) watch, watch->writesrc);
|
|
g_source_unref (watch->writesrc);
|
|
watch->writesrc = NULL;
|
|
}
|
|
if (watch->controlsrc) {
|
|
g_source_remove_child_source ((GSource *) watch, watch->controlsrc);
|
|
g_source_unref (watch->controlsrc);
|
|
watch->controlsrc = NULL;
|
|
}
|
|
|
|
if (watch->conn->input_stream) {
|
|
watch->readsrc =
|
|
g_pollable_input_stream_create_source (G_POLLABLE_INPUT_STREAM
|
|
(watch->conn->input_stream), NULL);
|
|
g_source_set_callback (watch->readsrc,
|
|
(GSourceFunc) gst_rtsp_source_dispatch_read, watch, NULL);
|
|
g_source_add_child_source ((GSource *) watch, watch->readsrc);
|
|
} else {
|
|
watch->readsrc = NULL;
|
|
}
|
|
|
|
/* we create and add the write source when we actually have something to
|
|
* write */
|
|
|
|
/* when write source is not added we add read source on the write socket
|
|
* instead to be able to detect when client closes get channel in tunneled
|
|
* mode */
|
|
if (watch->conn->control_stream) {
|
|
watch->controlsrc =
|
|
g_pollable_input_stream_create_source (G_POLLABLE_INPUT_STREAM
|
|
(watch->conn->control_stream), NULL);
|
|
g_source_set_callback (watch->controlsrc,
|
|
(GSourceFunc) gst_rtsp_source_dispatch_read_get_channel, watch, NULL);
|
|
g_source_add_child_source ((GSource *) watch, watch->controlsrc);
|
|
} else {
|
|
watch->controlsrc = NULL;
|
|
}
|
|
g_mutex_unlock (&watch->mutex);
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*/
|
|
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.
|
|
*/
|
|
void
|
|
gst_rtsp_watch_unref (GstRTSPWatch * watch)
|
|
{
|
|
g_return_if_fail (watch != NULL);
|
|
|
|
g_source_unref ((GSource *) watch);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_watch_set_send_backlog:
|
|
* @watch: a #GstRTSPWatch
|
|
* @bytes: maximum bytes
|
|
* @messages: maximum messages
|
|
*
|
|
* Set the maximum amount of bytes and messages that will be queued in @watch.
|
|
* When the maximum amounts are exceeded, gst_rtsp_watch_write_data() and
|
|
* gst_rtsp_watch_send_message() will return #GST_RTSP_ENOMEM.
|
|
*
|
|
* A value of 0 for @bytes or @messages means no limits.
|
|
*
|
|
* Since: 1.2
|
|
*/
|
|
void
|
|
gst_rtsp_watch_set_send_backlog (GstRTSPWatch * watch,
|
|
gsize bytes, guint messages)
|
|
{
|
|
g_return_if_fail (watch != NULL);
|
|
|
|
g_mutex_lock (&watch->mutex);
|
|
watch->max_bytes = bytes;
|
|
watch->max_messages = messages;
|
|
if (!IS_BACKLOG_FULL (watch))
|
|
g_cond_signal (&watch->queue_not_full);
|
|
g_mutex_unlock (&watch->mutex);
|
|
|
|
GST_DEBUG ("set backlog to bytes %" G_GSIZE_FORMAT ", messages %u",
|
|
bytes, messages);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_watch_get_send_backlog:
|
|
* @watch: a #GstRTSPWatch
|
|
* @bytes: (out) (allow-none): maximum bytes
|
|
* @messages: (out) (allow-none): maximum messages
|
|
*
|
|
* Get the maximum amount of bytes and messages that will be queued in @watch.
|
|
* See gst_rtsp_watch_set_send_backlog().
|
|
*
|
|
* Since: 1.2
|
|
*/
|
|
void
|
|
gst_rtsp_watch_get_send_backlog (GstRTSPWatch * watch,
|
|
gsize * bytes, guint * messages)
|
|
{
|
|
g_return_if_fail (watch != NULL);
|
|
|
|
g_mutex_lock (&watch->mutex);
|
|
if (bytes)
|
|
*bytes = watch->max_bytes;
|
|
if (messages)
|
|
*messages = watch->max_messages;
|
|
g_mutex_unlock (&watch->mutex);
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_watch_write_serialized_messages (GstRTSPWatch * watch,
|
|
GstRTSPSerializedMessage * messages, guint n_messages, guint * id)
|
|
{
|
|
GstRTSPResult res;
|
|
GMainContext *context = NULL;
|
|
GCancellable *cancellable;
|
|
gint i;
|
|
|
|
g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (messages != NULL, GST_RTSP_EINVAL);
|
|
|
|
g_mutex_lock (&watch->mutex);
|
|
if (watch->flushing)
|
|
goto flushing;
|
|
|
|
/* try to send the message synchronously first */
|
|
if (gst_queue_array_get_length (watch->messages) == 0) {
|
|
gint j, k;
|
|
GOutputVector *vectors;
|
|
GstMapInfo *map_infos;
|
|
gsize bytes_to_write, bytes_written;
|
|
guint n_vectors, n_memories, drop_messages;
|
|
|
|
for (i = 0, n_vectors = 0, n_memories = 0; i < n_messages; i++) {
|
|
n_vectors++;
|
|
if (messages[i].body_data) {
|
|
n_vectors++;
|
|
} else if (messages[i].body_buffer) {
|
|
n_vectors += gst_buffer_n_memory (messages[i].body_buffer);
|
|
n_memories += gst_buffer_n_memory (messages[i].body_buffer);
|
|
}
|
|
}
|
|
|
|
vectors = g_newa (GOutputVector, n_vectors);
|
|
map_infos = n_memories ? g_newa (GstMapInfo, n_memories) : NULL;
|
|
|
|
for (i = 0, j = 0, k = 0, bytes_to_write = 0; i < n_messages; i++) {
|
|
vectors[j].buffer = messages[i].data_is_data_header ?
|
|
messages[i].data_header : messages[i].data;
|
|
vectors[j].size = messages[i].data_size;
|
|
bytes_to_write += vectors[j].size;
|
|
j++;
|
|
|
|
if (messages[i].body_data) {
|
|
vectors[j].buffer = messages[i].body_data;
|
|
vectors[j].size = messages[i].body_data_size;
|
|
bytes_to_write += vectors[j].size;
|
|
j++;
|
|
} else if (messages[i].body_buffer) {
|
|
gint l, n;
|
|
|
|
n = gst_buffer_n_memory (messages[i].body_buffer);
|
|
for (l = 0; l < n; l++) {
|
|
GstMemory *mem = gst_buffer_peek_memory (messages[i].body_buffer, l);
|
|
|
|
gst_memory_map (mem, &map_infos[k], GST_MAP_READ);
|
|
vectors[j].buffer = map_infos[k].data;
|
|
vectors[j].size = map_infos[k].size;
|
|
bytes_to_write += vectors[j].size;
|
|
|
|
k++;
|
|
j++;
|
|
}
|
|
}
|
|
}
|
|
|
|
cancellable = get_cancellable (watch->conn);
|
|
|
|
res =
|
|
writev_bytes (watch->conn->output_stream, vectors, n_vectors,
|
|
&bytes_written, FALSE, cancellable);
|
|
g_assert (bytes_written == bytes_to_write || res != GST_RTSP_OK);
|
|
|
|
g_clear_object (&cancellable);
|
|
|
|
/* At this point we sent everything we could without blocking or
|
|
* error and updated the offsets inside the message accordingly */
|
|
|
|
/* First of all unmap all memories. This simplifies the code below */
|
|
for (k = 0; k < n_memories; k++) {
|
|
gst_memory_unmap (map_infos[k].memory, &map_infos[k]);
|
|
}
|
|
|
|
if (res != GST_RTSP_EINTR) {
|
|
/* actual error or done completely */
|
|
if (id != NULL)
|
|
*id = 0;
|
|
|
|
/* free everything */
|
|
for (i = 0, k = 0; i < n_messages; i++) {
|
|
gst_rtsp_serialized_message_clear (&messages[i]);
|
|
}
|
|
|
|
goto done;
|
|
}
|
|
|
|
/* not done, let's skip all messages that were sent already and free them */
|
|
for (i = 0, k = 0, drop_messages = 0; i < n_messages; i++) {
|
|
if (bytes_written >= messages[i].data_size) {
|
|
guint body_size;
|
|
|
|
/* all data of this message is sent, check body and otherwise
|
|
* skip the whole message for next time */
|
|
messages[i].data_offset = messages[i].data_size;
|
|
bytes_written -= messages[i].data_size;
|
|
|
|
if (messages[i].body_data) {
|
|
body_size = messages[i].body_data_size;
|
|
|
|
} else if (messages[i].body_buffer) {
|
|
body_size = gst_buffer_get_size (messages[i].body_buffer);
|
|
} else {
|
|
body_size = 0;
|
|
}
|
|
|
|
if (bytes_written >= body_size) {
|
|
/* body written, drop this message */
|
|
messages[i].body_offset = body_size;
|
|
bytes_written -= body_size;
|
|
drop_messages++;
|
|
|
|
gst_rtsp_serialized_message_clear (&messages[i]);
|
|
} else {
|
|
messages[i].body_offset = bytes_written;
|
|
bytes_written = 0;
|
|
}
|
|
} else {
|
|
/* Need to continue sending from the data of this message */
|
|
messages[i].data_offset = bytes_written;
|
|
bytes_written = 0;
|
|
}
|
|
}
|
|
|
|
g_assert (n_messages > drop_messages);
|
|
|
|
messages += drop_messages;
|
|
n_messages -= drop_messages;
|
|
}
|
|
|
|
/* check limits */
|
|
if (IS_BACKLOG_FULL (watch))
|
|
goto too_much_backlog;
|
|
|
|
for (i = 0; i < n_messages; i++) {
|
|
GstRTSPSerializedMessage local_message;
|
|
|
|
/* make a record with the data and id for sending async */
|
|
local_message = messages[i];
|
|
|
|
/* copy the body data or take an additional reference to the body buffer
|
|
* we don't own them here */
|
|
if (local_message.body_data) {
|
|
local_message.body_data =
|
|
g_memdup2 (local_message.body_data, local_message.body_data_size);
|
|
} else if (local_message.body_buffer) {
|
|
gst_buffer_ref (local_message.body_buffer);
|
|
}
|
|
local_message.borrowed = FALSE;
|
|
|
|
/* set an id for the very last message */
|
|
if (i == n_messages - 1) {
|
|
do {
|
|
/* make sure rec->id is never 0 */
|
|
local_message.id = ++watch->id;
|
|
} while (G_UNLIKELY (local_message.id == 0));
|
|
|
|
if (id != NULL)
|
|
*id = local_message.id;
|
|
} else {
|
|
local_message.id = 0;
|
|
}
|
|
|
|
/* add the record to a queue. */
|
|
gst_queue_array_push_tail_struct (watch->messages, &local_message);
|
|
watch->messages_bytes +=
|
|
(local_message.data_size - local_message.data_offset);
|
|
if (local_message.body_data)
|
|
watch->messages_bytes +=
|
|
(local_message.body_data_size - local_message.body_offset);
|
|
else if (local_message.body_buffer)
|
|
watch->messages_bytes +=
|
|
(gst_buffer_get_size (local_message.body_buffer) -
|
|
local_message.body_offset);
|
|
}
|
|
/* each message chunks is one unit */
|
|
watch->messages_count++;
|
|
|
|
/* make sure the main context will now also check for writability on the
|
|
* socket */
|
|
context = ((GSource *) watch)->context;
|
|
if (!watch->writesrc) {
|
|
/* remove the read source on the write socket, we will be able to detect
|
|
* errors while writing */
|
|
if (watch->controlsrc) {
|
|
g_source_remove_child_source ((GSource *) watch, watch->controlsrc);
|
|
g_source_unref (watch->controlsrc);
|
|
watch->controlsrc = NULL;
|
|
}
|
|
|
|
watch->writesrc =
|
|
g_pollable_output_stream_create_source (G_POLLABLE_OUTPUT_STREAM
|
|
(watch->conn->output_stream), NULL);
|
|
g_source_set_callback (watch->writesrc,
|
|
(GSourceFunc) gst_rtsp_source_dispatch_write, watch, NULL);
|
|
g_source_add_child_source ((GSource *) watch, watch->writesrc);
|
|
}
|
|
res = GST_RTSP_OK;
|
|
|
|
done:
|
|
g_mutex_unlock (&watch->mutex);
|
|
|
|
if (context)
|
|
g_main_context_wakeup (context);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG ("we are flushing");
|
|
g_mutex_unlock (&watch->mutex);
|
|
for (i = 0; i < n_messages; i++) {
|
|
gst_rtsp_serialized_message_clear (&messages[i]);
|
|
}
|
|
return GST_RTSP_EINTR;
|
|
}
|
|
too_much_backlog:
|
|
{
|
|
GST_WARNING ("too much backlog: max_bytes %" G_GSIZE_FORMAT ", current %"
|
|
G_GSIZE_FORMAT ", max_messages %u, current %u", watch->max_bytes,
|
|
watch->messages_bytes, watch->max_messages, watch->messages_count);
|
|
g_mutex_unlock (&watch->mutex);
|
|
for (i = 0; i < n_messages; i++) {
|
|
gst_rtsp_serialized_message_clear (&messages[i]);
|
|
}
|
|
return GST_RTSP_ENOMEM;
|
|
}
|
|
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_watch_write_data:
|
|
* @watch: a #GstRTSPWatch
|
|
* @data: (array length=size) (transfer full): the data to queue
|
|
* @size: the size of @data
|
|
* @id: (out) (allow-none): 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.
|
|
*
|
|
* If the amount of queued data exceeds the limits set with
|
|
* gst_rtsp_watch_set_send_backlog(), this function will return
|
|
* #GST_RTSP_ENOMEM.
|
|
*
|
|
* Returns: #GST_RTSP_OK on success. #GST_RTSP_ENOMEM when the backlog limits
|
|
* are reached. #GST_RTSP_EINTR when @watch was flushing.
|
|
*/
|
|
/* FIXME 2.0: This should've been static! */
|
|
GstRTSPResult
|
|
gst_rtsp_watch_write_data (GstRTSPWatch * watch, const guint8 * data,
|
|
guint size, guint * id)
|
|
{
|
|
GstRTSPSerializedMessage serialized_message;
|
|
|
|
memset (&serialized_message, 0, sizeof (serialized_message));
|
|
serialized_message.data = (guint8 *) data;
|
|
serialized_message.data_size = size;
|
|
|
|
return gst_rtsp_watch_write_serialized_messages (watch, &serialized_message,
|
|
1, id);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_watch_send_message:
|
|
* @watch: a #GstRTSPWatch
|
|
* @message: a #GstRTSPMessage
|
|
* @id: (out) (allow-none): 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.
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_watch_send_message (GstRTSPWatch * watch, GstRTSPMessage * message,
|
|
guint * id)
|
|
{
|
|
g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
|
|
|
|
return gst_rtsp_watch_send_messages (watch, message, 1, id);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_watch_send_messages:
|
|
* @watch: a #GstRTSPWatch
|
|
* @messages: (array length=n_messages): the messages to send
|
|
* @n_messages: the number of messages to send
|
|
* @id: (out) (allow-none): location for a message ID or %NULL
|
|
*
|
|
* Sends @messages using the connection of the @watch. If they cannot be sent
|
|
* immediately, they will be queued for transmission in @watch. The contents of
|
|
* @messages will then be serialized and transmitted when the connection of the
|
|
* @watch becomes writable. In case the @messages are queued, the ID returned in
|
|
* @id will be non-zero and used as the ID argument in the message_sent
|
|
* callback once the last message is sent. The callback will only be called
|
|
* once for the last message.
|
|
*
|
|
* Returns: #GST_RTSP_OK on success.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_watch_send_messages (GstRTSPWatch * watch, GstRTSPMessage * messages,
|
|
guint n_messages, guint * id)
|
|
{
|
|
GstRTSPSerializedMessage *serialized_messages;
|
|
gint i;
|
|
|
|
g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (messages != NULL || n_messages == 0, GST_RTSP_EINVAL);
|
|
|
|
serialized_messages = g_newa (GstRTSPSerializedMessage, n_messages);
|
|
memset (serialized_messages, 0,
|
|
sizeof (GstRTSPSerializedMessage) * n_messages);
|
|
|
|
for (i = 0; i < n_messages; i++) {
|
|
if (!serialize_message (watch->conn, &messages[i], &serialized_messages[i]))
|
|
goto error;
|
|
}
|
|
|
|
return gst_rtsp_watch_write_serialized_messages (watch, serialized_messages,
|
|
n_messages, id);
|
|
|
|
error:
|
|
for (i = 0; i < n_messages; i++) {
|
|
gst_rtsp_serialized_message_clear (&serialized_messages[i]);
|
|
}
|
|
|
|
return GST_RTSP_EINVAL;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_watch_wait_backlog_usec:
|
|
* @watch: a #GstRTSPWatch
|
|
* @timeout: a timeout in microseconds
|
|
*
|
|
* Wait until there is place in the backlog queue, @timeout is reached
|
|
* or @watch is set to flushing.
|
|
*
|
|
* If @timeout is 0 this function can block forever. If @timeout
|
|
* contains a valid timeout, this function will return %GST_RTSP_ETIMEOUT
|
|
* after the timeout expired.
|
|
*
|
|
* The typically use of this function is when gst_rtsp_watch_write_data
|
|
* returns %GST_RTSP_ENOMEM. The caller then calls this function to wait for
|
|
* free space in the backlog queue and try again.
|
|
*
|
|
* Returns: %GST_RTSP_OK when if there is room in queue.
|
|
* %GST_RTSP_ETIMEOUT when @timeout was reached.
|
|
* %GST_RTSP_EINTR when @watch is flushing
|
|
* %GST_RTSP_EINVAL when called with invalid parameters.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_watch_wait_backlog_usec (GstRTSPWatch * watch, gint64 timeout)
|
|
{
|
|
gint64 end_time;
|
|
|
|
g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
|
|
|
|
end_time = g_get_monotonic_time () + timeout;
|
|
|
|
g_mutex_lock (&watch->mutex);
|
|
if (watch->flushing)
|
|
goto flushing;
|
|
|
|
while (IS_BACKLOG_FULL (watch)) {
|
|
gboolean res;
|
|
|
|
res = g_cond_wait_until (&watch->queue_not_full, &watch->mutex, end_time);
|
|
if (watch->flushing)
|
|
goto flushing;
|
|
|
|
if (!res)
|
|
goto timeout;
|
|
}
|
|
g_mutex_unlock (&watch->mutex);
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG ("we are flushing");
|
|
g_mutex_unlock (&watch->mutex);
|
|
return GST_RTSP_EINTR;
|
|
}
|
|
timeout:
|
|
{
|
|
GST_DEBUG ("we timed out");
|
|
g_mutex_unlock (&watch->mutex);
|
|
return GST_RTSP_ETIMEOUT;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_watch_set_flushing:
|
|
* @watch: a #GstRTSPWatch
|
|
* @flushing: new flushing state
|
|
*
|
|
* When @flushing is %TRUE, abort a call to gst_rtsp_watch_wait_backlog()
|
|
* and make sure gst_rtsp_watch_write_data() returns immediately with
|
|
* #GST_RTSP_EINTR. And empty the queue.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
void
|
|
gst_rtsp_watch_set_flushing (GstRTSPWatch * watch, gboolean flushing)
|
|
{
|
|
g_return_if_fail (watch != NULL);
|
|
|
|
g_mutex_lock (&watch->mutex);
|
|
watch->flushing = flushing;
|
|
g_cond_signal (&watch->queue_not_full);
|
|
if (flushing) {
|
|
GstRTSPSerializedMessage *msg;
|
|
|
|
while ((msg = gst_queue_array_pop_head_struct (watch->messages))) {
|
|
gst_rtsp_serialized_message_clear (msg);
|
|
}
|
|
}
|
|
g_mutex_unlock (&watch->mutex);
|
|
}
|
|
|
|
|
|
#ifndef GST_DISABLE_DEPRECATED
|
|
G_GNUC_BEGIN_IGNORE_DEPRECATIONS
|
|
/* Deprecated */
|
|
#define TV_TO_USEC(tv) ((tv) ? ((tv)->tv_sec * G_USEC_PER_SEC + (tv)->tv_usec) : 0)
|
|
/**
|
|
* 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.
|
|
*
|
|
* Deprecated: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_connect (GstRTSPConnection * conn, GTimeVal * timeout)
|
|
{
|
|
return gst_rtsp_connection_connect_usec (conn, TV_TO_USEC (timeout));
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_connect_with_response:
|
|
* @conn: a #GstRTSPConnection
|
|
* @timeout: a GTimeVal timeout
|
|
* @response: a #GstRTSPMessage
|
|
*
|
|
* 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. If @conn is set to tunneled,
|
|
* @response will contain a response to the tunneling request messages.
|
|
*
|
|
* This function can be cancelled with gst_rtsp_connection_flush().
|
|
*
|
|
* Returns: #GST_RTSP_OK when a connection could be made.
|
|
*
|
|
* Since: 1.8
|
|
* Deprecated: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_connect_with_response (GstRTSPConnection * conn,
|
|
GTimeVal * timeout, GstRTSPMessage * response)
|
|
{
|
|
return gst_rtsp_connection_connect_with_response_usec (conn,
|
|
TV_TO_USEC (timeout), response);
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*
|
|
* Deprecated: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_read (GstRTSPConnection * conn, guint8 * data, guint size,
|
|
GTimeVal * timeout)
|
|
{
|
|
return gst_rtsp_connection_read_usec (conn, data, size, TV_TO_USEC (timeout));
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*
|
|
* Deprecated: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_write (GstRTSPConnection * conn, const guint8 * data,
|
|
guint size, GTimeVal * timeout)
|
|
{
|
|
return gst_rtsp_connection_write_usec (conn, data, size,
|
|
TV_TO_USEC (timeout));
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*
|
|
* Deprecated: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_send (GstRTSPConnection * conn, GstRTSPMessage * message,
|
|
GTimeVal * timeout)
|
|
{
|
|
return gst_rtsp_connection_send_usec (conn, message, TV_TO_USEC (timeout));
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_send_messages:
|
|
* @conn: a #GstRTSPConnection
|
|
* @messages: (array length=n_messages): the messages to send
|
|
* @n_messages: the number of messages to send
|
|
* @timeout: a timeout value or %NULL
|
|
*
|
|
* Attempt to send @messages 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.
|
|
*
|
|
* Since: 1.16
|
|
* Deprecated: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_send_messages (GstRTSPConnection * conn,
|
|
GstRTSPMessage * messages, guint n_messages, GTimeVal * timeout)
|
|
{
|
|
return gst_rtsp_connection_send_messages_usec (conn, messages, n_messages,
|
|
TV_TO_USEC (timeout));
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*
|
|
* Deprecated: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_receive (GstRTSPConnection * conn, GstRTSPMessage * message,
|
|
GTimeVal * timeout)
|
|
{
|
|
return gst_rtsp_connection_receive_usec (conn, message, TV_TO_USEC (timeout));
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*
|
|
* Deprecated: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_poll (GstRTSPConnection * conn, GstRTSPEvent events,
|
|
GstRTSPEvent * revents, GTimeVal * timeout)
|
|
{
|
|
return gst_rtsp_connection_poll_usec (conn, events, revents,
|
|
TV_TO_USEC (timeout));
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*
|
|
* Deprecated: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_connection_next_timeout (GstRTSPConnection * conn, GTimeVal * timeout)
|
|
{
|
|
gint64 tmptimeout = 0;
|
|
|
|
g_return_val_if_fail (timeout != NULL, GST_RTSP_EINVAL);
|
|
|
|
tmptimeout = gst_rtsp_connection_next_timeout_usec (conn);
|
|
|
|
timeout->tv_sec = tmptimeout / G_USEC_PER_SEC;
|
|
timeout->tv_usec = tmptimeout % G_USEC_PER_SEC;
|
|
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_rtsp_watch_wait_backlog:
|
|
* @watch: a #GstRTSPWatch
|
|
* @timeout: a GTimeVal timeout
|
|
*
|
|
* Wait until there is place in the backlog queue, @timeout is reached
|
|
* or @watch is set to flushing.
|
|
*
|
|
* 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.
|
|
*
|
|
* The typically use of this function is when gst_rtsp_watch_write_data
|
|
* returns %GST_RTSP_ENOMEM. The caller then calls this function to wait for
|
|
* free space in the backlog queue and try again.
|
|
*
|
|
* Returns: %GST_RTSP_OK when if there is room in queue.
|
|
* %GST_RTSP_ETIMEOUT when @timeout was reached.
|
|
* %GST_RTSP_EINTR when @watch is flushing
|
|
* %GST_RTSP_EINVAL when called with invalid parameters.
|
|
*
|
|
* Since: 1.4
|
|
* Deprecated: 1.18
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_watch_wait_backlog (GstRTSPWatch * watch, GTimeVal * timeout)
|
|
{
|
|
return gst_rtsp_watch_wait_backlog_usec (watch, TV_TO_USEC (timeout));
|
|
}
|
|
|
|
G_GNUC_END_IGNORE_DEPRECATIONS
|
|
#endif /* GST_DISABLE_DEPRECATED */
|