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gstreamer/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-client.c
Branko Subasic 341e55e9f2 gst-rtsp-server: fix race in rtsp-client 
When tunneling over HTTP, if connection on the second channel happens
before the control timer is created we may trigger an assert in
rtsp_ctrl_timeout_remove(). Avoid that by taking the priv->lock before
attaching the client thread to the context.

Fixes #1025

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1870>
2022-03-07 11:02:13 +00:00

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/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
* Copyright (C) 2015 Centricular Ltd
* Author: Sebastian Dröge <sebastian@centricular.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.
*/
/**
* SECTION:rtsp-client
* @short_description: A client connection state
* @see_also: #GstRTSPServer, #GstRTSPThreadPool
*
* The client object handles the connection with a client for as long as a TCP
* connection is open.
*
* A #GstRTSPClient is created by #GstRTSPServer when a new connection is
* accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
* #GstRTSPAuth and #GstRTSPThreadPool from the server.
*
* The client connection should be configured with the #GstRTSPConnection using
* gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
* using gst_rtsp_client_attach(). From then on the client will handle requests
* on the connection.
*
* Use gst_rtsp_client_session_filter() to iterate or modify all the
* #GstRTSPSession objects managed by the client object.
*
* Last reviewed on 2013-07-11 (1.0.0)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdio.h>
#include <string.h>
#include <gst/sdp/gstmikey.h>
#include <gst/rtsp/gstrtsp-enumtypes.h>
#include "rtsp-client.h"
#include "rtsp-sdp.h"
#include "rtsp-params.h"
#include "rtsp-server-internal.h"
typedef enum
{
TUNNEL_STATE_UNKNOWN,
TUNNEL_STATE_GET,
TUNNEL_STATE_POST
} GstRTSPTunnelState;
/* locking order:
* send_lock, lock, tunnels_lock
*/
struct _GstRTSPClientPrivate
{
GMutex lock; /* protects everything else */
GMutex send_lock;
GMutex watch_lock;
GstRTSPConnection *connection;
GstRTSPWatch *watch;
GMainContext *watch_context;
gchar *server_ip;
gboolean is_ipv6;
/* protected by send_lock */
GstRTSPClientSendFunc send_func;
gpointer send_data;
GDestroyNotify send_notify;
GstRTSPClientSendMessagesFunc send_messages_func;
gpointer send_messages_data;
GDestroyNotify send_messages_notify;
GArray *data_seqs;
GstRTSPSessionPool *session_pool;
gulong session_removed_id;
GstRTSPMountPoints *mount_points;
GstRTSPAuth *auth;
GstRTSPThreadPool *thread_pool;
/* used to cache the media in the last requested DESCRIBE so that
* we can pick it up in the next SETUP immediately */
gchar *path;
GstRTSPMedia *media;
GHashTable *transports;
GList *sessions;
guint sessions_cookie;
gboolean drop_backlog;
gint post_session_timeout;
guint content_length_limit;
gboolean had_session;
GSource *rtsp_ctrl_timeout;
guint rtsp_ctrl_timeout_cnt;
/* The version currently being used */
GstRTSPVersion version;
GHashTable *pipelined_requests; /* pipelined_request_id -> session_id */
GstRTSPTunnelState tstate;
};
typedef struct
{
guint8 channel;
guint seq;
} DataSeq;
static GMutex tunnels_lock;
static GHashTable *tunnels; /* protected by tunnels_lock */
#define WATCH_BACKLOG_SIZE 100
#define DEFAULT_SESSION_POOL NULL
#define DEFAULT_MOUNT_POINTS NULL
#define DEFAULT_DROP_BACKLOG TRUE
#define DEFAULT_POST_SESSION_TIMEOUT -1
#define RTSP_CTRL_CB_INTERVAL 1
#define RTSP_CTRL_TIMEOUT_VALUE 60
enum
{
PROP_0,
PROP_SESSION_POOL,
PROP_MOUNT_POINTS,
PROP_DROP_BACKLOG,
PROP_POST_SESSION_TIMEOUT,
PROP_LAST
};
enum
{
SIGNAL_CLOSED,
SIGNAL_NEW_SESSION,
SIGNAL_PRE_OPTIONS_REQUEST,
SIGNAL_OPTIONS_REQUEST,
SIGNAL_PRE_DESCRIBE_REQUEST,
SIGNAL_DESCRIBE_REQUEST,
SIGNAL_PRE_SETUP_REQUEST,
SIGNAL_SETUP_REQUEST,
SIGNAL_PRE_PLAY_REQUEST,
SIGNAL_PLAY_REQUEST,
SIGNAL_PRE_PAUSE_REQUEST,
SIGNAL_PAUSE_REQUEST,
SIGNAL_PRE_TEARDOWN_REQUEST,
SIGNAL_TEARDOWN_REQUEST,
SIGNAL_PRE_SET_PARAMETER_REQUEST,
SIGNAL_SET_PARAMETER_REQUEST,
SIGNAL_PRE_GET_PARAMETER_REQUEST,
SIGNAL_GET_PARAMETER_REQUEST,
SIGNAL_HANDLE_RESPONSE,
SIGNAL_SEND_MESSAGE,
SIGNAL_PRE_ANNOUNCE_REQUEST,
SIGNAL_ANNOUNCE_REQUEST,
SIGNAL_PRE_RECORD_REQUEST,
SIGNAL_RECORD_REQUEST,
SIGNAL_CHECK_REQUIREMENTS,
SIGNAL_LAST
};
GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
#define GST_CAT_DEFAULT rtsp_client_debug
static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
static void gst_rtsp_client_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec);
static void gst_rtsp_client_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec);
static void gst_rtsp_client_finalize (GObject * obj);
static void rtsp_ctrl_timeout_remove (GstRTSPClient * client);
static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
static gboolean handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPMedia * media, GstSDPMessage * sdp);
static gboolean default_configure_client_media (GstRTSPClient * client,
GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
static gboolean default_configure_client_transport (GstRTSPClient * client,
GstRTSPContext * ctx, GstRTSPTransport * ct);
static GstRTSPResult default_params_set (GstRTSPClient * client,
GstRTSPContext * ctx);
static GstRTSPResult default_params_get (GstRTSPClient * client,
GstRTSPContext * ctx);
static gchar *default_make_path_from_uri (GstRTSPClient * client,
const GstRTSPUrl * uri);
static void client_session_removed (GstRTSPSessionPool * pool,
GstRTSPSession * session, GstRTSPClient * client);
static GstRTSPStatusCode default_pre_signal_handler (GstRTSPClient * client,
GstRTSPContext * ctx);
static gboolean pre_signal_accumulator (GSignalInvocationHint * ihint,
GValue * return_accu, const GValue * handler_return, gpointer data);
G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
static void
gst_rtsp_client_class_init (GstRTSPClientClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->get_property = gst_rtsp_client_get_property;
gobject_class->set_property = gst_rtsp_client_set_property;
gobject_class->finalize = gst_rtsp_client_finalize;
klass->create_sdp = create_sdp;
klass->handle_sdp = handle_sdp;
klass->configure_client_media = default_configure_client_media;
klass->configure_client_transport = default_configure_client_transport;
klass->params_set = default_params_set;
klass->params_get = default_params_get;
klass->make_path_from_uri = default_make_path_from_uri;
klass->pre_options_request = default_pre_signal_handler;
klass->pre_describe_request = default_pre_signal_handler;
klass->pre_setup_request = default_pre_signal_handler;
klass->pre_play_request = default_pre_signal_handler;
klass->pre_pause_request = default_pre_signal_handler;
klass->pre_teardown_request = default_pre_signal_handler;
klass->pre_set_parameter_request = default_pre_signal_handler;
klass->pre_get_parameter_request = default_pre_signal_handler;
klass->pre_announce_request = default_pre_signal_handler;
klass->pre_record_request = default_pre_signal_handler;
g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
g_param_spec_object ("session-pool", "Session Pool",
"The session pool to use for client session",
GST_TYPE_RTSP_SESSION_POOL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
g_param_spec_object ("mount-points", "Mount Points",
"The mount points to use for client session",
GST_TYPE_RTSP_MOUNT_POINTS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG,
g_param_spec_boolean ("drop-backlog", "Drop Backlog",
"Drop data when the backlog queue is full",
DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClient::post-session-timeout:
*
* An extra tcp timeout ( > 0) after session timeout, in seconds.
* The tcp connection will be kept alive until this timeout happens to give
* the client a possibility to reuse the connection.
* 0 means that the connection will be closed immediately after the session
* timeout.
*
* Default value is -1 seconds, meaning that we let the system close
* the connection.
*
* Since: 1.18
*/
g_object_class_install_property (gobject_class, PROP_POST_SESSION_TIMEOUT,
g_param_spec_int ("post-session-timeout", "Post Session Timeout",
"An extra TCP connection timeout after session timeout", G_MININT,
G_MAXINT, DEFAULT_POST_SESSION_TIMEOUT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_rtsp_client_signals[SIGNAL_CLOSED] =
g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL, NULL,
G_TYPE_NONE, 0, G_TYPE_NONE);
gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL, NULL,
G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
/**
* GstRTSPClient::pre-options-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
* otherwise an appropriate return code
*
* Since: 1.12
*/
gst_rtsp_client_signals[SIGNAL_PRE_OPTIONS_REQUEST] =
g_signal_new ("pre-options-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
pre_options_request), pre_signal_accumulator, NULL, NULL,
GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::options-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*/
gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::pre-describe-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
* otherwise an appropriate return code
*
* Since: 1.12
*/
gst_rtsp_client_signals[SIGNAL_PRE_DESCRIBE_REQUEST] =
g_signal_new ("pre-describe-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
pre_describe_request), pre_signal_accumulator, NULL, NULL,
GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::describe-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*/
gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::pre-setup-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
* otherwise an appropriate return code
*
* Since: 1.12
*/
gst_rtsp_client_signals[SIGNAL_PRE_SETUP_REQUEST] =
g_signal_new ("pre-setup-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
pre_setup_request), pre_signal_accumulator, NULL, NULL,
GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::setup-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*/
gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::pre-play-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
* otherwise an appropriate return code
*
* Since: 1.12
*/
gst_rtsp_client_signals[SIGNAL_PRE_PLAY_REQUEST] =
g_signal_new ("pre-play-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
pre_play_request), pre_signal_accumulator, NULL,
NULL, GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::play-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*/
gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::pre-pause-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
* otherwise an appropriate return code
*
* Since: 1.12
*/
gst_rtsp_client_signals[SIGNAL_PRE_PAUSE_REQUEST] =
g_signal_new ("pre-pause-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
pre_pause_request), pre_signal_accumulator, NULL, NULL,
GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::pause-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*/
gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::pre-teardown-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
* otherwise an appropriate return code
*
* Since: 1.12
*/
gst_rtsp_client_signals[SIGNAL_PRE_TEARDOWN_REQUEST] =
g_signal_new ("pre-teardown-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
pre_teardown_request), pre_signal_accumulator, NULL, NULL,
GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::teardown-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*/
gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::pre-set-parameter-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
* otherwise an appropriate return code
*
* Since: 1.12
*/
gst_rtsp_client_signals[SIGNAL_PRE_SET_PARAMETER_REQUEST] =
g_signal_new ("pre-set-parameter-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
pre_set_parameter_request), pre_signal_accumulator, NULL, NULL,
GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::set-parameter-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*/
gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
set_parameter_request), NULL, NULL, NULL,
G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::pre-get-parameter-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
* otherwise an appropriate return code
*
* Since: 1.12
*/
gst_rtsp_client_signals[SIGNAL_PRE_GET_PARAMETER_REQUEST] =
g_signal_new ("pre-get-parameter-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
pre_get_parameter_request), pre_signal_accumulator, NULL, NULL,
GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::get-parameter-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*/
gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
get_parameter_request), NULL, NULL, NULL,
G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::handle-response:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*/
gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
handle_response), NULL, NULL, NULL,
G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::send-message:
* @client: The RTSP client
* @session: (type GstRtspServer.RTSPSession): The session
* @message: (type GstRtsp.RTSPMessage): The message
*/
gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] =
g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
send_message), NULL, NULL, NULL,
G_TYPE_NONE, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_POINTER);
/**
* GstRTSPClient::pre-announce-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
* otherwise an appropriate return code
*
* Since: 1.12
*/
gst_rtsp_client_signals[SIGNAL_PRE_ANNOUNCE_REQUEST] =
g_signal_new ("pre-announce-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
pre_announce_request), pre_signal_accumulator, NULL, NULL,
GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::announce-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*/
gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST] =
g_signal_new ("announce-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, announce_request),
NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::pre-record-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
* otherwise an appropriate return code
*
* Since: 1.12
*/
gst_rtsp_client_signals[SIGNAL_PRE_RECORD_REQUEST] =
g_signal_new ("pre-record-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
pre_record_request), pre_signal_accumulator, NULL, NULL,
GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::record-request:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
*/
gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST] =
g_signal_new ("record-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, record_request),
NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::check-requirements:
* @client: a #GstRTSPClient
* @ctx: (type GstRtspServer.RTSPContext): a #GstRTSPContext
* @arr: a NULL-terminated array of strings
*
* Returns: a newly allocated string with comma-separated list of
* unsupported options. An empty string must be returned if
* all options are supported.
*
* Since: 1.6
*/
gst_rtsp_client_signals[SIGNAL_CHECK_REQUIREMENTS] =
g_signal_new ("check-requirements", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
check_requirements), NULL, NULL, NULL,
G_TYPE_STRING, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_STRV);
tunnels =
g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
g_mutex_init (&tunnels_lock);
GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
}
static void
gst_rtsp_client_init (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = gst_rtsp_client_get_instance_private (client);
client->priv = priv;
g_mutex_init (&priv->lock);
g_mutex_init (&priv->send_lock);
g_mutex_init (&priv->watch_lock);
priv->data_seqs = g_array_new (FALSE, FALSE, sizeof (DataSeq));
priv->drop_backlog = DEFAULT_DROP_BACKLOG;
priv->post_session_timeout = DEFAULT_POST_SESSION_TIMEOUT;
priv->transports =
g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL,
g_object_unref);
priv->pipelined_requests = g_hash_table_new_full (g_str_hash,
g_str_equal, g_free, g_free);
priv->tstate = TUNNEL_STATE_UNKNOWN;
priv->content_length_limit = G_MAXUINT;
}
static GstRTSPFilterResult
filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
gpointer user_data)
{
gboolean *closed = user_data;
GstRTSPMedia *media;
guint i, n_streams;
gboolean is_all_udp = TRUE;
media = gst_rtsp_session_media_get_media (sessmedia);
n_streams = gst_rtsp_media_n_streams (media);
for (i = 0; i < n_streams; i++) {
GstRTSPStreamTransport *transport =
gst_rtsp_session_media_get_transport (sessmedia, i);
const GstRTSPTransport *rtsp_transport;
if (!transport)
continue;
rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
if (rtsp_transport
&& rtsp_transport->lower_transport != GST_RTSP_LOWER_TRANS_UDP
&& rtsp_transport->lower_transport != GST_RTSP_LOWER_TRANS_UDP_MCAST) {
is_all_udp = FALSE;
break;
}
}
if (!is_all_udp || gst_rtsp_media_is_stop_on_disconnect (media)) {
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
return GST_RTSP_FILTER_REMOVE;
} else {
*closed = FALSE;
return GST_RTSP_FILTER_KEEP;
}
}
static void
client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
{
GstRTSPClientPrivate *priv = client->priv;
g_mutex_lock (&priv->lock);
/* check if we already know about this session */
if (g_list_find (priv->sessions, session) == NULL) {
GST_INFO ("watching session %p", session);
priv->sessions = g_list_prepend (priv->sessions, g_object_ref (session));
priv->sessions_cookie++;
/* connect removed session handler, it will be disconnected when the last
* session gets removed */
if (priv->session_removed_id == 0)
priv->session_removed_id = g_signal_connect_data (priv->session_pool,
"session-removed", G_CALLBACK (client_session_removed),
g_object_ref (client), (GClosureNotify) g_object_unref, 0);
}
g_mutex_unlock (&priv->lock);
return;
}
/* should be called with lock */
static void
client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session,
GList * link)
{
GstRTSPClientPrivate *priv = client->priv;
GST_INFO ("client %p: unwatch session %p", client, session);
if (link == NULL) {
link = g_list_find (priv->sessions, session);
if (link == NULL)
return;
}
priv->sessions = g_list_delete_link (priv->sessions, link);
priv->sessions_cookie++;
/* if this was the last session, disconnect the handler.
* This will also drop the extra client ref */
if (!priv->sessions) {
g_signal_handler_disconnect (priv->session_pool, priv->session_removed_id);
priv->session_removed_id = 0;
}
if (!priv->drop_backlog) {
/* unlink all media managed in this session */
gst_rtsp_session_filter (session, filter_session_media, client);
}
/* remove the session */
g_object_unref (session);
}
static GstRTSPFilterResult
cleanup_session (GstRTSPClient * client, GstRTSPSession * sess,
gpointer user_data)
{
gboolean *closed = user_data;
GstRTSPClientPrivate *priv = client->priv;
if (priv->drop_backlog) {
/* unlink all media managed in this session. This needs to happen
* without the client lock, so we really want to do it here. */
gst_rtsp_session_filter (sess, filter_session_media, user_data);
}
if (*closed)
return GST_RTSP_FILTER_REMOVE;
else
return GST_RTSP_FILTER_KEEP;
}
static void
clean_cached_media (GstRTSPClient * client, gboolean unprepare)
{
GstRTSPClientPrivate *priv = client->priv;
if (priv->path) {
g_free (priv->path);
priv->path = NULL;
}
if (priv->media) {
if (unprepare)
gst_rtsp_media_unprepare (priv->media);
g_object_unref (priv->media);
priv->media = NULL;
}
}
/* A client is finalized when the connection is broken */
static void
gst_rtsp_client_finalize (GObject * obj)
{
GstRTSPClient *client = GST_RTSP_CLIENT (obj);
GstRTSPClientPrivate *priv = client->priv;
GST_INFO ("finalize client %p", client);
/* the watch and related state should be cleared before finalize
* as the watch actually holds a strong reference to the client */
g_assert (priv->watch == NULL);
g_assert (priv->rtsp_ctrl_timeout == NULL);
if (priv->watch_context) {
g_main_context_unref (priv->watch_context);
priv->watch_context = NULL;
}
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
/* all sessions should have been removed by now. We keep a ref to
* the client object for the session removed handler. The ref is
* dropped when the last session is removed from the list. */
g_assert (priv->sessions == NULL);
g_assert (priv->session_removed_id == 0);
g_array_unref (priv->data_seqs);
g_hash_table_unref (priv->transports);
g_hash_table_unref (priv->pipelined_requests);
if (priv->connection)
gst_rtsp_connection_free (priv->connection);
if (priv->session_pool) {
g_object_unref (priv->session_pool);
}
if (priv->mount_points)
g_object_unref (priv->mount_points);
if (priv->auth)
g_object_unref (priv->auth);
if (priv->thread_pool)
g_object_unref (priv->thread_pool);
clean_cached_media (client, TRUE);
g_free (priv->server_ip);
g_mutex_clear (&priv->lock);
g_mutex_clear (&priv->send_lock);
g_mutex_clear (&priv->watch_lock);
G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
}
static void
gst_rtsp_client_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
GstRTSPClientPrivate *priv = client->priv;
switch (propid) {
case PROP_SESSION_POOL:
g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
break;
case PROP_MOUNT_POINTS:
g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
break;
case PROP_DROP_BACKLOG:
g_value_set_boolean (value, priv->drop_backlog);
break;
case PROP_POST_SESSION_TIMEOUT:
g_value_set_int (value, priv->post_session_timeout);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
gst_rtsp_client_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
GstRTSPClientPrivate *priv = client->priv;
switch (propid) {
case PROP_SESSION_POOL:
gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
break;
case PROP_MOUNT_POINTS:
gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
break;
case PROP_DROP_BACKLOG:
g_mutex_lock (&priv->lock);
priv->drop_backlog = g_value_get_boolean (value);
g_mutex_unlock (&priv->lock);
break;
case PROP_POST_SESSION_TIMEOUT:
g_mutex_lock (&priv->lock);
priv->post_session_timeout = g_value_get_int (value);
g_mutex_unlock (&priv->lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
/**
* gst_rtsp_client_new:
*
* Create a new #GstRTSPClient instance.
*
* Returns: (transfer full): a new #GstRTSPClient
*/
GstRTSPClient *
gst_rtsp_client_new (void)
{
GstRTSPClient *result;
result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
return result;
}
static void
send_message (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPMessage * message, gboolean close)
{
GstRTSPClientPrivate *priv = client->priv;
gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
"GStreamer RTSP server");
/* remove any previous header */
gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
/* add the new session header for new session ids */
if (ctx->session) {
gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
gst_rtsp_session_get_header (ctx->session));
}
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (message);
}
if (close)
gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
if (ctx->request)
message->type_data.response.version =
ctx->request->type_data.request.version;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE],
0, ctx, message);
g_mutex_lock (&priv->send_lock);
if (priv->send_messages_func) {
priv->send_messages_func (client, message, 1, close, priv->send_data);
} else if (priv->send_func) {
priv->send_func (client, message, close, priv->send_data);
}
g_mutex_unlock (&priv->send_lock);
gst_rtsp_message_unset (message);
}
static void
send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
GstRTSPContext * ctx)
{
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
ctx->session = NULL;
send_message (client, ctx, ctx->response, FALSE);
}
static void
send_option_not_supported_response (GstRTSPClient * client,
GstRTSPContext * ctx, const gchar * unsupported_options)
{
GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED;
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
if (unsupported_options != NULL) {
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED,
unsupported_options);
}
ctx->session = NULL;
send_message (client, ctx, ctx->response, FALSE);
}
static gboolean
paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
{
if (path1 == NULL || path2 == NULL)
return FALSE;
if (strlen (path1) != len2)
return FALSE;
if (strncmp (path1, path2, len2))
return FALSE;
return TRUE;
}
/* this function is called to initially find the media for the DESCRIBE request
* but is cached for when the same client (without breaking the connection) is
* doing a setup for the exact same url. */
static GstRTSPMedia *
find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
gint * matched)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
gint path_len;
/* find the longest matching factory for the uri first */
if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
path, matched)))
goto no_factory;
ctx->factory = factory;
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
goto no_factory_access;
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
goto not_authorized;
if (matched)
path_len = *matched;
else
path_len = strlen (path);
if (!paths_are_equal (priv->path, path, path_len)) {
/* remove any previously cached values before we try to construct a new
* media for uri */
clean_cached_media (client, TRUE);
/* prepare the media and add it to the pipeline */
if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
goto no_media;
ctx->media = media;
if (!(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_RECORD)) {
GstRTSPThread *thread;
thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
GST_RTSP_THREAD_TYPE_MEDIA, ctx);
if (thread == NULL)
goto no_thread;
/* prepare the media */
if (!gst_rtsp_media_prepare (media, thread))
goto no_prepare;
}
/* now keep track of the uri and the media */
priv->path = g_strndup (path, path_len);
priv->media = media;
} else {
/* we have seen this path before, used cached media */
media = priv->media;
ctx->media = media;
GST_INFO ("reusing cached media %p for path %s", media, priv->path);
}
g_object_unref (factory);
ctx->factory = NULL;
if (media)
g_object_ref (media);
return media;
/* ERRORS */
no_factory:
{
GST_ERROR ("client %p: no factory for path %s", client, path);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return NULL;
}
no_factory_access:
{
g_object_unref (factory);
ctx->factory = NULL;
GST_ERROR ("client %p: not authorized to see factory path %s", client,
path);
/* error reply is already sent */
return NULL;
}
not_authorized:
{
g_object_unref (factory);
ctx->factory = NULL;
GST_ERROR ("client %p: not authorized for factory path %s", client, path);
/* error reply is already sent */
return NULL;
}
no_media:
{
GST_ERROR ("client %p: can't create media", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
g_object_unref (factory);
ctx->factory = NULL;
return NULL;
}
no_thread:
{
GST_ERROR ("client %p: can't create thread", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
ctx->media = NULL;
g_object_unref (factory);
ctx->factory = NULL;
return NULL;
}
no_prepare:
{
GST_ERROR ("client %p: can't prepare media", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
ctx->media = NULL;
g_object_unref (factory);
ctx->factory = NULL;
return NULL;
}
}
static inline DataSeq *
get_data_seq_element (GstRTSPClient * client, guint8 channel)
{
GstRTSPClientPrivate *priv = client->priv;
GArray *data_seqs = priv->data_seqs;
gint i = 0;
while (i < data_seqs->len) {
DataSeq *data_seq = &g_array_index (data_seqs, DataSeq, i);
if (data_seq->channel == channel)
return data_seq;
i++;
}
return NULL;
}
static void
add_data_seq (GstRTSPClient * client, guint8 channel)
{
GstRTSPClientPrivate *priv = client->priv;
DataSeq data_seq = {.channel = channel,.seq = 0 };
if (get_data_seq_element (client, channel) == NULL)
g_array_append_val (priv->data_seqs, data_seq);
}
static void
set_data_seq (GstRTSPClient * client, guint8 channel, guint seq)
{
DataSeq *data_seq;
data_seq = get_data_seq_element (client, channel);
g_assert_nonnull (data_seq);
data_seq->seq = seq;
}
static guint
get_data_seq (GstRTSPClient * client, guint8 channel)
{
DataSeq *data_seq;
data_seq = get_data_seq_element (client, channel);
g_assert_nonnull (data_seq);
return data_seq->seq;
}
static gboolean
get_data_channel (GstRTSPClient * client, guint seq, guint8 * channel)
{
GstRTSPClientPrivate *priv = client->priv;
GArray *data_seqs = priv->data_seqs;
gint i = 0;
while (i < data_seqs->len) {
DataSeq *data_seq = &g_array_index (data_seqs, DataSeq, i);
if (data_seq->seq == seq) {
*channel = data_seq->channel;
return TRUE;
}
i++;
}
return FALSE;
}
static gboolean
do_close (gpointer user_data)
{
GstRTSPClient *client = user_data;
gst_rtsp_client_close (client);
return G_SOURCE_REMOVE;
}
static gboolean
do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPMessage message = { 0 };
gboolean ret = TRUE;
gst_rtsp_message_init_data (&message, channel);
gst_rtsp_message_set_body_buffer (&message, buffer);
g_mutex_lock (&priv->send_lock);
if (get_data_seq (client, channel) != 0) {
GST_WARNING ("already a queued data message for channel %d", channel);
g_mutex_unlock (&priv->send_lock);
return FALSE;
}
if (priv->send_messages_func) {
ret =
priv->send_messages_func (client, &message, 1, FALSE, priv->send_data);
} else if (priv->send_func) {
ret = priv->send_func (client, &message, FALSE, priv->send_data);
}
g_mutex_unlock (&priv->send_lock);
gst_rtsp_message_unset (&message);
if (!ret) {
GSource *idle_src;
/* close in watch context */
idle_src = g_idle_source_new ();
g_source_set_callback (idle_src, do_close, client, NULL);
g_source_attach (idle_src, priv->watch_context);
g_source_unref (idle_src);
}
return ret;
}
static gboolean
do_check_back_pressure (guint8 channel, GstRTSPClient * client)
{
return get_data_seq (client, channel) != 0;
}
static gboolean
do_send_data_list (GstBufferList * buffer_list, guint8 channel,
GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
gboolean ret = TRUE;
guint i, n = gst_buffer_list_length (buffer_list);
GstRTSPMessage *messages;
g_mutex_lock (&priv->send_lock);
if (get_data_seq (client, channel) != 0) {
GST_WARNING ("already a queued data message for channel %d", channel);
g_mutex_unlock (&priv->send_lock);
return FALSE;
}
messages = g_newa (GstRTSPMessage, n);
memset (messages, 0, sizeof (GstRTSPMessage) * n);
for (i = 0; i < n; i++) {
GstBuffer *buffer = gst_buffer_list_get (buffer_list, i);
gst_rtsp_message_init_data (&messages[i], channel);
gst_rtsp_message_set_body_buffer (&messages[i], buffer);
}
if (priv->send_messages_func) {
ret =
priv->send_messages_func (client, messages, n, FALSE, priv->send_data);
} else if (priv->send_func) {
for (i = 0; i < n; i++) {
ret = priv->send_func (client, &messages[i], FALSE, priv->send_data);
if (!ret)
break;
}
}
g_mutex_unlock (&priv->send_lock);
for (i = 0; i < n; i++) {
gst_rtsp_message_unset (&messages[i]);
}
if (!ret) {
GSource *idle_src;
/* close in watch context */
idle_src = g_idle_source_new ();
g_source_set_callback (idle_src, do_close, client, NULL);
g_source_attach (idle_src, priv->watch_context);
g_source_unref (idle_src);
}
return ret;
}
/**
* gst_rtsp_client_close:
* @client: a #GstRTSPClient
*
* Close the connection of @client and remove all media it was managing.
*
* Since: 1.4
*/
void
gst_rtsp_client_close (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
GST_DEBUG ("client %p: closing connection", client);
g_mutex_lock (&priv->watch_lock);
/* Work around the lack of thread safety of gst_rtsp_connection_close */
if (priv->watch) {
gst_rtsp_watch_set_flushing (priv->watch, TRUE);
}
if (priv->connection) {
if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (&tunnels_lock);
}
gst_rtsp_connection_flush (priv->connection, TRUE);
gst_rtsp_connection_close (priv->connection);
}
if (priv->watch) {
GST_DEBUG ("client %p: destroying watch", client);
g_source_destroy ((GSource *) priv->watch);
priv->watch = NULL;
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
rtsp_ctrl_timeout_remove (client);
}
g_mutex_unlock (&priv->watch_lock);
}
static gchar *
default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
{
gchar *path;
if (uri->query) {
path = g_strconcat (uri->abspath, "?", uri->query, NULL);
} else {
/* normalize rtsp://<IP>:<PORT> to rtsp://<IP>:<PORT>/ */
path = g_strdup (uri->abspath[0] ? uri->abspath : "/");
}
return path;
}
/* Default signal handler function for all "pre-command" signals, like
* pre-options-request. It just returns the RTSP return code 200.
* Subclasses can override this to get another default behaviour.
*/
static GstRTSPStatusCode
default_pre_signal_handler (GstRTSPClient * client, GstRTSPContext * ctx)
{
GST_LOG_OBJECT (client, "returning GST_RTSP_STS_OK");
return GST_RTSP_STS_OK;
}
/* The pre-signal accumulator function checks the return value of the signal
* handlers. If any of them returns an RTSP status code that does not start
* with 2 it will return FALSE, no more signal handlers will be called, and
* this last RTSP status code will be the result of the signal emission.
*/
static gboolean
pre_signal_accumulator (GSignalInvocationHint * ihint, GValue * return_accu,
const GValue * handler_return, gpointer data)
{
GstRTSPStatusCode handler_value = g_value_get_enum (handler_return);
GstRTSPStatusCode accumulated_value = g_value_get_enum (return_accu);
if (handler_value < 200 || handler_value > 299) {
GST_DEBUG ("handler_value : %d, returning FALSE", handler_value);
g_value_set_enum (return_accu, handler_value);
return FALSE;
}
/* the accumulated value is initiated to 0 by GLib. if current handler value is
* bigger then use that instead
*
* FIXME: Should we prioritize the 2xx codes in a smarter way?
* Like, "201 Created" > "250 Low On Storage Space" > "200 OK"?
*/
if (handler_value > accumulated_value)
g_value_set_enum (return_accu, handler_value);
return TRUE;
}
/* The cleanup_transports function is called from handle_teardown_request() to
* remove any stream transports from the newly closed session that were added to
* priv->transports in handle_setup_request(). This is done to avoid forwarding
* data from the client on a channel that we just closed.
*/
static void
cleanup_transports (GstRTSPClient * client, GPtrArray * transports)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPStreamTransport *stream_transport;
const GstRTSPTransport *rtsp_transport;
guint i;
GST_LOG_OBJECT (client, "potentially removing %u transports",
transports->len);
for (i = 0; i < transports->len; i++) {
stream_transport = g_ptr_array_index (transports, i);
if (stream_transport == NULL) {
GST_LOG_OBJECT (client, "stream transport %u is NULL, continue", i);
continue;
}
rtsp_transport = gst_rtsp_stream_transport_get_transport (stream_transport);
if (rtsp_transport == NULL) {
GST_LOG_OBJECT (client, "RTSP transport %u is NULL, continue", i);
continue;
}
/* priv->transport only stores transports where RTP is tunneled over RTSP */
if (rtsp_transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
if (!g_hash_table_remove (priv->transports,
GINT_TO_POINTER (rtsp_transport->interleaved.min))) {
GST_WARNING_OBJECT (client,
"failed removing transport with key '%d' from priv->transports",
rtsp_transport->interleaved.min);
}
if (!g_hash_table_remove (priv->transports,
GINT_TO_POINTER (rtsp_transport->interleaved.max))) {
GST_WARNING_OBJECT (client,
"failed removing transport with key '%d' from priv->transports",
rtsp_transport->interleaved.max);
}
} else {
GST_LOG_OBJECT (client, "transport %u not RTP/RTSP, skip it", i);
}
}
}
static gboolean
handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPClientClass *klass;
GstRTSPSession *session;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStatusCode code;
gchar *path;
gint matched;
gboolean keep_session;
GstRTSPStatusCode sig_result;
GPtrArray *session_media_transports;
if (!ctx->session)
goto no_session;
session = ctx->session;
if (!ctx->uri)
goto no_uri;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
path = klass->make_path_from_uri (client, ctx->uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_dup_media (session, path, &matched);
if (!sessmedia)
goto not_found;
/* only aggregate control for now.. */
if (path[matched] != '\0')
goto no_aggregate;
g_free (path);
ctx->sessmedia = sessmedia;
media = gst_rtsp_session_media_get_media (sessmedia);
g_object_ref (media);
gst_rtsp_media_lock (media);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_TEARDOWN_REQUEST],
0, ctx, &sig_result);
if (sig_result != GST_RTSP_STS_OK) {
goto sig_failed;
}
/* get a reference to the transports in the session media so we can clean up
* our priv->transports before returning */
session_media_transports = gst_rtsp_session_media_get_transports (sessmedia);
/* we emit the signal before closing the connection */
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
0, ctx);
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
/* unmanage the media in the session, returns false if all media session
* are torn down. */
keep_session = gst_rtsp_session_release_media (session, sessmedia);
g_object_unref (sessmedia);
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
send_message (client, ctx, ctx->response, TRUE);
if (!keep_session) {
/* remove the session */
gst_rtsp_session_pool_remove (priv->session_pool, session);
}
gst_rtsp_media_unlock (media);
g_object_unref (media);
/* remove all transports that were present in the session media which we just
* unmanaged from the priv->transports array, so we do not try to handle data
* on channels that were just closed */
cleanup_transports (client, session_media_transports);
g_ptr_array_unref (session_media_transports);
return TRUE;
/* ERRORS */
no_session:
{
GST_ERROR ("client %p: no session", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri supplied", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: no media for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
g_free (path);
return FALSE;
}
no_aggregate:
{
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
g_free (path);
g_object_unref (sessmedia);
return FALSE;
}
sig_failed:
{
GST_ERROR ("client %p: pre signal returned error: %s", client,
gst_rtsp_status_as_text (sig_result));
send_generic_response (client, sig_result, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
g_object_unref (sessmedia);
return FALSE;
}
}
static GstRTSPResult
default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
res = gst_rtsp_params_set (client, ctx);
return res;
}
static GstRTSPResult
default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
res = gst_rtsp_params_get (client, ctx);
return res;
}
static gboolean
handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
guint8 *data;
guint size;
GstRTSPStatusCode sig_result;
g_signal_emit (client,
gst_rtsp_client_signals[SIGNAL_PRE_GET_PARAMETER_REQUEST], 0, ctx,
&sig_result);
if (sig_result != GST_RTSP_STS_OK) {
goto sig_failed;
}
res = gst_rtsp_message_get_body (ctx->request, &data, &size);
if (res != GST_RTSP_OK)
goto bad_request;
if (size == 0 || !data || strlen ((char *) data) == 0) {
if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0) {
GST_ERROR_OBJECT (client, "Using PLAY request for keep-alive is forbidden"
" in RTSP 2.0");
goto bad_request;
}
/* no body (or only '\0'), keep-alive request */
send_generic_response (client, GST_RTSP_STS_OK, ctx);
} else {
/* there is a body, handle the params */
res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
if (res != GST_RTSP_OK)
goto bad_request;
send_message (client, ctx, ctx->response, FALSE);
}
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
0, ctx);
return TRUE;
/* ERRORS */
sig_failed:
{
GST_ERROR ("client %p: pre signal returned error: %s", client,
gst_rtsp_status_as_text (sig_result));
send_generic_response (client, sig_result, ctx);
return FALSE;
}
bad_request:
{
GST_ERROR ("client %p: bad request", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
}
static gboolean
handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
guint8 *data;
guint size;
GstRTSPStatusCode sig_result;
g_signal_emit (client,
gst_rtsp_client_signals[SIGNAL_PRE_SET_PARAMETER_REQUEST], 0, ctx,
&sig_result);
if (sig_result != GST_RTSP_STS_OK) {
goto sig_failed;
}
res = gst_rtsp_message_get_body (ctx->request, &data, &size);
if (res != GST_RTSP_OK)
goto bad_request;
if (size == 0 || !data || strlen ((char *) data) == 0) {
/* no body (or only '\0'), keep-alive request */
send_generic_response (client, GST_RTSP_STS_OK, ctx);
} else {
/* there is a body, handle the params */
res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
if (res != GST_RTSP_OK)
goto bad_request;
send_message (client, ctx, ctx->response, FALSE);
}
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
0, ctx);
return TRUE;
/* ERRORS */
sig_failed:
{
GST_ERROR ("client %p: pre signal returned error: %s", client,
gst_rtsp_status_as_text (sig_result));
send_generic_response (client, sig_result, ctx);
return FALSE;
}
bad_request:
{
GST_ERROR ("client %p: bad request", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
}
static gboolean
handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
GstRTSPClientClass *klass;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStatusCode code;
GstRTSPState rtspstate;
gchar *path;
gint matched;
GstRTSPStatusCode sig_result;
guint i, n;
if (!(session = ctx->session))
goto no_session;
if (!ctx->uri)
goto no_uri;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
path = klass->make_path_from_uri (client, ctx->uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_dup_media (session, path, &matched);
if (!sessmedia)
goto not_found;
if (path[matched] != '\0')
goto no_aggregate;
g_free (path);
media = gst_rtsp_session_media_get_media (sessmedia);
g_object_ref (media);
gst_rtsp_media_lock (media);
n = gst_rtsp_media_n_streams (media);
for (i = 0; i < n; i++) {
GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET)
goto not_supported;
}
ctx->sessmedia = sessmedia;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_PAUSE_REQUEST], 0,
ctx, &sig_result);
if (sig_result != GST_RTSP_STS_OK) {
goto sig_failed;
}
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
/* the session state must be playing or recording */
if (rtspstate != GST_RTSP_STATE_PLAYING &&
rtspstate != GST_RTSP_STATE_RECORDING)
goto invalid_state;
/* then pause sending */
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
send_message (client, ctx, ctx->response, FALSE);
/* the state is now READY */
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
g_object_unref (sessmedia);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
return TRUE;
/* ERRORS */
no_session:
{
GST_ERROR ("client %p: no session", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri supplied", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: no media for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
g_free (path);
return FALSE;
}
no_aggregate:
{
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
g_object_unref (sessmedia);
g_free (path);
return FALSE;
}
sig_failed:
{
GST_ERROR ("client %p: pre signal returned error: %s", client,
gst_rtsp_status_as_text (sig_result));
send_generic_response (client, sig_result, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (sessmedia);
g_object_unref (media);
return FALSE;
}
invalid_state:
{
GST_ERROR ("client %p: not PLAYING or RECORDING", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
ctx);
gst_rtsp_media_unlock (media);
g_object_unref (sessmedia);
g_object_unref (media);
return FALSE;
}
not_supported:
{
GST_ERROR ("client %p: pausing not supported", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (sessmedia);
g_object_unref (media);
return FALSE;
}
}
/* convert @url and @path to a URL used as a content base for the factory
* located at @path */
static gchar *
make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
{
GstRTSPUrl tmp;
gchar *result;
const gchar *trail;
/* check for trailing '/' and append one */
trail = (path[strlen (path) - 1] != '/' ? "/" : "");
tmp = *url;
tmp.user = NULL;
tmp.passwd = NULL;
tmp.abspath = g_strdup_printf ("%s%s", path, trail);
tmp.query = NULL;
result = gst_rtsp_url_get_request_uri (&tmp);
g_free (tmp.abspath);
return result;
}
/* Check if the given header of type double is present and, if so,
* put it's value in the supplied variable.
*/
static GstRTSPStatusCode
parse_header_value_double (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPHeaderField header, gboolean * present, gdouble * value)
{
GstRTSPResult res;
gchar *str;
gchar *end;
res = gst_rtsp_message_get_header (ctx->request, header, &str, 0);
if (res == GST_RTSP_OK) {
*value = g_ascii_strtod (str, &end);
if (end == str)
goto parse_header_failed;
GST_DEBUG ("client %p: got '%s', value %f", client,
gst_rtsp_header_as_text (header), *value);
*present = TRUE;
} else {
*present = FALSE;
}
return GST_RTSP_STS_OK;
parse_header_failed:
{
GST_ERROR ("client %p: failed parsing '%s' header", client,
gst_rtsp_header_as_text (header));
return GST_RTSP_STS_BAD_REQUEST;
}
}
/* Parse scale and speed headers, if present, and set the rate to
* (rate * scale * speed) */
static GstRTSPStatusCode
parse_scale_and_speed (GstRTSPClient * client, GstRTSPContext * ctx,
gboolean * scale_present, gboolean * speed_present, gdouble * rate,
GstSeekFlags * flags)
{
gdouble scale = 1.0;
gdouble speed = 1.0;
GstRTSPStatusCode status;
GST_DEBUG ("got rate %f", *rate);
status = parse_header_value_double (client, ctx, GST_RTSP_HDR_SCALE,
scale_present, &scale);
if (status != GST_RTSP_STS_OK)
return status;
if (*scale_present) {
GST_DEBUG ("got Scale %f", scale);
if (scale == 0)
goto bad_scale_value;
*rate *= scale;
if (ABS (scale) != 1.0)
*flags |= GST_SEEK_FLAG_TRICKMODE;
}
GST_DEBUG ("rate after parsing Scale %f", *rate);
status = parse_header_value_double (client, ctx, GST_RTSP_HDR_SPEED,
speed_present, &speed);
if (status != GST_RTSP_STS_OK)
return status;
if (*speed_present) {
GST_DEBUG ("got Speed %f", speed);
if (speed <= 0)
goto bad_speed_value;
*rate *= speed;
}
GST_DEBUG ("rate after parsing Speed %f", *rate);
return status;
bad_scale_value:
{
GST_ERROR ("client %p: bad 'Scale' header value (%f)", client, scale);
return GST_RTSP_STS_BAD_REQUEST;
}
bad_speed_value:
{
GST_ERROR ("client %p: bad 'Speed' header value (%f)", client, speed);
return GST_RTSP_STS_BAD_REQUEST;
}
}
static GstRTSPStatusCode
setup_play_mode (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPRangeUnit * unit, gboolean * scale_present, gboolean * speed_present)
{
gchar *str;
GstRTSPResult res;
GstRTSPTimeRange *range = NULL;
gdouble rate = 1.0;
GstSeekFlags flags = GST_SEEK_FLAG_NONE;
GstRTSPClientClass *klass = GST_RTSP_CLIENT_GET_CLASS (client);
GstRTSPStatusCode rtsp_status_code;
GstClockTime trickmode_interval = 0;
gboolean enable_rate_control = TRUE;
/* parse the range header if we have one */
res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
if (res == GST_RTSP_OK) {
gchar *seek_style = NULL;
res = gst_rtsp_range_parse (str, &range);
if (res != GST_RTSP_OK)
goto parse_range_failed;
*unit = range->unit;
/* parse seek style header, if present */
res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_SEEK_STYLE,
&seek_style, 0);
if (res == GST_RTSP_OK) {
if (g_strcmp0 (seek_style, "RAP") == 0)
flags = GST_SEEK_FLAG_ACCURATE;
else if (g_strcmp0 (seek_style, "CoRAP") == 0)
flags = GST_SEEK_FLAG_KEY_UNIT;
else if (g_strcmp0 (seek_style, "First-Prior") == 0)
flags = GST_SEEK_FLAG_KEY_UNIT & GST_SEEK_FLAG_SNAP_BEFORE;
else if (g_strcmp0 (seek_style, "Next") == 0)
flags = GST_SEEK_FLAG_KEY_UNIT & GST_SEEK_FLAG_SNAP_AFTER;
else
GST_FIXME_OBJECT (client, "Add support for seek style %s", seek_style);
} else if (range->min.type == GST_RTSP_TIME_END) {
flags = GST_SEEK_FLAG_ACCURATE;
} else {
flags = GST_SEEK_FLAG_KEY_UNIT;
}
if (seek_style)
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_SEEK_STYLE,
seek_style);
} else {
flags = GST_SEEK_FLAG_ACCURATE;
}
/* check for scale and/or speed headers
* we will set the seek rate to (speed * scale) and let the media decide
* the resulting scale and speed. in the response we will use rate and applied
* rate from the resulting segment as values for the speed and scale headers
* respectively */
rtsp_status_code = parse_scale_and_speed (client, ctx, scale_present,
speed_present, &rate, &flags);
if (rtsp_status_code != GST_RTSP_STS_OK)
goto scale_speed_failed;
/* give the application a chance to tweak range, flags, or rate */
if (klass->adjust_play_mode != NULL) {
rtsp_status_code =
klass->adjust_play_mode (client, ctx, &range, &flags, &rate,
&trickmode_interval, &enable_rate_control);
if (rtsp_status_code != GST_RTSP_STS_OK)
goto adjust_play_mode_failed;
}
gst_rtsp_media_set_rate_control (ctx->media, enable_rate_control);
/* now do the seek with the seek options */
gst_rtsp_media_seek_trickmode (ctx->media, range, flags, rate,
trickmode_interval);
if (range != NULL)
gst_rtsp_range_free (range);
if (gst_rtsp_media_get_status (ctx->media) == GST_RTSP_MEDIA_STATUS_ERROR)
goto seek_failed;
return GST_RTSP_STS_OK;
parse_range_failed:
{
GST_ERROR ("client %p: failed parsing range header", client);
return GST_RTSP_STS_BAD_REQUEST;
}
scale_speed_failed:
{
if (range != NULL)
gst_rtsp_range_free (range);
GST_ERROR ("client %p: failed parsing Scale or Speed headers", client);
return rtsp_status_code;
}
adjust_play_mode_failed:
{
GST_ERROR ("client %p: sub class returned bad code (%d)", client,
rtsp_status_code);
if (range != NULL)
gst_rtsp_range_free (range);
return rtsp_status_code;
}
seek_failed:
{
GST_ERROR ("client %p: seek failed", client);
return GST_RTSP_STS_SERVICE_UNAVAILABLE;
}
}
static gboolean
handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
GstRTSPClientClass *klass;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStatusCode code;
GstRTSPUrl *uri;
gchar *str;
GstRTSPState rtspstate;
GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
gchar *path, *rtpinfo = NULL;
gint matched;
GstRTSPStatusCode sig_result;
GPtrArray *transports;
gboolean scale_present;
gboolean speed_present;
gdouble rate;
gdouble applied_rate;
if (!(session = ctx->session))
goto no_session;
if (!(uri = ctx->uri))
goto no_uri;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
path = klass->make_path_from_uri (client, uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_dup_media (session, path, &matched);
if (!sessmedia)
goto not_found;
if (path[matched] != '\0')
goto no_aggregate;
g_free (path);
ctx->sessmedia = sessmedia;
ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
g_object_ref (media);
gst_rtsp_media_lock (media);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_PLAY_REQUEST], 0,
ctx, &sig_result);
if (sig_result != GST_RTSP_STS_OK) {
goto sig_failed;
}
if (!(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_PLAY))
goto unsupported_mode;
/* the session state must be playing or ready */
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state;
/* update the pipeline */
transports = gst_rtsp_session_media_get_transports (sessmedia);
if (!gst_rtsp_media_complete_pipeline (media, transports)) {
g_ptr_array_unref (transports);
goto pipeline_error;
}
g_ptr_array_unref (transports);
/* in play we first unsuspend, media could be suspended from SDP or PAUSED */
if (!gst_rtsp_media_unsuspend (media))
goto unsuspend_failed;
code = setup_play_mode (client, ctx, &unit, &scale_present, &speed_present);
if (code != GST_RTSP_STS_OK)
goto invalid_mode;
/* grab RTPInfo from the media now */
if (gst_rtsp_media_has_completed_sender (media) &&
!(rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia)))
goto rtp_info_error;
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
/* add the RTP-Info header */
if (rtpinfo)
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
rtpinfo);
/* add the range */
str = gst_rtsp_media_get_range_string (media, TRUE, unit);
if (str)
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
if (gst_rtsp_media_has_completed_sender (media)) {
/* the scale and speed headers must always be added if they were present in
* the request. however, even if they were not, we still add them if
* applied_rate or rate deviate from the "normal", i.e. 1.0 */
if (!gst_rtsp_media_get_rates (media, &rate, &applied_rate))
goto get_rates_error;
g_assert (rate != 0 && applied_rate != 0);
if (scale_present || applied_rate != 1.0)
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_SCALE,
g_strdup_printf ("%1.3f", applied_rate));
if (speed_present || rate != 1.0)
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_SPEED,
g_strdup_printf ("%1.3f", rate));
}
if (klass->adjust_play_response) {
code = klass->adjust_play_response (client, ctx);
if (code != GST_RTSP_STS_OK)
goto adjust_play_response_failed;
}
send_message (client, ctx, ctx->response, FALSE);
/* start playing after sending the response */
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
g_object_unref (sessmedia);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
return TRUE;
/* ERRORS */
no_session:
{
GST_ERROR ("client %p: no session", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri supplied", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: media not found", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_aggregate:
{
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
g_object_unref (sessmedia);
g_free (path);
return FALSE;
}
sig_failed:
{
GST_ERROR ("client %p: pre signal returned error: %s", client,
gst_rtsp_status_as_text (sig_result));
send_generic_response (client, sig_result, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
g_object_unref (sessmedia);
return FALSE;
}
invalid_state:
{
GST_ERROR ("client %p: not PLAYING or READY", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
g_object_unref (sessmedia);
return FALSE;
}
pipeline_error:
{
GST_ERROR ("client %p: failed to configure the pipeline", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
g_object_unref (sessmedia);
return FALSE;
}
unsuspend_failed:
{
GST_ERROR ("client %p: unsuspend failed", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
g_object_unref (sessmedia);
return FALSE;
}
invalid_mode:
{
GST_ERROR ("client %p: seek failed", client);
send_generic_response (client, code, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
g_object_unref (sessmedia);
return FALSE;
}
unsupported_mode:
{
GST_ERROR ("client %p: media does not support PLAY", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
g_object_unref (sessmedia);
return FALSE;
}
get_rates_error:
{
GST_ERROR ("client %p: failed obtaining rate and applied_rate", client);
send_generic_response (client, GST_RTSP_STS_INTERNAL_SERVER_ERROR, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
g_object_unref (sessmedia);
return FALSE;
}
adjust_play_response_failed:
{
GST_ERROR ("client %p: failed to adjust play response", client);
send_generic_response (client, code, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
g_object_unref (sessmedia);
return FALSE;
}
rtp_info_error:
{
GST_ERROR ("client %p: failed to add RTP-Info", client);
send_generic_response (client, GST_RTSP_STS_INTERNAL_SERVER_ERROR, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
g_object_unref (sessmedia);
return FALSE;
}
}
static void
do_keepalive (GstRTSPSession * session)
{
GST_INFO ("keep session %p alive", session);
gst_rtsp_session_touch (session);
}
/* parse @transport and return a valid transport in @tr. only transports
* supported by @stream are returned. Returns FALSE if no valid transport
* was found. */
static gboolean
parse_transport (const char *transport, GstRTSPStream * stream,
GstRTSPTransport * tr)
{
gint i;
gboolean res;
gchar **transports;
res = FALSE;
gst_rtsp_transport_init (tr);
GST_DEBUG ("parsing transports %s", transport);
transports = g_strsplit (transport, ",", 0);
/* loop through the transports, try to parse */
for (i = 0; transports[i]; i++) {
g_strstrip (transports[i]);
res = gst_rtsp_transport_parse (transports[i], tr);
if (res != GST_RTSP_OK) {
/* no valid transport, search some more */
GST_WARNING ("could not parse transport %s", transports[i]);
goto next;
}
/* we have a transport, see if it's supported */
if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
GST_WARNING ("unsupported transport %s", transports[i]);
goto next;
}
/* we have a valid transport */
GST_INFO ("found valid transport %s", transports[i]);
res = TRUE;
break;
next:
gst_rtsp_transport_init (tr);
}
g_strfreev (transports);
return res;
}
static gboolean
default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
GstRTSPStream * stream, GstRTSPContext * ctx)
{
GstRTSPMessage *request = ctx->request;
gchar *blocksize_str;
if (!gst_rtsp_stream_is_sender (stream))
return TRUE;
if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
&blocksize_str, 0) == GST_RTSP_OK) {
guint64 blocksize;
gchar *end;
blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
if (end == blocksize_str)
goto parse_failed;
/* we don't want to change the mtu when this media
* can be shared because it impacts other clients */
if (gst_rtsp_media_is_shared (media))
goto done;
if (blocksize > G_MAXUINT)
blocksize = G_MAXUINT;
gst_rtsp_stream_set_mtu (stream, blocksize);
}
done:
return TRUE;
/* ERRORS */
parse_failed:
{
GST_ERROR_OBJECT (client, "failed to parse blocksize");
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
}
static gboolean
default_configure_client_transport (GstRTSPClient * client,
GstRTSPContext * ctx, GstRTSPTransport * ct)
{
GstRTSPClientPrivate *priv = client->priv;
/* we have a valid transport now, set the destination of the client. */
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST ||
ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP) {
/* allocate UDP ports */
GSocketFamily family;
gboolean use_client_settings = FALSE;
family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
if ((ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) &&
gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS) &&
(ct->destination != NULL)) {
if (!gst_rtsp_stream_verify_mcast_ttl (ctx->stream, ct->ttl))
goto error_ttl;
use_client_settings = TRUE;
}
/* We need to allocate the sockets for both families before starting
* multiudpsink, otherwise multiudpsink won't accept new clients with
* a different family.
*/
/* FIXME: could be more adequately solved by making it possible
* to set a socket on multiudpsink after it has already been started */
if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream,
G_SOCKET_FAMILY_IPV4, ct, use_client_settings)
&& family == G_SOCKET_FAMILY_IPV4)
goto error_allocating_ports;
if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream,
G_SOCKET_FAMILY_IPV6, ct, use_client_settings)
&& family == G_SOCKET_FAMILY_IPV6)
goto error_allocating_ports;
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
if (use_client_settings) {
/* FIXME: the address has been successfully allocated, however, in
* the use_client_settings case we need to verify that the allocated
* address is the one requested by the client and if this address is
* an allowed destination. Verifying this via the address pool in not
* the proper way as the address pool should only be used for choosing
* the server-selected address/port pairs. */
GSocket *rtp_socket;
guint ttl;
rtp_socket =
gst_rtsp_stream_get_rtp_multicast_socket (ctx->stream, family);
if (rtp_socket == NULL)
goto no_socket;
ttl = g_socket_get_multicast_ttl (rtp_socket);
g_object_unref (rtp_socket);
if (ct->ttl < ttl) {
/* use the maximum ttl that is requested by multicast clients */
GST_DEBUG ("requested ttl %u, but keeping ttl %u", ct->ttl, ttl);
ct->ttl = ttl;
}
} else {
GstRTSPAddress *addr = NULL;
g_free (ct->destination);
addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
if (addr == NULL)
goto no_address;
ct->destination = g_strdup (addr->address);
ct->port.min = addr->port;
ct->port.max = addr->port + addr->n_ports - 1;
ct->ttl = addr->ttl;
gst_rtsp_address_free (addr);
}
if (!gst_rtsp_stream_add_multicast_client_address (ctx->stream,
ct->destination, ct->port.min, ct->port.max, family))
goto error_mcast_transport;
} else {
GstRTSPUrl *url;
url = gst_rtsp_connection_get_url (priv->connection);
g_free (ct->destination);
ct->destination = g_strdup (url->host);
}
} else {
GstRTSPUrl *url;
url = gst_rtsp_connection_get_url (priv->connection);
g_free (ct->destination);
ct->destination = g_strdup (url->host);
if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
GSocket *sock;
GSocketAddress *addr;
sock = gst_rtsp_connection_get_read_socket (priv->connection);
if ((addr = g_socket_get_remote_address (sock, NULL))) {
/* our read port is the sender port of client */
ct->client_port.min =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
g_object_unref (addr);
}
if ((addr = g_socket_get_local_address (sock, NULL))) {
ct->server_port.max =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
g_object_unref (addr);
}
sock = gst_rtsp_connection_get_write_socket (priv->connection);
if ((addr = g_socket_get_remote_address (sock, NULL))) {
/* our write port is the receiver port of client */
ct->client_port.max =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
g_object_unref (addr);
}
if ((addr = g_socket_get_local_address (sock, NULL))) {
ct->server_port.min =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
g_object_unref (addr);
}
/* check if the client selected channels for TCP */
if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
&ct->interleaved);
}
/* alloc new channels if they are already taken */
while (g_hash_table_contains (priv->transports,
GINT_TO_POINTER (ct->interleaved.min))
|| g_hash_table_contains (priv->transports,
GINT_TO_POINTER (ct->interleaved.max))) {
gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
&ct->interleaved);
if (ct->interleaved.max > 255)
goto error_allocating_channels;
}
}
}
return TRUE;
/* ERRORS */
error_ttl:
{
GST_ERROR_OBJECT (client,
"Failed to allocate UDP ports: invalid ttl value");
return FALSE;
}
error_allocating_ports:
{
GST_ERROR_OBJECT (client, "Failed to allocate UDP ports");
return FALSE;
}
no_address:
{
GST_ERROR_OBJECT (client, "Failed to acquire address for stream");
return FALSE;
}
no_socket:
{
GST_ERROR_OBJECT (client, "Failed to get UDP socket");
return FALSE;
}
error_mcast_transport:
{
GST_ERROR_OBJECT (client, "Failed to add multicast client transport");
return FALSE;
}
error_allocating_channels:
{
GST_ERROR_OBJECT (client, "Failed to allocate interleaved channels");
return FALSE;
}
}
static GstRTSPTransport *
make_server_transport (GstRTSPClient * client, GstRTSPMedia * media,
GstRTSPContext * ctx, GstRTSPTransport * ct)
{
GstRTSPTransport *st;
GInetAddress *addr;
GSocketFamily family;
/* prepare the server transport */
gst_rtsp_transport_new (&st);
st->trans = ct->trans;
st->profile = ct->profile;
st->lower_transport = ct->lower_transport;
st->mode_play = ct->mode_play;
st->mode_record = ct->mode_record;
addr = g_inet_address_new_from_string (ct->destination);
if (!addr) {
GST_ERROR ("failed to get inet addr from client destination");
family = G_SOCKET_FAMILY_IPV4;
} else {
family = g_inet_address_get_family (addr);
g_object_unref (addr);
addr = NULL;
}
switch (st->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
st->client_port = ct->client_port;
gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
st->port = ct->port;
st->destination = g_strdup (ct->destination);
st->ttl = ct->ttl;
break;
case GST_RTSP_LOWER_TRANS_TCP:
st->interleaved = ct->interleaved;
st->client_port = ct->client_port;
st->server_port = ct->server_port;
default:
break;
}
if ((gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_PLAY))
gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
return st;
}
static void
rtsp_ctrl_timeout_remove_unlocked (GstRTSPClientPrivate * priv)
{
if (priv->rtsp_ctrl_timeout != NULL) {
GST_DEBUG ("rtsp control session removed timeout %p.",
priv->rtsp_ctrl_timeout);
g_source_destroy (priv->rtsp_ctrl_timeout);
g_source_unref (priv->rtsp_ctrl_timeout);
priv->rtsp_ctrl_timeout = NULL;
priv->rtsp_ctrl_timeout_cnt = 0;
}
}
static void
rtsp_ctrl_timeout_remove (GstRTSPClient * client)
{
g_mutex_lock (&client->priv->lock);
rtsp_ctrl_timeout_remove_unlocked (client->priv);
g_mutex_unlock (&client->priv->lock);
}
static void
rtsp_ctrl_timeout_destroy_notify (gpointer user_data)
{
GWeakRef *client_weak_ref = (GWeakRef *) user_data;
g_weak_ref_clear (client_weak_ref);
g_free (client_weak_ref);
}
static gboolean
rtsp_ctrl_timeout_cb (gpointer user_data)
{
gboolean res = G_SOURCE_CONTINUE;
GstRTSPClientPrivate *priv;
GWeakRef *client_weak_ref = (GWeakRef *) user_data;
GstRTSPClient *client = (GstRTSPClient *) g_weak_ref_get (client_weak_ref);
if (client == NULL) {
return G_SOURCE_REMOVE;
}
priv = client->priv;
g_mutex_lock (&priv->lock);
priv->rtsp_ctrl_timeout_cnt += RTSP_CTRL_CB_INTERVAL;
if ((priv->rtsp_ctrl_timeout_cnt > RTSP_CTRL_TIMEOUT_VALUE)
|| (priv->had_session
&& priv->rtsp_ctrl_timeout_cnt > priv->post_session_timeout)) {
GST_DEBUG ("rtsp control session timeout %p expired, closing client.",
priv->rtsp_ctrl_timeout);
rtsp_ctrl_timeout_remove_unlocked (client->priv);
res = G_SOURCE_REMOVE;
}
g_mutex_unlock (&priv->lock);
if (res == G_SOURCE_REMOVE) {
gst_rtsp_client_close (client);
}
g_object_unref (client);
return res;
}
static gchar *
stream_make_keymgmt (GstRTSPClient * client, const gchar * location,
GstRTSPStream * stream)
{
gchar *base64, *result = NULL;
GstMIKEYMessage *mikey_msg;
GstCaps *srtcpparams;
GstElement *rtcp_encoder;
gint srtcp_cipher, srtp_cipher;
gint srtcp_auth, srtp_auth;
GstBuffer *key;
GType ciphertype, authtype;
GEnumClass *cipher_enum, *auth_enum;
GEnumValue *srtcp_cipher_value, *srtp_cipher_value, *srtcp_auth_value,
*srtp_auth_value;
rtcp_encoder = gst_rtsp_stream_get_srtp_encoder (stream);
if (!rtcp_encoder)
goto done;
ciphertype = g_type_from_name ("GstSrtpCipherType");
authtype = g_type_from_name ("GstSrtpAuthType");
cipher_enum = g_type_class_ref (ciphertype);
auth_enum = g_type_class_ref (authtype);
/* We need to bring the encoder to READY so that it generates its key */
gst_element_set_state (rtcp_encoder, GST_STATE_READY);
g_object_get (rtcp_encoder, "rtcp-cipher", &srtcp_cipher, "rtcp-auth",
&srtcp_auth, "rtp-cipher", &srtp_cipher, "rtp-auth", &srtp_auth, "key",
&key, NULL);
g_object_unref (rtcp_encoder);
srtcp_cipher_value = g_enum_get_value (cipher_enum, srtcp_cipher);
srtp_cipher_value = g_enum_get_value (cipher_enum, srtp_cipher);
srtcp_auth_value = g_enum_get_value (auth_enum, srtcp_auth);
srtp_auth_value = g_enum_get_value (auth_enum, srtp_auth);
g_type_class_unref (cipher_enum);
g_type_class_unref (auth_enum);
srtcpparams = gst_caps_new_simple ("application/x-srtcp",
"srtcp-cipher", G_TYPE_STRING, srtcp_cipher_value->value_nick,
"srtcp-auth", G_TYPE_STRING, srtcp_auth_value->value_nick,
"srtp-cipher", G_TYPE_STRING, srtp_cipher_value->value_nick,
"srtp-auth", G_TYPE_STRING, srtp_auth_value->value_nick,
"srtp-key", GST_TYPE_BUFFER, key, NULL);
mikey_msg = gst_mikey_message_new_from_caps (srtcpparams);
if (mikey_msg) {
guint send_ssrc;
gst_rtsp_stream_get_ssrc (stream, &send_ssrc);
gst_mikey_message_add_cs_srtp (mikey_msg, 0, send_ssrc, 0);
base64 = gst_mikey_message_base64_encode (mikey_msg);
gst_mikey_message_unref (mikey_msg);
if (base64) {
result = gst_sdp_make_keymgmt (location, base64);
g_free (base64);
}
}
done:
return result;
}
static gboolean
handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
GstRTSPUrl *uri;
gchar *transport, *keymgmt;
GstRTSPTransport *ct, *st;
GstRTSPStatusCode code;
GstRTSPSession *session;
GstRTSPStreamTransport *trans;
gchar *trans_str;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStream *stream;
GstRTSPState rtspstate;
GstRTSPClientClass *klass;
gchar *path, *control = NULL;
gint matched;
gboolean new_session = FALSE;
GstRTSPStatusCode sig_result;
gchar *pipelined_request_id = NULL, *accept_range = NULL;
if (!ctx->uri)
goto no_uri;
uri = ctx->uri;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
path = klass->make_path_from_uri (client, uri);
/* parse the transport */
res =
gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
&transport, 0);
if (res != GST_RTSP_OK)
goto no_transport;
/* Handle Pipelined-requests if using >= 2.0 */
if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0)
gst_rtsp_message_get_header (ctx->request,
GST_RTSP_HDR_PIPELINED_REQUESTS, &pipelined_request_id, 0);
/* we create the session after parsing stuff so that we don't make
* a session for malformed requests */
if (priv->session_pool == NULL)
goto no_pool;
session = ctx->session;
if (session) {
g_object_ref (session);
/* get a handle to the configuration of the media in the session, this can
* return NULL if this is a new url to manage in this session. */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
} else {
/* we need a new media configuration in this session */
sessmedia = NULL;
}
/* we have no session media, find one and manage it */
if (sessmedia == NULL) {
/* get a handle to the configuration of the media in the session */
media = find_media (client, ctx, path, &matched);
/* need to suspend the media, if the protocol has changed */
if (media != NULL) {
gst_rtsp_media_lock (media);
gst_rtsp_media_suspend (media);
}
} else {
if ((media = gst_rtsp_session_media_get_media (sessmedia))) {
g_object_ref (media);
gst_rtsp_media_lock (media);
} else {
goto media_not_found;
}
}
/* no media, not found then */
if (media == NULL)
goto media_not_found_no_reply;
if (path[matched] == '\0') {
if (gst_rtsp_media_n_streams (media) == 1) {
stream = gst_rtsp_media_get_stream (media, 0);
} else {
goto control_not_found;
}
} else {
/* path is what matched. */
gchar *newpath = g_strndup (path, matched);
/* control is remainder */
if (matched == 1 && path[0] == '/')
control = g_strdup (&path[1]);
else
control = g_strdup (&path[matched + 1]);
g_free (path);
path = newpath;
/* find the stream now using the control part */
stream = gst_rtsp_media_find_stream (media, control);
}
if (stream == NULL)
goto stream_not_found;
/* now we have a uri identifying a valid media and stream */
ctx->stream = stream;
ctx->media = media;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_SETUP_REQUEST], 0,
ctx, &sig_result);
if (sig_result != GST_RTSP_STS_OK) {
goto sig_failed;
}
if (session == NULL) {
/* create a session if this fails we probably reached our session limit or
* something. */
if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
goto service_unavailable;
/* Pipelined requests should be cleared between sessions */
g_hash_table_remove_all (priv->pipelined_requests);
/* make sure this client is closed when the session is closed */
client_watch_session (client, session);
new_session = TRUE;
/* signal new session */
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
session);
ctx->session = session;
}
if (pipelined_request_id) {
g_hash_table_insert (client->priv->pipelined_requests,
g_strdup (pipelined_request_id),
g_strdup (gst_rtsp_session_get_sessionid (session)));
}
/* Remember that we had at least one session in the past */
priv->had_session = TRUE;
rtsp_ctrl_timeout_remove (client);
if (!klass->configure_client_media (client, media, stream, ctx))
goto configure_media_failed_no_reply;
gst_rtsp_transport_new (&ct);
/* parse and find a usable supported transport */
if (!parse_transport (transport, stream, ct))
goto unsupported_transports;
if ((ct->mode_play
&& !(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_PLAY)) || (ct->mode_record
&& !(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_RECORD)))
goto unsupported_mode;
/* parse the keymgmt */
if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
&keymgmt, 0) == GST_RTSP_OK) {
if (!gst_rtsp_stream_handle_keymgmt (ctx->stream, keymgmt))
goto keymgmt_error;
}
if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT_RANGES,
&accept_range, 0) == GST_RTSP_OK) {
GEnumValue *runit = NULL;
gint i;
gchar **valid_ranges;
GEnumClass *runit_class = g_type_class_ref (GST_TYPE_RTSP_RANGE_UNIT);
gst_rtsp_message_dump (ctx->request);
valid_ranges = g_strsplit (accept_range, ",", -1);
for (i = 0; valid_ranges[i]; i++) {
gchar *range = valid_ranges[i];
while (*range == ' ')
range++;
runit = g_enum_get_value_by_nick (runit_class, range);
if (runit)
break;
}
g_strfreev (valid_ranges);
g_type_class_unref (runit_class);
if (!runit)
goto unsupported_range_unit;
}
if (sessmedia == NULL) {
/* manage the media in our session now, if not done already */
sessmedia =
gst_rtsp_session_manage_media (session, path, g_object_ref (media));
/* if we stil have no media, error */
if (sessmedia == NULL)
goto sessmedia_unavailable;
/* don't cache media anymore */
clean_cached_media (client, FALSE);
}
ctx->sessmedia = sessmedia;
/* update the client transport */
if (!klass->configure_client_transport (client, ctx, ct))
goto unsupported_client_transport;
/* set in the session media transport */
trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
ctx->trans = trans;
/* configure the url used to set this transport, this we will use when
* generating the response for the PLAY request */
gst_rtsp_stream_transport_set_url (trans, uri);
/* configure keepalive for this transport */
gst_rtsp_stream_transport_set_keepalive (trans,
(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* our callbacks to send data on this TCP connection */
gst_rtsp_stream_transport_set_callbacks (trans,
(GstRTSPSendFunc) do_send_data,
(GstRTSPSendFunc) do_send_data, client, NULL);
gst_rtsp_stream_transport_set_list_callbacks (trans,
(GstRTSPSendListFunc) do_send_data_list,
(GstRTSPSendListFunc) do_send_data_list, client, NULL);
gst_rtsp_stream_transport_set_back_pressure_callback (trans,
(GstRTSPBackPressureFunc) do_check_back_pressure, client, NULL);
g_hash_table_insert (priv->transports,
GINT_TO_POINTER (ct->interleaved.min), trans);
g_object_ref (trans);
g_hash_table_insert (priv->transports,
GINT_TO_POINTER (ct->interleaved.max), trans);
g_object_ref (trans);
add_data_seq (client, ct->interleaved.min);
add_data_seq (client, ct->interleaved.max);
}
/* create and serialize the server transport */
st = make_server_transport (client, media, ctx, ct);
trans_str = gst_rtsp_transport_as_text (st);
gst_rtsp_transport_free (st);
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
trans_str);
g_free (trans_str);
if (pipelined_request_id)
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PIPELINED_REQUESTS,
pipelined_request_id);
if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0) {
GstClockTimeDiff seekable = gst_rtsp_media_seekable (media);
GString *media_properties = g_string_new (NULL);
if (seekable == -1)
g_string_append (media_properties,
"No-Seeking,Time-Progressing,Time-Duration=0.0");
else if (seekable == 0)
g_string_append (media_properties, "Beginning-Only");
else if (seekable == G_MAXINT64)
g_string_append (media_properties, "Random-Access");
else
g_string_append_printf (media_properties,
"Random-Access=%f, Unlimited, Immutable",
(gdouble) seekable / GST_SECOND);
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_MEDIA_PROPERTIES,
media_properties->str);
g_string_free (media_properties, TRUE);
/* TODO Check how Accept-Ranges should be filled */
gst_rtsp_message_add_header (ctx->request, GST_RTSP_HDR_ACCEPT_RANGES,
"npt, clock, smpte, clock");
}
send_message (client, ctx, ctx->response, FALSE);
/* update the state */
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
switch (rtspstate) {
case GST_RTSP_STATE_PLAYING:
case GST_RTSP_STATE_RECORDING:
case GST_RTSP_STATE_READY:
/* no state change */
break;
default:
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
break;
}
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
g_object_unref (session);
g_free (path);
g_free (control);
return TRUE;
/* ERRORS */
no_uri:
{
GST_ERROR ("client %p: no uri", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
no_transport:
{
GST_ERROR ("client %p: no transport", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
goto cleanup_path;
}
no_pool:
{
GST_ERROR ("client %p: no session pool configured", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
goto cleanup_path;
}
media_not_found_no_reply:
{
GST_ERROR ("client %p: media '%s' not found", client, path);
/* error reply is already sent */
goto cleanup_session;
}
media_not_found:
{
GST_ERROR ("client %p: media '%s' not found", client, path);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
goto cleanup_session;
}
control_not_found:
{
GST_ERROR ("client %p: no control in path '%s'", client, path);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
goto cleanup_session;
}
stream_not_found:
{
GST_ERROR ("client %p: stream '%s' not found", client,
GST_STR_NULL (control));
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
goto cleanup_session;
}
sig_failed:
{
GST_ERROR ("client %p: pre signal returned error: %s", client,
gst_rtsp_status_as_text (sig_result));
send_generic_response (client, sig_result, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
goto cleanup_path;
}
service_unavailable:
{
GST_ERROR ("client %p: can't create session", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
goto cleanup_session;
}
sessmedia_unavailable:
{
GST_ERROR ("client %p: can't create session media", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
goto cleanup_transport;
}
configure_media_failed_no_reply:
{
GST_ERROR ("client %p: configure_media failed", client);
gst_rtsp_media_unlock (media);
g_object_unref (media);
/* error reply is already sent */
goto cleanup_session;
}
unsupported_transports:
{
GST_ERROR ("client %p: unsupported transports", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
goto cleanup_transport;
}
unsupported_client_transport:
{
GST_ERROR ("client %p: unsupported client transport", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
goto cleanup_transport;
}
unsupported_mode:
{
GST_ERROR ("client %p: unsupported mode (media play: %d, media record: %d, "
"mode play: %d, mode record: %d)", client,
! !(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_PLAY),
! !(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_RECORD), ct->mode_play, ct->mode_record);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
goto cleanup_transport;
}
unsupported_range_unit:
{
GST_ERROR ("Client %p: does not support any range format we support",
client);
send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
goto cleanup_transport;
}
keymgmt_error:
{
GST_ERROR ("client %p: keymgmt error", client);
send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
goto cleanup_transport;
}
{
cleanup_transport:
gst_rtsp_transport_free (ct);
if (media) {
gst_rtsp_media_unlock (media);
g_object_unref (media);
}
cleanup_session:
if (new_session)
gst_rtsp_session_pool_remove (priv->session_pool, session);
if (session)
g_object_unref (session);
cleanup_path:
g_free (path);
g_free (control);
return FALSE;
}
}
static GstSDPMessage *
create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
{
GstRTSPClientPrivate *priv = client->priv;
GstSDPMessage *sdp;
GstSDPInfo info;
const gchar *proto;
guint64 session_id_tmp;
gchar session_id[21];
gst_sdp_message_new (&sdp);
/* some standard things first */
gst_sdp_message_set_version (sdp, "0");
if (priv->is_ipv6)
proto = "IP6";
else
proto = "IP4";
session_id_tmp = (((guint64) g_random_int ()) << 32) | g_random_int ();
g_snprintf (session_id, sizeof (session_id), "%" G_GUINT64_FORMAT,
session_id_tmp);
gst_sdp_message_set_origin (sdp, "-", session_id, "1", "IN", proto,
priv->server_ip);
gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
gst_sdp_message_set_information (sdp, "rtsp-server");
gst_sdp_message_add_time (sdp, "0", "0", NULL);
gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
gst_sdp_message_add_attribute (sdp, "type", "broadcast");
gst_sdp_message_add_attribute (sdp, "control", "*");
info.is_ipv6 = priv->is_ipv6;
info.server_ip = priv->server_ip;
/* create an SDP for the media object */
if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
goto no_sdp;
return sdp;
/* ERRORS */
no_sdp:
{
GST_ERROR ("client %p: could not create SDP", client);
gst_sdp_message_free (sdp);
return NULL;
}
}
/* for the describe we must generate an SDP */
static gboolean
handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
GstSDPMessage *sdp;
guint i;
gchar *path, *str;
GstRTSPMedia *media;
GstRTSPClientClass *klass;
GstRTSPStatusCode sig_result;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
if (!ctx->uri)
goto no_uri;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_DESCRIBE_REQUEST],
0, ctx, &sig_result);
if (sig_result != GST_RTSP_STS_OK) {
goto sig_failed;
}
/* check what kind of format is accepted, we don't really do anything with it
* and always return SDP for now. */
for (i = 0;; i++) {
gchar *accept;
res =
gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
&accept, i);
if (res == GST_RTSP_ENOTIMPL)
break;
if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
break;
}
if (!priv->mount_points)
goto no_mount_points;
if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
goto no_path;
/* find the media object for the uri */
if (!(media = find_media (client, ctx, path, NULL)))
goto no_media;
gst_rtsp_media_lock (media);
if (!(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_PLAY))
goto unsupported_mode;
/* create an SDP for the media object on this client */
if (!(sdp = klass->create_sdp (client, media)))
goto no_sdp;
/* we suspend after the describe */
gst_rtsp_media_suspend (media);
gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
"application/sdp");
/* content base for some clients that might screw up creating the setup uri */
str = make_base_url (client, ctx->uri, path);
g_free (path);
GST_INFO ("adding content-base: %s", str);
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
/* add SDP to the response body */
str = gst_sdp_message_as_text (sdp);
gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
gst_sdp_message_free (sdp);
send_message (client, ctx, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
0, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
return TRUE;
/* ERRORS */
sig_failed:
{
GST_ERROR ("client %p: pre signal returned error: %s", client,
gst_rtsp_status_as_text (sig_result));
send_generic_response (client, sig_result, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
no_mount_points:
{
GST_ERROR ("client %p: no mount points configured", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_path:
{
GST_ERROR ("client %p: can't find path for url", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_media:
{
GST_ERROR ("client %p: no media", client);
g_free (path);
/* error reply is already sent */
return FALSE;
}
unsupported_mode:
{
GST_ERROR ("client %p: media does not support DESCRIBE", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
g_free (path);
gst_rtsp_media_unlock (media);
g_object_unref (media);
return FALSE;
}
no_sdp:
{
GST_ERROR ("client %p: can't create SDP", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_free (path);
gst_rtsp_media_unlock (media);
g_object_unref (media);
return FALSE;
}
}
static gboolean
handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPMedia * media,
GstSDPMessage * sdp)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPThread *thread;
/* create an SDP for the media object */
if (!gst_rtsp_media_handle_sdp (media, sdp))
goto unhandled_sdp;
thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
GST_RTSP_THREAD_TYPE_MEDIA, ctx);
if (thread == NULL)
goto no_thread;
/* prepare the media */
if (!gst_rtsp_media_prepare (media, thread))
goto no_prepare;
return TRUE;
/* ERRORS */
unhandled_sdp:
{
GST_ERROR ("client %p: could not handle SDP", client);
return FALSE;
}
no_thread:
{
GST_ERROR ("client %p: can't create thread", client);
return FALSE;
}
no_prepare:
{
GST_ERROR ("client %p: can't prepare media", client);
return FALSE;
}
}
static gboolean
handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPClientClass *klass;
GstSDPResult sres;
GstSDPMessage *sdp;
GstRTSPMedia *media;
gchar *path, *cont = NULL;
guint8 *data;
guint size;
GstRTSPStatusCode sig_result;
guint i, n_streams;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
if (!ctx->uri)
goto no_uri;
if (!priv->mount_points)
goto no_mount_points;
/* check if reply is SDP */
gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_CONTENT_TYPE, &cont,
0);
/* could not be set but since the request returned OK, we assume it
* was SDP, else check it. */
if (cont) {
if (g_ascii_strcasecmp (cont, "application/sdp") != 0)
goto wrong_content_type;
}
/* get message body and parse as SDP */
gst_rtsp_message_get_body (ctx->request, &data, &size);
if (data == NULL || size == 0)
goto no_message;
GST_DEBUG ("client %p: parse SDP...", client);
gst_sdp_message_new (&sdp);
sres = gst_sdp_message_parse_buffer (data, size, sdp);
if (sres != GST_SDP_OK)
goto sdp_parse_failed;
if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
goto no_path;
/* find the media object for the uri */
if (!(media = find_media (client, ctx, path, NULL)))
goto no_media;
ctx->media = media;
gst_rtsp_media_lock (media);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_ANNOUNCE_REQUEST],
0, ctx, &sig_result);
if (sig_result != GST_RTSP_STS_OK) {
goto sig_failed;
}
if (!(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_RECORD))
goto unsupported_mode;
/* Tell client subclass about the media */
if (!klass->handle_sdp (client, ctx, media, sdp))
goto unhandled_sdp;
gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
n_streams = gst_rtsp_media_n_streams (media);
for (i = 0; i < n_streams; i++) {
GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
gchar *uri, *location, *keymgmt;
uri = gst_rtsp_url_get_request_uri (ctx->uri);
location = g_strdup_printf ("%s/stream=%d", uri, i);
keymgmt = stream_make_keymgmt (client, location, stream);
if (keymgmt)
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_KEYMGMT,
keymgmt);
g_free (location);
g_free (uri);
}
/* we suspend after the announce */
gst_rtsp_media_suspend (media);
send_message (client, ctx, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST],
0, ctx);
gst_sdp_message_free (sdp);
g_free (path);
gst_rtsp_media_unlock (media);
g_object_unref (media);
return TRUE;
no_uri:
{
GST_ERROR ("client %p: no uri", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
no_mount_points:
{
GST_ERROR ("client %p: no mount points configured", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_path:
{
GST_ERROR ("client %p: can't find path for url", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
gst_sdp_message_free (sdp);
return FALSE;
}
wrong_content_type:
{
GST_ERROR ("client %p: unknown content type", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
no_message:
{
GST_ERROR ("client %p: can't find SDP message", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
sdp_parse_failed:
{
GST_ERROR ("client %p: failed to parse SDP message", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
gst_sdp_message_free (sdp);
return FALSE;
}
no_media:
{
GST_ERROR ("client %p: no media", client);
g_free (path);
/* error reply is already sent */
gst_sdp_message_free (sdp);
return FALSE;
}
sig_failed:
{
GST_ERROR ("client %p: pre signal returned error: %s", client,
gst_rtsp_status_as_text (sig_result));
send_generic_response (client, sig_result, ctx);
gst_sdp_message_free (sdp);
gst_rtsp_media_unlock (media);
g_object_unref (media);
return FALSE;
}
unsupported_mode:
{
GST_ERROR ("client %p: media does not support ANNOUNCE", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
g_free (path);
gst_rtsp_media_unlock (media);
g_object_unref (media);
gst_sdp_message_free (sdp);
return FALSE;
}
unhandled_sdp:
{
GST_ERROR ("client %p: can't handle SDP", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_MEDIA_TYPE, ctx);
g_free (path);
gst_rtsp_media_unlock (media);
g_object_unref (media);
gst_sdp_message_free (sdp);
return FALSE;
}
}
static gboolean
handle_record_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
GstRTSPClientClass *klass;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPUrl *uri;
GstRTSPState rtspstate;
gchar *path;
gint matched;
GstRTSPStatusCode sig_result;
GPtrArray *transports;
if (!(session = ctx->session))
goto no_session;
if (!(uri = ctx->uri))
goto no_uri;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
path = klass->make_path_from_uri (client, uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
if (!sessmedia)
goto not_found;
if (path[matched] != '\0')
goto no_aggregate;
g_free (path);
ctx->sessmedia = sessmedia;
ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_RECORD_REQUEST], 0,
ctx, &sig_result);
if (sig_result != GST_RTSP_STS_OK) {
goto sig_failed;
}
if (!(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_RECORD))
goto unsupported_mode;
/* the session state must be playing or ready */
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state;
/* update the pipeline */
transports = gst_rtsp_session_media_get_transports (sessmedia);
if (!gst_rtsp_media_complete_pipeline (media, transports)) {
g_ptr_array_unref (transports);
goto pipeline_error;
}
g_ptr_array_unref (transports);
/* in record we first unsuspend, media could be suspended from SDP or PAUSED */
if (!gst_rtsp_media_unsuspend (media))
goto unsuspend_failed;
gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
send_message (client, ctx, ctx->response, FALSE);
/* start playing after sending the response */
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST], 0,
ctx);
return TRUE;
/* ERRORS */
no_session:
{
GST_ERROR ("client %p: no session", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri supplied", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: media not found", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_aggregate:
{
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
g_free (path);
return FALSE;
}
sig_failed:
{
GST_ERROR ("client %p: pre signal returned error: %s", client,
gst_rtsp_status_as_text (sig_result));
send_generic_response (client, sig_result, ctx);
return FALSE;
}
unsupported_mode:
{
GST_ERROR ("client %p: media does not support RECORD", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
return FALSE;
}
invalid_state:
{
GST_ERROR ("client %p: not PLAYING or READY", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
ctx);
return FALSE;
}
pipeline_error:
{
GST_ERROR ("client %p: failed to configure the pipeline", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
ctx);
return FALSE;
}
unsuspend_failed:
{
GST_ERROR ("client %p: unsuspend failed", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
return FALSE;
}
}
static gboolean
handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPVersion version)
{
GstRTSPMethod options;
gchar *str;
GstRTSPStatusCode sig_result;
options = GST_RTSP_DESCRIBE |
GST_RTSP_OPTIONS |
GST_RTSP_PAUSE |
GST_RTSP_PLAY |
GST_RTSP_SETUP |
GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
if (version < GST_RTSP_VERSION_2_0) {
options |= GST_RTSP_RECORD;
options |= GST_RTSP_ANNOUNCE;
}
str = gst_rtsp_options_as_text (options);
gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
g_free (str);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_OPTIONS_REQUEST], 0,
ctx, &sig_result);
if (sig_result != GST_RTSP_STS_OK) {
goto sig_failed;
}
send_message (client, ctx, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
0, ctx);
return TRUE;
/* ERRORS */
sig_failed:
{
GST_ERROR ("client %p: pre signal returned error: %s", client,
gst_rtsp_status_as_text (sig_result));
send_generic_response (client, sig_result, ctx);
gst_rtsp_message_free (ctx->response);
return FALSE;
}
}
/* remove duplicate and trailing '/' */
static void
sanitize_uri (GstRTSPUrl * uri)
{
gint i, len;
gchar *s, *d;
gboolean have_slash, prev_slash;
s = d = uri->abspath;
len = strlen (uri->abspath);
prev_slash = FALSE;
for (i = 0; i < len; i++) {
have_slash = s[i] == '/';
*d = s[i];
if (!have_slash || !prev_slash)
d++;
prev_slash = have_slash;
}
len = d - uri->abspath;
/* don't remove the first slash if that's the only thing left */
if (len > 1 && *(d - 1) == '/')
d--;
*d = '\0';
}
/* is called when the session is removed from its session pool. */
static void
client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
GSource *timer_src;
GST_INFO ("client %p: session %p removed", client, session);
g_mutex_lock (&priv->lock);
client_unwatch_session (client, session, NULL);
if (!priv->sessions && priv->rtsp_ctrl_timeout == NULL) {
if (priv->post_session_timeout > 0) {
GWeakRef *client_weak_ref = g_new (GWeakRef, 1);
timer_src = g_timeout_source_new_seconds (RTSP_CTRL_CB_INTERVAL);
g_weak_ref_init (client_weak_ref, client);
g_source_set_callback (timer_src, rtsp_ctrl_timeout_cb, client_weak_ref,
rtsp_ctrl_timeout_destroy_notify);
priv->rtsp_ctrl_timeout_cnt = 0;
g_source_attach (timer_src, priv->watch_context);
priv->rtsp_ctrl_timeout = timer_src;
GST_DEBUG ("rtsp control setting up connection timeout %p.",
priv->rtsp_ctrl_timeout);
g_mutex_unlock (&priv->lock);
} else if (priv->post_session_timeout == 0) {
g_mutex_unlock (&priv->lock);
gst_rtsp_client_close (client);
} else {
g_mutex_unlock (&priv->lock);
}
} else {
g_mutex_unlock (&priv->lock);
}
}
/* Check for Require headers. Returns TRUE if there are no Require headers,
* otherwise lets the application decide which headers are supported.
* By default all headers are unsupported.
* If there are unsupported options, FALSE will be returned together with
* a newly-allocated string of (comma-separated) unsupported options in
* the unsupported_reqs variable.
*
* There may be multiple Require headers, but we must send one single
* Unsupported header with all the unsupported options as response. If
* an incoming Require header contained a comma-separated list of options
* GstRtspConnection will already have split that list up into multiple
* headers.
*/
static gboolean
check_request_requirements (GstRTSPContext * ctx, gchar ** unsupported_reqs)
{
GstRTSPResult res;
GPtrArray *arr = NULL;
GstRTSPMessage *msg = ctx->request;
gchar *reqs = NULL;
gint i;
gchar *sig_result = NULL;
gboolean result = TRUE;
i = 0;
do {
res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
if (res == GST_RTSP_ENOTIMPL)
break;
if (arr == NULL)
arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
g_ptr_array_add (arr, g_strdup (reqs));
}
while (TRUE);
/* if we don't have any Require headers at all, all is fine */
if (i == 1)
return TRUE;
/* otherwise we've now processed at all the Require headers */
g_ptr_array_add (arr, NULL);
g_signal_emit (ctx->client,
gst_rtsp_client_signals[SIGNAL_CHECK_REQUIREMENTS], 0, ctx,
(gchar **) arr->pdata, &sig_result);
if (sig_result == NULL) {
/* no supported options, just report all of the required ones as
* unsupported */
*unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
result = FALSE;
goto done;
}
if (strlen (sig_result) == 0)
g_free (sig_result);
else {
*unsupported_reqs = sig_result;
result = FALSE;
}
done:
g_ptr_array_unref (arr);
return result;
}
static void
handle_request (GstRTSPClient * client, GstRTSPMessage * request)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPMethod method;
const gchar *uristr;
GstRTSPUrl *uri = NULL;
GstRTSPVersion version;
GstRTSPResult res;
GstRTSPSession *session = NULL;
GstRTSPContext sctx = { NULL }, *ctx;
GstRTSPMessage response = { 0 };
gchar *unsupported_reqs = NULL;
gchar *sessid = NULL, *pipelined_request_id = NULL;
if (!(ctx = gst_rtsp_context_get_current ())) {
ctx = &sctx;
ctx->auth = priv->auth;
gst_rtsp_context_push_current (ctx);
}
ctx->conn = priv->connection;
ctx->client = client;
ctx->request = request;
ctx->response = &response;
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (request);
}
gst_rtsp_message_parse_request (request, &method, &uristr, &version);
GST_INFO ("client %p: received a request %s %s %s", client,
gst_rtsp_method_as_text (method), uristr,
gst_rtsp_version_as_text (version));
/* we can only handle 1.0 requests */
if (version != GST_RTSP_VERSION_1_0 && version != GST_RTSP_VERSION_2_0)
goto not_supported;
ctx->method = method;
/* we always try to parse the url first */
if (strcmp (uristr, "*") == 0) {
/* special case where we have * as uri, keep uri = NULL */
} else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
/* check if the uristr is an absolute path <=> scheme and host information
* is missing */
gchar *scheme;
scheme = g_uri_parse_scheme (uristr);
if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
gchar *absolute_uristr = NULL;
GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
if (priv->server_ip == NULL) {
GST_WARNING_OBJECT (client, "host information missing");
goto bad_request;
}
absolute_uristr =
g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
g_free (absolute_uristr);
goto bad_request;
}
g_free (absolute_uristr);
} else {
g_free (scheme);
goto bad_request;
}
}
/* get the session if there is any */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_PIPELINED_REQUESTS,
&pipelined_request_id, 0);
if (res == GST_RTSP_OK) {
sessid = g_hash_table_lookup (client->priv->pipelined_requests,
pipelined_request_id);
if (!sessid)
res = GST_RTSP_ERROR;
}
if (res != GST_RTSP_OK)
res =
gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
if (res == GST_RTSP_OK) {
if (priv->session_pool == NULL)
goto no_pool;
/* we had a session in the request, find it again */
if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
goto session_not_found;
/* we add the session to the client list of watched sessions. When a session
* disappears because it times out, we will be notified. If all sessions are
* gone, we will close the connection */
client_watch_session (client, session);
}
/* sanitize the uri */
if (uri)
sanitize_uri (uri);
ctx->uri = uri;
ctx->session = session;
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
goto not_authorized;
/* handle any 'Require' headers */
if (!check_request_requirements (ctx, &unsupported_reqs))
goto unsupported_requirement;
/* now see what is asked and dispatch to a dedicated handler */
switch (method) {
case GST_RTSP_OPTIONS:
priv->version = version;
handle_options_request (client, ctx, version);
break;
case GST_RTSP_DESCRIBE:
handle_describe_request (client, ctx);
break;
case GST_RTSP_SETUP:
handle_setup_request (client, ctx);
break;
case GST_RTSP_PLAY:
handle_play_request (client, ctx);
break;
case GST_RTSP_PAUSE:
handle_pause_request (client, ctx);
break;
case GST_RTSP_TEARDOWN:
handle_teardown_request (client, ctx);
break;
case GST_RTSP_SET_PARAMETER:
handle_set_param_request (client, ctx);
break;
case GST_RTSP_GET_PARAMETER:
handle_get_param_request (client, ctx);
break;
case GST_RTSP_ANNOUNCE:
if (version >= GST_RTSP_VERSION_2_0)
goto invalid_command_for_version;
handle_announce_request (client, ctx);
break;
case GST_RTSP_RECORD:
if (version >= GST_RTSP_VERSION_2_0)
goto invalid_command_for_version;
handle_record_request (client, ctx);
break;
case GST_RTSP_REDIRECT:
goto not_implemented;
case GST_RTSP_INVALID:
default:
goto bad_request;
}
done:
if (ctx == &sctx)
gst_rtsp_context_pop_current (ctx);
if (session)
g_object_unref (session);
if (uri)
gst_rtsp_url_free (uri);
return;
/* ERRORS */
not_supported:
{
GST_ERROR ("client %p: version %d not supported", client, version);
send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
ctx);
goto done;
}
invalid_command_for_version:
{
GST_ERROR ("client %p: invalid command for version", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
goto done;
}
bad_request:
{
GST_ERROR ("client %p: bad request", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
goto done;
}
no_pool:
{
GST_ERROR ("client %p: no pool configured", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
goto done;
}
session_not_found:
{
GST_ERROR ("client %p: session not found", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
goto done;
}
not_authorized:
{
GST_ERROR ("client %p: not allowed", client);
/* error reply is already sent */
goto done;
}
unsupported_requirement:
{
GST_ERROR ("client %p: Required option is not supported (%s)", client,
unsupported_reqs);
send_option_not_supported_response (client, ctx, unsupported_reqs);
g_free (unsupported_reqs);
goto done;
}
not_implemented:
{
GST_ERROR ("client %p: method %d not implemented", client, method);
send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
goto done;
}
}
static void
handle_response (GstRTSPClient * client, GstRTSPMessage * response)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
GstRTSPSession *session = NULL;
GstRTSPContext sctx = { NULL }, *ctx;
gchar *sessid;
if (!(ctx = gst_rtsp_context_get_current ())) {
ctx = &sctx;
ctx->auth = priv->auth;
gst_rtsp_context_push_current (ctx);
}
ctx->conn = priv->connection;
ctx->client = client;
ctx->request = NULL;
ctx->uri = NULL;
ctx->method = GST_RTSP_INVALID;
ctx->response = response;
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (response);
}
GST_INFO ("client %p: received a response", client);
/* get the session if there is any */
res =
gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
if (res == GST_RTSP_OK) {
if (priv->session_pool == NULL)
goto no_pool;
/* we had a session in the request, find it again */
if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
goto session_not_found;
/* we add the session to the client list of watched sessions. When a session
* disappears because it times out, we will be notified. If all sessions are
* gone, we will close the connection */
client_watch_session (client, session);
}
ctx->session = session;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
0, ctx);
done:
if (ctx == &sctx)
gst_rtsp_context_pop_current (ctx);
if (session)
g_object_unref (session);
return;
no_pool:
{
GST_ERROR ("client %p: no pool configured", client);
goto done;
}
session_not_found:
{
GST_ERROR ("client %p: session not found", client);
goto done;
}
}
static void
handle_data (GstRTSPClient * client, GstRTSPMessage * message)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
guint8 channel;
guint8 *data;
guint size;
GstBuffer *buffer;
GstRTSPStreamTransport *trans;
/* find the stream for this message */
res = gst_rtsp_message_parse_data (message, &channel);
if (res != GST_RTSP_OK)
return;
gst_rtsp_message_get_body (message, &data, &size);
if (size < 2)
goto invalid_length;
gst_rtsp_message_steal_body (message, &data, &size);
/* Strip trailing \0 (which GstRTSPConnection adds) */
--size;
buffer = gst_buffer_new_wrapped (data, size);
trans =
g_hash_table_lookup (priv->transports, GINT_TO_POINTER ((gint) channel));
if (trans) {
GSocketAddress *addr;
/* Only create the socket address once for the transport, we don't really
* want to do that for every single packet.
*
* The netaddress meta is later used by the RTP stack to know where
* packets came from and allows us to match it again to a stream transport
*
* In theory we could use the remote socket address of the RTSP connection
* here, but this would fail with a custom configure_client_transport()
* implementation.
*/
if (!(addr =
g_object_get_data (G_OBJECT (trans), "rtsp-client.remote-addr"))) {
const GstRTSPTransport *tr;
GInetAddress *iaddr;
tr = gst_rtsp_stream_transport_get_transport (trans);
iaddr = g_inet_address_new_from_string (tr->destination);
if (iaddr) {
addr = g_inet_socket_address_new (iaddr, tr->client_port.min);
g_object_unref (iaddr);
g_object_set_data_full (G_OBJECT (trans), "rtsp-client.remote-addr",
addr, (GDestroyNotify) g_object_unref);
}
}
if (addr) {
gst_buffer_add_net_address_meta (buffer, addr);
}
/* dispatch to the stream based on the channel number */
GST_LOG_OBJECT (client, "%u bytes of data on channel %u", size, channel);
gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
} else {
GST_DEBUG_OBJECT (client, "received %u bytes of data for "
"unknown channel %u", size, channel);
gst_buffer_unref (buffer);
}
return;
/* ERRORS */
invalid_length:
{
GST_DEBUG ("client %p: Short message received, ignoring", client);
return;
}
}
/**
* gst_rtsp_client_set_session_pool:
* @client: a #GstRTSPClient
* @pool: (transfer none) (nullable): a #GstRTSPSessionPool
*
* Set @pool as the sessionpool for @client which it will use to find
* or allocate sessions. the sessionpool is usually inherited from the server
* that created the client but can be overridden later.
*/
void
gst_rtsp_client_set_session_pool (GstRTSPClient * client,
GstRTSPSessionPool * pool)
{
GstRTSPSessionPool *old;
GstRTSPClientPrivate *priv;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
if (pool)
g_object_ref (pool);
g_mutex_lock (&priv->lock);
old = priv->session_pool;
priv->session_pool = pool;
if (priv->session_removed_id) {
g_signal_handler_disconnect (old, priv->session_removed_id);
priv->session_removed_id = 0;
}
g_mutex_unlock (&priv->lock);
/* FIXME, should remove all sessions from the old pool for this client */
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_client_get_session_pool:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
*
* Returns: (transfer full) (nullable): a #GstRTSPSessionPool, unref after usage.
*/
GstRTSPSessionPool *
gst_rtsp_client_get_session_pool (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv;
GstRTSPSessionPool *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
priv = client->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->session_pool))
g_object_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_client_set_mount_points:
* @client: a #GstRTSPClient
* @mounts: (transfer none) (nullable): a #GstRTSPMountPoints
*
* Set @mounts as the mount points for @client which it will use to map urls
* to media streams. These mount points are usually inherited from the server that
* created the client but can be overriden later.
*/
void
gst_rtsp_client_set_mount_points (GstRTSPClient * client,
GstRTSPMountPoints * mounts)
{
GstRTSPClientPrivate *priv;
GstRTSPMountPoints *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
if (mounts)
g_object_ref (mounts);
g_mutex_lock (&priv->lock);
old = priv->mount_points;
priv->mount_points = mounts;
g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_client_get_mount_points:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
*
* Returns: (transfer full) (nullable): a #GstRTSPMountPoints, unref after usage.
*/
GstRTSPMountPoints *
gst_rtsp_client_get_mount_points (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv;
GstRTSPMountPoints *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
priv = client->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->mount_points))
g_object_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_client_set_content_length_limit:
* @client: a #GstRTSPClient
* @limit: Content-Length limit
*
* Configure @client to use the specified Content-Length limit.
*
* Define an appropriate request size limit and reject requests exceeding the
* limit with response status 413 Request Entity Too Large
*
* Since: 1.18
*/
void
gst_rtsp_client_set_content_length_limit (GstRTSPClient * client, guint limit)
{
GstRTSPClientPrivate *priv;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
g_mutex_lock (&priv->lock);
priv->content_length_limit = limit;
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_client_get_content_length_limit:
* @client: a #GstRTSPClient
*
* Get the Content-Length limit of @client.
*
* Returns: the Content-Length limit.
*
* Since: 1.18
*/
guint
gst_rtsp_client_get_content_length_limit (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv;
glong content_length_limit;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), -1);
priv = client->priv;
g_mutex_lock (&priv->lock);
content_length_limit = priv->content_length_limit;
g_mutex_unlock (&priv->lock);
return content_length_limit;
}
/**
* gst_rtsp_client_set_auth:
* @client: a #GstRTSPClient
* @auth: (transfer none) (nullable): a #GstRTSPAuth
*
* configure @auth to be used as the authentication manager of @client.
*/
void
gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
{
GstRTSPClientPrivate *priv;
GstRTSPAuth *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
if (auth)
g_object_ref (auth);
g_mutex_lock (&priv->lock);
old = priv->auth;
priv->auth = auth;
g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_client_get_auth:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPAuth used as the authentication manager of @client.
*
* Returns: (transfer full) (nullable): the #GstRTSPAuth of @client.
* g_object_unref() after usage.
*/
GstRTSPAuth *
gst_rtsp_client_get_auth (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv;
GstRTSPAuth *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
priv = client->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->auth))
g_object_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_client_set_thread_pool:
* @client: a #GstRTSPClient
* @pool: (transfer none) (nullable): a #GstRTSPThreadPool
*
* configure @pool to be used as the thread pool of @client.
*/
void
gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
GstRTSPThreadPool * pool)
{
GstRTSPClientPrivate *priv;
GstRTSPThreadPool *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
if (pool)
g_object_ref (pool);
g_mutex_lock (&priv->lock);
old = priv->thread_pool;
priv->thread_pool = pool;
g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_client_get_thread_pool:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPThreadPool used as the thread pool of @client.
*
* Returns: (transfer full) (nullable): the #GstRTSPThreadPool of @client. g_object_unref() after
* usage.
*/
GstRTSPThreadPool *
gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv;
GstRTSPThreadPool *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
priv = client->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->thread_pool))
g_object_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_client_set_connection:
* @client: a #GstRTSPClient
* @conn: (transfer full): a #GstRTSPConnection
*
* Set the #GstRTSPConnection of @client. This function takes ownership of
* @conn.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_client_set_connection (GstRTSPClient * client,
GstRTSPConnection * conn)
{
GstRTSPClientPrivate *priv;
GSocket *read_socket;
GSocketAddress *address;
GstRTSPUrl *url;
GError *error = NULL;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
g_return_val_if_fail (conn != NULL, FALSE);
priv = client->priv;
gst_rtsp_connection_set_content_length_limit (conn,
priv->content_length_limit);
read_socket = gst_rtsp_connection_get_read_socket (conn);
if (!(address = g_socket_get_local_address (read_socket, &error)))
goto no_address;
g_free (priv->server_ip);
/* keep the original ip that the client connected to */
if (G_IS_INET_SOCKET_ADDRESS (address)) {
GInetAddress *iaddr;
iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
/* socket might be ipv6 but adress still ipv4 */
priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
priv->server_ip = g_inet_address_to_string (iaddr);
g_object_unref (address);
} else {
priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
priv->server_ip = g_strdup ("unknown");
}
GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
priv->server_ip, priv->is_ipv6);
url = gst_rtsp_connection_get_url (conn);
GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
priv->connection = conn;
return TRUE;
/* ERRORS */
no_address:
{
GST_ERROR ("could not get local address %s", error->message);
g_error_free (error);
return FALSE;
}
}
/**
* gst_rtsp_client_get_connection:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPConnection of @client.
*
* Returns: (transfer none) (nullable): the #GstRTSPConnection of @client.
* The connection object returned remains valid until the client is freed.
*/
GstRTSPConnection *
gst_rtsp_client_get_connection (GstRTSPClient * client)
{
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
return client->priv->connection;
}
/**
* gst_rtsp_client_set_send_func:
* @client: a #GstRTSPClient
* @func: (scope notified): a #GstRTSPClientSendFunc
* @user_data: (closure): user data passed to @func
* @notify: (allow-none): called when @user_data is no longer in use
*
* Set @func as the callback that will be called when a new message needs to be
* sent to the client. @user_data is passed to @func and @notify is called when
* @user_data is no longer in use.
*
* By default, the client will send the messages on the #GstRTSPConnection that
* was configured with gst_rtsp_client_attach() was called.
*
* It is only allowed to set either a `send_func` or a `send_messages_func`
* but not both at the same time.
*/
void
gst_rtsp_client_set_send_func (GstRTSPClient * client,
GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
{
GstRTSPClientPrivate *priv;
GDestroyNotify old_notify;
gpointer old_data;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
g_mutex_lock (&priv->send_lock);
g_assert (func == NULL || priv->send_messages_func == NULL);
priv->send_func = func;
old_notify = priv->send_notify;
old_data = priv->send_data;
priv->send_notify = notify;
priv->send_data = user_data;
g_mutex_unlock (&priv->send_lock);
if (old_notify)
old_notify (old_data);
}
/**
* gst_rtsp_client_set_send_messages_func:
* @client: a #GstRTSPClient
* @func: (scope notified): a #GstRTSPClientSendMessagesFunc
* @user_data: (closure): user data passed to @func
* @notify: (allow-none): called when @user_data is no longer in use
*
* Set @func as the callback that will be called when new messages needs to be
* sent to the client. @user_data is passed to @func and @notify is called when
* @user_data is no longer in use.
*
* By default, the client will send the messages on the #GstRTSPConnection that
* was configured with gst_rtsp_client_attach() was called.
*
* It is only allowed to set either a `send_func` or a `send_messages_func`
* but not both at the same time.
*
* Since: 1.16
*/
void
gst_rtsp_client_set_send_messages_func (GstRTSPClient * client,
GstRTSPClientSendMessagesFunc func, gpointer user_data,
GDestroyNotify notify)
{
GstRTSPClientPrivate *priv;
GDestroyNotify old_notify;
gpointer old_data;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
g_mutex_lock (&priv->send_lock);
g_assert (func == NULL || priv->send_func == NULL);
priv->send_messages_func = func;
old_notify = priv->send_messages_notify;
old_data = priv->send_messages_data;
priv->send_messages_notify = notify;
priv->send_messages_data = user_data;
g_mutex_unlock (&priv->send_lock);
if (old_notify)
old_notify (old_data);
}
/**
* gst_rtsp_client_handle_message:
* @client: a #GstRTSPClient
* @message: (transfer none): an #GstRTSPMessage
*
* Let the client handle @message.
*
* Returns: a #GstRTSPResult.
*/
GstRTSPResult
gst_rtsp_client_handle_message (GstRTSPClient * client,
GstRTSPMessage * message)
{
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
switch (message->type) {
case GST_RTSP_MESSAGE_REQUEST:
handle_request (client, message);
break;
case GST_RTSP_MESSAGE_RESPONSE:
handle_response (client, message);
break;
case GST_RTSP_MESSAGE_DATA:
handle_data (client, message);
break;
default:
break;
}
return GST_RTSP_OK;
}
/**
* gst_rtsp_client_send_message:
* @client: a #GstRTSPClient
* @session: (allow-none) (transfer none): a #GstRTSPSession to send
* the message to or %NULL
* @message: (transfer none): The #GstRTSPMessage to send
*
* Send a message message to the remote end. @message must be a
* #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
*/
GstRTSPResult
gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPMessage * message)
{
GstRTSPContext sctx = { NULL }
, *ctx;
GstRTSPClientPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
priv = client->priv;
if (!(ctx = gst_rtsp_context_get_current ())) {
ctx = &sctx;
ctx->auth = priv->auth;
gst_rtsp_context_push_current (ctx);
}
ctx->conn = priv->connection;
ctx->client = client;
ctx->session = session;
send_message (client, ctx, message, FALSE);
if (ctx == &sctx)
gst_rtsp_context_pop_current (ctx);
return GST_RTSP_OK;
}
/**
* gst_rtsp_client_get_stream_transport:
*
* This is useful when providing a send function through
* gst_rtsp_client_set_send_func() when doing RTSP over TCP:
* the send function must call gst_rtsp_stream_transport_message_sent ()
* on the appropriate transport when data has been received for streaming
* to continue.
*
* Returns: (transfer none) (nullable): the #GstRTSPStreamTransport associated with @channel.
*
* Since: 1.18
*/
GstRTSPStreamTransport *
gst_rtsp_client_get_stream_transport (GstRTSPClient * self, guint8 channel)
{
return g_hash_table_lookup (self->priv->transports,
GINT_TO_POINTER ((gint) channel));
}
static gboolean
do_send_messages (GstRTSPClient * client, GstRTSPMessage * messages,
guint n_messages, gboolean close, gpointer user_data)
{
GstRTSPClientPrivate *priv = client->priv;
guint id = 0;
GstRTSPResult ret;
guint i;
/* send the message */
if (close)
GST_INFO ("client %p: sending close message", client);
ret = gst_rtsp_watch_send_messages (priv->watch, messages, n_messages, &id);
if (ret != GST_RTSP_OK)
goto error;
for (i = 0; i < n_messages; i++) {
if (gst_rtsp_message_get_type (&messages[i]) == GST_RTSP_MESSAGE_DATA) {
guint8 channel = 0;
GstRTSPResult r;
/* We assume that all data messages in the list are for the
* same channel */
r = gst_rtsp_message_parse_data (&messages[i], &channel);
if (r != GST_RTSP_OK) {
ret = r;
goto error;
}
/* check if the message has been queued for transmission in watch */
if (id) {
/* store the seq number so we can wait until it has been sent */
GST_DEBUG_OBJECT (client, "wait for message %d, channel %d", id,
channel);
set_data_seq (client, channel, id);
} else {
GstRTSPStreamTransport *trans;
trans =
g_hash_table_lookup (priv->transports,
GINT_TO_POINTER ((gint) channel));
if (trans) {
GST_DEBUG_OBJECT (client, "emit 'message-sent' signal");
g_mutex_unlock (&priv->send_lock);
gst_rtsp_stream_transport_message_sent (trans);
g_mutex_lock (&priv->send_lock);
}
}
break;
}
}
return ret == GST_RTSP_OK;
/* ERRORS */
error:
{
GST_DEBUG_OBJECT (client, "got error %d", ret);
return FALSE;
}
}
static GstRTSPResult
message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
gpointer user_data)
{
return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
}
static GstRTSPResult
message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GstRTSPClientPrivate *priv = client->priv;
GstRTSPStreamTransport *trans = NULL;
guint8 channel = 0;
g_mutex_lock (&priv->send_lock);
if (get_data_channel (client, cseq, &channel)) {
trans = g_hash_table_lookup (priv->transports, GINT_TO_POINTER (channel));
set_data_seq (client, channel, 0);
}
g_mutex_unlock (&priv->send_lock);
if (trans) {
GST_DEBUG_OBJECT (client, "emit 'message-sent' signal");
gst_rtsp_stream_transport_message_sent (trans);
}
return GST_RTSP_OK;
}
static GstRTSPResult
closed (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
GST_INFO ("client %p: connection closed", client);
if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (&tunnels_lock);
}
gst_rtsp_watch_set_flushing (watch, TRUE);
g_mutex_lock (&priv->watch_lock);
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
g_mutex_unlock (&priv->watch_lock);
return GST_RTSP_OK;
}
static GstRTSPResult
error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
gchar *str;
str = gst_rtsp_strresult (result);
GST_INFO ("client %p: received an error %s", client, str);
g_free (str);
return GST_RTSP_OK;
}
static GstRTSPResult
error_full (GstRTSPWatch * watch, GstRTSPResult result,
GstRTSPMessage * message, guint id, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
gchar *str;
GstRTSPContext sctx = { NULL }, *ctx;
GstRTSPClientPrivate *priv;
GstRTSPMessage response = { 0 };
priv = client->priv;
if (!(ctx = gst_rtsp_context_get_current ())) {
ctx = &sctx;
ctx->auth = priv->auth;
gst_rtsp_context_push_current (ctx);
}
ctx->conn = priv->connection;
ctx->client = client;
ctx->request = message;
ctx->method = GST_RTSP_INVALID;
ctx->response = &response;
/* only return error response if it is a request */
if (!message || message->type != GST_RTSP_MESSAGE_REQUEST)
goto done;
if (result == GST_RTSP_ENOMEM) {
send_generic_response (client, GST_RTSP_STS_REQUEST_ENTITY_TOO_LARGE, ctx);
goto done;
}
if (result == GST_RTSP_EPARSE) {
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
goto done;
}
done:
if (ctx == &sctx)
gst_rtsp_context_pop_current (ctx);
str = gst_rtsp_strresult (result);
GST_INFO
("client %p: error when handling message %p with id %d: %s",
client, message, id, str);
g_free (str);
return GST_RTSP_OK;
}
static gboolean
remember_tunnel (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
/* store client in the pending tunnels */
tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
if (tunnelid == NULL)
goto no_tunnelid;
GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
/* we can't have two clients connecting with the same tunnelid */
g_mutex_lock (&tunnels_lock);
if (g_hash_table_lookup (tunnels, tunnelid))
goto tunnel_existed;
g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
g_mutex_unlock (&tunnels_lock);
return TRUE;
/* ERRORS */
no_tunnelid:
{
GST_ERROR ("client %p: no tunnelid provided", client);
return FALSE;
}
tunnel_existed:
{
g_mutex_unlock (&tunnels_lock);
GST_ERROR ("client %p: tunnel session %s already existed", client,
tunnelid);
return FALSE;
}
}
static GstRTSPResult
tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GstRTSPClientPrivate *priv = client->priv;
GST_WARNING ("client %p: tunnel lost (connection %p)", client,
priv->connection);
/* ignore error, it'll only be a problem when the client does a POST again */
remember_tunnel (client);
return GST_RTSP_OK;
}
static GstRTSPStatusCode
handle_tunnel (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPClient *oclient;
GstRTSPClientPrivate *opriv;
const gchar *tunnelid;
tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
if (tunnelid == NULL)
goto no_tunnelid;
/* check for previous tunnel */
g_mutex_lock (&tunnels_lock);
oclient = g_hash_table_lookup (tunnels, tunnelid);
if (oclient == NULL) {
/* no previous tunnel, remember tunnel */
g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
g_mutex_unlock (&tunnels_lock);
GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
client, priv->connection);
} else {
/* merge both tunnels into the first client */
/* remove the old client from the table. ref before because removing it will
* remove the ref to it. */
g_object_ref (oclient);
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (&tunnels_lock);
opriv = oclient->priv;
g_mutex_lock (&opriv->watch_lock);
if (opriv->watch == NULL)
goto tunnel_closed;
if (opriv->tstate == priv->tstate)
goto tunnel_duplicate_id;
GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
oclient, opriv->connection, priv->connection);
gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
gst_rtsp_watch_reset (priv->watch);
gst_rtsp_watch_reset (opriv->watch);
g_mutex_unlock (&opriv->watch_lock);
g_object_unref (oclient);
/* the old client owns the tunnel now, the new one will be freed */
g_source_destroy ((GSource *) priv->watch);
priv->watch = NULL;
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
rtsp_ctrl_timeout_remove (client);
}
return GST_RTSP_STS_OK;
/* ERRORS */
no_tunnelid:
{
GST_ERROR ("client %p: no tunnelid provided", client);
return GST_RTSP_STS_SERVICE_UNAVAILABLE;
}
tunnel_closed:
{
GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
g_mutex_unlock (&opriv->watch_lock);
g_object_unref (oclient);
return GST_RTSP_STS_SERVICE_UNAVAILABLE;
}
tunnel_duplicate_id:
{
GST_ERROR ("client %p: tunnel session %s was duplicate", client, tunnelid);
g_mutex_unlock (&opriv->watch_lock);
g_object_unref (oclient);
return GST_RTSP_STS_BAD_REQUEST;
}
}
static GstRTSPStatusCode
tunnel_get (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GST_INFO ("client %p: tunnel get (connection %p)", client,
client->priv->connection);
g_mutex_lock (&client->priv->lock);
client->priv->tstate = TUNNEL_STATE_GET;
g_mutex_unlock (&client->priv->lock);
return handle_tunnel (client);
}
static GstRTSPResult
tunnel_post (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GST_INFO ("client %p: tunnel post (connection %p)", client,
client->priv->connection);
g_mutex_lock (&client->priv->lock);
client->priv->tstate = TUNNEL_STATE_POST;
g_mutex_unlock (&client->priv->lock);
if (handle_tunnel (client) != GST_RTSP_STS_OK)
return GST_RTSP_ERROR;
return GST_RTSP_OK;
}
static GstRTSPResult
tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
GstRTSPMessage * response, gpointer user_data)
{
GstRTSPClientClass *klass;
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
klass = GST_RTSP_CLIENT_GET_CLASS (client);
if (klass->tunnel_http_response) {
klass->tunnel_http_response (client, request, response);
}
return GST_RTSP_OK;
}
static GstRTSPWatchFuncs watch_funcs = {
message_received,
message_sent,
closed,
error,
tunnel_get,
tunnel_post,
error_full,
tunnel_lost,
tunnel_http_response
};
static void
client_watch_notify (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
gboolean closed = TRUE;
GST_INFO ("client %p: watch destroyed", client);
priv->watch = NULL;
/* remove all sessions if the media says so and so drop the extra client ref */
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
rtsp_ctrl_timeout_remove (client);
gst_rtsp_client_session_filter (client, cleanup_session, &closed);
if (closed)
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
g_object_unref (client);
}
/**
* gst_rtsp_client_attach:
* @client: a #GstRTSPClient
* @context: (allow-none): a #GMainContext
*
* Attaches @client to @context. When the mainloop for @context is run, the
* client will be dispatched. When @context is %NULL, the default context will be
* used).
*
* This function should be called when the client properties and urls are fully
* configured and the client is ready to start.
*
* Returns: the ID (greater than 0) for the source within the GMainContext.
*/
guint
gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
{
GstRTSPClientPrivate *priv;
GSource *timer_src;
guint res;
GWeakRef *client_weak_ref = g_new (GWeakRef, 1);
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
priv = client->priv;
g_return_val_if_fail (priv->connection != NULL, 0);
g_return_val_if_fail (priv->watch == NULL, 0);
g_return_val_if_fail (priv->watch_context == NULL, 0);
/* make sure noone will free the context before the watch is destroyed */
priv->watch_context = g_main_context_ref (context);
/* create watch for the connection and attach */
priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
g_object_ref (client), (GDestroyNotify) client_watch_notify);
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
gst_rtsp_client_set_send_messages_func (client, do_send_messages, priv->watch,
(GDestroyNotify) gst_rtsp_watch_unref);
gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
/* take the lock before attaching the client watch, so that the client thread
* can not access the control channel timer until it's properly in place */
g_mutex_lock (&priv->lock);
GST_INFO ("client %p: attaching to context %p", client, context);
res = gst_rtsp_watch_attach (priv->watch, context);
/* Setting up a timeout for the RTSP control channel until a session
* is up where it is handling timeouts. */
/* remove old timeout if any */
rtsp_ctrl_timeout_remove_unlocked (client->priv);
timer_src = g_timeout_source_new_seconds (RTSP_CTRL_CB_INTERVAL);
g_weak_ref_init (client_weak_ref, client);
g_source_set_callback (timer_src, rtsp_ctrl_timeout_cb, client_weak_ref,
rtsp_ctrl_timeout_destroy_notify);
g_source_attach (timer_src, priv->watch_context);
priv->rtsp_ctrl_timeout = timer_src;
GST_DEBUG ("rtsp control setting up session timeout %p.",
priv->rtsp_ctrl_timeout);
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_client_session_filter:
* @client: a #GstRTSPClient
* @func: (scope call) (allow-none): a callback
* @user_data: user data passed to @func
*
* Call @func for each session managed by @client. The result value of @func
* determines what happens to the session. @func will be called with @client
* locked so no further actions on @client can be performed from @func.
*
* If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
* @client.
*
* If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
*
* If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
* will also be added with an additional ref to the result #GList of this
* function..
*
* When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
*
* Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
* sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
* element in the #GList should be unreffed before the list is freed.
*/
GList *
gst_rtsp_client_session_filter (GstRTSPClient * client,
GstRTSPClientSessionFilterFunc func, gpointer user_data)
{
GstRTSPClientPrivate *priv;
GList *result, *walk, *next;
GHashTable *visited;
guint cookie;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
priv = client->priv;
result = NULL;
if (func)
visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
g_mutex_lock (&priv->lock);
restart:
cookie = priv->sessions_cookie;
for (walk = priv->sessions; walk; walk = next) {
GstRTSPSession *sess = walk->data;
GstRTSPFilterResult res;
gboolean changed;
next = g_list_next (walk);
if (func) {
/* only visit each session once */
if (g_hash_table_contains (visited, sess))
continue;
g_hash_table_add (visited, g_object_ref (sess));
g_mutex_unlock (&priv->lock);
res = func (client, sess, user_data);
g_mutex_lock (&priv->lock);
} else
res = GST_RTSP_FILTER_REF;
changed = (cookie != priv->sessions_cookie);
switch (res) {
case GST_RTSP_FILTER_REMOVE:
/* stop watching the session and pretend it went away, if the list was
* changed, we can't use the current list position, try to see if we
* still have the session */
client_unwatch_session (client, sess, changed ? NULL : walk);
cookie = priv->sessions_cookie;
break;
case GST_RTSP_FILTER_REF:
result = g_list_prepend (result, g_object_ref (sess));
break;
case GST_RTSP_FILTER_KEEP:
default:
break;
}
if (changed)
goto restart;
}
g_mutex_unlock (&priv->lock);
if (func)
g_hash_table_unref (visited);
return result;
}
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