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gstreamer/gst/rtsp-sink/gstrtspclientsink.c
Mathieu Duponchelle 6f308daa5f rtspclientsink: if OPEN failed, unqueue next command 
As READY_TO_PAUSED can no longer return async, the RECORD
command will be queued before the OPEN command fails
(for example in case the server could not be connected),
and record then waits for ever.

https://bugzilla.gnome.org/show_bug.cgi?id=793896
2018-02-27 20:37:14 +01:00

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/* GStreamer
* Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
* <2006> Lutz Mueller <lutz at topfrose dot de>
* <2015> Jan Schmidt <jan at centricular dot com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/*
* Unless otherwise indicated, Source Code is licensed under MIT license.
* See further explanation attached in License Statement (distributed in the file
* LICENSE).
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
* of the Software, and to permit persons to whom the Software is furnished to do
* so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
/**
* SECTION:element-rtspclientsink
*
* Makes a connection to an RTSP server and send data via RTSP RECORD.
* rtspclientsink strictly follows RFC 2326
*
* RTSP supports transport over TCP or UDP in unicast or multicast mode. By
* default rtspclientsink will negotiate a connection in the following order:
* UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
* protocols can be controlled with the #GstRTSPClientSink:protocols property.
*
* rtspclientsink will internally instantiate an RTP session manager element
* that will handle the RTCP messages to and from the server, jitter removal,
* and packet reordering.
* This feature is implemented using the gstrtpbin element.
*
* rtspclientsink accepts any stream for which there is an installed payloader,
* creates the payloader and manages payload-types, as well as RTX setup.
* The new-payloader signal is fired when a payloader is created, in case
* an app wants to do custom configuration (such as for MTU).
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 videotestsrc ! jpegenc ! rtspclientsink location=rtsp://some.server/url
* ]| Establish a connection to an RTSP server and send JPEG encoded video packets
* </refsect2>
*/
/* FIXMEs
* - Handle EOS properly and shutdown. The problem with EOS is we don't know
* when the server has received all data, so we don't know when to do teardown.
* At the moment, we forward EOS to the app as soon as we stop sending. Is there
* a way to know from the receiver that it's got all data? Some session timeout?
* - Implement extension support for Real / WMS if they support RECORD?
* - Add support for network clock synchronised streaming?
* - Fix crypto key nego so SAVP/SAVPF profiles work.
* - Test (&fix?) HTTP tunnel support
* - Add an address pool object for GstRTSPStreams to use for multicast
* - Test multicast UDP transport
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#ifdef HAVE_UNISTD_H
#include <unistd.h>
#endif /* HAVE_UNISTD_H */
#include <stdlib.h>
#include <string.h>
#include <stdio.h>
#include <stdarg.h>
#include <gst/net/gstnet.h>
#include <gst/sdp/gstsdpmessage.h>
#include <gst/sdp/gstmikey.h>
#include <gst/rtp/rtp.h>
#include "gstrtspclientsink.h"
typedef struct _GstRtspClientSinkPad GstRtspClientSinkPad;
typedef GstGhostPadClass GstRtspClientSinkPadClass;
struct _GstRtspClientSinkPad
{
GstGhostPad parent;
GstElement *custom_payloader;
};
enum
{
PROP_PAD_0,
PROP_PAD_PAYLOADER
};
static GType gst_rtsp_client_sink_pad_get_type (void);
G_DEFINE_TYPE (GstRtspClientSinkPad, gst_rtsp_client_sink_pad,
GST_TYPE_GHOST_PAD);
#define GST_TYPE_RTSP_CLIENT_SINK_PAD (gst_rtsp_client_sink_pad_get_type ())
#define GST_RTSP_CLIENT_SINK_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSP_CLIENT_SINK_PAD,GstRtspClientSinkPad))
static void
gst_rtsp_client_sink_pad_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtspClientSinkPad *pad;
pad = GST_RTSP_CLIENT_SINK_PAD (object);
switch (prop_id) {
case PROP_PAD_PAYLOADER:
GST_OBJECT_LOCK (pad);
if (pad->custom_payloader)
gst_object_unref (pad->custom_payloader);
pad->custom_payloader = g_value_get_object (value);
gst_object_ref_sink (pad->custom_payloader);
GST_OBJECT_UNLOCK (pad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtsp_client_sink_pad_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtspClientSinkPad *pad;
pad = GST_RTSP_CLIENT_SINK_PAD (object);
switch (prop_id) {
case PROP_PAD_PAYLOADER:
GST_OBJECT_LOCK (pad);
g_value_set_object (value, pad->custom_payloader);
GST_OBJECT_UNLOCK (pad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtsp_client_sink_pad_dispose (GObject * object)
{
GstRtspClientSinkPad *pad = GST_RTSP_CLIENT_SINK_PAD (object);
if (pad->custom_payloader)
gst_object_unref (pad->custom_payloader);
G_OBJECT_CLASS (gst_rtsp_client_sink_pad_parent_class)->dispose (object);
}
static void
gst_rtsp_client_sink_pad_class_init (GstRtspClientSinkPadClass * klass)
{
GObjectClass *gobject_klass;
gobject_klass = (GObjectClass *) klass;
gobject_klass->set_property = gst_rtsp_client_sink_pad_set_property;
gobject_klass->get_property = gst_rtsp_client_sink_pad_get_property;
gobject_klass->dispose = gst_rtsp_client_sink_pad_dispose;
g_object_class_install_property (gobject_klass, PROP_PAD_PAYLOADER,
g_param_spec_object ("payloader", "Payloader",
"The payloader element to use (NULL = default automatically selected)",
GST_TYPE_ELEMENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_rtsp_client_sink_pad_init (GstRtspClientSinkPad * pad)
{
}
static GstPad *
gst_rtsp_client_sink_pad_new (const GstPadTemplate * pad_tmpl,
const gchar * name)
{
GstRtspClientSinkPad *ret;
ret =
g_object_new (GST_TYPE_RTSP_CLIENT_SINK_PAD, "direction", GST_PAD_SINK,
"template", pad_tmpl, "name", name, NULL);
gst_ghost_pad_construct (GST_GHOST_PAD_CAST (ret));
return GST_PAD (ret);
}
GST_DEBUG_CATEGORY_STATIC (rtsp_client_sink_debug);
#define GST_CAT_DEFAULT (rtsp_client_sink_debug)
static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS_ANY); /* Actual caps come from available set of payloaders */
enum
{
SIGNAL_HANDLE_REQUEST,
SIGNAL_NEW_MANAGER,
SIGNAL_NEW_PAYLOADER,
SIGNAL_REQUEST_RTCP_KEY,
SIGNAL_ACCEPT_CERTIFICATE,
LAST_SIGNAL
};
enum _GstRTSPClientSinkNtpTimeSource
{
NTP_TIME_SOURCE_NTP,
NTP_TIME_SOURCE_UNIX,
NTP_TIME_SOURCE_RUNNING_TIME,
NTP_TIME_SOURCE_CLOCK_TIME
};
#define GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE (gst_rtsp_client_sink_ntp_time_source_get_type())
static GType
gst_rtsp_client_sink_ntp_time_source_get_type (void)
{
static GType ntp_time_source_type = 0;
static const GEnumValue ntp_time_source_values[] = {
{NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"},
{NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"},
{NTP_TIME_SOURCE_RUNNING_TIME,
"Running time based on pipeline clock",
"running-time"},
{NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"},
{0, NULL, NULL},
};
if (!ntp_time_source_type) {
ntp_time_source_type =
g_enum_register_static ("GstRTSPClientSinkNtpTimeSource",
ntp_time_source_values);
}
return ntp_time_source_type;
}
#define DEFAULT_LOCATION NULL
#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
#define DEFAULT_DEBUG FALSE
#define DEFAULT_RETRY 20
#define DEFAULT_TIMEOUT 5000000
#define DEFAULT_UDP_BUFFER_SIZE 0x80000
#define DEFAULT_TCP_TIMEOUT 20000000
#define DEFAULT_LATENCY_MS 2000
#define DEFAULT_DO_RTSP_KEEP_ALIVE TRUE
#define DEFAULT_PROXY NULL
#define DEFAULT_RTP_BLOCKSIZE 0
#define DEFAULT_USER_ID NULL
#define DEFAULT_USER_PW NULL
#define DEFAULT_PORT_RANGE NULL
#define DEFAULT_UDP_RECONNECT TRUE
#define DEFAULT_MULTICAST_IFACE NULL
#define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL
#define DEFAULT_TLS_DATABASE NULL
#define DEFAULT_TLS_INTERACTION NULL
#define DEFAULT_NTP_TIME_SOURCE NTP_TIME_SOURCE_NTP
#define DEFAULT_USER_AGENT "GStreamer/" PACKAGE_VERSION
#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
#define DEFAULT_RTX_TIME_MS 500
enum
{
PROP_0,
PROP_LOCATION,
PROP_PROTOCOLS,
PROP_DEBUG,
PROP_RETRY,
PROP_TIMEOUT,
PROP_TCP_TIMEOUT,
PROP_LATENCY,
PROP_RTX_TIME,
PROP_DO_RTSP_KEEP_ALIVE,
PROP_PROXY,
PROP_PROXY_ID,
PROP_PROXY_PW,
PROP_RTP_BLOCKSIZE,
PROP_USER_ID,
PROP_USER_PW,
PROP_PORT_RANGE,
PROP_UDP_BUFFER_SIZE,
PROP_UDP_RECONNECT,
PROP_MULTICAST_IFACE,
PROP_SDES,
PROP_TLS_VALIDATION_FLAGS,
PROP_TLS_DATABASE,
PROP_TLS_INTERACTION,
PROP_NTP_TIME_SOURCE,
PROP_USER_AGENT,
PROP_PROFILES
};
static void gst_rtsp_client_sink_finalize (GObject * object);
static void gst_rtsp_client_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtsp_client_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstClock *gst_rtsp_client_sink_provide_clock (GstElement * element);
static void gst_rtsp_client_sink_uri_handler_init (gpointer g_iface,
gpointer iface_data);
static gboolean gst_rtsp_client_sink_set_proxy (GstRTSPClientSink * rtsp,
const gchar * proxy);
static void gst_rtsp_client_sink_set_tcp_timeout (GstRTSPClientSink *
rtsp_client_sink, guint64 timeout);
static GstStateChangeReturn gst_rtsp_client_sink_change_state (GstElement *
element, GstStateChange transition);
static void gst_rtsp_client_sink_handle_message (GstBin * bin,
GstMessage * message);
static gboolean gst_rtsp_client_sink_setup_auth (GstRTSPClientSink * sink,
GstRTSPMessage * response);
static gboolean gst_rtsp_client_sink_loop_send_cmd (GstRTSPClientSink * sink,
gint cmd, gint mask);
static GstRTSPResult gst_rtsp_client_sink_open (GstRTSPClientSink * sink,
gboolean async);
static GstRTSPResult gst_rtsp_client_sink_record (GstRTSPClientSink * sink,
gboolean async);
static GstRTSPResult gst_rtsp_client_sink_pause (GstRTSPClientSink * sink,
gboolean async);
static GstRTSPResult gst_rtsp_client_sink_close (GstRTSPClientSink * sink,
gboolean async, gboolean only_close);
static gboolean gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink);
static gboolean gst_rtsp_client_sink_uri_set_uri (GstURIHandler * handler,
const gchar * uri, GError ** error);
static gchar *gst_rtsp_client_sink_uri_get_uri (GstURIHandler * handler);
static gboolean gst_rtsp_client_sink_loop (GstRTSPClientSink * sink);
static void gst_rtsp_client_sink_connection_flush (GstRTSPClientSink * sink,
gboolean flush);
static GstPad *gst_rtsp_client_sink_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
static void gst_rtsp_client_sink_release_pad (GstElement * element,
GstPad * pad);
/* commands we send to out loop to notify it of events */
#define CMD_OPEN (1 << 0)
#define CMD_RECORD (1 << 1)
#define CMD_PAUSE (1 << 2)
#define CMD_CLOSE (1 << 3)
#define CMD_WAIT (1 << 4)
#define CMD_RECONNECT (1 << 5)
#define CMD_LOOP (1 << 6)
/* mask for all commands */
#define CMD_ALL ((CMD_LOOP << 1) - 1)
#define GST_ELEMENT_PROGRESS(el, type, code, text) \
G_STMT_START { \
gchar *__txt = _gst_element_error_printf text; \
gst_element_post_message (GST_ELEMENT_CAST (el), \
gst_message_new_progress (GST_OBJECT_CAST (el), \
GST_PROGRESS_TYPE_ ##type, code, __txt)); \
g_free (__txt); \
} G_STMT_END
static guint gst_rtsp_client_sink_signals[LAST_SIGNAL] = { 0 };
/*********************************
* GstChildProxy implementation *
*********************************/
static GObject *
gst_rtsp_client_sink_child_proxy_get_child_by_index (GstChildProxy *
child_proxy, guint index)
{
GObject *obj;
GstRTSPClientSink *cs = GST_RTSP_CLIENT_SINK (child_proxy);
GST_OBJECT_LOCK (cs);
if ((obj = g_list_nth_data (GST_ELEMENT (cs)->sinkpads, index)))
g_object_ref (obj);
GST_OBJECT_UNLOCK (cs);
return obj;
}
static guint
gst_rtsp_client_sink_child_proxy_get_children_count (GstChildProxy *
child_proxy)
{
guint count = 0;
GST_OBJECT_LOCK (child_proxy);
count = GST_ELEMENT (child_proxy)->numsinkpads;
GST_OBJECT_UNLOCK (child_proxy);
GST_INFO_OBJECT (child_proxy, "Children Count: %d", count);
return count;
}
static void
gst_rtsp_client_sink_child_proxy_init (gpointer g_iface, gpointer iface_data)
{
GstChildProxyInterface *iface = g_iface;
GST_INFO ("intializing child proxy interface");
iface->get_child_by_index =
gst_rtsp_client_sink_child_proxy_get_child_by_index;
iface->get_children_count =
gst_rtsp_client_sink_child_proxy_get_children_count;
}
#define gst_rtsp_client_sink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstRTSPClientSink, gst_rtsp_client_sink, GST_TYPE_BIN,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
gst_rtsp_client_sink_uri_handler_init);
G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY,
gst_rtsp_client_sink_child_proxy_init);
);
#ifndef GST_DISABLE_GST_DEBUG
static inline const gchar *
cmd_to_string (guint cmd)
{
switch (cmd) {
case CMD_OPEN:
return "OPEN";
case CMD_RECORD:
return "RECORD";
case CMD_PAUSE:
return "PAUSE";
case CMD_CLOSE:
return "CLOSE";
case CMD_WAIT:
return "WAIT";
case CMD_RECONNECT:
return "RECONNECT";
case CMD_LOOP:
return "LOOP";
}
return "unknown";
}
#endif
static void
gst_rtsp_client_sink_class_init (GstRTSPClientSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBinClass *gstbin_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbin_class = (GstBinClass *) klass;
GST_DEBUG_CATEGORY_INIT (rtsp_client_sink_debug, "rtspclientsink", 0,
"RTSP sink element");
gobject_class->set_property = gst_rtsp_client_sink_set_property;
gobject_class->get_property = gst_rtsp_client_sink_get_property;
gobject_class->finalize = gst_rtsp_client_sink_finalize;
g_object_class_install_property (gobject_class, PROP_LOCATION,
g_param_spec_string ("location", "RTSP Location",
"Location of the RTSP url to read",
DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
g_param_spec_flags ("protocols", "Protocols",
"Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PROFILES,
g_param_spec_flags ("profiles", "Profiles",
"Allowed RTSP profiles", GST_TYPE_RTSP_PROFILE,
DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DEBUG,
g_param_spec_boolean ("debug", "Debug",
"Dump request and response messages to stdout",
DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RETRY,
g_param_spec_uint ("retry", "Retry",
"Max number of retries when allocating RTP ports.",
0, G_MAXUINT16, DEFAULT_RETRY,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TIMEOUT,
g_param_spec_uint64 ("timeout", "Timeout",
"Retry TCP transport after UDP timeout microseconds (0 = disabled)",
0, G_MAXUINT64, DEFAULT_TIMEOUT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
"Fail after timeout microseconds on TCP connections (0 = disabled)",
0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LATENCY,
g_param_spec_uint ("latency", "Buffer latency in ms",
"Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTX_TIME,
g_param_spec_uint ("rtx-time", "Retransmission buffer in ms",
"Amount of ms to buffer for retransmission. 0 disables retransmission",
0, G_MAXUINT, DEFAULT_RTX_TIME_MS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:do-rtsp-keep-alive:
*
* Enable RTSP keep alive support. Some old server don't like RTSP
* keep alive and then this property needs to be set to FALSE.
*/
g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
"Send RTSP keep alive packets, disable for old incompatible server.",
DEFAULT_DO_RTSP_KEEP_ALIVE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:proxy:
*
* Set the proxy parameters. This has to be a string of the format
* [http://][user:passwd@]host[:port].
*/
g_object_class_install_property (gobject_class, PROP_PROXY,
g_param_spec_string ("proxy", "Proxy",
"Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:proxy-id:
*
* Sets the proxy URI user id for authentication. If the URI set via the
* "proxy" property contains a user-id already, that will take precedence.
*
*/
g_object_class_install_property (gobject_class, PROP_PROXY_ID,
g_param_spec_string ("proxy-id", "proxy-id",
"HTTP proxy URI user id for authentication", "",
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:proxy-pw:
*
* Sets the proxy URI password for authentication. If the URI set via the
* "proxy" property contains a password already, that will take precedence.
*
*/
g_object_class_install_property (gobject_class, PROP_PROXY_PW,
g_param_spec_string ("proxy-pw", "proxy-pw",
"HTTP proxy URI user password for authentication", "",
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:rtp-blocksize:
*
* RTP package size to suggest to server.
*/
g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
"RTP package size to suggest to server (0 = disabled)",
0, 65536, DEFAULT_RTP_BLOCKSIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_USER_ID,
g_param_spec_string ("user-id", "user-id",
"RTSP location URI user id for authentication", DEFAULT_USER_ID,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_USER_PW,
g_param_spec_string ("user-pw", "user-pw",
"RTSP location URI user password for authentication", DEFAULT_USER_PW,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:port-range:
*
* Configure the client port numbers that can be used to receive
* RTCP.
*/
g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
g_param_spec_string ("port-range", "Port range",
"Client port range that can be used to receive RTCP data, "
"eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink:udp-buffer-size:
*
* Size of the kernel UDP receive buffer in bytes.
*/
g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
"Size of the kernel UDP receive buffer in bytes, 0=default",
0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
"Reconnect to the server if RTSP connection is closed when doing UDP",
DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
g_param_spec_string ("multicast-iface", "Multicast Interface",
"The network interface on which to join the multicast group",
DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SDES,
g_param_spec_boxed ("sdes", "SDES",
"The SDES items of this session",
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink::tls-validation-flags:
*
* TLS certificate validation flags used to validate server
* certificate.
*
*/
g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
"TLS certificate validation flags used to validate the server certificate",
G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink::tls-database:
*
* TLS database with anchor certificate authorities used to validate
* the server certificate.
*
*/
g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
g_param_spec_object ("tls-database", "TLS database",
"TLS database with anchor certificate authorities used to validate the server certificate",
G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink::tls-interaction:
*
* A #GTlsInteraction object to be used when the connection or certificate
* database need to interact with the user. This will be used to prompt the
* user for passwords where necessary.
*
*/
g_object_class_install_property (gobject_class, PROP_TLS_INTERACTION,
g_param_spec_object ("tls-interaction", "TLS interaction",
"A GTlsInteraction object to prompt the user for password or certificate",
G_TYPE_TLS_INTERACTION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink::ntp-time-source:
*
* allows to select the time source that should be used
* for the NTP time in outgoing packets
*
*/
g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
g_param_spec_enum ("ntp-time-source", "NTP Time Source",
"NTP time source for RTCP packets",
GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE, DEFAULT_NTP_TIME_SOURCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink::user-agent:
*
* The string to set in the User-Agent header.
*
*/
g_object_class_install_property (gobject_class, PROP_USER_AGENT,
g_param_spec_string ("user-agent", "User Agent",
"The User-Agent string to send to the server",
DEFAULT_USER_AGENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPClientSink::handle-request:
* @rtsp_client_sink: a #GstRTSPClientSink
* @request: a #GstRTSPMessage
* @response: a #GstRTSPMessage
*
* Handle a server request in @request and prepare @response.
*
* This signal is called from the streaming thread, you should therefore not
* do any state changes on @rtsp_client_sink because this might deadlock. If you want
* to modify the state as a result of this signal, post a
* #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
* in some other way.
*
*/
gst_rtsp_client_sink_signals[SIGNAL_HANDLE_REQUEST] =
g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
G_TYPE_POINTER, G_TYPE_POINTER);
/**
* GstRTSPClientSink::new-manager:
* @rtsp_client_sink: a #GstRTSPClientSink
* @manager: a #GstElement
*
* Emitted after a new manager (like rtpbin) was created and the default
* properties were configured.
*
*/
gst_rtsp_client_sink_signals[SIGNAL_NEW_MANAGER] =
g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
/**
* GstRTSPClientSink::new-payloader:
* @rtsp_client_sink: a #GstRTSPClientSink
* @payloader: a #GstElement
*
* Emitted after a new RTP payloader was created and the default
* properties were configured.
*
*/
gst_rtsp_client_sink_signals[SIGNAL_NEW_PAYLOADER] =
g_signal_new_class_handler ("new-payloader", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
/**
* GstRTSPClientSink::request-rtcp-key:
* @rtsp_client_sink: a #GstRTSPClientSink
* @num: the stream number
*
* Signal emitted to get the crypto parameters relevant to the RTCP
* stream. User should provide the key and the RTCP encryption ciphers
* and authentication, and return them wrapped in a GstCaps.
*
*/
gst_rtsp_client_sink_signals[SIGNAL_REQUEST_RTCP_KEY] =
g_signal_new ("request-rtcp-key", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
/**
* GstRTSPClientSink::accept-certificate:
* @rtsp_client_sink: a #GstRTSPClientSink
* @peer_cert: the peer's #GTlsCertificate
* @errors: the problems with @peer_cert
* @user_data: user data set when the signal handler was connected.
*
* This will directly map to #GTlsConnection 's "accept-certificate"
* signal and be performed after the default checks of #GstRTSPConnection
* (checking against the #GTlsDatabase with the given #GTlsCertificateFlags)
* have failed. If no #GTlsDatabase is set on this connection, only this
* signal will be emitted.
*
* Since: 1.14
*/
gst_rtsp_client_sink_signals[SIGNAL_ACCEPT_CERTIFICATE] =
g_signal_new ("accept-certificate", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, g_signal_accumulator_true_handled, NULL, NULL,
G_TYPE_BOOLEAN, 3, G_TYPE_TLS_CONNECTION, G_TYPE_TLS_CERTIFICATE,
G_TYPE_TLS_CERTIFICATE_FLAGS);
gstelement_class->provide_clock = gst_rtsp_client_sink_provide_clock;
gstelement_class->change_state = gst_rtsp_client_sink_change_state;
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_request_new_pad);
gstelement_class->release_pad =
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_release_pad);
gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
&rtptemplate, GST_TYPE_RTSP_CLIENT_SINK_PAD);
gst_element_class_set_static_metadata (gstelement_class,
"RTSP RECORD client", "Sink/Network",
"Send data over the network via RTSP RECORD(RFC 2326)",
"Jan Schmidt <jan@centricular.com>");
gstbin_class->handle_message = gst_rtsp_client_sink_handle_message;
}
static void
gst_rtsp_client_sink_init (GstRTSPClientSink * sink)
{
sink->conninfo.location = g_strdup (DEFAULT_LOCATION);
sink->protocols = DEFAULT_PROTOCOLS;
sink->debug = DEFAULT_DEBUG;
sink->retry = DEFAULT_RETRY;
sink->udp_timeout = DEFAULT_TIMEOUT;
gst_rtsp_client_sink_set_tcp_timeout (sink, DEFAULT_TCP_TIMEOUT);
sink->latency = DEFAULT_LATENCY_MS;
sink->rtx_time = DEFAULT_RTX_TIME_MS;
sink->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
gst_rtsp_client_sink_set_proxy (sink, DEFAULT_PROXY);
sink->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
sink->user_id = g_strdup (DEFAULT_USER_ID);
sink->user_pw = g_strdup (DEFAULT_USER_PW);
sink->client_port_range.min = 0;
sink->client_port_range.max = 0;
sink->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
sink->udp_reconnect = DEFAULT_UDP_RECONNECT;
sink->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
sink->sdes = NULL;
sink->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
sink->tls_database = DEFAULT_TLS_DATABASE;
sink->tls_interaction = DEFAULT_TLS_INTERACTION;
sink->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
sink->user_agent = g_strdup (DEFAULT_USER_AGENT);
sink->profiles = DEFAULT_PROFILES;
/* protects the streaming thread in interleaved mode or the polling
* thread in UDP mode. */
g_rec_mutex_init (&sink->stream_rec_lock);
/* protects our state changes from multiple invocations */
g_rec_mutex_init (&sink->state_rec_lock);
g_mutex_init (&sink->send_lock);
g_mutex_init (&sink->preroll_lock);
g_cond_init (&sink->preroll_cond);
sink->state = GST_RTSP_STATE_INVALID;
g_mutex_init (&sink->conninfo.send_lock);
g_mutex_init (&sink->conninfo.recv_lock);
g_mutex_init (&sink->block_streams_lock);
g_cond_init (&sink->block_streams_cond);
g_mutex_init (&sink->open_conn_lock);
g_cond_init (&sink->open_conn_cond);
sink->internal_bin = (GstBin *) gst_bin_new ("rtspbin");
gst_element_set_locked_state (GST_ELEMENT_CAST (sink->internal_bin), TRUE);
gst_bin_add (GST_BIN (sink), GST_ELEMENT_CAST (sink->internal_bin));
sink->next_dyn_pt = 96;
gst_sdp_message_init (&sink->cursdp);
GST_OBJECT_FLAG_SET (sink, GST_ELEMENT_FLAG_SINK);
}
static void
gst_rtsp_client_sink_finalize (GObject * object)
{
GstRTSPClientSink *rtsp_client_sink;
rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
gst_sdp_message_uninit (&rtsp_client_sink->cursdp);
g_free (rtsp_client_sink->conninfo.location);
gst_rtsp_url_free (rtsp_client_sink->conninfo.url);
g_free (rtsp_client_sink->conninfo.url_str);
g_free (rtsp_client_sink->user_id);
g_free (rtsp_client_sink->user_pw);
g_free (rtsp_client_sink->multi_iface);
g_free (rtsp_client_sink->user_agent);
if (rtsp_client_sink->uri_sdp) {
gst_sdp_message_free (rtsp_client_sink->uri_sdp);
rtsp_client_sink->uri_sdp = NULL;
}
if (rtsp_client_sink->provided_clock)
gst_object_unref (rtsp_client_sink->provided_clock);
if (rtsp_client_sink->sdes)
gst_structure_free (rtsp_client_sink->sdes);
if (rtsp_client_sink->tls_database)
g_object_unref (rtsp_client_sink->tls_database);
if (rtsp_client_sink->tls_interaction)
g_object_unref (rtsp_client_sink->tls_interaction);
/* free locks */
g_rec_mutex_clear (&rtsp_client_sink->stream_rec_lock);
g_rec_mutex_clear (&rtsp_client_sink->state_rec_lock);
g_mutex_clear (&rtsp_client_sink->conninfo.send_lock);
g_mutex_clear (&rtsp_client_sink->conninfo.recv_lock);
g_mutex_clear (&rtsp_client_sink->send_lock);
g_mutex_clear (&rtsp_client_sink->preroll_lock);
g_cond_clear (&rtsp_client_sink->preroll_cond);
g_mutex_clear (&rtsp_client_sink->block_streams_lock);
g_cond_clear (&rtsp_client_sink->block_streams_cond);
g_mutex_clear (&rtsp_client_sink->open_conn_lock);
g_cond_clear (&rtsp_client_sink->open_conn_cond);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_payloader_filter_func (GstPluginFeature * feature, gpointer user_data)
{
GstElementFactory *factory = NULL;
const gchar *klass;
if (!GST_IS_ELEMENT_FACTORY (feature))
return FALSE;
factory = GST_ELEMENT_FACTORY (feature);
if (gst_plugin_feature_get_rank (feature) == GST_RANK_NONE)
return FALSE;
if (!gst_element_factory_list_is_type (factory,
GST_ELEMENT_FACTORY_TYPE_PAYLOADER))
return FALSE;
klass =
gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_KLASS);
if (strstr (klass, "Codec") == NULL)
return FALSE;
if (strstr (klass, "RTP") == NULL)
return FALSE;
return TRUE;
}
static gint
compare_ranks (GstPluginFeature * f1, GstPluginFeature * f2)
{
gint diff;
const gchar *rname1, *rname2;
GstRank rank1, rank2;
rname1 = gst_plugin_feature_get_name (f1);
rname2 = gst_plugin_feature_get_name (f2);
rank1 = gst_plugin_feature_get_rank (f1);
rank2 = gst_plugin_feature_get_rank (f2);
/* HACK: Prefer rtpmp4apay over rtpmp4gpay */
if (g_str_equal (rname1, "rtpmp4apay"))
rank1 = GST_RANK_SECONDARY + 1;
if (g_str_equal (rname2, "rtpmp4apay"))
rank2 = GST_RANK_SECONDARY + 1;
diff = rank2 - rank1;
if (diff != 0)
return diff;
diff = strcmp (rname2, rname1);
return diff;
}
static GList *
gst_rtsp_client_sink_get_factories (void)
{
static GList *payloader_factories = NULL;
if (g_once_init_enter (&payloader_factories)) {
GList *all_factories;
all_factories =
gst_registry_feature_filter (gst_registry_get (),
gst_rtp_payloader_filter_func, FALSE, NULL);
all_factories = g_list_sort (all_factories, (GCompareFunc) compare_ranks);
g_once_init_leave (&payloader_factories, all_factories);
}
return payloader_factories;
}
static GstCaps *
gst_rtsp_client_sink_get_payloader_caps (GstElementFactory * factory)
{
const GList *tmp;
GstCaps *caps = gst_caps_new_empty ();
for (tmp = gst_element_factory_get_static_pad_templates (factory);
tmp; tmp = g_list_next (tmp)) {
GstStaticPadTemplate *template = tmp->data;
if (template->direction == GST_PAD_SINK) {
GstCaps *static_caps = gst_static_pad_template_get_caps (template);
GST_LOG ("Found pad template %s on factory %s",
template->name_template, gst_plugin_feature_get_name (factory));
if (static_caps)
caps = gst_caps_merge (caps, static_caps);
/* Early out, any is absorbing */
if (gst_caps_is_any (caps))
goto out;
}
}
out:
return caps;
}
static GstCaps *
gst_rtsp_client_sink_get_all_payloaders_caps (void)
{
/* Cached caps result */
static GstCaps *ret;
if (g_once_init_enter (&ret)) {
GList *factories, *cur;
GstCaps *caps = gst_caps_new_empty ();
factories = gst_rtsp_client_sink_get_factories ();
for (cur = factories; cur != NULL; cur = g_list_next (cur)) {
GstElementFactory *factory = GST_ELEMENT_FACTORY (cur->data);
GstCaps *payloader_caps =
gst_rtsp_client_sink_get_payloader_caps (factory);
caps = gst_caps_merge (caps, payloader_caps);
/* Early out, any is absorbing */
if (gst_caps_is_any (caps))
goto out;
}
out:
g_once_init_leave (&ret, caps);
}
/* Return cached result */
return gst_caps_ref (ret);
}
static GstElement *
gst_rtsp_client_sink_make_payloader (GstCaps * caps)
{
GList *factories, *cur;
factories = gst_rtsp_client_sink_get_factories ();
for (cur = factories; cur != NULL; cur = g_list_next (cur)) {
GstElementFactory *factory = GST_ELEMENT_FACTORY (cur->data);
const GList *tmp;
for (tmp = gst_element_factory_get_static_pad_templates (factory);
tmp; tmp = g_list_next (tmp)) {
GstStaticPadTemplate *template = tmp->data;
if (template->direction == GST_PAD_SINK) {
GstCaps *static_caps = gst_static_pad_template_get_caps (template);
GstElement *payloader = NULL;
if (gst_caps_can_intersect (static_caps, caps)) {
GST_DEBUG ("caps %" GST_PTR_FORMAT " intersects with template %"
GST_PTR_FORMAT " for payloader %s", caps, static_caps,
gst_plugin_feature_get_name (factory));
payloader = gst_element_factory_create (factory, NULL);
}
gst_caps_unref (static_caps);
if (payloader)
return payloader;
}
}
}
return NULL;
}
static GstRTSPStream *
gst_rtsp_client_sink_create_stream (GstRTSPClientSink * sink,
GstRTSPStreamContext * context, GstElement * payloader, GstPad * pad)
{
GstRTSPStream *stream = NULL;
guint pt, aux_pt;
GST_OBJECT_LOCK (sink);
g_object_get (G_OBJECT (payloader), "pt", &pt, NULL);
if (pt >= 96 && pt <= sink->next_dyn_pt) {
/* Payloader has a dynamic PT, but one that's already used */
/* FIXME: Create a caps->ptmap instead? */
pt = sink->next_dyn_pt;
if (pt > 127)
goto no_free_pt;
GST_DEBUG_OBJECT (sink, "Assigning pt %u to stream %d", pt, context->index);
sink->next_dyn_pt++;
} else {
GST_DEBUG_OBJECT (sink, "Keeping existing pt %u for stream %d",
pt, context->index);
}
aux_pt = sink->next_dyn_pt;
if (aux_pt > 127)
goto no_free_pt;
sink->next_dyn_pt++;
GST_OBJECT_UNLOCK (sink);
g_object_set (G_OBJECT (payloader), "pt", pt, NULL);
stream = gst_rtsp_stream_new (context->index, payloader, pad);
gst_rtsp_stream_set_client_side (stream, TRUE);
gst_rtsp_stream_set_retransmission_time (stream,
(GstClockTime) (sink->rtx_time) * GST_MSECOND);
gst_rtsp_stream_set_protocols (stream, sink->protocols);
gst_rtsp_stream_set_profiles (stream, sink->profiles);
gst_rtsp_stream_set_retransmission_pt (stream, aux_pt);
gst_rtsp_stream_set_buffer_size (stream, sink->udp_buffer_size);
if (sink->rtp_blocksize > 0)
gst_rtsp_stream_set_mtu (stream, sink->rtp_blocksize);
gst_rtsp_stream_set_multicast_iface (stream, sink->multi_iface);
#if 0
if (priv->pool)
gst_rtsp_stream_set_address_pool (stream, priv->pool);
#endif
return stream;
no_free_pt:
GST_OBJECT_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, NO_SPACE_LEFT, (NULL),
("Ran out of dynamic payload types."));
return NULL;
}
static GstPadProbeReturn
handle_payloader_block (GstPad * pad, GstPadProbeInfo * info,
GstRTSPStreamContext * context)
{
GstRTSPClientSink *sink = context->parent;
GST_INFO_OBJECT (sink, "Block on pad %" GST_PTR_FORMAT, pad);
g_mutex_lock (&sink->preroll_lock);
context->prerolled = TRUE;
g_cond_broadcast (&sink->preroll_cond);
g_mutex_unlock (&sink->preroll_lock);
GST_INFO_OBJECT (sink, "Announced preroll on pad %" GST_PTR_FORMAT, pad);
return GST_PAD_PROBE_OK;
}
static gboolean
gst_rtsp_client_sink_setup_payloader (GstRTSPClientSink * sink, GstPad * pad,
GstCaps * caps)
{
GstRTSPStreamContext *context;
GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad);
GstElement *payloader;
GstPad *sinkpad, *srcpad, *ghostsink;
context = gst_pad_get_element_private (pad);
if (cspad->custom_payloader) {
payloader = cspad->custom_payloader;
} else {
/* Find the payloader. */
payloader = gst_rtsp_client_sink_make_payloader (caps);
}
if (payloader == NULL)
return FALSE;
GST_DEBUG_OBJECT (sink, "Configuring payloader %" GST_PTR_FORMAT
" for pad %" GST_PTR_FORMAT, payloader, pad);
sinkpad = gst_element_get_static_pad (payloader, "sink");
if (sinkpad == NULL)
goto no_sinkpad;
srcpad = gst_element_get_static_pad (payloader, "src");
if (srcpad == NULL)
goto no_srcpad;
gst_bin_add (GST_BIN (sink->internal_bin), payloader);
ghostsink = gst_ghost_pad_new (NULL, sinkpad);
gst_pad_set_active (ghostsink, TRUE);
gst_element_add_pad (GST_ELEMENT (sink->internal_bin), ghostsink);
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_NEW_PAYLOADER], 0,
payloader);
GST_RTSP_STATE_LOCK (sink);
context->payloader_block_id =
gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM,
(GstPadProbeCallback) handle_payloader_block, context, NULL);
context->payloader = payloader;
payloader = gst_object_ref (payloader);
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), ghostsink);
gst_object_unref (GST_OBJECT (sinkpad));
GST_RTSP_STATE_UNLOCK (sink);
gst_element_sync_state_with_parent (payloader);
gst_object_unref (payloader);
gst_object_unref (GST_OBJECT (srcpad));
return TRUE;
no_sinkpad:
GST_ERROR_OBJECT (sink,
"Could not find sink pad on payloader %" GST_PTR_FORMAT, payloader);
if (!cspad->custom_payloader)
gst_object_unref (payloader);
return FALSE;
no_srcpad:
GST_ERROR_OBJECT (sink,
"Could not find src pad on payloader %" GST_PTR_FORMAT, payloader);
gst_object_unref (GST_OBJECT (sinkpad));
gst_object_unref (payloader);
return TRUE;
}
static gboolean
gst_rtsp_client_sink_sinkpad_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
if (target == NULL) {
GstCaps *caps;
/* No target yet - choose a payloader and configure it */
gst_event_parse_caps (event, &caps);
GST_DEBUG_OBJECT (parent,
"Have set caps event on pad %" GST_PTR_FORMAT
" caps %" GST_PTR_FORMAT, pad, caps);
if (!gst_rtsp_client_sink_setup_payloader (GST_RTSP_CLIENT_SINK (parent),
pad, caps)) {
GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad);
GST_ELEMENT_ERROR (parent, CORE, NEGOTIATION,
("Could not create payloader"),
("Custom payloader: %p, caps: %" GST_PTR_FORMAT,
cspad->custom_payloader, caps));
gst_event_unref (event);
return FALSE;
}
} else {
gst_object_unref (target);
}
}
return gst_pad_event_default (pad, parent, event);
}
static gboolean
gst_rtsp_client_sink_sinkpad_query (GstPad * pad, GstObject * parent,
GstQuery * query)
{
if (GST_QUERY_TYPE (query) == GST_QUERY_CAPS) {
GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
if (target == NULL) {
GstRtspClientSinkPad *cspad = GST_RTSP_CLIENT_SINK_PAD (pad);
GstCaps *caps;
if (cspad->custom_payloader) {
GstPad *sinkpad =
gst_element_get_static_pad (cspad->custom_payloader, "sink");
if (sinkpad) {
caps = gst_pad_query_caps (sinkpad, NULL);
gst_object_unref (sinkpad);
} else {
GST_ELEMENT_ERROR (parent, CORE, NEGOTIATION, (NULL),
("Custom payloaders are expected to expose a sink pad named 'sink'"));
return FALSE;
}
} else {
/* No target yet - return the union of all payloader caps */
caps = gst_rtsp_client_sink_get_all_payloaders_caps ();
}
GST_TRACE_OBJECT (parent, "Returning payloader caps %" GST_PTR_FORMAT,
caps);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
return TRUE;
}
gst_object_unref (target);
}
return gst_pad_query_default (pad, parent, query);
}
static GstPad *
gst_rtsp_client_sink_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
{
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
GstPad *pad;
GstRTSPStreamContext *context;
guint idx = (guint) - 1;
gchar *tmpname;
g_mutex_lock (&sink->preroll_lock);
if (sink->streams_collected) {
GST_WARNING_OBJECT (element, "Can't add streams to a running session");
g_mutex_unlock (&sink->preroll_lock);
return NULL;
}
g_mutex_unlock (&sink->preroll_lock);
GST_OBJECT_LOCK (sink);
if (name) {
if (!sscanf (name, "sink_%u", &idx)) {
GST_OBJECT_UNLOCK (sink);
GST_ERROR_OBJECT (element, "Invalid sink pad name %s", name);
return NULL;
}
if (idx >= sink->next_pad_id)
sink->next_pad_id = idx + 1;
}
if (idx == (guint) - 1) {
idx = sink->next_pad_id;
sink->next_pad_id++;
}
GST_OBJECT_UNLOCK (sink);
tmpname = g_strdup_printf ("sink_%u", idx);
pad = gst_rtsp_client_sink_pad_new (templ, tmpname);
g_free (tmpname);
GST_DEBUG_OBJECT (element, "Creating request pad %" GST_PTR_FORMAT, pad);
gst_pad_set_event_function (pad,
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_sinkpad_event));
gst_pad_set_query_function (pad,
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_sinkpad_query));
context = g_new0 (GstRTSPStreamContext, 1);
context->parent = sink;
context->index = idx;
gst_pad_set_element_private (pad, context);
/* The rest of the context is configured on a caps set */
gst_pad_set_active (pad, TRUE);
gst_element_add_pad (element, pad);
gst_child_proxy_child_added (GST_CHILD_PROXY (element), G_OBJECT (pad),
GST_PAD_NAME (pad));
(void) gst_rtsp_client_sink_get_factories ();
g_mutex_init (&context->conninfo.send_lock);
g_mutex_init (&context->conninfo.recv_lock);
GST_RTSP_STATE_LOCK (sink);
sink->contexts = g_list_prepend (sink->contexts, context);
GST_RTSP_STATE_UNLOCK (sink);
return pad;
}
static void
gst_rtsp_client_sink_release_pad (GstElement * element, GstPad * pad)
{
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
GstRTSPStreamContext *context;
context = gst_pad_get_element_private (pad);
GST_RTSP_STATE_LOCK (sink);
sink->contexts = g_list_remove (sink->contexts, context);
GST_RTSP_STATE_UNLOCK (sink);
/* FIXME: Shut down and clean up streaming on this pad,
* do teardown if needed */
GST_LOG_OBJECT (sink,
"Cleaning up payloader and stream for released pad %" GST_PTR_FORMAT,
pad);
if (context->stream_transport) {
gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
gst_object_unref (context->stream_transport);
context->stream_transport = NULL;
}
if (context->stream) {
if (context->joined) {
gst_rtsp_stream_leave_bin (context->stream,
GST_BIN (sink->internal_bin), sink->rtpbin);
context->joined = FALSE;
}
gst_object_unref (context->stream);
context->stream = NULL;
}
if (context->srtcpparams)
gst_caps_unref (context->srtcpparams);
g_free (context->conninfo.location);
context->conninfo.location = NULL;
g_mutex_clear (&context->conninfo.send_lock);
g_mutex_clear (&context->conninfo.recv_lock);
g_free (context);
gst_element_remove_pad (element, pad);
}
static GstClock *
gst_rtsp_client_sink_provide_clock (GstElement * element)
{
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
GstClock *clock;
if ((clock = sink->provided_clock) != NULL)
gst_object_ref (clock);
return clock;
}
/* a proxy string of the format [user:passwd@]host[:port] */
static gboolean
gst_rtsp_client_sink_set_proxy (GstRTSPClientSink * rtsp, const gchar * proxy)
{
gchar *p, *at, *col;
g_free (rtsp->proxy_user);
rtsp->proxy_user = NULL;
g_free (rtsp->proxy_passwd);
rtsp->proxy_passwd = NULL;
g_free (rtsp->proxy_host);
rtsp->proxy_host = NULL;
rtsp->proxy_port = 0;
p = (gchar *) proxy;
if (p == NULL)
return TRUE;
/* we allow http:// in front but ignore it */
if (g_str_has_prefix (p, "http://"))
p += 7;
at = strchr (p, '@');
if (at) {
/* look for user:passwd */
col = strchr (proxy, ':');
if (col == NULL || col > at)
return FALSE;
rtsp->proxy_user = g_strndup (p, col - p);
col++;
rtsp->proxy_passwd = g_strndup (col, at - col);
/* move to host */
p = at + 1;
} else {
if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
}
}
col = strchr (p, ':');
if (col) {
/* everything before the colon is the hostname */
rtsp->proxy_host = g_strndup (p, col - p);
p = col + 1;
rtsp->proxy_port = strtoul (p, (char **) &p, 10);
} else {
rtsp->proxy_host = g_strdup (p);
rtsp->proxy_port = 8080;
}
return TRUE;
}
static void
gst_rtsp_client_sink_set_tcp_timeout (GstRTSPClientSink * rtsp_client_sink,
guint64 timeout)
{
rtsp_client_sink->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
rtsp_client_sink->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
if (timeout != 0)
rtsp_client_sink->ptcp_timeout = &rtsp_client_sink->tcp_timeout;
else
rtsp_client_sink->ptcp_timeout = NULL;
}
static void
gst_rtsp_client_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTSPClientSink *rtsp_client_sink;
rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
switch (prop_id) {
case PROP_LOCATION:
gst_rtsp_client_sink_uri_set_uri (GST_URI_HANDLER (rtsp_client_sink),
g_value_get_string (value), NULL);
break;
case PROP_PROTOCOLS:
rtsp_client_sink->protocols = g_value_get_flags (value);
break;
case PROP_PROFILES:
rtsp_client_sink->profiles = g_value_get_flags (value);
break;
case PROP_DEBUG:
rtsp_client_sink->debug = g_value_get_boolean (value);
break;
case PROP_RETRY:
rtsp_client_sink->retry = g_value_get_uint (value);
break;
case PROP_TIMEOUT:
rtsp_client_sink->udp_timeout = g_value_get_uint64 (value);
break;
case PROP_TCP_TIMEOUT:
gst_rtsp_client_sink_set_tcp_timeout (rtsp_client_sink,
g_value_get_uint64 (value));
break;
case PROP_LATENCY:
rtsp_client_sink->latency = g_value_get_uint (value);
break;
case PROP_RTX_TIME:
rtsp_client_sink->rtx_time = g_value_get_uint (value);
break;
case PROP_DO_RTSP_KEEP_ALIVE:
rtsp_client_sink->do_rtsp_keep_alive = g_value_get_boolean (value);
break;
case PROP_PROXY:
gst_rtsp_client_sink_set_proxy (rtsp_client_sink,
g_value_get_string (value));
break;
case PROP_PROXY_ID:
if (rtsp_client_sink->prop_proxy_id)
g_free (rtsp_client_sink->prop_proxy_id);
rtsp_client_sink->prop_proxy_id = g_value_dup_string (value);
break;
case PROP_PROXY_PW:
if (rtsp_client_sink->prop_proxy_pw)
g_free (rtsp_client_sink->prop_proxy_pw);
rtsp_client_sink->prop_proxy_pw = g_value_dup_string (value);
break;
case PROP_RTP_BLOCKSIZE:
rtsp_client_sink->rtp_blocksize = g_value_get_uint (value);
break;
case PROP_USER_ID:
if (rtsp_client_sink->user_id)
g_free (rtsp_client_sink->user_id);
rtsp_client_sink->user_id = g_value_dup_string (value);
break;
case PROP_USER_PW:
if (rtsp_client_sink->user_pw)
g_free (rtsp_client_sink->user_pw);
rtsp_client_sink->user_pw = g_value_dup_string (value);
break;
case PROP_PORT_RANGE:
{
const gchar *str;
str = g_value_get_string (value);
if (!str || !sscanf (str, "%u-%u",
&rtsp_client_sink->client_port_range.min,
&rtsp_client_sink->client_port_range.max)) {
rtsp_client_sink->client_port_range.min = 0;
rtsp_client_sink->client_port_range.max = 0;
}
break;
}
case PROP_UDP_BUFFER_SIZE:
rtsp_client_sink->udp_buffer_size = g_value_get_int (value);
break;
case PROP_UDP_RECONNECT:
rtsp_client_sink->udp_reconnect = g_value_get_boolean (value);
break;
case PROP_MULTICAST_IFACE:
g_free (rtsp_client_sink->multi_iface);
if (g_value_get_string (value) == NULL)
rtsp_client_sink->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
else
rtsp_client_sink->multi_iface = g_value_dup_string (value);
break;
case PROP_SDES:
rtsp_client_sink->sdes = g_value_dup_boxed (value);
break;
case PROP_TLS_VALIDATION_FLAGS:
rtsp_client_sink->tls_validation_flags = g_value_get_flags (value);
break;
case PROP_TLS_DATABASE:
g_clear_object (&rtsp_client_sink->tls_database);
rtsp_client_sink->tls_database = g_value_dup_object (value);
break;
case PROP_TLS_INTERACTION:
g_clear_object (&rtsp_client_sink->tls_interaction);
rtsp_client_sink->tls_interaction = g_value_dup_object (value);
break;
case PROP_NTP_TIME_SOURCE:
rtsp_client_sink->ntp_time_source = g_value_get_enum (value);
break;
case PROP_USER_AGENT:
g_free (rtsp_client_sink->user_agent);
rtsp_client_sink->user_agent = g_value_dup_string (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtsp_client_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRTSPClientSink *rtsp_client_sink;
rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
switch (prop_id) {
case PROP_LOCATION:
g_value_set_string (value, rtsp_client_sink->conninfo.location);
break;
case PROP_PROTOCOLS:
g_value_set_flags (value, rtsp_client_sink->protocols);
break;
case PROP_PROFILES:
g_value_set_flags (value, rtsp_client_sink->profiles);
break;
case PROP_DEBUG:
g_value_set_boolean (value, rtsp_client_sink->debug);
break;
case PROP_RETRY:
g_value_set_uint (value, rtsp_client_sink->retry);
break;
case PROP_TIMEOUT:
g_value_set_uint64 (value, rtsp_client_sink->udp_timeout);
break;
case PROP_TCP_TIMEOUT:
{
guint64 timeout;
timeout = rtsp_client_sink->tcp_timeout.tv_sec * G_USEC_PER_SEC +
rtsp_client_sink->tcp_timeout.tv_usec;
g_value_set_uint64 (value, timeout);
break;
}
case PROP_LATENCY:
g_value_set_uint (value, rtsp_client_sink->latency);
break;
case PROP_RTX_TIME:
g_value_set_uint (value, rtsp_client_sink->rtx_time);
break;
case PROP_DO_RTSP_KEEP_ALIVE:
g_value_set_boolean (value, rtsp_client_sink->do_rtsp_keep_alive);
break;
case PROP_PROXY:
{
gchar *str;
if (rtsp_client_sink->proxy_host) {
str =
g_strdup_printf ("%s:%d", rtsp_client_sink->proxy_host,
rtsp_client_sink->proxy_port);
} else {
str = NULL;
}
g_value_take_string (value, str);
break;
}
case PROP_PROXY_ID:
g_value_set_string (value, rtsp_client_sink->prop_proxy_id);
break;
case PROP_PROXY_PW:
g_value_set_string (value, rtsp_client_sink->prop_proxy_pw);
break;
case PROP_RTP_BLOCKSIZE:
g_value_set_uint (value, rtsp_client_sink->rtp_blocksize);
break;
case PROP_USER_ID:
g_value_set_string (value, rtsp_client_sink->user_id);
break;
case PROP_USER_PW:
g_value_set_string (value, rtsp_client_sink->user_pw);
break;
case PROP_PORT_RANGE:
{
gchar *str;
if (rtsp_client_sink->client_port_range.min != 0) {
str = g_strdup_printf ("%u-%u", rtsp_client_sink->client_port_range.min,
rtsp_client_sink->client_port_range.max);
} else {
str = NULL;
}
g_value_take_string (value, str);
break;
}
case PROP_UDP_BUFFER_SIZE:
g_value_set_int (value, rtsp_client_sink->udp_buffer_size);
break;
case PROP_UDP_RECONNECT:
g_value_set_boolean (value, rtsp_client_sink->udp_reconnect);
break;
case PROP_MULTICAST_IFACE:
g_value_set_string (value, rtsp_client_sink->multi_iface);
break;
case PROP_SDES:
g_value_set_boxed (value, rtsp_client_sink->sdes);
break;
case PROP_TLS_VALIDATION_FLAGS:
g_value_set_flags (value, rtsp_client_sink->tls_validation_flags);
break;
case PROP_TLS_DATABASE:
g_value_set_object (value, rtsp_client_sink->tls_database);
break;
case PROP_TLS_INTERACTION:
g_value_set_object (value, rtsp_client_sink->tls_interaction);
break;
case PROP_NTP_TIME_SOURCE:
g_value_set_enum (value, rtsp_client_sink->ntp_time_source);
break;
case PROP_USER_AGENT:
g_value_set_string (value, rtsp_client_sink->user_agent);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static const gchar *
get_aggregate_control (GstRTSPClientSink * sink)
{
const gchar *base;
if (sink->control)
base = sink->control;
else if (sink->content_base)
base = sink->content_base;
else if (sink->conninfo.url_str)
base = sink->conninfo.url_str;
else
base = "/";
return base;
}
static void
gst_rtsp_client_sink_cleanup (GstRTSPClientSink * sink)
{
GList *walk;
GST_DEBUG_OBJECT (sink, "cleanup");
gst_element_set_state (GST_ELEMENT (sink->internal_bin), GST_STATE_NULL);
/* Clean up any left over stream objects */
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) (walk->data);
if (context->stream_transport) {
gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
gst_object_unref (context->stream_transport);
context->stream_transport = NULL;
}
if (context->stream) {
if (context->joined) {
gst_rtsp_stream_leave_bin (context->stream,
GST_BIN (sink->internal_bin), sink->rtpbin);
context->joined = FALSE;
}
gst_object_unref (context->stream);
context->stream = NULL;
}
if (context->srtcpparams) {
gst_caps_unref (context->srtcpparams);
context->srtcpparams = NULL;
}
g_free (context->conninfo.location);
context->conninfo.location = NULL;
}
if (sink->rtpbin) {
gst_element_set_state (sink->rtpbin, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (sink->internal_bin), sink->rtpbin);
sink->rtpbin = NULL;
}
g_free (sink->content_base);
sink->content_base = NULL;
g_free (sink->control);
sink->control = NULL;
if (sink->range)
gst_rtsp_range_free (sink->range);
sink->range = NULL;
/* don't clear the SDP when it was used in the url */
if (sink->uri_sdp && !sink->from_sdp) {
gst_sdp_message_free (sink->uri_sdp);
sink->uri_sdp = NULL;
}
if (sink->provided_clock) {
gst_object_unref (sink->provided_clock);
sink->provided_clock = NULL;
}
g_free (sink->server_ip);
sink->server_ip = NULL;
sink->next_pad_id = 0;
sink->next_dyn_pt = 96;
}
static GstRTSPResult
gst_rtsp_client_sink_connection_send (GstRTSPClientSink * sink,
GstRTSPConnInfo * conninfo, GstRTSPMessage * message, GTimeVal * timeout)
{
GstRTSPResult ret;
if (conninfo->connection) {
g_mutex_lock (&conninfo->send_lock);
ret = gst_rtsp_connection_send (conninfo->connection, message, timeout);
g_mutex_unlock (&conninfo->send_lock);
} else {
ret = GST_RTSP_ERROR;
}
return ret;
}
static GstRTSPResult
gst_rtsp_client_sink_connection_receive (GstRTSPClientSink * sink,
GstRTSPConnInfo * conninfo, GstRTSPMessage * message, GTimeVal * timeout)
{
GstRTSPResult ret;
if (conninfo->connection) {
g_mutex_lock (&conninfo->recv_lock);
ret = gst_rtsp_connection_receive (conninfo->connection, message, timeout);
g_mutex_unlock (&conninfo->recv_lock);
} else {
ret = GST_RTSP_ERROR;
}
return ret;
}
static gboolean
accept_certificate_cb (GTlsConnection * conn, GTlsCertificate * peer_cert,
GTlsCertificateFlags errors, gpointer user_data)
{
GstRTSPClientSink *sink = user_data;
gboolean accept = FALSE;
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_ACCEPT_CERTIFICATE],
0, conn, peer_cert, errors, &accept);
return accept;
}
static GstRTSPResult
gst_rtsp_conninfo_connect (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
gboolean async)
{
GstRTSPResult res;
if (info->connection == NULL) {
if (info->url == NULL) {
GST_DEBUG_OBJECT (sink, "parsing uri (%s)...", info->location);
if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
goto parse_error;
}
/* create connection */
GST_DEBUG_OBJECT (sink, "creating connection (%s)...", info->location);
if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
goto could_not_create;
if (info->url_str)
g_free (info->url_str);
info->url_str = gst_rtsp_url_get_request_uri (info->url);
GST_DEBUG_OBJECT (sink, "sanitized uri %s", info->url_str);
if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
sink->tls_validation_flags))
GST_WARNING_OBJECT (sink, "Unable to set TLS validation flags");
if (sink->tls_database)
gst_rtsp_connection_set_tls_database (info->connection,
sink->tls_database);
if (sink->tls_interaction)
gst_rtsp_connection_set_tls_interaction (info->connection,
sink->tls_interaction);
gst_rtsp_connection_set_accept_certificate_func (info->connection,
accept_certificate_cb, sink, NULL);
}
if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
gst_rtsp_connection_set_tunneled (info->connection, TRUE);
if (sink->proxy_host) {
GST_DEBUG_OBJECT (sink, "setting proxy %s:%d", sink->proxy_host,
sink->proxy_port);
gst_rtsp_connection_set_proxy (info->connection, sink->proxy_host,
sink->proxy_port);
}
}
if (!info->connected) {
/* connect */
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "connect",
("Connecting to %s", info->location));
GST_DEBUG_OBJECT (sink, "connecting (%s)...", info->location);
if ((res =
gst_rtsp_connection_connect (info->connection,
sink->ptcp_timeout)) < 0)
goto could_not_connect;
info->connected = TRUE;
}
return GST_RTSP_OK;
/* ERRORS */
parse_error:
{
GST_ERROR_OBJECT (sink, "No valid RTSP URL was provided");
return res;
}
could_not_create:
{
gchar *str = gst_rtsp_strresult (res);
GST_ERROR_OBJECT (sink, "Could not create connection. (%s)", str);
g_free (str);
return res;
}
could_not_connect:
{
gchar *str = gst_rtsp_strresult (res);
GST_ERROR_OBJECT (sink, "Could not connect to server. (%s)", str);
g_free (str);
return res;
}
}
static GstRTSPResult
gst_rtsp_conninfo_close (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
gboolean free)
{
GST_RTSP_STATE_LOCK (sink);
if (info->connected) {
GST_DEBUG_OBJECT (sink, "closing connection...");
gst_rtsp_connection_close (info->connection);
info->connected = FALSE;
}
if (free && info->connection) {
/* free connection */
GST_DEBUG_OBJECT (sink, "freeing connection...");
gst_rtsp_connection_free (info->connection);
info->connection = NULL;
}
GST_RTSP_STATE_UNLOCK (sink);
return GST_RTSP_OK;
}
static GstRTSPResult
gst_rtsp_conninfo_reconnect (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
gboolean async)
{
GstRTSPResult res;
GST_DEBUG_OBJECT (sink, "reconnecting connection...");
gst_rtsp_conninfo_close (sink, info, FALSE);
res = gst_rtsp_conninfo_connect (sink, info, async);
return res;
}
static void
gst_rtsp_client_sink_connection_flush (GstRTSPClientSink * sink, gboolean flush)
{
GList *walk;
GST_DEBUG_OBJECT (sink, "set flushing %d", flush);
g_mutex_lock (&sink->preroll_lock);
if (sink->conninfo.connection && sink->conninfo.flushing != flush) {
GST_DEBUG_OBJECT (sink, "connection flush");
gst_rtsp_connection_flush (sink->conninfo.connection, flush);
sink->conninfo.flushing = flush;
}
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
GST_DEBUG_OBJECT (sink, "stream %p flush", stream);
gst_rtsp_connection_flush (stream->conninfo.connection, flush);
stream->conninfo.flushing = flush;
}
}
g_cond_broadcast (&sink->preroll_cond);
g_mutex_unlock (&sink->preroll_lock);
}
static GstRTSPResult
gst_rtsp_client_sink_init_request (GstRTSPClientSink * sink,
GstRTSPMessage * msg, GstRTSPMethod method, const gchar * uri)
{
GstRTSPResult res;
res = gst_rtsp_message_init_request (msg, method, uri);
if (res < 0)
return res;
/* set user-agent */
if (sink->user_agent)
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_USER_AGENT,
sink->user_agent);
return res;
}
/* FIXME, handle server request, reply with OK, for now */
static GstRTSPResult
gst_rtsp_client_sink_handle_request (GstRTSPClientSink * sink,
GstRTSPConnInfo * conninfo, GstRTSPMessage * request)
{
GstRTSPMessage response = { 0 };
GstRTSPResult res;
GST_DEBUG_OBJECT (sink, "got server request message");
if (sink->debug)
gst_rtsp_message_dump (request);
/* default implementation, send OK */
GST_DEBUG_OBJECT (sink, "prepare OK reply");
res =
gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
request);
if (res < 0)
goto send_error;
/* let app parse and reply */
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_HANDLE_REQUEST],
0, request, &response);
if (sink->debug)
gst_rtsp_message_dump (&response);
res = gst_rtsp_client_sink_connection_send (sink, conninfo, &response, NULL);
if (res < 0)
goto send_error;
gst_rtsp_message_unset (&response);
return GST_RTSP_OK;
/* ERRORS */
send_error:
{
gst_rtsp_message_unset (&response);
return res;
}
}
/* send server keep-alive */
static GstRTSPResult
gst_rtsp_client_sink_send_keep_alive (GstRTSPClientSink * sink)
{
GstRTSPMessage request = { 0 };
GstRTSPResult res;
GstRTSPMethod method;
const gchar *control;
if (sink->do_rtsp_keep_alive == FALSE) {
GST_DEBUG_OBJECT (sink, "do-rtsp-keep-alive is FALSE, not sending.");
gst_rtsp_connection_reset_timeout (sink->conninfo.connection);
return GST_RTSP_OK;
}
GST_DEBUG_OBJECT (sink, "creating server keep-alive");
/* find a method to use for keep-alive */
if (sink->methods & GST_RTSP_GET_PARAMETER)
method = GST_RTSP_GET_PARAMETER;
else
method = GST_RTSP_OPTIONS;
control = get_aggregate_control (sink);
if (control == NULL)
goto no_control;
res = gst_rtsp_client_sink_init_request (sink, &request, method, control);
if (res < 0)
goto send_error;
if (sink->debug)
gst_rtsp_message_dump (&request);
res =
gst_rtsp_client_sink_connection_send (sink, &sink->conninfo,
&request, NULL);
if (res < 0)
goto send_error;
gst_rtsp_connection_reset_timeout (sink->conninfo.connection);
gst_rtsp_message_unset (&request);
return GST_RTSP_OK;
/* ERRORS */
no_control:
{
GST_WARNING_OBJECT (sink, "no control url to send keepalive");
return GST_RTSP_OK;
}
send_error:
{
gchar *str = gst_rtsp_strresult (res);
gst_rtsp_message_unset (&request);
GST_ELEMENT_WARNING (sink, RESOURCE, WRITE, (NULL),
("Could not send keep-alive. (%s)", str));
g_free (str);
return res;
}
}
static GstFlowReturn
gst_rtsp_client_sink_loop_rx (GstRTSPClientSink * sink)
{
GstRTSPResult res;
GstRTSPMessage message = { 0 };
gint retry = 0;
while (TRUE) {
GTimeVal tv_timeout;
/* get the next timeout interval */
gst_rtsp_connection_next_timeout (sink->conninfo.connection, &tv_timeout);
GST_DEBUG_OBJECT (sink, "doing receive with timeout %d seconds",
(gint) tv_timeout.tv_sec);
gst_rtsp_message_unset (&message);
/* we should continue reading the TCP socket because the server might
* send us requests. When the session timeout expires, we need to send a
* keep-alive request to keep the session open. */
res =
gst_rtsp_client_sink_connection_receive (sink,
&sink->conninfo, &message, &tv_timeout);
switch (res) {
case GST_RTSP_OK:
GST_DEBUG_OBJECT (sink, "we received a server message");
break;
case GST_RTSP_EINTR:
/* we got interrupted, see what we have to do */
goto interrupt;
case GST_RTSP_ETIMEOUT:
/* send keep-alive, ignore the result, a warning will be posted. */
GST_DEBUG_OBJECT (sink, "timeout, sending keep-alive");
if ((res =
gst_rtsp_client_sink_send_keep_alive (sink)) == GST_RTSP_EINTR)
goto interrupt;
continue;
case GST_RTSP_EEOF:
/* server closed the connection. not very fatal for UDP, reconnect and
* see what happens. */
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
("The server closed the connection."));
if (sink->udp_reconnect) {
if ((res =
gst_rtsp_conninfo_reconnect (sink, &sink->conninfo,
FALSE)) < 0)
goto connect_error;
} else {
goto server_eof;
}
continue;
break;
case GST_RTSP_ENET:
GST_DEBUG_OBJECT (sink, "An ethernet problem occured.");
default:
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
("Unhandled return value %d.", res));
goto receive_error;
}
switch (message.type) {
case GST_RTSP_MESSAGE_REQUEST:
/* server sends us a request message, handle it */
res =
gst_rtsp_client_sink_handle_request (sink,
&sink->conninfo, &message);
if (res == GST_RTSP_EEOF)
goto server_eof;
else if (res < 0)
goto handle_request_failed;
break;
case GST_RTSP_MESSAGE_RESPONSE:
/* we ignore response and data messages */
GST_DEBUG_OBJECT (sink, "ignoring response message");
if (sink->debug)
gst_rtsp_message_dump (&message);
if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
GST_DEBUG_OBJECT (sink, "but is Unauthorized response ...");
if (gst_rtsp_client_sink_setup_auth (sink, &message) && !(retry++)) {
GST_DEBUG_OBJECT (sink, "so retrying keep-alive");
if ((res =
gst_rtsp_client_sink_send_keep_alive (sink)) ==
GST_RTSP_EINTR)
goto interrupt;
}
} else {
retry = 0;
}
break;
case GST_RTSP_MESSAGE_DATA:
/* we ignore response and data messages */
GST_DEBUG_OBJECT (sink, "ignoring data message");
break;
default:
GST_WARNING_OBJECT (sink, "ignoring unknown message type %d",
message.type);
break;
}
}
g_assert_not_reached ();
/* we get here when the connection got interrupted */
interrupt:
{
gst_rtsp_message_unset (&message);
GST_DEBUG_OBJECT (sink, "got interrupted");
return GST_FLOW_FLUSHING;
}
connect_error:
{
gchar *str = gst_rtsp_strresult (res);
GstFlowReturn ret;
sink->conninfo.connected = FALSE;
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ_WRITE, (NULL),
("Could not connect to server. (%s)", str));
g_free (str);
ret = GST_FLOW_ERROR;
} else {
ret = GST_FLOW_FLUSHING;
}
return ret;
}
receive_error:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
g_free (str);
return GST_FLOW_ERROR;
}
handle_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GstFlowReturn ret;
gst_rtsp_message_unset (&message);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Could not handle server message. (%s)", str));
g_free (str);
ret = GST_FLOW_ERROR;
} else {
ret = GST_FLOW_FLUSHING;
}
return ret;
}
server_eof:
{
GST_DEBUG_OBJECT (sink, "we got an eof from the server");
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
("The server closed the connection."));
sink->conninfo.connected = FALSE;
gst_rtsp_message_unset (&message);
return GST_FLOW_EOS;
}
}
static GstRTSPResult
gst_rtsp_client_sink_reconnect (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPResult res = GST_RTSP_OK;
gboolean restart = FALSE;
GST_DEBUG_OBJECT (sink, "doing reconnect");
GST_FIXME_OBJECT (sink, "Reconnection is not yet implemented");
/* no need to restart, we're done */
if (!restart)
goto done;
/* we can try only TCP now */
sink->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
/* close and cleanup our state */
if ((res = gst_rtsp_client_sink_close (sink, async, FALSE)) < 0)
goto done;
/* see if we have TCP left to try. Also don't try TCP when we were configured
* with an SDP. */
if (!(sink->protocols & GST_RTSP_LOWER_TRANS_TCP) || sink->from_sdp)
goto no_protocols;
/* We post a warning message now to inform the user
* that nothing happened. It's most likely a firewall thing. */
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
("Could not receive any UDP packets for %.4f seconds, maybe your "
"firewall is blocking it. Retrying using a TCP connection.",
gst_guint64_to_gdouble (sink->udp_timeout / 1000000.0)));
/* open new connection using tcp */
if (gst_rtsp_client_sink_open (sink, async) < 0)
goto open_failed;
/* start recording */
if (gst_rtsp_client_sink_record (sink, async) < 0)
goto play_failed;
done:
return res;
/* ERRORS */
no_protocols:
{
sink->cur_protocols = 0;
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not receive any UDP packets for %.4f seconds, maybe your "
"firewall is blocking it. No other protocols to try.",
gst_guint64_to_gdouble (sink->udp_timeout / 1000000.0)));
return GST_RTSP_ERROR;
}
open_failed:
{
GST_DEBUG_OBJECT (sink, "open failed");
return GST_RTSP_OK;
}
play_failed:
{
GST_DEBUG_OBJECT (sink, "play failed");
return GST_RTSP_OK;
}
}
static void
gst_rtsp_client_sink_loop_start_cmd (GstRTSPClientSink * sink, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (sink, START, "open", ("Opening Stream"));
break;
case CMD_RECORD:
GST_ELEMENT_PROGRESS (sink, START, "request", ("Sending RECORD request"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (sink, START, "request", ("Sending PAUSE request"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (sink, START, "close", ("Closing Stream"));
break;
default:
break;
}
}
static void
gst_rtsp_client_sink_loop_complete_cmd (GstRTSPClientSink * sink, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (sink, COMPLETE, "open", ("Opened Stream"));
break;
case CMD_RECORD:
GST_ELEMENT_PROGRESS (sink, COMPLETE, "request", ("Sent RECORD request"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (sink, COMPLETE, "request", ("Sent PAUSE request"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (sink, COMPLETE, "close", ("Closed Stream"));
break;
default:
break;
}
}
static void
gst_rtsp_client_sink_loop_cancel_cmd (GstRTSPClientSink * sink, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (sink, CANCELED, "open", ("Open canceled"));
break;
case CMD_RECORD:
GST_ELEMENT_PROGRESS (sink, CANCELED, "request", ("RECORD canceled"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (sink, CANCELED, "request", ("PAUSE canceled"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (sink, CANCELED, "close", ("Close canceled"));
break;
default:
break;
}
}
static void
gst_rtsp_client_sink_loop_error_cmd (GstRTSPClientSink * sink, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (sink, ERROR, "open", ("Open failed"));
break;
case CMD_RECORD:
GST_ELEMENT_PROGRESS (sink, ERROR, "request", ("RECORD failed"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (sink, ERROR, "request", ("PAUSE failed"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (sink, ERROR, "close", ("Close failed"));
break;
default:
break;
}
}
static void
gst_rtsp_client_sink_loop_end_cmd (GstRTSPClientSink * sink, gint cmd,
GstRTSPResult ret)
{
if (ret == GST_RTSP_OK)
gst_rtsp_client_sink_loop_complete_cmd (sink, cmd);
else if (ret == GST_RTSP_EINTR)
gst_rtsp_client_sink_loop_cancel_cmd (sink, cmd);
else
gst_rtsp_client_sink_loop_error_cmd (sink, cmd);
}
static gboolean
gst_rtsp_client_sink_loop_send_cmd (GstRTSPClientSink * sink, gint cmd,
gint mask)
{
gint old;
gboolean flushed = FALSE;
/* start new request */
gst_rtsp_client_sink_loop_start_cmd (sink, cmd);
GST_DEBUG_OBJECT (sink, "sending cmd %s", cmd_to_string (cmd));
GST_OBJECT_LOCK (sink);
old = sink->pending_cmd;
if (old == CMD_RECONNECT) {
GST_DEBUG_OBJECT (sink, "ignore, we were reconnecting");
cmd = CMD_RECONNECT;
}
if (old != CMD_WAIT) {
sink->pending_cmd = CMD_WAIT;
GST_OBJECT_UNLOCK (sink);
/* cancel previous request */
GST_DEBUG_OBJECT (sink, "cancel previous request %s", cmd_to_string (old));
gst_rtsp_client_sink_loop_cancel_cmd (sink, old);
GST_OBJECT_LOCK (sink);
}
sink->pending_cmd = cmd;
/* interrupt if allowed */
if (sink->busy_cmd & mask) {
GST_DEBUG_OBJECT (sink, "connection flush busy %s",
cmd_to_string (sink->busy_cmd));
gst_rtsp_client_sink_connection_flush (sink, TRUE);
flushed = TRUE;
} else {
GST_DEBUG_OBJECT (sink, "not interrupting busy cmd %s",
cmd_to_string (sink->busy_cmd));
}
if (sink->task)
gst_task_start (sink->task);
GST_OBJECT_UNLOCK (sink);
return flushed;
}
static gboolean
gst_rtsp_client_sink_loop (GstRTSPClientSink * sink)
{
GstFlowReturn ret;
if (!sink->conninfo.connection || !sink->conninfo.connected)
goto no_connection;
ret = gst_rtsp_client_sink_loop_rx (sink);
if (ret != GST_FLOW_OK)
goto pause;
return TRUE;
/* ERRORS */
no_connection:
{
GST_WARNING_OBJECT (sink, "we are not connected");
ret = GST_FLOW_FLUSHING;
goto pause;
}
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (sink, "pausing task, reason %s", reason);
gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT, CMD_LOOP);
return FALSE;
}
}
#ifndef GST_DISABLE_GST_DEBUG
static const gchar *
gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
{
gint index = 0;
while (method != 0) {
index++;
method >>= 1;
}
switch (index) {
case 0:
return "None";
case 1:
return "Basic";
case 2:
return "Digest";
}
return "Unknown";
}
#endif
/* Parse a WWW-Authenticate Response header and determine the
* available authentication methods
*
* This code should also cope with the fact that each WWW-Authenticate
* header can contain multiple challenge methods + tokens
*
* At the moment, for Basic auth, we just do a minimal check and don't
* even parse out the realm */
static void
gst_rtsp_client_sink_parse_auth_hdr (GstRTSPMessage * response,
GstRTSPAuthMethod * methods, GstRTSPConnection * conn, gboolean * stale)
{
GstRTSPAuthCredential **credentials, **credential;
g_return_if_fail (response != NULL);
g_return_if_fail (methods != NULL);
g_return_if_fail (stale != NULL);
credentials =
gst_rtsp_message_parse_auth_credentials (response,
GST_RTSP_HDR_WWW_AUTHENTICATE);
if (!credentials)
return;
credential = credentials;
while (*credential) {
if ((*credential)->scheme == GST_RTSP_AUTH_BASIC) {
*methods |= GST_RTSP_AUTH_BASIC;
} else if ((*credential)->scheme == GST_RTSP_AUTH_DIGEST) {
GstRTSPAuthParam **param = (*credential)->params;
*methods |= GST_RTSP_AUTH_DIGEST;
gst_rtsp_connection_clear_auth_params (conn);
*stale = FALSE;
while (*param) {
if (strcmp ((*param)->name, "stale") == 0
&& g_ascii_strcasecmp ((*param)->value, "TRUE") == 0)
*stale = TRUE;
gst_rtsp_connection_set_auth_param (conn, (*param)->name,
(*param)->value);
param++;
}
}
credential++;
}
gst_rtsp_auth_credentials_free (credentials);
}
/**
* gst_rtsp_client_sink_setup_auth:
* @src: the rtsp source
*
* Configure a username and password and auth method on the
* connection object based on a response we received from the
* peer.
*
* Currently, this requires that a username and password were supplied
* in the uri. In the future, they may be requested on demand by sending
* a message up the bus.
*
* Returns: TRUE if authentication information could be set up correctly.
*/
static gboolean
gst_rtsp_client_sink_setup_auth (GstRTSPClientSink * sink,
GstRTSPMessage * response)
{
gchar *user = NULL;
gchar *pass = NULL;
GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
GstRTSPAuthMethod method;
GstRTSPResult auth_result;
GstRTSPUrl *url;
GstRTSPConnection *conn;
gboolean stale = FALSE;
conn = sink->conninfo.connection;
/* Identify the available auth methods and see if any are supported */
gst_rtsp_client_sink_parse_auth_hdr (response, &avail_methods, conn, &stale);
if (avail_methods == GST_RTSP_AUTH_NONE)
goto no_auth_available;
/* For digest auth, if the response indicates that the session
* data are stale, we just update them in the connection object and
* return TRUE to retry the request */
if (stale)
sink->tried_url_auth = FALSE;
url = gst_rtsp_connection_get_url (conn);
/* Do we have username and password available? */
if (url != NULL && !sink->tried_url_auth && url->user != NULL
&& url->passwd != NULL) {
user = url->user;
pass = url->passwd;
sink->tried_url_auth = TRUE;
GST_DEBUG_OBJECT (sink,
"Attempting authentication using credentials from the URL");
} else {
user = sink->user_id;
pass = sink->user_pw;
GST_DEBUG_OBJECT (sink,
"Attempting authentication using credentials from the properties");
}
/* FIXME: If the url didn't contain username and password or we tried them
* already, request a username and passwd from the application via some kind
* of credentials request message */
/* If we don't have a username and passwd at this point, bail out. */
if (user == NULL || pass == NULL)
goto no_user_pass;
/* Try to configure for each available authentication method, strongest to
* weakest */
for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
/* Check if this method is available on the server */
if ((method & avail_methods) == 0)
continue;
/* Pass the credentials to the connection to try on the next request */
auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
/* INVAL indicates an invalid username/passwd were supplied, so we'll just
* ignore it and end up retrying later */
if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
GST_DEBUG_OBJECT (sink, "Attempting %s authentication",
gst_rtsp_auth_method_to_string (method));
break;
}
}
if (method == GST_RTSP_AUTH_NONE)
goto no_auth_available;
return TRUE;
no_auth_available:
{
/* Output an error indicating that we couldn't connect because there were
* no supported authentication protocols */
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ, (NULL),
("No supported authentication protocol was found"));
return FALSE;
}
no_user_pass:
{
/* We don't fire an error message, we just return FALSE and let the
* normal NOT_AUTHORIZED error be propagated */
return FALSE;
}
}
static GstRTSPResult
gst_rtsp_client_sink_try_send (GstRTSPClientSink * sink,
GstRTSPConnInfo * conninfo, GstRTSPMessage * request,
GstRTSPMessage * response, GstRTSPStatusCode * code)
{
GstRTSPResult res;
GstRTSPStatusCode thecode;
gchar *content_base = NULL;
gint try = 0;
again:
GST_DEBUG_OBJECT (sink, "sending message");
if (sink->debug)
gst_rtsp_message_dump (request);
g_mutex_lock (&sink->send_lock);
res =
gst_rtsp_client_sink_connection_send (sink, conninfo, request,
sink->ptcp_timeout);
if (res < 0) {
g_mutex_unlock (&sink->send_lock);
goto send_error;
}
gst_rtsp_connection_reset_timeout (conninfo->connection);
/* See if we should handle the response */
if (response == NULL) {
g_mutex_unlock (&sink->send_lock);
return GST_RTSP_OK;
}
next:
res =
gst_rtsp_client_sink_connection_receive (sink, conninfo, response,
sink->ptcp_timeout);
g_mutex_unlock (&sink->send_lock);
if (res < 0)
goto receive_error;
if (sink->debug)
gst_rtsp_message_dump (response);
switch (response->type) {
case GST_RTSP_MESSAGE_REQUEST:
res = gst_rtsp_client_sink_handle_request (sink, conninfo, response);
if (res == GST_RTSP_EEOF)
goto server_eof;
else if (res < 0)
goto handle_request_failed;
g_mutex_lock (&sink->send_lock);
goto next;
case GST_RTSP_MESSAGE_RESPONSE:
/* ok, a response is good */
GST_DEBUG_OBJECT (sink, "received response message");
break;
case GST_RTSP_MESSAGE_DATA:
/* we ignore data messages */
GST_DEBUG_OBJECT (sink, "ignoring data message");
g_mutex_lock (&sink->send_lock);
goto next;
default:
GST_WARNING_OBJECT (sink, "ignoring unknown message type %d",
response->type);
g_mutex_lock (&sink->send_lock);
goto next;
}
thecode = response->type_data.response.code;
GST_DEBUG_OBJECT (sink, "got response message %d", thecode);
/* if the caller wanted the result code, we store it. */
if (code)
*code = thecode;
/* If the request didn't succeed, bail out before doing any more */
if (thecode != GST_RTSP_STS_OK)
return GST_RTSP_OK;
/* store new content base if any */
gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
&content_base, 0);
if (content_base) {
g_free (sink->content_base);
sink->content_base = g_strdup (content_base);
}
return GST_RTSP_OK;
/* ERRORS */
send_error:
{
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "send interrupted");
}
g_free (str);
return res;
}
receive_error:
{
switch (res) {
case GST_RTSP_EEOF:
GST_WARNING_OBJECT (sink, "server closed connection");
if ((try == 0) && !sink->interleaved && sink->udp_reconnect) {
try++;
/* if reconnect succeeds, try again */
if ((res =
gst_rtsp_conninfo_reconnect (sink, &sink->conninfo,
FALSE)) == 0)
goto again;
}
/* only try once after reconnect, then fallthrough and error out */
default:
{
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "receive interrupted");
}
g_free (str);
break;
}
}
return res;
}
handle_request_failed:
{
/* ERROR was posted */
gst_rtsp_message_unset (response);
return res;
}
server_eof:
{
GST_DEBUG_OBJECT (sink, "we got an eof from the server");
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
("The server closed the connection."));
gst_rtsp_message_unset (response);
return res;
}
}
static void
gst_rtsp_client_sink_set_state (GstRTSPClientSink * sink, GstState state)
{
GST_DEBUG_OBJECT (sink, "Setting internal state to %s",
gst_element_state_get_name (state));
gst_element_set_state (GST_ELEMENT (sink->internal_bin), state);
}
/**
* gst_rtsp_client_sink_send:
* @src: the rtsp source
* @conn: the connection to send on
* @request: must point to a valid request
* @response: must point to an empty #GstRTSPMessage
* @code: an optional code result
*
* send @request and retrieve the response in @response. optionally @code can be
* non-NULL in which case it will contain the status code of the response.
*
* If This function returns #GST_RTSP_OK, @response will contain a valid response
* message that should be cleaned with gst_rtsp_message_unset() after usage.
*
* If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
* @response message) if the response code was not 200 (OK).
*
* If the attempt results in an authentication failure, then this will attempt
* to retrieve authentication credentials via gst_rtsp_client_sink_setup_auth and retry
* the request.
*
* Returns: #GST_RTSP_OK if the processing was successful.
*/
static GstRTSPResult
gst_rtsp_client_sink_send (GstRTSPClientSink * sink, GstRTSPConnInfo * conninfo,
GstRTSPMessage * request, GstRTSPMessage * response,
GstRTSPStatusCode * code)
{
GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
GstRTSPResult res = GST_RTSP_ERROR;
gint count;
gboolean retry;
GstRTSPMethod method = GST_RTSP_INVALID;
count = 0;
do {
retry = FALSE;
/* make sure we don't loop forever */
if (count++ > 8)
break;
/* save method so we can disable it when the server complains */
method = request->type_data.request.method;
if ((res =
gst_rtsp_client_sink_try_send (sink, conninfo, request, response,
&int_code)) < 0)
goto error;
switch (int_code) {
case GST_RTSP_STS_UNAUTHORIZED:
if (gst_rtsp_client_sink_setup_auth (sink, response)) {
/* Try the request/response again after configuring the auth info
* and loop again */
retry = TRUE;
}
break;
default:
break;
}
} while (retry == TRUE);
/* If the user requested the code, let them handle errors, otherwise
* post an error below */
if (code != NULL)
*code = int_code;
else if (int_code != GST_RTSP_STS_OK)
goto error_response;
return res;
/* ERRORS */
error:
{
GST_DEBUG_OBJECT (sink, "got error %d", res);
return res;
}
error_response:
{
res = GST_RTSP_ERROR;
switch (response->type_data.response.code) {
case GST_RTSP_STS_NOT_FOUND:
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, (NULL), ("%s",
response->type_data.response.reason));
break;
case GST_RTSP_STS_UNAUTHORIZED:
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_AUTHORIZED, (NULL), ("%s",
response->type_data.response.reason));
break;
case GST_RTSP_STS_MOVED_PERMANENTLY:
case GST_RTSP_STS_MOVE_TEMPORARILY:
{
gchar *new_location;
GstRTSPLowerTrans transports;
GST_DEBUG_OBJECT (sink, "got redirection");
/* if we don't have a Location Header, we must error */
if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
&new_location, 0) < 0)
break;
/* When we receive a redirect result, we go back to the INIT state after
* parsing the new URI. The caller should do the needed steps to issue
* a new setup when it detects this state change. */
GST_DEBUG_OBJECT (sink, "redirection to %s", new_location);
/* save current transports */
if (sink->conninfo.url)
transports = sink->conninfo.url->transports;
else
transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
gst_rtsp_client_sink_uri_set_uri (GST_URI_HANDLER (sink), new_location,
NULL);
/* set old transports */
if (sink->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
sink->conninfo.url->transports = transports;
sink->need_redirect = TRUE;
sink->state = GST_RTSP_STATE_INIT;
res = GST_RTSP_OK;
break;
}
case GST_RTSP_STS_NOT_ACCEPTABLE:
case GST_RTSP_STS_NOT_IMPLEMENTED:
case GST_RTSP_STS_METHOD_NOT_ALLOWED:
GST_WARNING_OBJECT (sink, "got NOT IMPLEMENTED, disable method %s",
gst_rtsp_method_as_text (method));
sink->methods &= ~method;
res = GST_RTSP_OK;
break;
default:
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Got error response: %d (%s).", response->type_data.response.code,
response->type_data.response.reason));
break;
}
/* if we return ERROR we should unset the response ourselves */
if (res == GST_RTSP_ERROR)
gst_rtsp_message_unset (response);
return res;
}
}
/* parse the response and collect all the supported methods. We need this
* information so that we don't try to send an unsupported request to the
* server.
*/
static gboolean
gst_rtsp_client_sink_parse_methods (GstRTSPClientSink * sink,
GstRTSPMessage * response)
{
GstRTSPHeaderField field;
gchar *respoptions;
gint indx = 0;
/* reset supported methods */
sink->methods = 0;
/* Try Allow Header first */
field = GST_RTSP_HDR_ALLOW;
while (TRUE) {
respoptions = NULL;
gst_rtsp_message_get_header (response, field, &respoptions, indx);
if (indx == 0 && !respoptions) {
/* if no Allow header was found then try the Public header... */
field = GST_RTSP_HDR_PUBLIC;
gst_rtsp_message_get_header (response, field, &respoptions, indx);
}
if (!respoptions)
break;
sink->methods |= gst_rtsp_options_from_text (respoptions);
indx++;
}
if (sink->methods == 0) {
/* neither Allow nor Public are required, assume the server supports
* at least SETUP. */
GST_DEBUG_OBJECT (sink, "could not get OPTIONS");
sink->methods = GST_RTSP_SETUP;
}
/* Even if the server replied, and didn't say it supports
* RECORD|ANNOUNCE, try anyway by assuming it does */
sink->methods |= GST_RTSP_ANNOUNCE | GST_RTSP_RECORD;
if (!(sink->methods & GST_RTSP_SETUP))
goto no_setup;
return TRUE;
/* ERRORS */
no_setup:
{
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ, (NULL),
("Server does not support SETUP."));
return FALSE;
}
}
static GstRTSPResult
gst_rtsp_client_sink_connect_to_server (GstRTSPClientSink * sink,
gboolean async)
{
GstRTSPResult res;
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GSocket *conn_socket;
GSocketAddress *sa;
GInetAddress *ia;
sink->need_redirect = FALSE;
/* can't continue without a valid url */
if (G_UNLIKELY (sink->conninfo.url == NULL)) {
res = GST_RTSP_EINVAL;
goto no_url;
}
sink->tried_url_auth = FALSE;
if ((res = gst_rtsp_conninfo_connect (sink, &sink->conninfo, async)) < 0)
goto connect_failed;
conn_socket = gst_rtsp_connection_get_read_socket (sink->conninfo.connection);
sa = g_socket_get_remote_address (conn_socket, NULL);
ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
sink->server_ip = g_inet_address_to_string (ia);
g_object_unref (sa);
/* create OPTIONS */
GST_DEBUG_OBJECT (sink, "create options...");
res =
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_OPTIONS,
sink->conninfo.url_str);
if (res < 0)
goto create_request_failed;
/* send OPTIONS */
GST_DEBUG_OBJECT (sink, "send options...");
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "open",
("Retrieving server options"));
if ((res =
gst_rtsp_client_sink_send (sink, &sink->conninfo, &request,
&response, NULL)) < 0)
goto send_error;
/* parse OPTIONS */
if (!gst_rtsp_client_sink_parse_methods (sink, &response))
goto methods_error;
/* FIXME: Do we need to handle REDIRECT responses for OPTIONS? */
/* clean up any messages */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
return res;
/* ERRORS */
no_url:
{
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, (NULL),
("No valid RTSP URL was provided"));
goto cleanup_error;
}
connect_failed:
{
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ_WRITE, (NULL),
("Failed to connect. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "connect interrupted");
}
g_free (str);
goto cleanup_error;
}
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto cleanup_error;
}
send_error:
{
/* Don't post a message - the rtsp_send method will have
* taken care of it because we passed NULL for the response code */
goto cleanup_error;
}
methods_error:
{
/* error was posted */
res = GST_RTSP_ERROR;
goto cleanup_error;
}
cleanup_error:
{
if (sink->conninfo.connection) {
GST_DEBUG_OBJECT (sink, "free connection");
gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
}
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
return res;
}
}
static GstRTSPResult
gst_rtsp_client_sink_open (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPResult ret;
sink->methods =
GST_RTSP_SETUP | GST_RTSP_RECORD | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
g_mutex_lock (&sink->open_conn_lock);
sink->open_conn_start = TRUE;
g_cond_broadcast (&sink->open_conn_cond);
GST_DEBUG_OBJECT (sink, "connection to server started");
g_mutex_unlock (&sink->open_conn_lock);
if ((ret = gst_rtsp_client_sink_connect_to_server (sink, async)) < 0)
goto open_failed;
if (async)
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_OPEN, ret);
return ret;
/* ERRORS */
open_failed:
{
GST_WARNING_OBJECT (sink, "Failed to connect to server");
sink->open_error = TRUE;
if (async)
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_OPEN, ret);
return ret;
}
}
static GstRTSPResult
gst_rtsp_client_sink_close (GstRTSPClientSink * sink, gboolean async,
gboolean only_close)
{
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPResult res = GST_RTSP_OK;
GList *walk;
const gchar *control;
GST_DEBUG_OBJECT (sink, "TEARDOWN...");
gst_rtsp_client_sink_set_state (sink, GST_STATE_NULL);
if (sink->state < GST_RTSP_STATE_READY) {
GST_DEBUG_OBJECT (sink, "not ready, doing cleanup");
goto close;
}
if (only_close)
goto close;
/* construct a control url */
control = get_aggregate_control (sink);
if (!(sink->methods & (GST_RTSP_RECORD | GST_RTSP_TEARDOWN)))
goto not_supported;
/* stop streaming */
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
if (context->stream_transport)
gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
if (context->joined) {
gst_rtsp_stream_leave_bin (context->stream, GST_BIN (sink->internal_bin),
sink->rtpbin);
context->joined = FALSE;
}
}
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
const gchar *setup_url;
GstRTSPConnInfo *info;
GST_DEBUG_OBJECT (sink, "Looking at stream %p for teardown",
context->stream);
/* try aggregate control first but do non-aggregate control otherwise */
if (control)
setup_url = control;
else if ((setup_url = context->conninfo.location) == NULL) {
GST_DEBUG_OBJECT (sink, "Skipping TEARDOWN stream %p - no setup URL",
context->stream);
continue;
}
if (sink->conninfo.connection) {
info = &sink->conninfo;
} else if (context->conninfo.connection) {
info = &context->conninfo;
} else {
continue;
}
if (!info->connected)
goto next;
/* do TEARDOWN */
GST_DEBUG_OBJECT (sink, "Sending teardown for stream %p at URL %s",
context->stream, setup_url);
res =
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_TEARDOWN,
setup_url);
if (res < 0)
goto create_request_failed;
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "close", ("Closing stream"));
if ((res =
gst_rtsp_client_sink_send (sink, info, &request,
&response, NULL)) < 0)
goto send_error;
/* FIXME, parse result? */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
next:
/* early exit when we did aggregate control */
if (control)
break;
}
close:
/* close connections */
GST_DEBUG_OBJECT (sink, "closing connection...");
gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
gst_rtsp_conninfo_close (sink, &stream->conninfo, TRUE);
}
/* cleanup */
gst_rtsp_client_sink_cleanup (sink);
sink->state = GST_RTSP_STATE_INVALID;
if (async)
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_CLOSE, res);
return res;
/* ERRORS */
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto close;
}
send_error:
{
gchar *str = gst_rtsp_strresult (res);
gst_rtsp_message_unset (&request);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "TEARDOWN interrupted");
}
g_free (str);
goto close;
}
not_supported:
{
GST_DEBUG_OBJECT (sink,
"TEARDOWN and PLAY not supported, can't do TEARDOWN");
goto close;
}
}
static gboolean
gst_rtsp_client_sink_configure_manager (GstRTSPClientSink * sink)
{
GstElement *rtpbin;
GstStateChangeReturn ret;
rtpbin = sink->rtpbin;
if (rtpbin == NULL) {
GObjectClass *klass;
rtpbin = gst_element_factory_make ("rtpbin", NULL);
if (rtpbin == NULL)
goto no_rtpbin;
gst_bin_add (GST_BIN_CAST (sink->internal_bin), rtpbin);
sink->rtpbin = rtpbin;
/* Any more settings we should configure on rtpbin here? */
g_object_set (sink->rtpbin, "latency", sink->latency, NULL);
klass = G_OBJECT_GET_CLASS (G_OBJECT (rtpbin));
if (g_object_class_find_property (klass, "ntp-time-source")) {
g_object_set (sink->rtpbin, "ntp-time-source", sink->ntp_time_source,
NULL);
}
if (sink->sdes && g_object_class_find_property (klass, "sdes")) {
g_object_set (sink->rtpbin, "sdes", sink->sdes, NULL);
}
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_NEW_MANAGER], 0,
sink->rtpbin);
}
ret = gst_element_set_state (rtpbin, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_manager_failure;
return TRUE;
no_rtpbin:
{
GST_WARNING ("no rtpbin element");
g_warning ("failed to create element 'rtpbin', check your installation");
return FALSE;
}
start_manager_failure:
{
GST_DEBUG_OBJECT (sink, "could not start session manager");
gst_bin_remove (GST_BIN_CAST (sink->internal_bin), rtpbin);
return FALSE;
}
}
static GstElement *
request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPClientSink * sink)
{
GstRTSPStream *stream = NULL;
GstElement *ret = NULL;
GList *walk;
GST_RTSP_STATE_LOCK (sink);
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
if (sessid == gst_rtsp_stream_get_index (context->stream)) {
stream = context->stream;
break;
}
}
if (stream != NULL) {
GST_DEBUG_OBJECT (sink, "Creating aux sender for stream %u", sessid);
ret = gst_rtsp_stream_request_aux_sender (stream, sessid);
}
GST_RTSP_STATE_UNLOCK (sink);
return ret;
}
static gboolean
gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink)
{
GstRTSPStreamContext *context;
GList *walk;
const gchar *base;
gboolean has_slash;
GST_DEBUG_OBJECT (sink, "Collecting stream information");
if (!gst_rtsp_client_sink_configure_manager (sink))
return FALSE;
base = get_aggregate_control (sink);
/* check if the base ends with / */
has_slash = g_str_has_suffix (base, "/");
g_mutex_lock (&sink->preroll_lock);
while (sink->contexts == NULL && !sink->conninfo.flushing) {
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
}
g_mutex_unlock (&sink->preroll_lock);
/* FIXME: Need different locking - need to protect against pad releases
* and potential state changes ruining things here */
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstPad *srcpad;
context = (GstRTSPStreamContext *) walk->data;
if (context->stream)
continue;
g_mutex_lock (&sink->preroll_lock);
while (!context->prerolled && !sink->conninfo.flushing) {
GST_DEBUG_OBJECT (sink, "Waiting for caps on stream %d", context->index);
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
}
if (sink->conninfo.flushing) {
g_mutex_unlock (&sink->preroll_lock);
break;
}
g_mutex_unlock (&sink->preroll_lock);
if (context->payloader == NULL)
continue;
srcpad = gst_element_get_static_pad (context->payloader, "src");
GST_DEBUG_OBJECT (sink, "Creating stream object for stream %d",
context->index);
context->stream =
gst_rtsp_client_sink_create_stream (sink, context, context->payloader,
srcpad);
/* concatenate the two strings, insert / when not present */
g_free (context->conninfo.location);
context->conninfo.location =
g_strdup_printf ("%s%sstream=%d", base, has_slash ? "" : "/",
context->index);
if (sink->rtx_time > 0) {
/* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
g_signal_connect (sink->rtpbin, "request-aux-sender",
(GCallback) request_aux_sender, sink);
}
if (!gst_rtsp_stream_join_bin (context->stream,
GST_BIN (sink->internal_bin), sink->rtpbin, GST_STATE_PAUSED)) {
goto join_bin_failed;
}
context->joined = TRUE;
/* Block the stream, as it does not have any transport parts yet */
gst_rtsp_stream_set_blocked (context->stream, TRUE);
/* Let the stream object receive data */
gst_pad_remove_probe (srcpad, context->payloader_block_id);
gst_object_unref (srcpad);
}
/* Now wait for the preroll of the rtp bin */
g_mutex_lock (&sink->preroll_lock);
while (!sink->prerolled && !sink->conninfo.flushing) {
GST_LOG_OBJECT (sink, "Waiting for preroll before continuing");
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
}
GST_LOG_OBJECT (sink, "Marking streams as collected");
sink->streams_collected = TRUE;
g_mutex_unlock (&sink->preroll_lock);
return TRUE;
join_bin_failed:
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not start stream %d", context->index));
return FALSE;
}
static GstRTSPResult
gst_rtsp_client_sink_create_transports_string (GstRTSPClientSink * sink,
GstRTSPStreamContext * context, GSocketFamily family,
GstRTSPLowerTrans protocols, GstRTSPProfile profiles, gchar ** transports)
{
GString *result;
GstRTSPStream *stream = context->stream;
gboolean first = TRUE;
/* the default RTSP transports */
result = g_string_new ("RTP");
while (profiles != 0) {
if (!first)
g_string_append (result, ",RTP");
if (profiles & GST_RTSP_PROFILE_SAVPF) {
g_string_append (result, "/SAVPF");
profiles &= ~GST_RTSP_PROFILE_SAVPF;
} else if (profiles & GST_RTSP_PROFILE_SAVP) {
g_string_append (result, "/SAVP");
profiles &= ~GST_RTSP_PROFILE_SAVP;
} else if (profiles & GST_RTSP_PROFILE_AVPF) {
g_string_append (result, "/AVPF");
profiles &= ~GST_RTSP_PROFILE_AVPF;
} else if (profiles & GST_RTSP_PROFILE_AVP) {
g_string_append (result, "/AVP");
profiles &= ~GST_RTSP_PROFILE_AVP;
} else {
GST_WARNING_OBJECT (sink, "Unimplemented profile(s) 0x%x", profiles);
break;
}
if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
GstRTSPRange ports;
GST_DEBUG_OBJECT (sink, "adding UDP unicast");
gst_rtsp_stream_get_server_port (stream, &ports, family);
g_string_append_printf (result, "/UDP;unicast;client_port=%d-%d",
ports.min, ports.max);
} else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
GstRTSPAddress *addr =
gst_rtsp_stream_get_multicast_address (stream, family);
if (addr) {
GST_DEBUG_OBJECT (sink, "adding UDP multicast");
g_string_append_printf (result, "/UDP;multicast;client_port=%d-%d",
addr->port, addr->port + addr->n_ports - 1);
gst_rtsp_address_free (addr);
}
} else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
GST_DEBUG_OBJECT (sink, "adding TCP");
g_string_append_printf (result, "/TCP;unicast;interleaved=%d-%d",
sink->free_channel, sink->free_channel + 1);
}
g_string_append (result, ";mode=RECORD");
/* FIXME: Support appending too:
if (sink->append)
g_string_append (result, ";append");
*/
first = FALSE;
}
if (first) {
/* No valid transport could be constructed */
GST_ERROR_OBJECT (sink, "No supported profiles configured");
goto fail;
}
*transports = g_string_free (result, FALSE);
GST_DEBUG_OBJECT (sink, "prepared transports %s", GST_STR_NULL (*transports));
return GST_RTSP_OK;
fail:
g_string_free (result, TRUE);
return GST_RTSP_ERROR;
}
static GstCaps *
signal_get_srtcp_params (GstRTSPClientSink * sink,
GstRTSPStreamContext * context)
{
GstCaps *caps = NULL;
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_REQUEST_RTCP_KEY], 0,
context->index, &caps);
if (caps != NULL)
GST_DEBUG_OBJECT (sink, "SRTP parameters received");
return caps;
}
static gchar *
gst_rtsp_client_sink_stream_make_keymgmt (GstRTSPClientSink * sink,
GstRTSPStreamContext * context)
{
gchar *base64, *result = NULL;
GstMIKEYMessage *mikey_msg;
context->srtcpparams = signal_get_srtcp_params (sink, context);
if (context->srtcpparams == NULL)
context->srtcpparams = gst_rtsp_stream_get_caps (context->stream);
mikey_msg = gst_mikey_message_new_from_caps (context->srtcpparams);
if (mikey_msg) {
guint send_ssrc;
/* add policy '0' for our SSRC */
gst_rtsp_stream_get_ssrc (context->stream, &send_ssrc);
GST_LOG_OBJECT (sink, "Stream %p ssrc %x", context->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 (context->conninfo.location, base64);
g_free (base64);
}
}
return result;
}
/* masks to be kept in sync with the hardcoded protocol order of preference
* in code below */
static const guint protocol_masks[] = {
GST_RTSP_LOWER_TRANS_UDP,
GST_RTSP_LOWER_TRANS_UDP_MCAST,
GST_RTSP_LOWER_TRANS_TCP,
0
};
/* Same for profile_masks */
static const guint profile_masks[] = {
GST_RTSP_PROFILE_SAVPF,
GST_RTSP_PROFILE_SAVP,
GST_RTSP_PROFILE_AVPF,
GST_RTSP_PROFILE_AVP,
0
};
static gboolean
do_send_data (GstBuffer * buffer, guint8 channel,
GstRTSPStreamContext * context)
{
GstRTSPClientSink *sink = context->parent;
GstRTSPMessage message = { 0 };
GstRTSPResult res = GST_RTSP_OK;
GstMapInfo map_info;
guint8 *data;
guint usize;
gst_rtsp_message_init_data (&message, channel);
/* FIXME, need some sort of iovec RTSPMessage here */
if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
return FALSE;
gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
res =
gst_rtsp_client_sink_try_send (sink, &sink->conninfo, &message,
NULL, NULL);
gst_rtsp_message_steal_body (&message, &data, &usize);
gst_buffer_unmap (buffer, &map_info);
gst_rtsp_message_unset (&message);
return res == GST_RTSP_OK;
}
static GstRTSPResult
gst_rtsp_client_sink_setup_streams (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPResult res = GST_RTSP_ERROR;
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPLowerTrans protocols;
GstRTSPStatusCode code;
GSocketFamily family;
GSocketAddress *sa;
GSocket *conn_socket;
GstRTSPUrl *url;
GList *walk;
gchar *hval;
if (sink->conninfo.connection) {
url = gst_rtsp_connection_get_url (sink->conninfo.connection);
/* we initially allow all configured lower transports. based on the URL
* transports and the replies from the server we narrow them down. */
protocols = url->transports & sink->cur_protocols;
} else {
url = NULL;
protocols = sink->cur_protocols;
}
if (protocols == 0)
goto no_protocols;
GST_RTSP_STATE_LOCK (sink);
if (G_UNLIKELY (sink->contexts == NULL))
goto no_streams;
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
GstRTSPStream *stream;
GstRTSPConnInfo *info;
GstRTSPProfile profiles;
GstRTSPProfile cur_profile;
gchar *transports;
gint retry = 0;
guint profile_mask = 0;
guint mask = 0;
GstCaps *caps;
const GstSDPMedia *media;
stream = context->stream;
profiles = gst_rtsp_stream_get_profiles (stream);
caps = gst_rtsp_stream_get_caps (stream);
if (caps == NULL) {
GST_DEBUG_OBJECT (sink, "skipping stream %p, no caps", stream);
continue;
}
gst_caps_unref (caps);
media = gst_sdp_message_get_media (&sink->cursdp, context->sdp_index);
if (media == NULL) {
GST_DEBUG_OBJECT (sink, "skipping stream %p, no SDP info", stream);
continue;
}
/* skip setup if we have no URL for it */
if (context->conninfo.location == NULL) {
GST_DEBUG_OBJECT (sink, "skipping stream %p, no setup", stream);
continue;
}
if (sink->conninfo.connection == NULL) {
if (!gst_rtsp_conninfo_connect (sink, &context->conninfo, async)) {
GST_DEBUG_OBJECT (sink, "skipping stream %p, failed to connect",
stream);
continue;
}
info = &context->conninfo;
} else {
info = &sink->conninfo;
}
GST_DEBUG_OBJECT (sink, "doing setup of stream %p with %s", stream,
context->conninfo.location);
conn_socket = gst_rtsp_connection_get_read_socket (info->connection);
sa = g_socket_get_local_address (conn_socket, NULL);
family = g_socket_address_get_family (sa);
g_object_unref (sa);
next_protocol:
/* first selectable profile */
while (profile_masks[profile_mask]
&& !(profiles & profile_masks[profile_mask]))
profile_mask++;
if (!profile_masks[profile_mask])
goto no_profiles;
/* first selectable protocol */
while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
mask++;
if (!protocol_masks[mask])
goto no_protocols;
retry:
GST_DEBUG_OBJECT (sink, "protocols = 0x%x, protocol mask = 0x%x", protocols,
protocol_masks[mask]);
/* create a string with first transport in line */
transports = NULL;
cur_profile = profiles & profile_masks[profile_mask];
res = gst_rtsp_client_sink_create_transports_string (sink, context, family,
protocols & protocol_masks[mask], cur_profile, &transports);
if (res < 0 || transports == NULL)
goto setup_transport_failed;
if (strlen (transports) == 0) {
g_free (transports);
GST_DEBUG_OBJECT (sink, "no transports found");
mask++;
profile_mask = 0;
goto next_protocol;
}
GST_DEBUG_OBJECT (sink, "transport is %s", GST_STR_NULL (transports));
/* create SETUP request */
res =
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_SETUP,
context->conninfo.location);
if (res < 0) {
g_free (transports);
goto create_request_failed;
}
/* select transport */
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
/* set up keys */
if (cur_profile == GST_RTSP_PROFILE_SAVP ||
cur_profile == GST_RTSP_PROFILE_SAVPF) {
hval = gst_rtsp_client_sink_stream_make_keymgmt (sink, context);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_KEYMGMT, hval);
}
/* if the user wants a non default RTP packet size we add the blocksize
* parameter */
if (sink->rtp_blocksize > 0) {
hval = g_strdup_printf ("%d", sink->rtp_blocksize);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
}
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "request", ("SETUP stream %d",
context->index));
{
GstRTSPTransport *transport;
gst_rtsp_transport_new (&transport);
if (gst_rtsp_transport_parse (transports, transport) != GST_RTSP_OK)
goto parse_transport_failed;
if (transport->lower_transport != GST_RTSP_LOWER_TRANS_TCP) {
if (!gst_rtsp_stream_allocate_udp_sockets (stream, family, transport,
FALSE)) {
gst_rtsp_transport_free (transport);
goto allocate_udp_ports_failed;
}
}
if (!gst_rtsp_stream_complete_stream (stream, transport)) {
gst_rtsp_transport_free (transport);
goto complete_stream_failed;
}
gst_rtsp_transport_free (transport);
gst_rtsp_stream_set_blocked (stream, FALSE);
}
/* handle the code ourselves */
res = gst_rtsp_client_sink_send (sink, info, &request, &response, &code);
if (res < 0)
goto send_error;
switch (code) {
case GST_RTSP_STS_OK:
break;
case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
/* Try another profile. If no more, move to the next protocol */
profile_mask++;
while (profile_masks[profile_mask]
&& !(profiles & profile_masks[profile_mask]))
profile_mask++;
if (profile_masks[profile_mask])
goto retry;
/* select next available protocol, give up on this stream if none */
/* Reset profiles to try: */
profile_mask = 0;
mask++;
while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
mask++;
if (!protocol_masks[mask])
continue;
else
goto retry;
default:
goto response_error;
}
/* parse response transport */
{
gchar *resptrans = NULL;
GstRTSPTransport *transport;
gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
&resptrans, 0);
if (!resptrans) {
goto no_transport;
}
gst_rtsp_transport_new (&transport);
/* parse transport, go to next stream on parse error */
if (gst_rtsp_transport_parse (resptrans, transport) != GST_RTSP_OK) {
GST_WARNING_OBJECT (sink, "failed to parse transport %s", resptrans);
goto next;
}
/* update allowed transports for other streams. once the transport of
* one stream has been determined, we make sure that all other streams
* are configured in the same way */
switch (transport->lower_transport) {
case GST_RTSP_LOWER_TRANS_TCP:
GST_DEBUG_OBJECT (sink, "stream %p as TCP interleaved", stream);
protocols = GST_RTSP_LOWER_TRANS_TCP;
sink->interleaved = TRUE;
/* update free channels */
sink->free_channel =
MAX (transport->interleaved.min, sink->free_channel);
sink->free_channel =
MAX (transport->interleaved.max, sink->free_channel);
sink->free_channel++;
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
/* only allow multicast for other streams */
GST_DEBUG_OBJECT (sink, "stream %p as UDP multicast", stream);
protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
break;
case GST_RTSP_LOWER_TRANS_UDP:
/* only allow unicast for other streams */
GST_DEBUG_OBJECT (sink, "stream %p as UDP unicast", stream);
protocols = GST_RTSP_LOWER_TRANS_UDP;
/* Update transport with server destination if not provided by the server */
if (transport->destination == NULL) {
transport->destination = g_strdup (sink->server_ip);
}
break;
default:
GST_DEBUG_OBJECT (sink, "stream %p unknown transport %d", stream,
transport->lower_transport);
break;
}
if (!retry) {
GST_DEBUG ("Configuring the stream transport for stream %d",
context->index);
if (context->stream_transport == NULL)
context->stream_transport =
gst_rtsp_stream_transport_new (stream, transport);
else
gst_rtsp_stream_transport_set_transport (context->stream_transport,
transport);
if (transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* our callbacks to send data on this TCP connection */
gst_rtsp_stream_transport_set_callbacks (context->stream_transport,
(GstRTSPSendFunc) do_send_data,
(GstRTSPSendFunc) do_send_data, context, NULL);
}
/* The stream_transport now owns the transport */
transport = NULL;
gst_rtsp_stream_transport_set_active (context->stream_transport, TRUE);
}
next:
if (transport)
gst_rtsp_transport_free (transport);
/* clean up used RTSP messages */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
}
}
GST_RTSP_STATE_UNLOCK (sink);
/* store the transport protocol that was configured */
sink->cur_protocols = protocols;
return res;
no_streams:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("SDP contains no streams"));
return GST_RTSP_ERROR;
}
setup_transport_failed:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not setup transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
no_profiles:
{
GST_RTSP_STATE_UNLOCK (sink);
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not connect to server, no profiles left"));
return GST_RTSP_ERROR;
}
no_protocols:
{
GST_RTSP_STATE_UNLOCK (sink);
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
("Could not connect to server, no protocols left"));
return GST_RTSP_ERROR;
}
no_transport:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Server did not select transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto cleanup_error;
}
parse_transport_failed:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not parse transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
allocate_udp_ports_failed:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not parse transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
complete_stream_failed:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not parse transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
send_error:
{
gchar *str = gst_rtsp_strresult (res);
GST_RTSP_STATE_UNLOCK (sink);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "send interrupted");
}
g_free (str);
goto cleanup_error;
}
response_error:
{
const gchar *str = gst_rtsp_status_as_text (code);
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Error (%d): %s", code, GST_STR_NULL (str)));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
cleanup_error:
{
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
return res;
}
}
static GstRTSPResult
gst_rtsp_client_sink_ensure_open (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPResult res = GST_RTSP_OK;
if (sink->state < GST_RTSP_STATE_READY) {
res = GST_RTSP_ERROR;
if (sink->open_error) {
GST_DEBUG_OBJECT (sink, "the stream was in error");
goto done;
}
if (async)
gst_rtsp_client_sink_loop_start_cmd (sink, CMD_OPEN);
if ((res = gst_rtsp_client_sink_open (sink, async)) < 0) {
GST_DEBUG_OBJECT (sink, "failed to open stream");
goto done;
}
}
done:
return res;
}
static GstRTSPResult
gst_rtsp_client_sink_record (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPResult res = GST_RTSP_OK;
GstSDPMessage *sdp;
guint sdp_index = 0;
GstSDPInfo info = { 0, };
const gchar *proto;
gchar *sess_id, *client_ip, *str;
GSocketAddress *sa;
GInetAddress *ia;
GSocket *conn_socket;
GList *walk;
g_mutex_lock (&sink->preroll_lock);
if (sink->state == GST_RTSP_STATE_PLAYING) {
/* Already recording, don't send another request */
GST_LOG_OBJECT (sink, "Already in RECORD. Skipping duplicate request.");
g_mutex_unlock (&sink->preroll_lock);
goto done;
}
g_mutex_unlock (&sink->preroll_lock);
/* Collect all our input streams and create
* stream objects before actually returning.
* The streams are blocked at this point as we do not have any transport
* parts yet. */
gst_rtsp_client_sink_collect_streams (sink);
g_mutex_lock (&sink->block_streams_lock);
/* Wait for streams to be blocked */
while (!sink->streams_blocked) {
GST_DEBUG_OBJECT (sink, "waiting for streams to be blocked");
g_cond_wait (&sink->block_streams_cond, &sink->block_streams_lock);
}
g_mutex_unlock (&sink->block_streams_lock);
/* Send announce, then setup for all streams */
gst_sdp_message_init (&sink->cursdp);
sdp = &sink->cursdp;
/* some standard things first */
gst_sdp_message_set_version (sdp, "0");
/* session ID doesn't have to be super-unique in this case */
sess_id = g_strdup_printf ("%u", g_random_int ());
if (sink->conninfo.connection == NULL)
return GST_RTSP_ERROR;
conn_socket = gst_rtsp_connection_get_read_socket (sink->conninfo.connection);
sa = g_socket_get_local_address (conn_socket, NULL);
ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
client_ip = g_inet_address_to_string (ia);
if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV6) {
info.is_ipv6 = TRUE;
proto = "IP6";
} else if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV4)
proto = "IP4";
else
g_assert_not_reached ();
g_object_unref (sa);
/* FIXME: Should this actually be the server's IP or ours? */
info.server_ip = sink->server_ip;
gst_sdp_message_set_origin (sdp, "-", sess_id, "1", "IN", proto, client_ip);
gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
gst_sdp_message_set_information (sdp, "rtspclientsink");
gst_sdp_message_add_time (sdp, "0", "0", NULL);
gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
/* add stream */
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
gst_rtsp_sdp_from_stream (sdp, &info, context->stream);
context->sdp_index = sdp_index++;
}
g_free (sess_id);
g_free (client_ip);
/* send ANNOUNCE request */
GST_DEBUG_OBJECT (sink, "create ANNOUNCE request...");
res =
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_ANNOUNCE,
sink->conninfo.url_str);
if (res < 0)
goto create_request_failed;
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CONTENT_TYPE,
"application/sdp");
/* add SDP to the request body */
str = gst_sdp_message_as_text (sdp);
gst_rtsp_message_take_body (&request, (guint8 *) str, strlen (str));
/* send ANNOUNCE */
GST_DEBUG_OBJECT (sink, "sending announce...");
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "record",
("Sending server stream info"));
if ((res =
gst_rtsp_client_sink_send (sink, &sink->conninfo, &request,
&response, NULL)) < 0)
goto send_error;
/* send setup for all streams */
if ((res = gst_rtsp_client_sink_setup_streams (sink, async)) < 0)
goto setup_failed;
res = gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_RECORD,
sink->conninfo.url_str);
if (res < 0)
goto create_request_failed;
#if 0 /* FIXME: Configure a range based on input segments? */
if (src->need_range) {
hval = gen_range_header (src, segment);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
}
if (segment->rate != 1.0) {
gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
g_ascii_dtostr (hval, sizeof (hval), segment->rate);
if (src->skip)
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
else
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
}
#endif
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "record", ("Starting recording"));
if ((res =
gst_rtsp_client_sink_send (sink, &sink->conninfo, &request,
&response, NULL)) < 0)
goto send_error;
#if 0 /* FIXME: Check if servers return these for record: */
/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
* for the RTP packets. If this is not present, we assume all starts from 0...
* This is info for the RTP session manager that we pass to it in caps. */
hval_idx = 0;
while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
&hval, hval_idx++) == GST_RTSP_OK)
gst_rtspsrc_parse_rtpinfo (src, hval);
/* some servers indicate RTCP parameters in PLAY response,
* rather than properly in SDP */
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
&hval, 0) == GST_RTSP_OK)
gst_rtspsrc_handle_rtcp_interval (src, hval);
#endif
gst_rtsp_client_sink_set_state (sink, GST_STATE_PLAYING);
sink->state = GST_RTSP_STATE_PLAYING;
/* clean up any messages */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
done:
return res;
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto cleanup_error;
}
send_error:
{
/* Don't post a message - the rtsp_send method will have
* taken care of it because we passed NULL for the response code */
goto cleanup_error;
}
setup_failed:
{
GST_ERROR_OBJECT (sink, "setup failed");
goto cleanup_error;
}
cleanup_error:
{
if (sink->conninfo.connection) {
GST_DEBUG_OBJECT (sink, "free connection");
gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
}
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
return res;
}
}
static GstRTSPResult
gst_rtsp_client_sink_pause (GstRTSPClientSink * sink, gboolean async)
{
GstRTSPResult res = GST_RTSP_OK;
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GList *walk;
const gchar *control;
GST_DEBUG_OBJECT (sink, "PAUSE...");
if ((res = gst_rtsp_client_sink_ensure_open (sink, async)) < 0)
goto open_failed;
if (!(sink->methods & GST_RTSP_PAUSE))
goto not_supported;
if (sink->state == GST_RTSP_STATE_READY)
goto was_paused;
if (!sink->conninfo.connection || !sink->conninfo.connected)
goto no_connection;
/* construct a control url */
control = get_aggregate_control (sink);
/* loop over the streams. We might exit the loop early when we could do an
* aggregate control */
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
GstRTSPConnInfo *info;
const gchar *setup_url;
/* try aggregate control first but do non-aggregate control otherwise */
if (control)
setup_url = control;
else if ((setup_url = stream->conninfo.location) == NULL)
continue;
if (sink->conninfo.connection) {
info = &sink->conninfo;
} else if (stream->conninfo.connection) {
info = &stream->conninfo;
} else {
continue;
}
if (async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "request",
("Sending PAUSE request"));
if ((res =
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_PAUSE,
setup_url)) < 0)
goto create_request_failed;
if ((res =
gst_rtsp_client_sink_send (sink, info, &request, &response,
NULL)) < 0)
goto send_error;
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
/* exit early when we did agregate control */
if (control)
break;
}
/* change element states now */
gst_rtsp_client_sink_set_state (sink, GST_STATE_PAUSED);
no_connection:
sink->state = GST_RTSP_STATE_READY;
done:
if (async)
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_PAUSE, res);
return res;
/* ERRORS */
open_failed:
{
GST_DEBUG_OBJECT (sink, "failed to open stream");
goto done;
}
not_supported:
{
GST_DEBUG_OBJECT (sink, "PAUSE is not supported");
goto done;
}
was_paused:
{
GST_DEBUG_OBJECT (sink, "we were already PAUSED");
goto done;
}
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto done;
}
send_error:
{
gchar *str = gst_rtsp_strresult (res);
gst_rtsp_message_unset (&request);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (sink, "PAUSE interrupted");
}
g_free (str);
goto done;
}
}
static void
gst_rtsp_client_sink_handle_message (GstBin * bin, GstMessage * message)
{
GstRTSPClientSink *rtsp_client_sink;
rtsp_client_sink = GST_RTSP_CLIENT_SINK (bin);
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ELEMENT:
{
const GstStructure *s = gst_message_get_structure (message);
if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
gboolean ignore_timeout;
GST_DEBUG_OBJECT (bin, "timeout on UDP port");
GST_OBJECT_LOCK (rtsp_client_sink);
ignore_timeout = rtsp_client_sink->ignore_timeout;
rtsp_client_sink->ignore_timeout = TRUE;
GST_OBJECT_UNLOCK (rtsp_client_sink);
/* we only act on the first udp timeout message, others are irrelevant
* and can be ignored. */
if (!ignore_timeout)
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_RECONNECT,
CMD_LOOP);
/* eat and free */
gst_message_unref (message);
return;
} else if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
/* An RTSPStream has prerolled */
GST_DEBUG_OBJECT (rtsp_client_sink, "received GstRTSPStreamBlocking");
g_mutex_lock (&rtsp_client_sink->block_streams_lock);
rtsp_client_sink->streams_blocked = TRUE;
g_cond_broadcast (&rtsp_client_sink->block_streams_cond);
g_mutex_unlock (&rtsp_client_sink->block_streams_lock);
}
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
case GST_MESSAGE_ASYNC_START:{
GstObject *sender;
sender = GST_MESSAGE_SRC (message);
GST_LOG_OBJECT (rtsp_client_sink,
"Have async-start from %" GST_PTR_FORMAT, sender);
if (sender == GST_OBJECT (rtsp_client_sink->internal_bin)) {
GST_LOG_OBJECT (rtsp_client_sink, "child bin is now ASYNC");
}
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
case GST_MESSAGE_ASYNC_DONE:
{
GstObject *sender;
gboolean need_async_done;
sender = GST_MESSAGE_SRC (message);
GST_LOG_OBJECT (rtsp_client_sink, "Have async-done from %" GST_PTR_FORMAT,
sender);
g_mutex_lock (&rtsp_client_sink->preroll_lock);
if (sender == GST_OBJECT_CAST (rtsp_client_sink->internal_bin)) {
GST_LOG_OBJECT (rtsp_client_sink, "child bin is no longer ASYNC");
}
need_async_done = rtsp_client_sink->in_async;
if (rtsp_client_sink->in_async) {
rtsp_client_sink->in_async = FALSE;
g_cond_broadcast (&rtsp_client_sink->preroll_cond);
}
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
if (need_async_done) {
GST_DEBUG_OBJECT (rtsp_client_sink, "Posting ASYNC-DONE");
gst_element_post_message (GST_ELEMENT_CAST (rtsp_client_sink),
gst_message_new_async_done (GST_OBJECT_CAST (rtsp_client_sink),
GST_CLOCK_TIME_NONE));
}
break;
}
case GST_MESSAGE_ERROR:
{
GstObject *sender;
sender = GST_MESSAGE_SRC (message);
GST_DEBUG_OBJECT (rtsp_client_sink, "got error from %s",
GST_ELEMENT_NAME (sender));
/* FIXME: Ignore errors on RTCP? */
/* fatal but not our message, forward */
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
case GST_MESSAGE_STATE_CHANGED:
{
if (GST_MESSAGE_SRC (message) ==
(GstObject *) rtsp_client_sink->internal_bin) {
GstState newstate, pending;
gst_message_parse_state_changed (message, NULL, &newstate, &pending);
g_mutex_lock (&rtsp_client_sink->preroll_lock);
rtsp_client_sink->prerolled = (newstate >= GST_STATE_PAUSED)
&& pending == GST_STATE_VOID_PENDING;
g_cond_broadcast (&rtsp_client_sink->preroll_cond);
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
GST_DEBUG_OBJECT (bin,
"Internal bin changed state to %s (pending %s). Prerolled now %d",
gst_element_state_get_name (newstate),
gst_element_state_get_name (pending), rtsp_client_sink->prerolled);
}
/* fallthrough */
}
default:
{
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
}
}
/* the thread where everything happens */
static void
gst_rtsp_client_sink_thread (GstRTSPClientSink * sink)
{
gint cmd;
GST_OBJECT_LOCK (sink);
cmd = sink->pending_cmd;
if (cmd == CMD_RECONNECT || cmd == CMD_RECORD || cmd == CMD_PAUSE
|| cmd == CMD_LOOP || cmd == CMD_OPEN)
sink->pending_cmd = CMD_LOOP;
else
sink->pending_cmd = CMD_WAIT;
GST_DEBUG_OBJECT (sink, "got command %s", cmd_to_string (cmd));
/* we got the message command, so ensure communication is possible again */
gst_rtsp_client_sink_connection_flush (sink, FALSE);
sink->busy_cmd = cmd;
GST_OBJECT_UNLOCK (sink);
switch (cmd) {
case CMD_OPEN:
if (gst_rtsp_client_sink_open (sink, TRUE) == GST_RTSP_ERROR)
gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT,
CMD_ALL & ~CMD_CLOSE);
break;
case CMD_RECORD:
gst_rtsp_client_sink_record (sink, TRUE);
break;
case CMD_PAUSE:
gst_rtsp_client_sink_pause (sink, TRUE);
break;
case CMD_CLOSE:
gst_rtsp_client_sink_close (sink, TRUE, FALSE);
break;
case CMD_LOOP:
gst_rtsp_client_sink_loop (sink);
break;
case CMD_RECONNECT:
gst_rtsp_client_sink_reconnect (sink, FALSE);
break;
default:
break;
}
GST_OBJECT_LOCK (sink);
/* and go back to sleep */
if (sink->pending_cmd == CMD_WAIT) {
if (sink->task)
gst_task_pause (sink->task);
}
/* reset waiting */
sink->busy_cmd = CMD_WAIT;
GST_OBJECT_UNLOCK (sink);
}
static gboolean
gst_rtsp_client_sink_start (GstRTSPClientSink * sink)
{
GST_DEBUG_OBJECT (sink, "starting");
sink->streams_collected = FALSE;
gst_element_set_locked_state (GST_ELEMENT (sink->internal_bin), TRUE);
gst_rtsp_client_sink_set_state (sink, GST_STATE_READY);
GST_OBJECT_LOCK (sink);
sink->pending_cmd = CMD_WAIT;
if (sink->task == NULL) {
sink->task =
gst_task_new ((GstTaskFunction) gst_rtsp_client_sink_thread, sink,
NULL);
if (sink->task == NULL)
goto task_error;
gst_task_set_lock (sink->task, GST_RTSP_STREAM_GET_LOCK (sink));
}
GST_OBJECT_UNLOCK (sink);
return TRUE;
/* ERRORS */
task_error:
{
GST_OBJECT_UNLOCK (sink);
GST_ERROR_OBJECT (sink, "failed to create task");
return FALSE;
}
}
static gboolean
gst_rtsp_client_sink_stop (GstRTSPClientSink * sink)
{
GstTask *task;
GST_DEBUG_OBJECT (sink, "stopping");
/* also cancels pending task */
gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT, CMD_ALL & ~CMD_CLOSE);
GST_OBJECT_LOCK (sink);
if ((task = sink->task)) {
sink->task = NULL;
GST_OBJECT_UNLOCK (sink);
gst_task_stop (task);
/* make sure it is not running */
GST_RTSP_STREAM_LOCK (sink);
GST_RTSP_STREAM_UNLOCK (sink);
/* now wait for the task to finish */
gst_task_join (task);
/* and free the task */
gst_object_unref (GST_OBJECT (task));
GST_OBJECT_LOCK (sink);
}
GST_OBJECT_UNLOCK (sink);
/* ensure synchronously all is closed and clean */
gst_rtsp_client_sink_close (sink, FALSE, TRUE);
return TRUE;
}
static GstStateChangeReturn
gst_rtsp_client_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstRTSPClientSink *rtsp_client_sink;
GstStateChangeReturn ret;
rtsp_client_sink = GST_RTSP_CLIENT_SINK (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!gst_rtsp_client_sink_start (rtsp_client_sink))
goto start_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* init some state */
rtsp_client_sink->cur_protocols = rtsp_client_sink->protocols;
/* first attempt, don't ignore timeouts */
rtsp_client_sink->ignore_timeout = FALSE;
rtsp_client_sink->open_error = FALSE;
gst_rtsp_client_sink_set_state (rtsp_client_sink, GST_STATE_PAUSED);
g_mutex_lock (&rtsp_client_sink->preroll_lock);
if (rtsp_client_sink->in_async) {
GST_DEBUG_OBJECT (rtsp_client_sink, "Posting ASYNC-START");
gst_element_post_message (GST_ELEMENT_CAST (rtsp_client_sink),
gst_message_new_async_start (GST_OBJECT_CAST (rtsp_client_sink)));
}
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
/* fall-through */
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* unblock the tcp tasks and make the loop waiting */
if (gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_WAIT,
CMD_LOOP)) {
/* make sure it is waiting before we send PLAY below */
GST_RTSP_STREAM_LOCK (rtsp_client_sink);
GST_RTSP_STREAM_UNLOCK (rtsp_client_sink);
}
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtsp_client_sink_set_state (rtsp_client_sink, GST_STATE_READY);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
goto done;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* Return ASYNC and preroll input streams */
g_mutex_lock (&rtsp_client_sink->preroll_lock);
if (rtsp_client_sink->in_async)
ret = GST_STATE_CHANGE_ASYNC;
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_OPEN, 0);
/* CMD_OPEN has been scheduled. Wait until the sink thread starts
* opening connection to the server */
g_mutex_lock (&rtsp_client_sink->open_conn_lock);
while (!rtsp_client_sink->open_conn_start) {
GST_DEBUG_OBJECT (rtsp_client_sink,
"wait for connection to be started");
g_cond_wait (&rtsp_client_sink->open_conn_cond,
&rtsp_client_sink->open_conn_lock);
}
rtsp_client_sink->open_conn_start = FALSE;
g_mutex_unlock (&rtsp_client_sink->open_conn_lock);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
GST_DEBUG_OBJECT (rtsp_client_sink,
"Switching to playing -sending RECORD");
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_RECORD, 0);
ret = GST_STATE_CHANGE_SUCCESS;
break;
}
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* send pause request and keep the idle task around */
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_PAUSE,
CMD_LOOP);
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_CLOSE,
CMD_PAUSE);
ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_rtsp_client_sink_stop (rtsp_client_sink);
ret = GST_STATE_CHANGE_SUCCESS;
break;
default:
break;
}
done:
return ret;
start_failed:
{
GST_DEBUG_OBJECT (rtsp_client_sink, "start failed");
return GST_STATE_CHANGE_FAILURE;
}
}
/*** GSTURIHANDLER INTERFACE *************************************************/
static GstURIType
gst_rtsp_client_sink_uri_get_type (GType type)
{
return GST_URI_SINK;
}
static const gchar *const *
gst_rtsp_client_sink_uri_get_protocols (GType type)
{
static const gchar *protocols[] =
{ "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
"rtsps", "rtspsu", "rtspst", "rtspsh", NULL
};
return protocols;
}
static gchar *
gst_rtsp_client_sink_uri_get_uri (GstURIHandler * handler)
{
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (handler);
/* FIXME: make thread-safe */
return g_strdup (sink->conninfo.location);
}
static gboolean
gst_rtsp_client_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri,
GError ** error)
{
GstRTSPClientSink *sink;
GstRTSPResult res;
GstSDPResult sres;
GstRTSPUrl *newurl = NULL;
GstSDPMessage *sdp = NULL;
sink = GST_RTSP_CLIENT_SINK (handler);
/* same URI, we're fine */
if (sink->conninfo.location && uri && !strcmp (uri, sink->conninfo.location))
goto was_ok;
if (g_str_has_prefix (uri, "rtsp-sdp://")) {
sres = gst_sdp_message_new (&sdp);
if (sres < 0)
goto sdp_failed;
GST_DEBUG_OBJECT (sink, "parsing SDP message");
sres = gst_sdp_message_parse_uri (uri, sdp);
if (sres < 0)
goto invalid_sdp;
} else {
/* try to parse */
GST_DEBUG_OBJECT (sink, "parsing URI");
if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
goto parse_error;
}
/* if worked, free previous and store new url object along with the original
* location. */
GST_DEBUG_OBJECT (sink, "configuring URI");
g_free (sink->conninfo.location);
sink->conninfo.location = g_strdup (uri);
gst_rtsp_url_free (sink->conninfo.url);
sink->conninfo.url = newurl;
g_free (sink->conninfo.url_str);
if (newurl)
sink->conninfo.url_str = gst_rtsp_url_get_request_uri (sink->conninfo.url);
else
sink->conninfo.url_str = NULL;
if (sink->uri_sdp)
gst_sdp_message_free (sink->uri_sdp);
sink->uri_sdp = sdp;
sink->from_sdp = sdp != NULL;
GST_DEBUG_OBJECT (sink, "set uri: %s", GST_STR_NULL (uri));
GST_DEBUG_OBJECT (sink, "request uri is: %s",
GST_STR_NULL (sink->conninfo.url_str));
return TRUE;
/* Special cases */
was_ok:
{
GST_DEBUG_OBJECT (sink, "URI was ok: '%s'", GST_STR_NULL (uri));
return TRUE;
}
sdp_failed:
{
GST_ERROR_OBJECT (sink, "Could not create new SDP (%d)", sres);
g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
"Could not create SDP");
return FALSE;
}
invalid_sdp:
{
GST_ERROR_OBJECT (sink, "Not a valid SDP (%d) '%s'", sres,
GST_STR_NULL (uri));
gst_sdp_message_free (sdp);
g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
"Invalid SDP");
return FALSE;
}
parse_error:
{
GST_ERROR_OBJECT (sink, "Not a valid RTSP url '%s' (%d)",
GST_STR_NULL (uri), res);
g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
"Invalid RTSP URI");
return FALSE;
}
}
static void
gst_rtsp_client_sink_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_rtsp_client_sink_uri_get_type;
iface->get_protocols = gst_rtsp_client_sink_uri_get_protocols;
iface->get_uri = gst_rtsp_client_sink_uri_get_uri;
iface->set_uri = gst_rtsp_client_sink_uri_set_uri;
}
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