mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-03 13:16:04 +00:00
7446839e0d
This commit adds capability to add custom headers to any http requests during http tunnel mode. If header exist new header will replace old. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5268>
5339 lines
151 KiB
C
5339 lines
151 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 struct
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{
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gchar *key;
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gchar *value;
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} GstRTSPExtraHttpHeader;
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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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/* HTTP tunneling */
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GArray *extra_http_headers;
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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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newconn->extra_http_headers =
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g_array_new (FALSE, FALSE, sizeof (GstRTSPExtraHttpHeader));
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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) (nullable): 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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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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*conn = NULL;
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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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/* 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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stream = G_IO_STREAM (g_socket_connection_factory_create_connection (socket));
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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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*conn = newconn;
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return GST_RTSP_OK;
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|
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/* ERRORS */
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getnameinfo_failed:
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{
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GST_ERROR ("failed to get local address: %s", err->message);
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|
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) (nullable): 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);
|
|
|
|
*conn = NULL;
|
|
|
|
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: (nullable): 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) (nullable): 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: (nullable): 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) (nullable): 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 void
|
|
add_extra_headers (GstRTSPMessage * msg, GArray * headers)
|
|
{
|
|
for (int i = 0; i < headers->len; i++) {
|
|
GstRTSPExtraHttpHeader *hdr =
|
|
&g_array_index (headers, GstRTSPExtraHttpHeader, i);
|
|
|
|
/* Remove any existing header */
|
|
gst_rtsp_message_remove_header_by_name (msg, hdr->key, -1);
|
|
|
|
/* and add... */
|
|
gst_rtsp_message_add_header_by_name (msg, hdr->key, hdr->value);
|
|
}
|
|
}
|
|
|
|
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);
|
|
add_extra_headers (msg, conn->extra_http_headers);
|
|
|
|
/* 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);
|
|
add_extra_headers (msg, conn->extra_http_headers);
|
|
|
|
/* 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: (array length=size): 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) {
|
|
/* only process this as a separate header if there is more than just
|
|
* trailing whitespace after this */
|
|
for (gchar * curr_char = next_value; *curr_char != '\0';
|
|
curr_char++) {
|
|
if (!g_ascii_isspace (*curr_char)) {
|
|
next_value = comma;
|
|
comma = NULL;
|
|
break;
|
|
}
|
|
}
|
|
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: (array length=size): 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: (transfer none): 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);
|
|
|
|
for (gint i = 0; i < conn->extra_http_headers->len; i++) {
|
|
GstRTSPExtraHttpHeader *header =
|
|
&g_array_index (conn->extra_http_headers, GstRTSPExtraHttpHeader, i);
|
|
|
|
g_free (header->key);
|
|
g_free (header->value);
|
|
}
|
|
g_array_free (conn->extra_http_headers, TRUE);
|
|
g_free (conn);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_poll_usec:
|
|
* @conn: a #GstRTSPConnection
|
|
* @events: a bitmask of #GstRTSPEvent flags to check
|
|
* @revents: (out caller-allocates): 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) (nullable): 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) (nullable): 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: (nullable): 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_add_extra_http_request_header:
|
|
* @conn: a #GstRTSPConnection
|
|
* @key: HTTP header name
|
|
* @value: HTTP header value
|
|
*
|
|
* Add header to be appended to any HTTP request made by connection.
|
|
* If the header already exists then the old header is replaced by the new header.
|
|
*
|
|
* Only applicable in HTTP tunnel mode.
|
|
*
|
|
* Since: 1.24
|
|
*/
|
|
void
|
|
gst_rtsp_connection_add_extra_http_request_header (GstRTSPConnection * conn,
|
|
const gchar * key, const gchar * value)
|
|
{
|
|
GstRTSPExtraHttpHeader header;
|
|
|
|
header.key = strdup (key);
|
|
header.value = strdup (value);
|
|
|
|
g_array_append_val (conn->extra_http_headers, header);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_connection_do_tunnel:
|
|
* @conn: a #GstRTSPConnection
|
|
* @conn2: (nullable): 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: (transfer full): 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: (nullable): 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) (optional): 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) (optional): 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) (optional): 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: (array length=size): 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: (array length=size): 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: (transfer none): 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: (out): 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 */
|