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gstreamer/subprojects/gst-plugins-base/gst-libs/gst/rtsp/gstrtspconnection.c

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/* GStreamer
* Copyright (C) <2005-2009> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/*
* Unless otherwise indicated, Source Code is licensed under MIT license.
* See further explanation attached in License Statement (distributed in the file
* LICENSE).
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
* of the Software, and to permit persons to whom the Software is furnished to do
* so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
/**
* SECTION:gstrtspconnection
* @title: GstRTSPConnection
* @short_description: manage RTSP connections
* @see_also: gstrtspurl
*
* This object manages the RTSP connection to the server. It provides function
* to receive and send bytes and messages.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <stdio.h>
#include <errno.h>
#include <stdlib.h>
#include <string.h>
#include <time.h>
/* we include this here to get the G_OS_* defines */
#include <glib.h>
#include <gst/gst.h>
#include <gst/base/base.h>
/* necessary for IP_TOS define */
#include <gio/gnetworking.h>
#include "gstrtspconnection.h"
#ifdef IP_TOS
union gst_sockaddr
{
struct sockaddr sa;
struct sockaddr_in sa_in;
struct sockaddr_in6 sa_in6;
struct sockaddr_storage sa_stor;
};
#endif
typedef struct
{
gint state;
guint save;
guchar out[3]; /* the size must be evenly divisible by 3 */
guint cout;
guint coutl;
} DecodeCtx;
typedef struct
{
/* If %TRUE we only own data and none of the
* other fields
*/
gboolean borrowed;
/* Header or full message */
guint8 *data;
guint data_size;
gboolean data_is_data_header;
/* Payload following data, if any */
guint8 *body_data;
guint body_data_size;
/* or */
GstBuffer *body_buffer;
/* DATA packet header statically allocated for above */
guint8 data_header[4];
/* all below only for async writing */
guint data_offset; /* == data_size when done */
guint body_offset; /* into body_data or the buffer */
/* ID of the message for notification */
guint id;
} GstRTSPSerializedMessage;
static void
gst_rtsp_serialized_message_clear (GstRTSPSerializedMessage * msg)
{
if (!msg->borrowed) {
g_free (msg->body_data);
gst_buffer_replace (&msg->body_buffer, NULL);
}
g_free (msg->data);
}
#ifdef MSG_NOSIGNAL
#define SEND_FLAGS MSG_NOSIGNAL
#else
#define SEND_FLAGS 0
#endif
typedef enum
{
TUNNEL_STATE_NONE,
TUNNEL_STATE_GET,
TUNNEL_STATE_POST,
TUNNEL_STATE_COMPLETE
} GstRTSPTunnelState;
#define TUNNELID_LEN 24
struct _GstRTSPConnection
{
/*< private > */
/* URL for the remote connection */
GstRTSPUrl *url;
GstRTSPVersion version;
gboolean server;
GSocketClient *client;
GIOStream *stream0;
GIOStream *stream1;
GInputStream *input_stream;
GOutputStream *output_stream;
/* this is a read source we add on the write socket in tunneled mode to be
* able to detect when client disconnects the GET channel */
GInputStream *control_stream;
/* connection state */
GSocket *read_socket;
GSocket *write_socket;
GSocket *socket0, *socket1;
gboolean read_socket_used;
gboolean write_socket_used;
GMutex socket_use_mutex;
gboolean manual_http;
gboolean may_cancel;
GCancellable *cancellable;
gchar tunnelid[TUNNELID_LEN];
gboolean tunneled;
gboolean ignore_x_server_reply;
GstRTSPTunnelState tstate;
/* the remote and local ip */
gchar *remote_ip;
gchar *local_ip;
gint read_ahead;
gchar *initial_buffer;
gsize initial_buffer_offset;
gboolean remember_session_id; /* remember the session id or not */
/* Session state */
gint cseq; /* sequence number */
gchar session_id[512]; /* session id */
gint timeout; /* session timeout in seconds */
GTimer *timer; /* timeout timer */
/* Authentication */
GstRTSPAuthMethod auth_method;
gchar *username;
gchar *passwd;
GHashTable *auth_params;
guint content_length_limit;
/* TLS */
GTlsDatabase *tls_database;
GTlsInteraction *tls_interaction;
GstRTSPConnectionAcceptCertificateFunc accept_certificate_func;
GDestroyNotify accept_certificate_destroy_notify;
gpointer accept_certificate_user_data;
DecodeCtx ctx;
DecodeCtx *ctxp;
gchar *proxy_host;
guint proxy_port;
};
enum
{
STATE_START = 0,
STATE_DATA_HEADER,
STATE_DATA_BODY,
STATE_READ_LINES,
STATE_END,
STATE_LAST
};
enum
{
READ_AHEAD_EOH = -1, /* end of headers */
READ_AHEAD_CRLF = -2,
READ_AHEAD_CRLFCR = -3
};
/* a structure for constructing RTSPMessages */
typedef struct
{
gint state;
GstRTSPResult status;
guint8 buffer[4096];
guint offset;
guint line;
guint8 *body_data;
guint body_len;
} GstRTSPBuilder;
/* function prototypes */
static void add_auth_header (GstRTSPConnection * conn,
GstRTSPMessage * message);
static void
build_reset (GstRTSPBuilder * builder)
{
g_free (builder->body_data);
memset (builder, 0, sizeof (GstRTSPBuilder));
}
static GstRTSPResult
gst_rtsp_result_from_g_io_error (GError * error, GstRTSPResult default_res)
{
if (error == NULL)
return GST_RTSP_OK;
if (error->domain != G_IO_ERROR)
return default_res;
switch (error->code) {
case G_IO_ERROR_TIMED_OUT:
return GST_RTSP_ETIMEOUT;
case G_IO_ERROR_INVALID_ARGUMENT:
return GST_RTSP_EINVAL;
case G_IO_ERROR_CANCELLED:
case G_IO_ERROR_WOULD_BLOCK:
return GST_RTSP_EINTR;
default:
return default_res;
}
}
static gboolean
tls_accept_certificate (GTlsConnection * conn, GTlsCertificate * peer_cert,
GTlsCertificateFlags errors, GstRTSPConnection * rtspconn)
{
GError *error = NULL;
gboolean accept = FALSE;
if (rtspconn->tls_database) {
GSocketConnectable *peer_identity;
GTlsCertificateFlags validation_flags;
GST_DEBUG ("TLS peer certificate not accepted, checking user database...");
peer_identity =
g_tls_client_connection_get_server_identity (G_TLS_CLIENT_CONNECTION
(conn));
errors =
g_tls_database_verify_chain (rtspconn->tls_database, peer_cert,
G_TLS_DATABASE_PURPOSE_AUTHENTICATE_SERVER, peer_identity,
g_tls_connection_get_interaction (conn), G_TLS_DATABASE_VERIFY_NONE,
NULL, &error);
if (error)
goto verify_error;
validation_flags = gst_rtsp_connection_get_tls_validation_flags (rtspconn);
accept = ((errors & validation_flags) == 0);
if (accept)
GST_DEBUG ("Peer certificate accepted");
else
GST_DEBUG ("Peer certificate not accepted (errors: 0x%08X)", errors);
}
if (!accept && rtspconn->accept_certificate_func) {
accept =
rtspconn->accept_certificate_func (conn, peer_cert, errors,
rtspconn->accept_certificate_user_data);
GST_DEBUG ("Peer certificate %saccepted by accept-certificate function",
accept ? "" : "not ");
}
return accept;
/* ERRORS */
verify_error:
{
GST_ERROR ("An error occurred while verifying the peer certificate: %s",
error->message);
g_clear_error (&error);
return FALSE;
}
}
static void
socket_client_event (GSocketClient * client, GSocketClientEvent event,
GSocketConnectable * connectable, GTlsConnection * connection,
GstRTSPConnection * rtspconn)
{
if (event == G_SOCKET_CLIENT_TLS_HANDSHAKING) {
GST_DEBUG ("TLS handshaking about to start...");
g_signal_connect (connection, "accept-certificate",
(GCallback) tls_accept_certificate, rtspconn);
g_tls_connection_set_interaction (connection, rtspconn->tls_interaction);
}
}
/**
* gst_rtsp_connection_create:
* @url: a #GstRTSPUrl
* @conn: (out) (transfer full): storage for a #GstRTSPConnection
*
* Create a newly allocated #GstRTSPConnection from @url and store it in @conn.
* The connection will not yet attempt to connect to @url, use
* gst_rtsp_connection_connect().
*
* A copy of @url will be made.
*
* Returns: #GST_RTSP_OK when @conn contains a valid connection.
*/
GstRTSPResult
gst_rtsp_connection_create (const GstRTSPUrl * url, GstRTSPConnection ** conn)
{
GstRTSPConnection *newconn;
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (url != NULL, GST_RTSP_EINVAL);
newconn = g_new0 (GstRTSPConnection, 1);
newconn->may_cancel = TRUE;
newconn->cancellable = g_cancellable_new ();
newconn->client = g_socket_client_new ();
if (url->transports & GST_RTSP_LOWER_TRANS_TLS)
g_socket_client_set_tls (newconn->client, TRUE);
g_signal_connect (newconn->client, "event", (GCallback) socket_client_event,
newconn);
newconn->url = gst_rtsp_url_copy (url);
newconn->timer = g_timer_new ();
newconn->timeout = 60;
newconn->cseq = 1; /* RFC 7826: "it is RECOMMENDED to start at 0.",
but some servers don't copy values <1 due to bugs. */
newconn->remember_session_id = TRUE;
newconn->auth_method = GST_RTSP_AUTH_NONE;
newconn->username = NULL;
newconn->passwd = NULL;
newconn->auth_params = NULL;
newconn->version = 0;
newconn->content_length_limit = G_MAXUINT;
*conn = newconn;
return GST_RTSP_OK;
}
static gboolean
collect_addresses (GSocket * socket, gchar ** ip, guint16 * port,
gboolean remote, GError ** error)
{
GSocketAddress *addr;
if (remote)
addr = g_socket_get_remote_address (socket, error);
else
addr = g_socket_get_local_address (socket, error);
if (!addr)
return FALSE;
if (ip)
*ip = g_inet_address_to_string (g_inet_socket_address_get_address
(G_INET_SOCKET_ADDRESS (addr)));
if (port)
*port = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
g_object_unref (addr);
return TRUE;
}
/**
* gst_rtsp_connection_create_from_socket:
* @socket: a #GSocket
* @ip: the IP address of the other end
* @port: the port used by the other end
* @initial_buffer: data already read from @fd
* @conn: (out) (transfer full): storage for a #GstRTSPConnection
*
* Create a new #GstRTSPConnection for handling communication on the existing
* socket @socket. The @initial_buffer contains zero terminated data already
* read from @socket which should be used before starting to read new data.
*
* Returns: #GST_RTSP_OK when @conn contains a valid connection.
*/
/* FIXME 2.0 We don't need the ip and port since they can be got from the
* GSocket */
GstRTSPResult
gst_rtsp_connection_create_from_socket (GSocket * socket, const gchar * ip,
guint16 port, const gchar * initial_buffer, GstRTSPConnection ** conn)
{
GstRTSPConnection *newconn = NULL;
GstRTSPUrl *url;
GstRTSPResult res;
GError *err = NULL;
gchar *local_ip;
GIOStream *stream;
g_return_val_if_fail (G_IS_SOCKET (socket), GST_RTSP_EINVAL);
g_return_val_if_fail (ip != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
if (!collect_addresses (socket, &local_ip, NULL, FALSE, &err))
goto getnameinfo_failed;
/* create a url for the client address */
url = g_new0 (GstRTSPUrl, 1);
url->host = g_strdup (ip);
url->port = port;
/* now create the connection object */
GST_RTSP_CHECK (gst_rtsp_connection_create (url, &newconn), newconn_failed);
gst_rtsp_url_free (url);
stream = G_IO_STREAM (g_socket_connection_factory_create_connection (socket));
/* both read and write initially */
newconn->server = TRUE;
newconn->socket0 = socket;
newconn->stream0 = stream;
newconn->write_socket = newconn->read_socket = newconn->socket0;
newconn->read_socket_used = FALSE;
newconn->write_socket_used = FALSE;
g_mutex_init (&newconn->socket_use_mutex);
newconn->input_stream = g_io_stream_get_input_stream (stream);
newconn->output_stream = g_io_stream_get_output_stream (stream);
newconn->control_stream = NULL;
newconn->remote_ip = g_strdup (ip);
newconn->local_ip = local_ip;
newconn->initial_buffer = g_strdup (initial_buffer);
*conn = newconn;
return GST_RTSP_OK;
/* ERRORS */
getnameinfo_failed:
{
GST_ERROR ("failed to get local address: %s", err->message);
res = gst_rtsp_result_from_g_io_error (err, GST_RTSP_ERROR);
g_clear_error (&err);
return res;
}
newconn_failed:
{
GST_ERROR ("failed to make connection");
g_free (local_ip);
gst_rtsp_url_free (url);
return res;
}
}
/**
* gst_rtsp_connection_accept:
* @socket: a socket
* @conn: (out) (transfer full): storage for a #GstRTSPConnection
* @cancellable: a #GCancellable to cancel the operation
*
* Accept a new connection on @socket and create a new #GstRTSPConnection for
* handling communication on new socket.
*
* Returns: #GST_RTSP_OK when @conn contains a valid connection.
*/
GstRTSPResult
gst_rtsp_connection_accept (GSocket * socket, GstRTSPConnection ** conn,
GCancellable * cancellable)
{
GError *err = NULL;
gchar *ip;
guint16 port;
GSocket *client_sock;
GstRTSPResult ret;
g_return_val_if_fail (G_IS_SOCKET (socket), GST_RTSP_EINVAL);
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
client_sock = g_socket_accept (socket, cancellable, &err);
if (!client_sock)
goto accept_failed;
/* get the remote ip address and port */
if (!collect_addresses (client_sock, &ip, &port, TRUE, &err))
goto getnameinfo_failed;
ret =
gst_rtsp_connection_create_from_socket (client_sock, ip, port, NULL,
conn);
g_object_unref (client_sock);
g_free (ip);
return ret;
/* ERRORS */
accept_failed:
{
GST_DEBUG ("Accepting client failed: %s", err->message);
ret = gst_rtsp_result_from_g_io_error (err, GST_RTSP_ESYS);
g_clear_error (&err);
return ret;
}
getnameinfo_failed:
{
GST_DEBUG ("getnameinfo failed: %s", err->message);
ret = gst_rtsp_result_from_g_io_error (err, GST_RTSP_ERROR);
g_clear_error (&err);
if (!g_socket_close (client_sock, &err)) {
GST_DEBUG ("Closing socket failed: %s", err->message);
g_clear_error (&err);
}
g_object_unref (client_sock);
return ret;
}
}
/**
* gst_rtsp_connection_get_tls:
* @conn: a #GstRTSPConnection
* @error: #GError for error reporting, or NULL to ignore.
*
* Get the TLS connection of @conn.
*
* For client side this will return the #GTlsClientConnection when connected
* over TLS.
*
* For server side connections, this function will create a GTlsServerConnection
* when called the first time and will return that same connection on subsequent
* calls. The server is then responsible for configuring the TLS connection.
*
* Returns: (transfer none): the TLS connection for @conn.
*
* Since: 1.2
*/
GTlsConnection *
gst_rtsp_connection_get_tls (GstRTSPConnection * conn, GError ** error)
{
GTlsConnection *result;
if (G_IS_TLS_CONNECTION (conn->stream0)) {
/* we already had one, return it */
result = G_TLS_CONNECTION (conn->stream0);
} else if (conn->server) {
/* no TLS connection but we are server, make one */
result = (GTlsConnection *)
g_tls_server_connection_new (conn->stream0, NULL, error);
if (result) {
g_object_unref (conn->stream0);
conn->stream0 = G_IO_STREAM (result);
conn->input_stream = g_io_stream_get_input_stream (conn->stream0);
conn->output_stream = g_io_stream_get_output_stream (conn->stream0);
}
} else {
/* client */
result = NULL;
g_set_error (error, GST_LIBRARY_ERROR, GST_LIBRARY_ERROR_FAILED,
"client not connected with TLS");
}
return result;
}
/**
* gst_rtsp_connection_set_tls_validation_flags:
* @conn: a #GstRTSPConnection
* @flags: the validation flags.
*
* Sets the TLS validation flags to be used to verify the peer
* certificate when a TLS connection is established.
*
* 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);
if (res)
g_socket_client_set_tls_validation_flags (conn->client, flags);
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.
*
* Returns: the validationg flags.
*
* Since: 1.2.1
*/
GTlsCertificateFlags
gst_rtsp_connection_get_tls_validation_flags (GstRTSPConnection * conn)
{
g_return_val_if_fail (conn != NULL, 0);
return g_socket_client_get_tls_validation_flags (conn->client);
}
/**
* gst_rtsp_connection_set_tls_database:
* @conn: a #GstRTSPConnection
* @database: a #GTlsDatabase
*
* Sets the anchor certificate authorities database. This certificate
* database will be used to verify the server's certificate in case it
* can't be verified with the default certificate database first.
*
* Since: 1.4
*/
void
gst_rtsp_connection_set_tls_database (GstRTSPConnection * conn,
GTlsDatabase * database)
{
GTlsDatabase *old_db;
g_return_if_fail (conn != NULL);
if (database)
g_object_ref (database);
old_db = conn->tls_database;
conn->tls_database = database;
if (old_db)
g_object_unref (old_db);
}
/**
* gst_rtsp_connection_get_tls_database:
* @conn: a #GstRTSPConnection
*
* Gets the anchor certificate authorities database that will be used
* after a server certificate can't be verified with the default
* certificate database.
*
* Returns: (transfer full): the anchor certificate authorities database, or NULL if no
* database has been previously set. Use g_object_unref() to release the
* certificate database.
*
* Since: 1.4
*/
GTlsDatabase *
gst_rtsp_connection_get_tls_database (GstRTSPConnection * conn)
{
GTlsDatabase *result;
g_return_val_if_fail (conn != NULL, NULL);
if ((result = conn->tls_database))
g_object_ref (result);
return result;
}
/**
* gst_rtsp_connection_set_tls_interaction:
* @conn: a #GstRTSPConnection
* @interaction: a #GTlsInteraction
*
* Sets a #GTlsInteraction object to be used when the connection or certificate
* database need to interact with the user. This will be used to prompt the
* user for passwords where necessary.
*
* Since: 1.6
*/
void
gst_rtsp_connection_set_tls_interaction (GstRTSPConnection * conn,
GTlsInteraction * interaction)
{
GTlsInteraction *old_interaction;
g_return_if_fail (conn != NULL);
if (interaction)
g_object_ref (interaction);
old_interaction = conn->tls_interaction;
conn->tls_interaction = interaction;
if (old_interaction)
g_object_unref (old_interaction);
}
/**
* gst_rtsp_connection_get_tls_interaction:
* @conn: a #GstRTSPConnection
*
* Gets a #GTlsInteraction object to be used when the connection or certificate
* database need to interact with the user. This will be used to prompt the
* user for passwords where necessary.
*
* Returns: (transfer full): a reference on the #GTlsInteraction. Use
* g_object_unref() to release.
*
* Since: 1.6
*/
GTlsInteraction *
gst_rtsp_connection_get_tls_interaction (GstRTSPConnection * conn)
{
GTlsInteraction *result;
g_return_val_if_fail (conn != NULL, NULL);
if ((result = conn->tls_interaction))
g_object_ref (result);
return result;
}
/**
* gst_rtsp_connection_set_accept_certificate_func:
* @conn: a #GstRTSPConnection
* @func: a #GstRTSPConnectionAcceptCertificateFunc to check certificates
* @destroy_notify: #GDestroyNotify for @user_data
* @user_data: User data passed to @func
*
* Sets a custom accept-certificate function for checking certificates for
* validity. This will directly map to #GTlsConnection 's "accept-certificate"
* signal and be performed after the default checks of #GstRTSPConnection
* (checking against the #GTlsDatabase with the given #GTlsCertificateFlags)
* have failed. If no #GTlsDatabase is set on this connection, only @func will
* be called.
*
* Since: 1.14
*/
void
gst_rtsp_connection_set_accept_certificate_func (GstRTSPConnection * conn,
GstRTSPConnectionAcceptCertificateFunc func,
gpointer user_data, GDestroyNotify destroy_notify)
{
if (conn->accept_certificate_destroy_notify)
conn->
accept_certificate_destroy_notify (conn->accept_certificate_user_data);
conn->accept_certificate_func = func;
conn->accept_certificate_user_data = user_data;
conn->accept_certificate_destroy_notify = destroy_notify;
}
static gchar *
get_tunneled_connection_uri_strdup (GstRTSPUrl * url, guint16 port)
{
const gchar *pre_host = "";
const gchar *post_host = "";
if (url->family == GST_RTSP_FAM_INET6) {
pre_host = "[";
post_host = "]";
}
return g_strdup_printf ("http://%s%s%s:%d%s%s%s", pre_host, url->host,
post_host, port, url->abspath, url->query ? "?" : "",
url->query ? url->query : "");
}
static GstRTSPResult
setup_tunneling (GstRTSPConnection * conn, gint64 timeout, gchar * uri,
GstRTSPMessage * response)
{
gint i;
GstRTSPResult res;
gchar *value;
guint16 url_port;
GstRTSPMessage *msg;
gboolean old_http;
GstRTSPUrl *url;
GError *error = NULL;
GSocketConnection *connection;
GSocket *socket;
gchar *connection_uri = NULL;
gchar *request_uri = NULL;
gchar *host = NULL;
url = conn->url;
gst_rtsp_url_get_port (url, &url_port);
host = g_strdup_printf ("%s:%d", url->host, url_port);
/* create a random sessionid */
for (i = 0; i < TUNNELID_LEN; i++)
conn->tunnelid[i] = g_random_int_range ('a', 'z');
conn->tunnelid[TUNNELID_LEN - 1] = '\0';
/* create the GET request for the read connection */
GST_RTSP_CHECK (gst_rtsp_message_new_request (&msg, GST_RTSP_GET, uri),
no_message);
msg->type = GST_RTSP_MESSAGE_HTTP_REQUEST;
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_X_SESSIONCOOKIE,
conn->tunnelid);
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_ACCEPT,
"application/x-rtsp-tunnelled");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CACHE_CONTROL, "no-cache");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_PRAGMA, "no-cache");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_HOST, host);
/* we need to temporarily set conn->tunneled to FALSE to prevent the HTTP
* request from being base64 encoded */
conn->tunneled = FALSE;
GST_RTSP_CHECK (gst_rtsp_connection_send_usec (conn, msg, timeout),
write_failed);
gst_rtsp_message_free (msg);
conn->tunneled = TRUE;
/* receive the response to the GET request */
/* we need to temporarily set manual_http to TRUE since
* gst_rtsp_connection_receive() will treat the HTTP response as a parsing
* failure otherwise */
old_http = conn->manual_http;
conn->manual_http = TRUE;
GST_RTSP_CHECK (gst_rtsp_connection_receive_usec (conn, response, timeout),
read_failed);
conn->manual_http = old_http;
if (response->type != GST_RTSP_MESSAGE_HTTP_RESPONSE ||
response->type_data.response.code != GST_RTSP_STS_OK)
goto wrong_result;
if (!conn->ignore_x_server_reply &&
gst_rtsp_message_get_header (response, GST_RTSP_HDR_X_SERVER_IP_ADDRESS,
&value, 0) == GST_RTSP_OK) {
g_free (url->host);
url->host = g_strdup (value);
g_free (conn->remote_ip);
conn->remote_ip = g_strdup (value);
}
connection_uri = get_tunneled_connection_uri_strdup (url, url_port);
/* connect to the host/port */
if (conn->proxy_host) {
connection = g_socket_client_connect_to_host (conn->client,
conn->proxy_host, conn->proxy_port, conn->cancellable, &error);
request_uri = g_strdup (connection_uri);
} else {
connection = g_socket_client_connect_to_uri (conn->client,
connection_uri, 0, conn->cancellable, &error);
request_uri =
g_strdup_printf ("%s%s%s", url->abspath,
url->query ? "?" : "", url->query ? url->query : "");
}
if (connection == NULL)
goto connect_failed;
socket = g_socket_connection_get_socket (connection);
/* get remote address */
g_free (conn->remote_ip);
conn->remote_ip = NULL;
if (!collect_addresses (socket, &conn->remote_ip, NULL, TRUE, &error))
goto remote_address_failed;
/* this is now our writing socket */
conn->stream1 = G_IO_STREAM (connection);
conn->socket1 = socket;
conn->write_socket = conn->socket1;
conn->output_stream = g_io_stream_get_output_stream (conn->stream1);
conn->control_stream = NULL;
/* create the POST request for the write connection */
GST_RTSP_CHECK (gst_rtsp_message_new_request (&msg, GST_RTSP_POST,
request_uri), no_message);
msg->type = GST_RTSP_MESSAGE_HTTP_REQUEST;
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_X_SESSIONCOOKIE,
conn->tunnelid);
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_ACCEPT,
"application/x-rtsp-tunnelled");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CONTENT_TYPE,
"application/x-rtsp-tunnelled");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CACHE_CONTROL, "no-cache");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_PRAGMA, "no-cache");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_EXPIRES,
"Sun, 9 Jan 1972 00:00:00 GMT");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CONTENT_LENGTH, "32767");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_HOST, host);
/* we need to temporarily set conn->tunneled to FALSE to prevent the HTTP
* request from being base64 encoded */
conn->tunneled = FALSE;
GST_RTSP_CHECK (gst_rtsp_connection_send_usec (conn, msg, timeout),
write_failed);
gst_rtsp_message_free (msg);
conn->tunneled = TRUE;
exit:
g_free (connection_uri);
g_free (request_uri);
g_free (host);
return res;
/* ERRORS */
no_message:
{
GST_ERROR ("failed to create request (%d)", res);
goto exit;
}
write_failed:
{
GST_ERROR ("write failed (%d)", res);
gst_rtsp_message_free (msg);
conn->tunneled = TRUE;
goto exit;
}
read_failed:
{
GST_ERROR ("read failed (%d)", res);
conn->manual_http = FALSE;
goto exit;
}
wrong_result:
{
GST_ERROR ("got failure response %d %s",
response->type_data.response.code, response->type_data.response.reason);
res = GST_RTSP_ERROR;
goto exit;
}
connect_failed:
{
GST_ERROR ("failed to connect: %s", error->message);
res = gst_rtsp_result_from_g_io_error (error, GST_RTSP_ERROR);
g_clear_error (&error);
goto exit;
}
remote_address_failed:
{
GST_ERROR ("failed to resolve address: %s", error->message);
res = gst_rtsp_result_from_g_io_error (error, GST_RTSP_ERROR);
g_object_unref (connection);
g_clear_error (&error);
return res;
}
}
/**
* gst_rtsp_connection_connect_with_response_usec:
* @conn: a #GstRTSPConnection
* @timeout: a timeout in microseconds
* @response: a #GstRTSPMessage
*
* Attempt to connect to the url of @conn made with
* gst_rtsp_connection_create(). If @timeout is 0 this function can block
* forever. If @timeout contains a valid timeout, this function will return
* #GST_RTSP_ETIMEOUT after the timeout expired. If @conn is set to tunneled,
* @response will contain a response to the tunneling request messages.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK when a connection could be made.
*
* Since: 1.18
*/
GstRTSPResult
gst_rtsp_connection_connect_with_response_usec (GstRTSPConnection * conn,
gint64 timeout, GstRTSPMessage * response)
{
GstRTSPResult res;
GSocketConnection *connection;
GSocket *socket;
GError *error = NULL;
gchar *connection_uri, *request_uri, *remote_ip;
GstClockTime to;
guint16 url_port;
GstRTSPUrl *url;
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);
}
if (conn->proxy_host) {
connection = g_socket_client_connect_to_host (conn->client,
conn->proxy_host, conn->proxy_port, conn->cancellable, &error);
request_uri = g_strdup (connection_uri);
} else {
connection = g_socket_client_connect_to_uri (conn->client,
connection_uri, url_port, conn->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 : "");
}
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:
{
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! */
#if GLIB_CHECK_VERSION(2, 59, 1)
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;
}
}
#else
static GstRTSPResult
writev_bytes (GOutputStream * stream, GOutputVector * vectors, gint n_vectors,
gsize * bytes_written, gboolean block, GCancellable * cancellable)
{
gsize _bytes_written = 0;
guint written;
gint i;
GstRTSPResult res = GST_RTSP_OK;
for (i = 0; i < n_vectors; i++) {
written = 0;
res =
write_bytes (stream, vectors[i].buffer, &written, vectors[i].size,
block, cancellable);
_bytes_written += written;
if (G_UNLIKELY (res != GST_RTSP_OK))
break;
}
*bytes_written = _bytes_written;
return res;
}
#endif
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;
if (block)
r = g_input_stream_read (conn->input_stream, (gchar *) & buffer[out],
count, conn->may_cancel ? conn->cancellable : NULL, err);
else
r = g_pollable_input_stream_read_nonblocking (G_POLLABLE_INPUT_STREAM
(conn->input_stream), (gchar *) & buffer[out], count,
conn->may_cancel ? conn->cancellable : NULL, err);
if (G_UNLIKELY (r < 0)) {
if (out == 0) {
/* propagate the error */
out = r;
} else {
/* we have some data ignore error */
g_clear_error (err);
}
} else
out += r;
}
return out;
}
static gint
fill_bytes (GstRTSPConnection * conn, guint8 * buffer, guint size,
gboolean block, GError ** err)
{
DecodeCtx *ctx = conn->ctxp;
gint out = 0;
if (ctx) {
while (size > 0) {
guint8 in[sizeof (ctx->out) * 4 / 3];
gint r;
while (size > 0 && ctx->cout < ctx->coutl) {
/* we have some leftover bytes */
*buffer++ = ctx->out[ctx->cout++];
size--;
out++;
}
/* got what we needed? */
if (size == 0)
break;
/* try to read more bytes */
r = fill_raw_bytes (conn, in, sizeof (in), block, err);
if (r <= 0) {
if (out == 0) {
out = r;
} else {
/* we have some data ignore error */
g_clear_error (err);
}
break;
}
ctx->cout = 0;
ctx->coutl =
g_base64_decode_step ((gchar *) in, r, ctx->out, &ctx->state,
&ctx->save);
}
} else {
out = fill_raw_bytes (conn, buffer, size, block, err);
}
return out;
}
static GstRTSPResult
read_bytes (GstRTSPConnection * conn, guint8 * buffer, guint * idx, guint size,
gboolean block)
{
guint left;
gint r;
GstRTSPResult res;
GError *err = NULL;
if (G_UNLIKELY (*idx > size))
return GST_RTSP_ERROR;
left = size - *idx;
while (left) {
r = fill_bytes (conn, &buffer[*idx], left, block, &err);
if (G_UNLIKELY (r <= 0))
goto error;
left -= r;
*idx += r;
}
return GST_RTSP_OK;
/* ERRORS */
error:
{
if (G_UNLIKELY (r == 0))
return GST_RTSP_EEOF;
GST_DEBUG ("%s", err->message);
res = gst_rtsp_result_from_g_io_error (err, GST_RTSP_ESYS);
g_clear_error (&err);
return res;
}
}
/* The code below tries to handle clients using \r, \n or \r\n to indicate the
* end of a line. It even does its best to handle clients which mix them (even
* though this is a really stupid idea (tm).) It also handles Line White Space
* (LWS), where a line end followed by whitespace is considered LWS. This is
* the method used in RTSP (and HTTP) to break long lines.
*/
static GstRTSPResult
read_line (GstRTSPConnection * conn, guint8 * buffer, guint * idx, guint size,
gboolean block)
{
GstRTSPResult res;
while (TRUE) {
guint8 c;
guint i;
if (conn->read_ahead == READ_AHEAD_EOH) {
/* the last call to read_line() already determined that we have reached
* the end of the headers, so convey that information now */
conn->read_ahead = 0;
break;
} else if (conn->read_ahead == READ_AHEAD_CRLF) {
/* the last call to read_line() left off after having read \r\n */
c = '\n';
} else if (conn->read_ahead == READ_AHEAD_CRLFCR) {
/* the last call to read_line() left off after having read \r\n\r */
c = '\r';
} else if (conn->read_ahead != 0) {
/* the last call to read_line() left us with a character to start with */
c = (guint8) conn->read_ahead;
conn->read_ahead = 0;
} else {
/* read the next character */
i = 0;
res = read_bytes (conn, &c, &i, 1, block);
if (G_UNLIKELY (res != GST_RTSP_OK))
return res;
}
/* special treatment of line endings */
if (c == '\r' || c == '\n') {
guint8 read_ahead;
retry:
/* need to read ahead one more character to know what to do... */
i = 0;
res = read_bytes (conn, &read_ahead, &i, 1, block);
if (G_UNLIKELY (res != GST_RTSP_OK))
return res;
if (read_ahead == ' ' || read_ahead == '\t') {
if (conn->read_ahead == READ_AHEAD_CRLFCR) {
/* got \r\n\r followed by whitespace, treat it as a normal line
* followed by one starting with LWS */
conn->read_ahead = read_ahead;
break;
} else {
/* got LWS, change the line ending to a space and continue */
c = ' ';
conn->read_ahead = read_ahead;
}
} else if (conn->read_ahead == READ_AHEAD_CRLFCR) {
if (read_ahead == '\r' || read_ahead == '\n') {
/* got \r\n\r\r or \r\n\r\n, treat it as the end of the headers */
conn->read_ahead = READ_AHEAD_EOH;
break;
} else {
/* got \r\n\r followed by something else, this is not really
* supported since we have probably just eaten the first character
* of the body or the next message, so just ignore the second \r
* and live with it... */
conn->read_ahead = read_ahead;
break;
}
} else if (conn->read_ahead == READ_AHEAD_CRLF) {
if (read_ahead == '\r') {
/* got \r\n\r so far, need one more character... */
conn->read_ahead = READ_AHEAD_CRLFCR;
goto retry;
} else if (read_ahead == '\n') {
/* got \r\n\n, treat it as the end of the headers */
conn->read_ahead = READ_AHEAD_EOH;
break;
} else {
/* found the end of a line, keep read_ahead for the next line */
conn->read_ahead = read_ahead;
break;
}
} else if (c == read_ahead) {
/* got double \r or \n, treat it as the end of the headers */
conn->read_ahead = READ_AHEAD_EOH;
break;
} else if (c == '\r' && read_ahead == '\n') {
/* got \r\n so far, still need more to know what to do... */
conn->read_ahead = READ_AHEAD_CRLF;
goto retry;
} else {
/* found the end of a line, keep read_ahead for the next line */
conn->read_ahead = read_ahead;
break;
}
}
if (G_LIKELY (*idx < size - 1))
buffer[(*idx)++] = c;
}
buffer[*idx] = '\0';
return GST_RTSP_OK;
}
static void
set_read_socket_timeout (GstRTSPConnection * conn, gint64 timeout)
{
GstClockTime to_nsecs;
guint to_secs;
g_mutex_lock (&conn->socket_use_mutex);
g_assert (!conn->read_socket_used);
conn->read_socket_used = TRUE;
to_nsecs = timeout * 1000;
to_secs = (to_nsecs + GST_SECOND - 1) / GST_SECOND;
if (to_secs > g_socket_get_timeout (conn->read_socket)) {
g_socket_set_timeout (conn->read_socket, to_secs);
}
g_mutex_unlock (&conn->socket_use_mutex);
}
static void
set_write_socket_timeout (GstRTSPConnection * conn, gint64 timeout)
{
GstClockTime to_nsecs;
guint to_secs;
g_mutex_lock (&conn->socket_use_mutex);
g_assert (!conn->write_socket_used);
conn->write_socket_used = TRUE;
to_nsecs = timeout * 1000;
to_secs = (to_nsecs + GST_SECOND - 1) / GST_SECOND;
if (to_secs > g_socket_get_timeout (conn->write_socket)) {
g_socket_set_timeout (conn->write_socket, to_secs);
}
g_mutex_unlock (&conn->socket_use_mutex);
}
static void
clear_read_socket_timeout (GstRTSPConnection * conn)
{
g_mutex_lock (&conn->socket_use_mutex);
conn->read_socket_used = FALSE;
if (conn->read_socket != conn->write_socket || !conn->write_socket_used) {
g_socket_set_timeout (conn->read_socket, 0);
}
g_mutex_unlock (&conn->socket_use_mutex);
}
static void
clear_write_socket_timeout (GstRTSPConnection * conn)
{
g_mutex_lock (&conn->socket_use_mutex);
conn->write_socket_used = FALSE;
if (conn->write_socket != conn->read_socket || !conn->read_socket_used) {
g_socket_set_timeout (conn->write_socket, 0);
}
g_mutex_unlock (&conn->socket_use_mutex);
}
/**
* gst_rtsp_connection_write_usec:
* @conn: a #GstRTSPConnection
* @data: the data to write
* @size: the size of @data
* @timeout: a timeout value or 0
*
* Attempt to write @size bytes of @data to the connected @conn, blocking up to
* the specified @timeout. @timeout can be 0, in which case this function
* might block forever.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK on success.
*
* Since: 1.18
*/
/* FIXME 2.0: This should've been static! */
GstRTSPResult
gst_rtsp_connection_write_usec (GstRTSPConnection * conn, const guint8 * data,
guint size, gint64 timeout)
{
guint offset;
GstRTSPResult res;
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);
res =
write_bytes (conn->output_stream, data, &offset, size, TRUE,
conn->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;
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);
res =
writev_bytes (conn->output_stream, vectors, n_vectors, &bytes_written,
TRUE, conn->cancellable);
clear_write_socket_timeout (conn);
g_assert (bytes_written == bytes_to_write || res != GST_RTSP_OK);
/* free everything */
for (i = 0, k = 0; i < n_messages; i++) {
if (serialized_messages[i].body_buffer) {
gint l, n;
n = gst_buffer_n_memory (serialized_messages[i].body_buffer);
for (l = 0; l < n; l++) {
GstMemory *mem =
gst_buffer_peek_memory (serialized_messages[i].body_buffer, l);
gst_memory_unmap (mem, &map_infos[k]);
k++;
}
}
g_free (serialized_messages[i].data);
}
return res;
no_message:
{
for (i = 0; i < n_messages; i++) {
gst_rtsp_serialized_message_clear (&serialized_messages[i]);
}
g_warning ("Wrong message");
return GST_RTSP_EINVAL;
}
}
static GstRTSPResult
parse_string (gchar * dest, gint size, gchar ** src)
{
GstRTSPResult res = GST_RTSP_OK;
gint idx;
idx = 0;
/* skip spaces */
while (g_ascii_isspace (**src))
(*src)++;
while (!g_ascii_isspace (**src) && **src != '\0') {
if (idx < size - 1)
dest[idx++] = **src;
else
res = GST_RTSP_EPARSE;
(*src)++;
}
if (size > 0)
dest[idx] = '\0';
return res;
}
static GstRTSPResult
parse_protocol_version (gchar * protocol, GstRTSPMsgType * type,
GstRTSPVersion * version)
{
GstRTSPVersion rversion;
GstRTSPResult res = GST_RTSP_OK;
gchar *ver;
if (G_LIKELY ((ver = strchr (protocol, '/')) != NULL)) {
guint major;
guint minor;
gchar dummychar;
*ver++ = '\0';
/* the version number must be formatted as X.Y with nothing following */
if (sscanf (ver, "%u.%u%c", &major, &minor, &dummychar) != 2)
res = GST_RTSP_EPARSE;
rversion = major * 0x10 + minor;
if (g_ascii_strcasecmp (protocol, "RTSP") == 0) {
if (rversion != GST_RTSP_VERSION_1_0 && rversion != GST_RTSP_VERSION_2_0) {
*version = GST_RTSP_VERSION_INVALID;
res = GST_RTSP_ERROR;
}
} else if (g_ascii_strcasecmp (protocol, "HTTP") == 0) {
if (*type == GST_RTSP_MESSAGE_REQUEST)
*type = GST_RTSP_MESSAGE_HTTP_REQUEST;
else if (*type == GST_RTSP_MESSAGE_RESPONSE)
*type = GST_RTSP_MESSAGE_HTTP_RESPONSE;
if (rversion != GST_RTSP_VERSION_1_0 &&
rversion != GST_RTSP_VERSION_1_1 && rversion != GST_RTSP_VERSION_2_0)
res = GST_RTSP_ERROR;
} else
res = GST_RTSP_EPARSE;
} else
res = GST_RTSP_EPARSE;
if (res == GST_RTSP_OK)
*version = rversion;
return res;
}
static GstRTSPResult
parse_response_status (guint8 * buffer, GstRTSPMessage * msg)
{
GstRTSPResult res = GST_RTSP_OK;
GstRTSPResult res2;
gchar versionstr[20];
gchar codestr[4];
gint code;
gchar *bptr;
bptr = (gchar *) buffer;
if (parse_string (versionstr, sizeof (versionstr), &bptr) != GST_RTSP_OK)
res = GST_RTSP_EPARSE;
if (parse_string (codestr, sizeof (codestr), &bptr) != GST_RTSP_OK)
res = GST_RTSP_EPARSE;
code = atoi (codestr);
if (G_UNLIKELY (*codestr == '\0' || code < 0 || code >= 600))
res = GST_RTSP_EPARSE;
while (g_ascii_isspace (*bptr))
bptr++;
if (G_UNLIKELY (gst_rtsp_message_init_response (msg, code, bptr,
NULL) != GST_RTSP_OK))
res = GST_RTSP_EPARSE;
res2 = parse_protocol_version (versionstr, &msg->type,
&msg->type_data.response.version);
if (G_LIKELY (res == GST_RTSP_OK))
res = res2;
return res;
}
static GstRTSPResult
parse_request_line (guint8 * buffer, GstRTSPMessage * msg)
{
GstRTSPResult res = GST_RTSP_OK;
GstRTSPResult res2;
gchar versionstr[20];
gchar methodstr[20];
gchar urlstr[4096];
gchar *bptr;
GstRTSPMethod method;
bptr = (gchar *) buffer;
if (parse_string (methodstr, sizeof (methodstr), &bptr) != GST_RTSP_OK)
res = GST_RTSP_EPARSE;
method = gst_rtsp_find_method (methodstr);
if (parse_string (urlstr, sizeof (urlstr), &bptr) != GST_RTSP_OK)
res = GST_RTSP_EPARSE;
if (G_UNLIKELY (*urlstr == '\0'))
res = GST_RTSP_EPARSE;
if (parse_string (versionstr, sizeof (versionstr), &bptr) != GST_RTSP_OK)
res = GST_RTSP_EPARSE;
if (G_UNLIKELY (*bptr != '\0'))
res = GST_RTSP_EPARSE;
if (G_UNLIKELY (gst_rtsp_message_init_request (msg, method,
urlstr) != GST_RTSP_OK))
res = GST_RTSP_EPARSE;
res2 = parse_protocol_version (versionstr, &msg->type,
&msg->type_data.request.version);
if (G_LIKELY (res == GST_RTSP_OK))
res = res2;
if (G_LIKELY (msg->type == GST_RTSP_MESSAGE_REQUEST)) {
/* GET and POST are not allowed as RTSP methods */
if (msg->type_data.request.method == GST_RTSP_GET ||
msg->type_data.request.method == GST_RTSP_POST) {
msg->type_data.request.method = GST_RTSP_INVALID;
if (res == GST_RTSP_OK)
res = GST_RTSP_ERROR;
}
} else if (msg->type == GST_RTSP_MESSAGE_HTTP_REQUEST) {
/* only GET and POST are allowed as HTTP methods */
if (msg->type_data.request.method != GST_RTSP_GET &&
msg->type_data.request.method != GST_RTSP_POST) {
msg->type_data.request.method = GST_RTSP_INVALID;
if (res == GST_RTSP_OK)
res = GST_RTSP_ERROR;
}
}
return res;
}
/* parsing lines means reading a Key: Value pair */
static GstRTSPResult
parse_line (guint8 * buffer, GstRTSPMessage * msg)
{
GstRTSPHeaderField field;
gchar *line = (gchar *) buffer;
gchar *field_name = NULL;
gchar *value;
if ((value = strchr (line, ':')) == NULL || value == line)
goto parse_error;
/* trim space before the colon */
if (value[-1] == ' ')
value[-1] = '\0';
/* replace the colon with a NUL */
*value++ = '\0';
/* find the header */
field = gst_rtsp_find_header_field (line);
/* custom header not present in the list of pre-defined headers */
if (field == GST_RTSP_HDR_INVALID)
field_name = line;
/* split up the value in multiple key:value pairs if it contains comma(s) */
while (*value != '\0') {
gchar *next_value;
gchar *comma = NULL;
gboolean quoted = FALSE;
guint comment = 0;
/* trim leading space */
if (*value == ' ')
value++;
/* for headers which may not appear multiple times, and thus may not
* contain multiple values on the same line, we can short-circuit the loop
* below and the entire value results in just one key:value pair*/
if (!gst_rtsp_header_allow_multiple (field))
next_value = value + strlen (value);
else
next_value = value;
/* find the next value, taking special care of quotes and comments */
while (*next_value != '\0') {
if ((quoted || comment != 0) && *next_value == '\\' &&
next_value[1] != '\0')
next_value++;
else if (comment == 0 && *next_value == '"')
quoted = !quoted;
else if (!quoted && *next_value == '(')
comment++;
else if (comment != 0 && *next_value == ')')
comment--;
else if (!quoted && comment == 0) {
/* To quote RFC 2068: "User agents MUST take special care in parsing
* the WWW-Authenticate field value if it contains more than one
* challenge, or if more than one WWW-Authenticate header field is
* provided, since the contents of a challenge may itself contain a
* comma-separated list of authentication parameters."
*
* What this means is that we cannot just look for an unquoted comma
* when looking for multiple values in Proxy-Authenticate and
* WWW-Authenticate headers. Instead we need to look for the sequence
* "comma [space] token space token" before we can split after the
* comma...
*/
if (field == GST_RTSP_HDR_PROXY_AUTHENTICATE ||
field == GST_RTSP_HDR_WWW_AUTHENTICATE) {
if (*next_value == ',') {
if (next_value[1] == ' ') {
/* skip any space following the comma so we do not mistake it for
* separating between two tokens */
next_value++;
}
comma = next_value;
} else if (*next_value == ' ' && next_value[1] != ',' &&
next_value[1] != '=' && comma != NULL) {
next_value = comma;
comma = NULL;
break;
}
} else if (*next_value == ',')
break;
}
next_value++;
}
if (msg->type == GST_RTSP_MESSAGE_REQUEST && field == GST_RTSP_HDR_SESSION) {
/* The timeout parameter is only allowed in a session response header
* but some clients send it as part of the session request header.
* Ignore everything from the semicolon to the end of the line. */
next_value = value;
while (*next_value != '\0') {
if (*next_value == ';') {
break;
}
next_value++;
}
}
/* trim space */
if (value != next_value && next_value[-1] == ' ')
next_value[-1] = '\0';
if (*next_value != '\0')
*next_value++ = '\0';
/* add the key:value pair */
if (*value != '\0') {
if (field != GST_RTSP_HDR_INVALID)
gst_rtsp_message_add_header (msg, field, value);
else
gst_rtsp_message_add_header_by_name (msg, field_name, value);
}
value = next_value;
}
return GST_RTSP_OK;
/* ERRORS */
parse_error:
{
return GST_RTSP_EPARSE;
}
}
/* convert all consecutive whitespace to a single space */
static void
normalize_line (guint8 * buffer)
{
while (*buffer) {
if (g_ascii_isspace (*buffer)) {
guint8 *tmp;
*buffer++ = ' ';
for (tmp = buffer; g_ascii_isspace (*tmp); tmp++) {
}
if (buffer != tmp)
memmove (buffer, tmp, strlen ((gchar *) tmp) + 1);
} else {
buffer++;
}
}
}
static gboolean
cseq_validation (GstRTSPConnection * conn, GstRTSPMessage * message)
{
gchar *cseq_header;
gint64 cseq = 0;
GstRTSPResult res;
if (message->type == GST_RTSP_MESSAGE_RESPONSE ||
message->type == GST_RTSP_MESSAGE_REQUEST) {
if ((res = gst_rtsp_message_get_header (message, GST_RTSP_HDR_CSEQ,
&cseq_header, 0)) != GST_RTSP_OK) {
/* rfc2326 This field MUST be present in all RTSP req and resp */
goto invalid_format;
}
errno = 0;
cseq = g_ascii_strtoll (cseq_header, NULL, 10);
if (errno != 0 || cseq < 0) {
/* CSeq has no valid value */
goto invalid_format;
}
if (message->type == GST_RTSP_MESSAGE_RESPONSE &&
(conn->cseq == 0 || conn->cseq < cseq)) {
/* Response CSeq can't be higher than the number of outgoing requests
* neither is a response valid if no request has been made */
goto invalid_format;
}
}
return GST_RTSP_OK;
invalid_format:
{
return GST_RTSP_EPARSE;
}
}
/* returns:
* GST_RTSP_OK when a complete message was read.
* GST_RTSP_EEOF: when the read socket is closed
* GST_RTSP_EINTR: when more data is needed.
* GST_RTSP_..: some other error occurred.
*/
static GstRTSPResult
build_next (GstRTSPBuilder * builder, GstRTSPMessage * message,
GstRTSPConnection * conn, gboolean block)
{
GstRTSPResult res;
while (TRUE) {
switch (builder->state) {
case STATE_START:
{
guint8 c;
builder->offset = 0;
res =
read_bytes (conn, (guint8 *) builder->buffer, &builder->offset, 1,
block);
if (res != GST_RTSP_OK)
goto done;
c = builder->buffer[0];
/* we have 1 bytes now and we can see if this is a data message or
* not */
if (c == '$') {
/* data message, prepare for the header */
builder->state = STATE_DATA_HEADER;
conn->may_cancel = FALSE;
} else if (c == '\n' || c == '\r') {
/* skip \n and \r */
builder->offset = 0;
} else {
builder->line = 0;
builder->state = STATE_READ_LINES;
conn->may_cancel = FALSE;
}
break;
}
case STATE_DATA_HEADER:
{
res =
read_bytes (conn, (guint8 *) builder->buffer, &builder->offset, 4,
block);
if (res != GST_RTSP_OK)
goto done;
gst_rtsp_message_init_data (message, builder->buffer[1]);
builder->body_len = (builder->buffer[2] << 8) | builder->buffer[3];
builder->body_data = g_malloc (builder->body_len + 1);
builder->body_data[builder->body_len] = '\0';
builder->offset = 0;
builder->state = STATE_DATA_BODY;
break;
}
case STATE_DATA_BODY:
{
res =
read_bytes (conn, builder->body_data, &builder->offset,
builder->body_len, block);
if (res != GST_RTSP_OK)
goto done;
/* we have the complete body now, store in the message adjusting the
* length to include the trailing '\0' */
gst_rtsp_message_take_body (message,
(guint8 *) builder->body_data, builder->body_len + 1);
builder->body_data = NULL;
builder->body_len = 0;
builder->state = STATE_END;
break;
}
case STATE_READ_LINES:
{
res = read_line (conn, builder->buffer, &builder->offset,
sizeof (builder->buffer), block);
if (res != GST_RTSP_OK)
goto done;
/* we have a regular response */
if (builder->buffer[0] == '\0') {
gchar *hdrval;
gint64 content_length_parsed = 0;
/* empty line, end of message header */
/* see if there is a Content-Length header, but ignore it if this
* is a POST request with an x-sessioncookie header */
if (gst_rtsp_message_get_header (message,
GST_RTSP_HDR_CONTENT_LENGTH, &hdrval, 0) == GST_RTSP_OK &&
(message->type != GST_RTSP_MESSAGE_HTTP_REQUEST ||
message->type_data.request.method != GST_RTSP_POST ||
gst_rtsp_message_get_header (message,
GST_RTSP_HDR_X_SESSIONCOOKIE, NULL, 0) != GST_RTSP_OK)) {
/* there is, prepare to read the body */
errno = 0;
content_length_parsed = g_ascii_strtoll (hdrval, NULL, 10);
if (errno != 0 || content_length_parsed < 0) {
res = GST_RTSP_EPARSE;
goto invalid_body_len;
} else if (content_length_parsed > conn->content_length_limit) {
res = GST_RTSP_ENOMEM;
goto invalid_body_len;
}
builder->body_len = content_length_parsed;
builder->body_data = g_try_malloc (builder->body_len + 1);
/* we can't do much here, we need the length to know how many bytes
* we need to read next and when allocation fails, we can't read the payload. */
if (builder->body_data == NULL) {
res = GST_RTSP_ENOMEM;
goto invalid_body_len;
}
builder->body_data[builder->body_len] = '\0';
builder->offset = 0;
builder->state = STATE_DATA_BODY;
} else {
builder->state = STATE_END;
}
break;
}
/* we have a line */
normalize_line (builder->buffer);
if (builder->line == 0) {
/* first line, check for response status */
if (memcmp (builder->buffer, "RTSP", 4) == 0 ||
memcmp (builder->buffer, "HTTP", 4) == 0) {
builder->status = parse_response_status (builder->buffer, message);
} else {
builder->status = parse_request_line (builder->buffer, message);
}
} else {
/* else just parse the line */
res = parse_line (builder->buffer, message);
if (res != GST_RTSP_OK)
builder->status = res;
}
if (builder->status != GST_RTSP_OK) {
res = builder->status;
goto invalid_format;
}
builder->line++;
builder->offset = 0;
break;
}
case STATE_END:
{
gchar *session_cookie;
gchar *session_id;
conn->may_cancel = TRUE;
if ((res = cseq_validation (conn, message)) != GST_RTSP_OK) {
/* message don't comply with rfc2326 regarding CSeq */
goto invalid_format;
}
if (message->type == GST_RTSP_MESSAGE_DATA) {
/* data messages don't have headers */
res = GST_RTSP_OK;
goto done;
}
/* save the tunnel session in the connection */
if (message->type == GST_RTSP_MESSAGE_HTTP_REQUEST &&
!conn->manual_http &&
conn->tstate == TUNNEL_STATE_NONE &&
gst_rtsp_message_get_header (message, GST_RTSP_HDR_X_SESSIONCOOKIE,
&session_cookie, 0) == GST_RTSP_OK) {
strncpy (conn->tunnelid, session_cookie, TUNNELID_LEN);
conn->tunnelid[TUNNELID_LEN - 1] = '\0';
conn->tunneled = TRUE;
}
/* save session id in the connection for further use */
if (message->type == GST_RTSP_MESSAGE_RESPONSE &&
gst_rtsp_message_get_header (message, GST_RTSP_HDR_SESSION,
&session_id, 0) == GST_RTSP_OK) {
gint maxlen, i;
maxlen = sizeof (conn->session_id) - 1;
/* the sessionid can have attributes marked with ;
* Make sure we strip them */
for (i = 0; i < maxlen && session_id[i] != '\0'; i++) {
if (session_id[i] == ';') {
maxlen = i;
/* parse timeout */
do {
i++;
} while (g_ascii_isspace (session_id[i]));
if (g_str_has_prefix (&session_id[i], "timeout=")) {
gint to;
/* if we parsed something valid, configure */
if ((to = atoi (&session_id[i + 8])) > 0)
conn->timeout = to;
}
break;
}
}
/* make sure to not overflow */
if (conn->remember_session_id) {
strncpy (conn->session_id, session_id, maxlen);
conn->session_id[maxlen] = '\0';
}
}
res = builder->status;
goto done;
}
default:
res = GST_RTSP_ERROR;
goto done;
}
}
done:
conn->may_cancel = TRUE;
return res;
/* ERRORS */
invalid_body_len:
{
conn->may_cancel = TRUE;
GST_DEBUG ("could not allocate body");
return res;
}
invalid_format:
{
conn->may_cancel = TRUE;
GST_DEBUG ("could not parse");
return res;
}
}
/**
* gst_rtsp_connection_read_usec:
* @conn: a #GstRTSPConnection
* @data: the data to read
* @size: the size of @data
* @timeout: a timeout value in microseconds
*
* Attempt to read @size bytes into @data from the connected @conn, blocking up to
* the specified @timeout. @timeout can be 0, in which case this function
* might block forever.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK on success.
*
* Since: 1.18
*/
GstRTSPResult
gst_rtsp_connection_read_usec (GstRTSPConnection * conn, guint8 * data,
guint size, gint64 timeout)
{
guint offset;
GstRTSPResult res;
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (data != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (conn->read_socket != NULL, GST_RTSP_EINVAL);
if (G_UNLIKELY (size == 0))
return GST_RTSP_OK;
offset = 0;
/* configure timeout if any */
set_read_socket_timeout (conn, timeout);
res = read_bytes (conn, data, &offset, size, TRUE);
clear_read_socket_timeout (conn);
return res;
}
static GstRTSPMessage *
gen_tunnel_reply (GstRTSPConnection * conn, GstRTSPStatusCode code,
const GstRTSPMessage * request)
{
GstRTSPMessage *msg;
GstRTSPResult res;
if (gst_rtsp_status_as_text (code) == NULL)
code = GST_RTSP_STS_INTERNAL_SERVER_ERROR;
GST_RTSP_CHECK (gst_rtsp_message_new_response (&msg, code, NULL, request),
no_message);
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_SERVER,
"GStreamer RTSP Server");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CONNECTION, "close");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CACHE_CONTROL, "no-store");
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_PRAGMA, "no-cache");
if (code == GST_RTSP_STS_OK) {
/* add the local ip address to the tunnel reply, this is where the client
* should send the POST request to */
if (conn->local_ip)
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_X_SERVER_IP_ADDRESS,
conn->local_ip);
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CONTENT_TYPE,
"application/x-rtsp-tunnelled");
}
return msg;
/* ERRORS */
no_message:
{
return NULL;
}
}
/**
* gst_rtsp_connection_receive_usec:
* @conn: a #GstRTSPConnection
* @message: the message to read
* @timeout: a timeout value or 0
*
* Attempt to read into @message from the connected @conn, blocking up to
* the specified @timeout. @timeout can be 0, in which case this function
* might block forever.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK on success.
*
* Since: 1.18
*/
GstRTSPResult
gst_rtsp_connection_receive_usec (GstRTSPConnection * conn,
GstRTSPMessage * message, gint64 timeout)
{
GstRTSPResult res;
GstRTSPBuilder builder;
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (conn->read_socket != NULL, GST_RTSP_EINVAL);
/* configure timeout if any */
set_read_socket_timeout (conn, timeout);
memset (&builder, 0, sizeof (GstRTSPBuilder));
res = build_next (&builder, message, conn, TRUE);
clear_read_socket_timeout (conn);
if (G_UNLIKELY (res != GST_RTSP_OK))
goto read_error;
if (!conn->manual_http) {
if (message->type == GST_RTSP_MESSAGE_HTTP_REQUEST) {
if (conn->tstate == TUNNEL_STATE_NONE &&
message->type_data.request.method == GST_RTSP_GET) {
GstRTSPMessage *response;
conn->tstate = TUNNEL_STATE_GET;
/* tunnel GET request, we can reply now */
response = gen_tunnel_reply (conn, GST_RTSP_STS_OK, message);
res = gst_rtsp_connection_send_usec (conn, response, timeout);
gst_rtsp_message_free (response);
if (res == GST_RTSP_OK)
res = GST_RTSP_ETGET;
goto cleanup;
} else if (conn->tstate == TUNNEL_STATE_NONE &&
message->type_data.request.method == GST_RTSP_POST) {
conn->tstate = TUNNEL_STATE_POST;
/* tunnel POST request, the caller now has to link the two
* connections. */
res = GST_RTSP_ETPOST;
goto cleanup;
} else {
res = GST_RTSP_EPARSE;
goto cleanup;
}
} else if (message->type == GST_RTSP_MESSAGE_HTTP_RESPONSE) {
res = GST_RTSP_EPARSE;
goto cleanup;
}
}
/* we have a message here */
build_reset (&builder);
return GST_RTSP_OK;
/* ERRORS */
read_error:
cleanup:
{
build_reset (&builder);
gst_rtsp_message_unset (message);
return res;
}
}
/**
* gst_rtsp_connection_close:
* @conn: a #GstRTSPConnection
*
* Close the connected @conn. After this call, the connection is in the same
* state as when it was first created.
*
* Returns: #GST_RTSP_OK on success.
*/
GstRTSPResult
gst_rtsp_connection_close (GstRTSPConnection * conn)
{
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
/* last unref closes the connection we don't want to explicitly close here
* because these sockets might have been provided at construction */
if (conn->stream0) {
g_object_unref (conn->stream0);
conn->stream0 = NULL;
conn->socket0 = NULL;
}
if (conn->stream1) {
g_object_unref (conn->stream1);
conn->stream1 = NULL;
conn->socket1 = NULL;
}
/* these were owned by the stream */
conn->input_stream = NULL;
conn->output_stream = NULL;
conn->control_stream = NULL;
g_free (conn->remote_ip);
conn->remote_ip = NULL;
g_free (conn->local_ip);
conn->local_ip = NULL;
conn->read_ahead = 0;
g_free (conn->initial_buffer);
conn->initial_buffer = NULL;
conn->initial_buffer_offset = 0;
conn->write_socket = NULL;
conn->read_socket = NULL;
conn->write_socket_used = FALSE;
conn->read_socket_used = FALSE;
conn->tunneled = FALSE;
conn->tstate = TUNNEL_STATE_NONE;
conn->ctxp = NULL;
g_free (conn->username);
conn->username = NULL;
g_free (conn->passwd);
conn->passwd = NULL;
gst_rtsp_connection_clear_auth_params (conn);
conn->timeout = 60;
conn->cseq = 0;
conn->session_id[0] = '\0';
return GST_RTSP_OK;
}
/**
* gst_rtsp_connection_free:
* @conn: a #GstRTSPConnection
*
* Close and free @conn.
*
* Returns: #GST_RTSP_OK on success.
*/
GstRTSPResult
gst_rtsp_connection_free (GstRTSPConnection * conn)
{
GstRTSPResult res;
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
res = gst_rtsp_connection_close (conn);
if (conn->cancellable)
g_object_unref (conn->cancellable);
if (conn->client)
g_object_unref (conn->client);
if (conn->tls_database)
g_object_unref (conn->tls_database);
if (conn->tls_interaction)
g_object_unref (conn->tls_interaction);
if (conn->accept_certificate_destroy_notify)
conn->
accept_certificate_destroy_notify (conn->accept_certificate_user_data);
g_timer_destroy (conn->timer);
gst_rtsp_url_free (conn->url);
g_free (conn->proxy_host);
g_free (conn);
return res;
}
/**
* gst_rtsp_connection_poll_usec:
* @conn: a #GstRTSPConnection
* @events: a bitmask of #GstRTSPEvent flags to check
* @revents: location for result flags
* @timeout: a timeout in microseconds
*
* Wait up to the specified @timeout for the connection to become available for
* at least one of the operations specified in @events. When the function returns
* with #GST_RTSP_OK, @revents will contain a bitmask of available operations on
* @conn.
*
* @timeout can be 0, in which case this function might block forever.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK on success.
*
* Since: 1.18
*/
GstRTSPResult
gst_rtsp_connection_poll_usec (GstRTSPConnection * conn, GstRTSPEvent events,
GstRTSPEvent * revents, gint64 timeout)
{
GMainContext *ctx;
GSource *rs, *ws, *ts;
GIOCondition condition;
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);
}
if (events & GST_RTSP_EV_READ) {
rs = g_socket_create_source (conn->read_socket, G_IO_IN | G_IO_PRI,
conn->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,
conn->cancellable);
g_source_set_dummy_callback (ws);
g_source_attach (ws, ctx);
g_source_unref (ws);
}
/* 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) {
g_cancellable_cancel (conn->cancellable);
} else {
g_object_unref (conn->cancellable);
conn->cancellable = g_cancellable_new ();
}
return GST_RTSP_OK;
}
/**
* gst_rtsp_connection_set_proxy:
* @conn: a #GstRTSPConnection
* @host: the proxy host
* @port: the proxy port
*
* Set the proxy host and port.
*
* Returns: #GST_RTSP_OK.
*/
GstRTSPResult
gst_rtsp_connection_set_proxy (GstRTSPConnection * conn,
const gchar * host, guint port)
{
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
g_free (conn->proxy_host);
conn->proxy_host = g_strdup (host);
conn->proxy_port = port;
return GST_RTSP_OK;
}
/**
* gst_rtsp_connection_set_auth:
* @conn: a #GstRTSPConnection
* @method: authentication method
* @user: the user
* @pass: the password
*
* Configure @conn for authentication mode @method with @user and @pass as the
* user and password respectively.
*
* Returns: #GST_RTSP_OK.
*/
GstRTSPResult
gst_rtsp_connection_set_auth (GstRTSPConnection * conn,
GstRTSPAuthMethod method, const gchar * user, const gchar * pass)
{
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
if (method == GST_RTSP_AUTH_DIGEST && ((user == NULL || pass == NULL)
|| g_strrstr (user, ":") != NULL))
return GST_RTSP_EINVAL;
/* Make sure the username and passwd are being set for authentication */
if (method == GST_RTSP_AUTH_NONE && (user == NULL || pass == NULL))
return GST_RTSP_EINVAL;
/* ":" chars are not allowed in usernames for basic auth */
if (method == GST_RTSP_AUTH_BASIC && g_strrstr (user, ":") != NULL)
return GST_RTSP_EINVAL;
g_free (conn->username);
g_free (conn->passwd);
conn->auth_method = method;
conn->username = g_strdup (user);
conn->passwd = g_strdup (pass);
return GST_RTSP_OK;
}
/**
* str_case_hash:
* @key: ASCII string to hash
*
* Hashes @key in a case-insensitive manner.
*
* Returns: the hash code.
**/
static guint
str_case_hash (gconstpointer key)
{
const char *p = key;
guint h = g_ascii_toupper (*p);
if (h)
for (p += 1; *p != '\0'; p++)
h = (h << 5) - h + g_ascii_toupper (*p);
return h;
}
/**
* str_case_equal:
* @v1: an ASCII string
* @v2: another ASCII string
*
* Compares @v1 and @v2 in a case-insensitive manner
*
* Returns: %TRUE if they are equal (modulo case)
**/
static gboolean
str_case_equal (gconstpointer v1, gconstpointer v2)
{
const char *string1 = v1;
const char *string2 = v2;
return g_ascii_strcasecmp (string1, string2) == 0;
}
/**
* gst_rtsp_connection_set_auth_param:
* @conn: a #GstRTSPConnection
* @param: authentication directive
* @value: value
*
* Setup @conn with authentication directives. This is not necessary for
* methods #GST_RTSP_AUTH_NONE and #GST_RTSP_AUTH_BASIC. For
* #GST_RTSP_AUTH_DIGEST, directives should be taken from the digest challenge
* in the WWW-Authenticate response header and can include realm, domain,
* nonce, opaque, stale, algorithm, qop as per RFC2617.
*/
void
gst_rtsp_connection_set_auth_param (GstRTSPConnection * conn,
const gchar * param, const gchar * value)
{
g_return_if_fail (conn != NULL);
g_return_if_fail (param != NULL);
if (conn->auth_params == NULL) {
conn->auth_params =
g_hash_table_new_full (str_case_hash, str_case_equal, g_free, g_free);
}
g_hash_table_insert (conn->auth_params, g_strdup (param), g_strdup (value));
}
/**
* gst_rtsp_connection_clear_auth_params:
* @conn: a #GstRTSPConnection
*
* Clear the list of authentication directives stored in @conn.
*/
void
gst_rtsp_connection_clear_auth_params (GstRTSPConnection * conn)
{
g_return_if_fail (conn != NULL);
if (conn->auth_params != NULL) {
g_hash_table_destroy (conn->auth_params);
conn->auth_params = NULL;
}
}
static GstRTSPResult
set_qos_dscp (GSocket * socket, guint qos_dscp)
{
#ifndef IP_TOS
GST_FIXME ("IP_TOS socket option is not defined, not setting dscp");
return GST_RTSP_OK;
#else
gint fd;
union gst_sockaddr sa;
socklen_t slen = sizeof (sa);
gint af;
gint tos;
if (!socket)
return GST_RTSP_OK;
fd = g_socket_get_fd (socket);
if (getsockname (fd, &sa.sa, &slen) < 0)
goto no_getsockname;
af = sa.sa.sa_family;
/* if this is an IPv4-mapped address then do IPv4 QoS */
if (af == AF_INET6) {
if (IN6_IS_ADDR_V4MAPPED (&sa.sa_in6.sin6_addr))
af = AF_INET;
}
/* extract and shift 6 bits of the DSCP */
tos = (qos_dscp & 0x3f) << 2;
#ifdef G_OS_WIN32
# define SETSOCKOPT_ARG4_TYPE const char *
#else
# define SETSOCKOPT_ARG4_TYPE const void *
#endif
switch (af) {
case AF_INET:
if (setsockopt (fd, IPPROTO_IP, IP_TOS, (SETSOCKOPT_ARG4_TYPE) & tos,
sizeof (tos)) < 0)
goto no_setsockopt;
break;
case AF_INET6:
#ifdef IPV6_TCLASS
if (setsockopt (fd, IPPROTO_IPV6, IPV6_TCLASS,
(SETSOCKOPT_ARG4_TYPE) & tos, sizeof (tos)) < 0)
goto no_setsockopt;
break;
#endif
default:
goto wrong_family;
}
return GST_RTSP_OK;
/* ERRORS */
no_getsockname:
no_setsockopt:
{
return GST_RTSP_ESYS;
}
wrong_family:
{
return GST_RTSP_ERROR;
}
#endif
}
/**
* gst_rtsp_connection_set_qos_dscp:
* @conn: a #GstRTSPConnection
* @qos_dscp: DSCP value
*
* Configure @conn to use the specified DSCP value.
*
* Returns: #GST_RTSP_OK on success.
*/
GstRTSPResult
gst_rtsp_connection_set_qos_dscp (GstRTSPConnection * conn, guint qos_dscp)
{
GstRTSPResult res;
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (conn->read_socket != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (conn->write_socket != NULL, GST_RTSP_EINVAL);
res = set_qos_dscp (conn->socket0, qos_dscp);
if (res == GST_RTSP_OK)
res = set_qos_dscp (conn->socket1, qos_dscp);
return res;
}
/**
* gst_rtsp_connection_set_content_length_limit:
* @conn: a #GstRTSPConnection
* @limit: Content-Length limit
*
* Configure @conn to use the specified Content-Length limit.
* Both requests and responses are validated. If content-length is
* exceeded, ENOMEM error will be returned.
*
* Since: 1.18
*/
void
gst_rtsp_connection_set_content_length_limit (GstRTSPConnection * conn,
guint limit)
{
g_return_if_fail (conn != NULL);
conn->content_length_limit = limit;
}
/**
* gst_rtsp_connection_get_url:
* @conn: a #GstRTSPConnection
*
* Retrieve the URL of the other end of @conn.
*
* Returns: The URL. This value remains valid until the
* connection is freed.
*/
GstRTSPUrl *
gst_rtsp_connection_get_url (const GstRTSPConnection * conn)
{
g_return_val_if_fail (conn != NULL, NULL);
return conn->url;
}
/**
* gst_rtsp_connection_get_ip:
* @conn: a #GstRTSPConnection
*
* Retrieve the IP address of the other end of @conn.
*
* Returns: The IP address as a string. this value remains valid until the
* connection is closed.
*/
const gchar *
gst_rtsp_connection_get_ip (const GstRTSPConnection * conn)
{
g_return_val_if_fail (conn != NULL, NULL);
return conn->remote_ip;
}
/**
* gst_rtsp_connection_set_ip:
* @conn: a #GstRTSPConnection
* @ip: an ip address
*
* Set the IP address of the server.
*/
void
gst_rtsp_connection_set_ip (GstRTSPConnection * conn, const gchar * ip)
{
g_return_if_fail (conn != NULL);
g_free (conn->remote_ip);
conn->remote_ip = g_strdup (ip);
}
/**
* gst_rtsp_connection_get_read_socket:
* @conn: a #GstRTSPConnection
*
* Get the file descriptor for reading.
*
* Returns: (transfer none): the file descriptor used for reading or %NULL on
* error. The file descriptor remains valid until the connection is closed.
*/
GSocket *
gst_rtsp_connection_get_read_socket (const GstRTSPConnection * conn)
{
g_return_val_if_fail (conn != NULL, NULL);
g_return_val_if_fail (conn->read_socket != NULL, NULL);
return conn->read_socket;
}
/**
* gst_rtsp_connection_get_write_socket:
* @conn: a #GstRTSPConnection
*
* Get the file descriptor for writing.
*
* Returns: (transfer none): the file descriptor used for writing or NULL on
* error. The file descriptor remains valid until the connection is closed.
*/
GSocket *
gst_rtsp_connection_get_write_socket (const GstRTSPConnection * conn)
{
g_return_val_if_fail (conn != NULL, NULL);
g_return_val_if_fail (conn->write_socket != NULL, NULL);
return conn->write_socket;
}
/**
* gst_rtsp_connection_set_http_mode:
* @conn: a #GstRTSPConnection
* @enable: %TRUE to enable manual HTTP mode
*
* By setting the HTTP mode to %TRUE the message parsing will support HTTP
* messages in addition to the RTSP messages. It will also disable the
* automatic handling of setting up an HTTP tunnel.
*/
void
gst_rtsp_connection_set_http_mode (GstRTSPConnection * conn, gboolean enable)
{
g_return_if_fail (conn != NULL);
conn->manual_http = enable;
}
/**
* gst_rtsp_connection_set_tunneled:
* @conn: a #GstRTSPConnection
* @tunneled: the new state
*
* Set the HTTP tunneling state of the connection. This must be configured before
* the @conn is connected.
*/
void
gst_rtsp_connection_set_tunneled (GstRTSPConnection * conn, gboolean tunneled)
{
g_return_if_fail (conn != NULL);
g_return_if_fail (conn->read_socket == NULL);
g_return_if_fail (conn->write_socket == NULL);
conn->tunneled = tunneled;
}
/**
* gst_rtsp_connection_is_tunneled:
* @conn: a #GstRTSPConnection
*
* Get the tunneling state of the connection.
*
* Returns: if @conn is using HTTP tunneling.
*/
gboolean
gst_rtsp_connection_is_tunneled (const GstRTSPConnection * conn)
{
g_return_val_if_fail (conn != NULL, FALSE);
return conn->tunneled;
}
/**
* gst_rtsp_connection_get_tunnelid:
* @conn: a #GstRTSPConnection
*
* Get the tunnel session id the connection.
*
* Returns: returns a non-empty string if @conn is being tunneled over HTTP.
*/
const gchar *
gst_rtsp_connection_get_tunnelid (const GstRTSPConnection * conn)
{
g_return_val_if_fail (conn != NULL, NULL);
if (!conn->tunneled)
return NULL;
return conn->tunnelid;
}
/**
* gst_rtsp_connection_set_ignore_x_server_reply:
* @conn: a #GstRTSPConnection
* @ignore: %TRUE to ignore the x-server-ip-address header reply or %FALSE to
* comply with it (%FALSE is the default).
*
* Set whether to ignore the x-server-ip-address header reply or not. If the
* header is ignored, the original address will be used instead.
*
* Since: 1.20
*/
void
gst_rtsp_connection_set_ignore_x_server_reply (GstRTSPConnection * conn,
gboolean ignore)
{
g_return_if_fail (conn != NULL);
conn->ignore_x_server_reply = ignore;
}
/**
* gst_rtsp_connection_get_ignore_x_server_reply:
* @conn: a #GstRTSPConnection
*
* Get the ignore_x_server_reply value.
*
* Returns: returns %TRUE if the x-server-ip-address header reply will be
* ignored, else returns %FALSE
*
* Since: 1.20
*/
gboolean
gst_rtsp_connection_get_ignore_x_server_reply (const GstRTSPConnection * conn)
{
g_return_val_if_fail (conn != NULL, FALSE);
return conn->ignore_x_server_reply;
}
/**
* gst_rtsp_connection_do_tunnel:
* @conn: a #GstRTSPConnection
* @conn2: a #GstRTSPConnection or %NULL
*
* If @conn received the first tunnel connection and @conn2 received
* the second tunnel connection, link the two connections together so that
* @conn manages the tunneled connection.
*
* After this call, @conn2 cannot be used anymore and must be freed with
* gst_rtsp_connection_free().
*
* If @conn2 is %NULL then only the base64 decoding context will be setup for
* @conn.
*
* Returns: return GST_RTSP_OK on success.
*/
GstRTSPResult
gst_rtsp_connection_do_tunnel (GstRTSPConnection * conn,
GstRTSPConnection * conn2)
{
g_return_val_if_fail (conn != NULL, GST_RTSP_EINVAL);
if (conn2 != NULL) {
GstRTSPTunnelState ts1 = conn->tstate;
GstRTSPTunnelState ts2 = conn2->tstate;
g_return_val_if_fail ((ts1 == TUNNEL_STATE_GET && ts2 == TUNNEL_STATE_POST)
|| (ts1 == TUNNEL_STATE_POST && ts2 == TUNNEL_STATE_GET),
GST_RTSP_EINVAL);
g_return_val_if_fail (!memcmp (conn2->tunnelid, conn->tunnelid,
TUNNELID_LEN), GST_RTSP_EINVAL);
/* both connections have socket0 as the read/write socket */
if (ts1 == TUNNEL_STATE_GET) {
/* conn2 is the HTTP POST channel. take its socket and set it as read
* socket in conn */
conn->socket1 = conn2->socket0;
conn->stream1 = conn2->stream0;
conn->input_stream = conn2->input_stream;
conn->control_stream = g_io_stream_get_input_stream (conn->stream0);
conn2->output_stream = NULL;
} else {
/* conn2 is the HTTP GET channel. take its socket and set it as write
* socket in conn */
conn->socket1 = conn->socket0;
conn->stream1 = conn->stream0;
conn->socket0 = conn2->socket0;
conn->stream0 = conn2->stream0;
conn->output_stream = conn2->output_stream;
conn->control_stream = g_io_stream_get_input_stream (conn->stream0);
}
/* clean up some of the state of conn2 */
g_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;
/* 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;
/* 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++;
}
}
}
res =
writev_bytes (watch->conn->output_stream, vectors, n_vectors,
&bytes_written, FALSE, watch->conn->cancellable);
g_assert (bytes_written == bytes_to_write || res != GST_RTSP_OK);
/* First unmap all memories here, this simplifies the code below
* as we don't have to skip all memories that were already written
* before */
for (i = 0; i < n_mmap; i++) {
gst_memory_unmap (map_infos[i].memory, &map_infos[i]);
}
if (bytes_written == bytes_to_write) {
/* fast path, just unmap all memories, free memory, drop all messages and notify them */
l = 0;
while ((msg = gst_queue_array_pop_head_struct (watch->messages))) {
if (msg->id) {
ids[l] = msg->id;
l++;
}
gst_rtsp_serialized_message_clear (msg);
}
g_assert (watch->messages_bytes >= bytes_written);
watch->messages_bytes -= bytes_written;
} else if (bytes_written > 0) {
/* not done, let's skip all messages that were sent already and free them */
for (i = 0, drop_messages = 0; i < n_messages; i++) {
msg = gst_queue_array_peek_nth_struct (watch->messages, i);
if (bytes_written >= msg->data_size - msg->data_offset) {
guint body_size;
/* all data of this message is sent, check body and otherwise
* skip the whole message for next time */
bytes_written -= (msg->data_size - msg->data_offset);
watch->messages_bytes -= (msg->data_size - msg->data_offset);
msg->data_offset = msg->data_size;
if (msg->body_data) {
body_size = msg->body_data_size;
} else if (msg->body_buffer) {
body_size = gst_buffer_get_size (msg->body_buffer);
} else {
body_size = 0;
}
if (bytes_written + msg->body_offset >= body_size) {
/* body written, drop this message */
bytes_written -= body_size - msg->body_offset;
watch->messages_bytes -= body_size - msg->body_offset;
msg->body_offset = body_size;
drop_messages++;
if (msg->id) {
ids[l] = msg->id;
l++;
}
gst_rtsp_serialized_message_clear (msg);
} else {
msg->body_offset += bytes_written;
watch->messages_bytes -= bytes_written;
bytes_written = 0;
}
} else {
/* Need to continue sending from the data of this message */
msg->data_offset += bytes_written;
watch->messages_bytes -= bytes_written;
bytes_written = 0;
}
}
while (drop_messages > 0) {
msg = gst_queue_array_pop_head_struct (watch->messages);
g_assert (msg);
drop_messages--;
}
g_assert (watch->messages_bytes >= bytes_written);
watch->messages_bytes -= bytes_written;
}
if (!IS_BACKLOG_FULL (watch))
g_cond_signal (&watch->queue_not_full);
g_mutex_unlock (&watch->mutex);
/* notify all messages that were successfully written */
if (ids) {
while (*ids) {
/* only decrease the counter for messages that have an id. Only
* the last message of a messages chunk is counted */
watch->messages_count--;
if (watch->funcs.message_sent)
watch->funcs.message_sent (watch, *ids, watch->user_data);
ids++;
}
}
if (res == GST_RTSP_EINTR) {
goto write_blocked;
} else if (G_UNLIKELY (res != GST_RTSP_OK)) {
goto write_error;
}
g_mutex_lock (&watch->mutex);
} while (TRUE);
g_mutex_unlock (&watch->mutex);
write_blocked:
return TRUE;
/* ERRORS */
eof:
{
return FALSE;
}
write_error:
{
if (watch->funcs.error_full) {
guint i, n_messages;
n_messages = gst_queue_array_get_length (watch->messages);
for (i = 0; i < n_messages; i++) {
GstRTSPSerializedMessage *msg =
gst_queue_array_peek_nth_struct (watch->messages, i);
if (msg->id)
watch->funcs.error_full (watch, res, NULL, msg->id, watch->user_data);
}
} else if (watch->funcs.error) {
watch->funcs.error (watch, res, watch->user_data);
}
return FALSE;
}
}
static void
gst_rtsp_source_finalize (GSource * source)
{
GstRTSPWatch *watch = (GstRTSPWatch *) source;
GstRTSPSerializedMessage *msg;
if (watch->notify)
watch->notify (watch->user_data);
build_reset (&watch->builder);
gst_rtsp_message_unset (&watch->message);
while ((msg = gst_queue_array_pop_head_struct (watch->messages))) {
gst_rtsp_serialized_message_clear (msg);
}
gst_queue_array_free (watch->messages);
watch->messages = NULL;
watch->messages_bytes = 0;
watch->messages_count = 0;
g_cond_clear (&watch->queue_not_full);
if (watch->readsrc)
g_source_unref (watch->readsrc);
if (watch->writesrc)
g_source_unref (watch->writesrc);
if (watch->controlsrc)
g_source_unref (watch->controlsrc);
g_mutex_clear (&watch->mutex);
}
static GSourceFuncs gst_rtsp_source_funcs = {
gst_rtsp_source_prepare,
gst_rtsp_source_check,
gst_rtsp_source_dispatch,
gst_rtsp_source_finalize,
NULL,
NULL
};
/**
* gst_rtsp_watch_new: (skip)
* @conn: a #GstRTSPConnection
* @funcs: watch functions
* @user_data: user data to pass to @funcs
* @notify: notify when @user_data is not referenced anymore
*
* Create a watch object for @conn. The functions provided in @funcs will be
* called with @user_data when activity happened on the watch.
*
* The new watch is usually created so that it can be attached to a
* maincontext with gst_rtsp_watch_attach().
*
* @conn must exist for the entire lifetime of the watch.
*
* Returns: a #GstRTSPWatch that can be used for asynchronous RTSP
* communication. Free with gst_rtsp_watch_unref () after usage.
*/
GstRTSPWatch *
gst_rtsp_watch_new (GstRTSPConnection * conn,
GstRTSPWatchFuncs * funcs, gpointer user_data, GDestroyNotify notify)
{
GstRTSPWatch *result;
g_return_val_if_fail (conn != NULL, NULL);
g_return_val_if_fail (funcs != NULL, NULL);
g_return_val_if_fail (conn->read_socket != NULL, NULL);
g_return_val_if_fail (conn->write_socket != NULL, NULL);
result = (GstRTSPWatch *) g_source_new (&gst_rtsp_source_funcs,
sizeof (GstRTSPWatch));
result->conn = conn;
result->builder.state = STATE_START;
g_mutex_init (&result->mutex);
result->messages =
gst_queue_array_new_for_struct (sizeof (GstRTSPSerializedMessage), 10);
g_cond_init (&result->queue_not_full);
gst_rtsp_watch_reset (result);
result->keep_running = TRUE;
result->flushing = FALSE;
result->funcs = *funcs;
result->user_data = user_data;
result->notify = notify;
return result;
}
/**
* gst_rtsp_watch_reset:
* @watch: a #GstRTSPWatch
*
* Reset @watch, this is usually called after gst_rtsp_connection_do_tunnel()
* when the file descriptors of the connection might have changed.
*/
void
gst_rtsp_watch_reset (GstRTSPWatch * watch)
{
g_mutex_lock (&watch->mutex);
if (watch->readsrc) {
g_source_remove_child_source ((GSource *) watch, watch->readsrc);
g_source_unref (watch->readsrc);
}
if (watch->writesrc) {
g_source_remove_child_source ((GSource *) watch, watch->writesrc);
g_source_unref (watch->writesrc);
watch->writesrc = NULL;
}
if (watch->controlsrc) {
g_source_remove_child_source ((GSource *) watch, watch->controlsrc);
g_source_unref (watch->controlsrc);
watch->controlsrc = NULL;
}
if (watch->conn->input_stream) {
watch->readsrc =
g_pollable_input_stream_create_source (G_POLLABLE_INPUT_STREAM
(watch->conn->input_stream), NULL);
g_source_set_callback (watch->readsrc,
(GSourceFunc) gst_rtsp_source_dispatch_read, watch, NULL);
g_source_add_child_source ((GSource *) watch, watch->readsrc);
} else {
watch->readsrc = NULL;
}
/* we create and add the write source when we actually have something to
* write */
/* when write source is not added we add read source on the write socket
* instead to be able to detect when client closes get channel in tunneled
* mode */
if (watch->conn->control_stream) {
watch->controlsrc =
g_pollable_input_stream_create_source (G_POLLABLE_INPUT_STREAM
(watch->conn->control_stream), NULL);
g_source_set_callback (watch->controlsrc,
(GSourceFunc) gst_rtsp_source_dispatch_read_get_channel, watch, NULL);
g_source_add_child_source ((GSource *) watch, watch->controlsrc);
} else {
watch->controlsrc = NULL;
}
g_mutex_unlock (&watch->mutex);
}
/**
* gst_rtsp_watch_attach:
* @watch: a #GstRTSPWatch
* @context: a GMainContext (if NULL, the default context will be used)
*
* Adds a #GstRTSPWatch to a context so that it will be executed within that context.
*
* Returns: the ID (greater than 0) for the watch within the GMainContext.
*/
guint
gst_rtsp_watch_attach (GstRTSPWatch * watch, GMainContext * context)
{
g_return_val_if_fail (watch != NULL, 0);
return g_source_attach ((GSource *) watch, context);
}
/**
* gst_rtsp_watch_unref:
* @watch: a #GstRTSPWatch
*
* Decreases the reference count of @watch by one. If the resulting reference
* count is zero the watch and associated memory will be destroyed.
*/
void
gst_rtsp_watch_unref (GstRTSPWatch * watch)
{
g_return_if_fail (watch != NULL);
g_source_unref ((GSource *) watch);
}
/**
* gst_rtsp_watch_set_send_backlog:
* @watch: a #GstRTSPWatch
* @bytes: maximum bytes
* @messages: maximum messages
*
* Set the maximum amount of bytes and messages that will be queued in @watch.
* When the maximum amounts are exceeded, gst_rtsp_watch_write_data() and
* gst_rtsp_watch_send_message() will return #GST_RTSP_ENOMEM.
*
* A value of 0 for @bytes or @messages means no limits.
*
* Since: 1.2
*/
void
gst_rtsp_watch_set_send_backlog (GstRTSPWatch * watch,
gsize bytes, guint messages)
{
g_return_if_fail (watch != NULL);
g_mutex_lock (&watch->mutex);
watch->max_bytes = bytes;
watch->max_messages = messages;
if (!IS_BACKLOG_FULL (watch))
g_cond_signal (&watch->queue_not_full);
g_mutex_unlock (&watch->mutex);
GST_DEBUG ("set backlog to bytes %" G_GSIZE_FORMAT ", messages %u",
bytes, messages);
}
/**
* gst_rtsp_watch_get_send_backlog:
* @watch: a #GstRTSPWatch
* @bytes: (out) (allow-none): maximum bytes
* @messages: (out) (allow-none): maximum messages
*
* Get the maximum amount of bytes and messages that will be queued in @watch.
* See gst_rtsp_watch_set_send_backlog().
*
* Since: 1.2
*/
void
gst_rtsp_watch_get_send_backlog (GstRTSPWatch * watch,
gsize * bytes, guint * messages)
{
g_return_if_fail (watch != NULL);
g_mutex_lock (&watch->mutex);
if (bytes)
*bytes = watch->max_bytes;
if (messages)
*messages = watch->max_messages;
g_mutex_unlock (&watch->mutex);
}
static GstRTSPResult
gst_rtsp_watch_write_serialized_messages (GstRTSPWatch * watch,
GstRTSPSerializedMessage * messages, guint n_messages, guint * id)
{
GstRTSPResult res;
GMainContext *context = NULL;
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++;
}
}
}
res =
writev_bytes (watch->conn->output_stream, vectors, n_vectors,
&bytes_written, FALSE, watch->conn->cancellable);
g_assert (bytes_written == bytes_to_write || res != GST_RTSP_OK);
/* At this point we sent everything we could without blocking or
* error and updated the offsets inside the message accordingly */
/* First of all unmap all memories. This simplifies the code below */
for (k = 0; k < n_memories; k++) {
gst_memory_unmap (map_infos[k].memory, &map_infos[k]);
}
if (res != GST_RTSP_EINTR) {
/* actual error or done completely */
if (id != NULL)
*id = 0;
/* free everything */
for (i = 0, k = 0; i < n_messages; i++) {
gst_rtsp_serialized_message_clear (&messages[i]);
}
goto done;
}
/* not done, let's skip all messages that were sent already and free them */
for (i = 0, k = 0, drop_messages = 0; i < n_messages; i++) {
if (bytes_written >= messages[i].data_size) {
guint body_size;
/* all data of this message is sent, check body and otherwise
* skip the whole message for next time */
messages[i].data_offset = messages[i].data_size;
bytes_written -= messages[i].data_size;
if (messages[i].body_data) {
body_size = messages[i].body_data_size;
} else if (messages[i].body_buffer) {
body_size = gst_buffer_get_size (messages[i].body_buffer);
} else {
body_size = 0;
}
if (bytes_written >= body_size) {
/* body written, drop this message */
messages[i].body_offset = body_size;
bytes_written -= body_size;
drop_messages++;
gst_rtsp_serialized_message_clear (&messages[i]);
} else {
messages[i].body_offset = bytes_written;
bytes_written = 0;
}
} else {
/* Need to continue sending from the data of this message */
messages[i].data_offset = bytes_written;
bytes_written = 0;
}
}
g_assert (n_messages > drop_messages);
messages += drop_messages;
n_messages -= drop_messages;
}
/* check limits */
if (IS_BACKLOG_FULL (watch))
goto too_much_backlog;
for (i = 0; i < n_messages; i++) {
GstRTSPSerializedMessage local_message;
/* make a record with the data and id for sending async */
local_message = messages[i];
/* copy the body data or take an additional reference to the body buffer
* we don't own them here */
if (local_message.body_data) {
local_message.body_data =
g_memdup2 (local_message.body_data, local_message.body_data_size);
} else if (local_message.body_buffer) {
gst_buffer_ref (local_message.body_buffer);
}
local_message.borrowed = FALSE;
/* set an id for the very last message */
if (i == n_messages - 1) {
do {
/* make sure rec->id is never 0 */
local_message.id = ++watch->id;
} while (G_UNLIKELY (local_message.id == 0));
if (id != NULL)
*id = local_message.id;
} else {
local_message.id = 0;
}
/* add the record to a queue. */
gst_queue_array_push_tail_struct (watch->messages, &local_message);
watch->messages_bytes +=
(local_message.data_size - local_message.data_offset);
if (local_message.body_data)
watch->messages_bytes +=
(local_message.body_data_size - local_message.body_offset);
else if (local_message.body_buffer)
watch->messages_bytes +=
(gst_buffer_get_size (local_message.body_buffer) -
local_message.body_offset);
}
/* each message chunks is one unit */
watch->messages_count++;
/* make sure the main context will now also check for writability on the
* socket */
context = ((GSource *) watch)->context;
if (!watch->writesrc) {
/* remove the read source on the write socket, we will be able to detect
* errors while writing */
if (watch->controlsrc) {
g_source_remove_child_source ((GSource *) watch, watch->controlsrc);
g_source_unref (watch->controlsrc);
watch->controlsrc = NULL;
}
watch->writesrc =
g_pollable_output_stream_create_source (G_POLLABLE_OUTPUT_STREAM
(watch->conn->output_stream), NULL);
g_source_set_callback (watch->writesrc,
(GSourceFunc) gst_rtsp_source_dispatch_write, watch, NULL);
g_source_add_child_source ((GSource *) watch, watch->writesrc);
}
res = GST_RTSP_OK;
done:
g_mutex_unlock (&watch->mutex);
if (context)
g_main_context_wakeup (context);
return res;
/* ERRORS */
flushing:
{
GST_DEBUG ("we are flushing");
g_mutex_unlock (&watch->mutex);
for (i = 0; i < n_messages; i++) {
gst_rtsp_serialized_message_clear (&messages[i]);
}
return GST_RTSP_EINTR;
}
too_much_backlog:
{
GST_WARNING ("too much backlog: max_bytes %" G_GSIZE_FORMAT ", current %"
G_GSIZE_FORMAT ", max_messages %u, current %u", watch->max_bytes,
watch->messages_bytes, watch->max_messages, watch->messages_count);
g_mutex_unlock (&watch->mutex);
for (i = 0; i < n_messages; i++) {
gst_rtsp_serialized_message_clear (&messages[i]);
}
return GST_RTSP_ENOMEM;
}
return GST_RTSP_OK;
}
/**
* gst_rtsp_watch_write_data:
* @watch: a #GstRTSPWatch
* @data: (array length=size) (transfer full): the data to queue
* @size: the size of @data
* @id: (out) (allow-none): location for a message ID or %NULL
*
* Write @data using the connection of the @watch. If it cannot be sent
* immediately, it will be queued for transmission in @watch. The contents of
* @message will then be serialized and transmitted when the connection of the
* @watch becomes writable. In case the @message is queued, the ID returned in
* @id will be non-zero and used as the ID argument in the message_sent
* callback.
*
* This function will take ownership of @data and g_free() it after use.
*
* If the amount of queued data exceeds the limits set with
* gst_rtsp_watch_set_send_backlog(), this function will return
* #GST_RTSP_ENOMEM.
*
* Returns: #GST_RTSP_OK on success. #GST_RTSP_ENOMEM when the backlog limits
* are reached. #GST_RTSP_EINTR when @watch was flushing.
*/
/* FIXME 2.0: This should've been static! */
GstRTSPResult
gst_rtsp_watch_write_data (GstRTSPWatch * watch, const guint8 * data,
guint size, guint * id)
{
GstRTSPSerializedMessage serialized_message;
memset (&serialized_message, 0, sizeof (serialized_message));
serialized_message.data = (guint8 *) data;
serialized_message.data_size = size;
return gst_rtsp_watch_write_serialized_messages (watch, &serialized_message,
1, id);
}
/**
* gst_rtsp_watch_send_message:
* @watch: a #GstRTSPWatch
* @message: a #GstRTSPMessage
* @id: (out) (allow-none): location for a message ID or %NULL
*
* Send a @message using the connection of the @watch. If it cannot be sent
* immediately, it will be queued for transmission in @watch. The contents of
* @message will then be serialized and transmitted when the connection of the
* @watch becomes writable. In case the @message is queued, the ID returned in
* @id will be non-zero and used as the ID argument in the message_sent
* callback.
*
* Returns: #GST_RTSP_OK on success.
*/
GstRTSPResult
gst_rtsp_watch_send_message (GstRTSPWatch * watch, GstRTSPMessage * message,
guint * id)
{
g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
return gst_rtsp_watch_send_messages (watch, message, 1, id);
}
/**
* gst_rtsp_watch_send_messages:
* @watch: a #GstRTSPWatch
* @messages: (array length=n_messages): the messages to send
* @n_messages: the number of messages to send
* @id: (out) (allow-none): location for a message ID or %NULL
*
* Sends @messages using the connection of the @watch. If they cannot be sent
* immediately, they will be queued for transmission in @watch. The contents of
* @messages will then be serialized and transmitted when the connection of the
* @watch becomes writable. In case the @messages are queued, the ID returned in
* @id will be non-zero and used as the ID argument in the message_sent
* callback once the last message is sent. The callback will only be called
* once for the last message.
*
* Returns: #GST_RTSP_OK on success.
*
* Since: 1.16
*/
GstRTSPResult
gst_rtsp_watch_send_messages (GstRTSPWatch * watch, GstRTSPMessage * messages,
guint n_messages, guint * id)
{
GstRTSPSerializedMessage *serialized_messages;
gint i;
g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (messages != NULL || n_messages == 0, GST_RTSP_EINVAL);
serialized_messages = g_newa (GstRTSPSerializedMessage, n_messages);
memset (serialized_messages, 0,
sizeof (GstRTSPSerializedMessage) * n_messages);
for (i = 0; i < n_messages; i++) {
if (!serialize_message (watch->conn, &messages[i], &serialized_messages[i]))
goto error;
}
return gst_rtsp_watch_write_serialized_messages (watch, serialized_messages,
n_messages, id);
error:
for (i = 0; i < n_messages; i++) {
gst_rtsp_serialized_message_clear (&serialized_messages[i]);
}
return GST_RTSP_EINVAL;
}
/**
* gst_rtsp_watch_wait_backlog_usec:
* @watch: a #GstRTSPWatch
* @timeout: a timeout in microseconds
*
* Wait until there is place in the backlog queue, @timeout is reached
* or @watch is set to flushing.
*
* If @timeout is 0 this function can block forever. If @timeout
* contains a valid timeout, this function will return %GST_RTSP_ETIMEOUT
* after the timeout expired.
*
* The typically use of this function is when gst_rtsp_watch_write_data
* returns %GST_RTSP_ENOMEM. The caller then calls this function to wait for
* free space in the backlog queue and try again.
*
* Returns: %GST_RTSP_OK when if there is room in queue.
* %GST_RTSP_ETIMEOUT when @timeout was reached.
* %GST_RTSP_EINTR when @watch is flushing
* %GST_RTSP_EINVAL when called with invalid parameters.
*
* Since: 1.18
*/
GstRTSPResult
gst_rtsp_watch_wait_backlog_usec (GstRTSPWatch * watch, gint64 timeout)
{
gint64 end_time;
g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
end_time = g_get_monotonic_time () + timeout;
g_mutex_lock (&watch->mutex);
if (watch->flushing)
goto flushing;
while (IS_BACKLOG_FULL (watch)) {
gboolean res;
res = g_cond_wait_until (&watch->queue_not_full, &watch->mutex, end_time);
if (watch->flushing)
goto flushing;
if (!res)
goto timeout;
}
g_mutex_unlock (&watch->mutex);
return GST_RTSP_OK;
/* ERRORS */
flushing:
{
GST_DEBUG ("we are flushing");
g_mutex_unlock (&watch->mutex);
return GST_RTSP_EINTR;
}
timeout:
{
GST_DEBUG ("we timed out");
g_mutex_unlock (&watch->mutex);
return GST_RTSP_ETIMEOUT;
}
}
/**
* gst_rtsp_watch_set_flushing:
* @watch: a #GstRTSPWatch
* @flushing: new flushing state
*
* When @flushing is %TRUE, abort a call to gst_rtsp_watch_wait_backlog()
* and make sure gst_rtsp_watch_write_data() returns immediately with
* #GST_RTSP_EINTR. And empty the queue.
*
* Since: 1.4
*/
void
gst_rtsp_watch_set_flushing (GstRTSPWatch * watch, gboolean flushing)
{
g_return_if_fail (watch != NULL);
g_mutex_lock (&watch->mutex);
watch->flushing = flushing;
g_cond_signal (&watch->queue_not_full);
if (flushing) {
GstRTSPSerializedMessage *msg;
while ((msg = gst_queue_array_pop_head_struct (watch->messages))) {
gst_rtsp_serialized_message_clear (msg);
}
}
g_mutex_unlock (&watch->mutex);
}
#ifndef GST_DISABLE_DEPRECATED
G_GNUC_BEGIN_IGNORE_DEPRECATIONS
/* Deprecated */
#define TV_TO_USEC(tv) ((tv) ? ((tv)->tv_sec * G_USEC_PER_SEC + (tv)->tv_usec) : 0)
/**
* gst_rtsp_connection_connect:
* @conn: a #GstRTSPConnection
* @timeout: a GTimeVal timeout
*
* Attempt to connect to the url of @conn made with
* gst_rtsp_connection_create(). If @timeout is %NULL this function can block
* forever. If @timeout contains a valid timeout, this function will return
* #GST_RTSP_ETIMEOUT after the timeout expired.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK when a connection could be made.
*
* Deprecated: 1.18
*/
GstRTSPResult
gst_rtsp_connection_connect (GstRTSPConnection * conn, GTimeVal * timeout)
{
return gst_rtsp_connection_connect_usec (conn, TV_TO_USEC (timeout));
}
/**
* gst_rtsp_connection_connect_with_response:
* @conn: a #GstRTSPConnection
* @timeout: a GTimeVal timeout
* @response: a #GstRTSPMessage
*
* Attempt to connect to the url of @conn made with
* gst_rtsp_connection_create(). If @timeout is %NULL this function can block
* forever. If @timeout contains a valid timeout, this function will return
* #GST_RTSP_ETIMEOUT after the timeout expired. If @conn is set to tunneled,
* @response will contain a response to the tunneling request messages.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK when a connection could be made.
*
* Since: 1.8
* Deprecated: 1.18
*/
GstRTSPResult
gst_rtsp_connection_connect_with_response (GstRTSPConnection * conn,
GTimeVal * timeout, GstRTSPMessage * response)
{
return gst_rtsp_connection_connect_with_response_usec (conn,
TV_TO_USEC (timeout), response);
}
/**
* gst_rtsp_connection_read:
* @conn: a #GstRTSPConnection
* @data: the data to read
* @size: the size of @data
* @timeout: a timeout value or %NULL
*
* Attempt to read @size bytes into @data from the connected @conn, blocking up to
* the specified @timeout. @timeout can be %NULL, in which case this function
* might block forever.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK on success.
*
* Deprecated: 1.18
*/
GstRTSPResult
gst_rtsp_connection_read (GstRTSPConnection * conn, guint8 * data, guint size,
GTimeVal * timeout)
{
return gst_rtsp_connection_read_usec (conn, data, size, TV_TO_USEC (timeout));
}
/**
* gst_rtsp_connection_write:
* @conn: a #GstRTSPConnection
* @data: the data to write
* @size: the size of @data
* @timeout: a timeout value or %NULL
*
* Attempt to write @size bytes of @data to the connected @conn, blocking up to
* the specified @timeout. @timeout can be %NULL, in which case this function
* might block forever.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK on success.
*
* Deprecated: 1.18
*/
GstRTSPResult
gst_rtsp_connection_write (GstRTSPConnection * conn, const guint8 * data,
guint size, GTimeVal * timeout)
{
return gst_rtsp_connection_write_usec (conn, data, size,
TV_TO_USEC (timeout));
}
/**
* gst_rtsp_connection_send:
* @conn: a #GstRTSPConnection
* @message: the message to send
* @timeout: a timeout value or %NULL
*
* Attempt to send @message to the connected @conn, blocking up to
* the specified @timeout. @timeout can be %NULL, in which case this function
* might block forever.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK on success.
*
* Deprecated: 1.18
*/
GstRTSPResult
gst_rtsp_connection_send (GstRTSPConnection * conn, GstRTSPMessage * message,
GTimeVal * timeout)
{
return gst_rtsp_connection_send_usec (conn, message, TV_TO_USEC (timeout));
}
/**
* gst_rtsp_connection_send_messages:
* @conn: a #GstRTSPConnection
* @messages: (array length=n_messages): the messages to send
* @n_messages: the number of messages to send
* @timeout: a timeout value or %NULL
*
* Attempt to send @messages to the connected @conn, blocking up to
* the specified @timeout. @timeout can be %NULL, in which case this function
* might block forever.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK on success.
*
* Since: 1.16
* Deprecated: 1.18
*/
GstRTSPResult
gst_rtsp_connection_send_messages (GstRTSPConnection * conn,
GstRTSPMessage * messages, guint n_messages, GTimeVal * timeout)
{
return gst_rtsp_connection_send_messages_usec (conn, messages, n_messages,
TV_TO_USEC (timeout));
}
/**
* gst_rtsp_connection_receive:
* @conn: a #GstRTSPConnection
* @message: the message to read
* @timeout: a timeout value or %NULL
*
* Attempt to read into @message from the connected @conn, blocking up to
* the specified @timeout. @timeout can be %NULL, in which case this function
* might block forever.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK on success.
*
* Deprecated: 1.18
*/
GstRTSPResult
gst_rtsp_connection_receive (GstRTSPConnection * conn, GstRTSPMessage * message,
GTimeVal * timeout)
{
return gst_rtsp_connection_receive_usec (conn, message, TV_TO_USEC (timeout));
}
/**
* gst_rtsp_connection_poll:
* @conn: a #GstRTSPConnection
* @events: a bitmask of #GstRTSPEvent flags to check
* @revents: location for result flags
* @timeout: a timeout
*
* Wait up to the specified @timeout for the connection to become available for
* at least one of the operations specified in @events. When the function returns
* with #GST_RTSP_OK, @revents will contain a bitmask of available operations on
* @conn.
*
* @timeout can be %NULL, in which case this function might block forever.
*
* This function can be cancelled with gst_rtsp_connection_flush().
*
* Returns: #GST_RTSP_OK on success.
*
* Deprecated: 1.18
*/
GstRTSPResult
gst_rtsp_connection_poll (GstRTSPConnection * conn, GstRTSPEvent events,
GstRTSPEvent * revents, GTimeVal * timeout)
{
return gst_rtsp_connection_poll_usec (conn, events, revents,
TV_TO_USEC (timeout));
}
/**
* gst_rtsp_connection_next_timeout:
* @conn: a #GstRTSPConnection
* @timeout: a timeout
*
* Calculate the next timeout for @conn, storing the result in @timeout.
*
* Returns: #GST_RTSP_OK.
*
* Deprecated: 1.18
*/
GstRTSPResult
gst_rtsp_connection_next_timeout (GstRTSPConnection * conn, GTimeVal * timeout)
{
gint64 tmptimeout = 0;
g_return_val_if_fail (timeout != NULL, GST_RTSP_EINVAL);
tmptimeout = gst_rtsp_connection_next_timeout_usec (conn);
timeout->tv_sec = tmptimeout / G_USEC_PER_SEC;
timeout->tv_usec = tmptimeout % G_USEC_PER_SEC;
return GST_RTSP_OK;
}
/**
* gst_rtsp_watch_wait_backlog:
* @watch: a #GstRTSPWatch
* @timeout: a GTimeVal timeout
*
* Wait until there is place in the backlog queue, @timeout is reached
* or @watch is set to flushing.
*
* If @timeout is %NULL this function can block forever. If @timeout
* contains a valid timeout, this function will return %GST_RTSP_ETIMEOUT
* after the timeout expired.
*
* The typically use of this function is when gst_rtsp_watch_write_data
* returns %GST_RTSP_ENOMEM. The caller then calls this function to wait for
* free space in the backlog queue and try again.
*
* Returns: %GST_RTSP_OK when if there is room in queue.
* %GST_RTSP_ETIMEOUT when @timeout was reached.
* %GST_RTSP_EINTR when @watch is flushing
* %GST_RTSP_EINVAL when called with invalid parameters.
*
* Since: 1.4
* Deprecated: 1.18
*/
GstRTSPResult
gst_rtsp_watch_wait_backlog (GstRTSPWatch * watch, GTimeVal * timeout)
{
return gst_rtsp_watch_wait_backlog_usec (watch, TV_TO_USEC (timeout));
}
G_GNUC_END_IGNORE_DEPRECATIONS
#endif /* GST_DISABLE_DEPRECATED */
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