blob: ce17a0e4d3ae8a371492f2247d2c5181b1c2531e [file] [log] [blame]
/* GStreamer
* Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
* <2006> Lutz Mueller <lutz at topfrose dot de>
*
* 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-rtspsrc
*
* Makes a connection to an RTSP server and read the data.
* rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
* RealMedia/Quicktime/Microsoft extensions.
*
* RTSP supports transport over TCP or UDP in unicast or multicast mode. By
* default rtspsrc 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 #GstRTSPSrc:protocols property.
*
* rtspsrc currently understands SDP as the format of the session description.
* For each stream listed in the SDP a new rtp_stream\%d pad will be created
* with caps derived from the SDP media description. This is a caps of mime type
* "application/x-rtp" that can be connected to any available RTP depayloader
* element.
*
* rtspsrc will internally instantiate an RTP session manager element
* that will handle the RTCP messages to and from the server, jitter removal,
* packet reordering along with providing a clock for the pipeline.
* This feature is implemented using the gstrtpbin element.
*
* rtspsrc acts like a live source and will therefore only generate data in the
* PLAYING state.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 rtspsrc location=rtsp://some.server/url ! fakesink
* ]| Establish a connection to an RTSP server and send the raw RTP packets to a
* fakesink.
* </refsect2>
*/
#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 "gst/gst-i18n-plugin.h"
#include "gstrtspsrc.h"
GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
#define GST_CAT_DEFAULT (rtspsrc_debug)
static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
/* templates used internally */
static GstStaticPadTemplate anysrctemplate =
GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS_ANY);
static GstStaticPadTemplate anysinktemplate =
GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
GST_PAD_SINK,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS_ANY);
enum
{
SIGNAL_HANDLE_REQUEST,
SIGNAL_ON_SDP,
SIGNAL_SELECT_STREAM,
SIGNAL_NEW_MANAGER,
SIGNAL_REQUEST_RTCP_KEY,
SIGNAL_ACCEPT_CERTIFICATE,
SIGNAL_BEFORE_SEND,
SIGNAL_PUSH_BACKCHANNEL_BUFFER,
LAST_SIGNAL
};
enum _GstRtspSrcRtcpSyncMode
{
RTCP_SYNC_ALWAYS,
RTCP_SYNC_INITIAL,
RTCP_SYNC_RTP
};
enum _GstRtspSrcBufferMode
{
BUFFER_MODE_NONE,
BUFFER_MODE_SLAVE,
BUFFER_MODE_BUFFER,
BUFFER_MODE_AUTO,
BUFFER_MODE_SYNCED
};
#define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
static GType
gst_rtsp_src_buffer_mode_get_type (void)
{
static GType buffer_mode_type = 0;
static const GEnumValue buffer_modes[] = {
{BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
{BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
{BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
{BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
{BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks", "synced"},
{0, NULL, NULL},
};
if (!buffer_mode_type) {
buffer_mode_type =
g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
}
return buffer_mode_type;
}
enum _GstRtspSrcNtpTimeSource
{
NTP_TIME_SOURCE_NTP,
NTP_TIME_SOURCE_UNIX,
NTP_TIME_SOURCE_RUNNING_TIME,
NTP_TIME_SOURCE_CLOCK_TIME
};
#define DEBUG_RTSP(__self,msg) gst_rtspsrc_print_rtsp_message (__self, msg)
#define DEBUG_SDP(__self,msg) gst_rtspsrc_print_sdp_message (__self, msg)
#define GST_TYPE_RTSP_SRC_NTP_TIME_SOURCE (gst_rtsp_src_ntp_time_source_get_type())
static GType
gst_rtsp_src_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 ("GstRTSPSrcNtpTimeSource",
ntp_time_source_values);
}
return ntp_time_source_type;
}
enum _GstRtspBackchannel
{
BACKCHANNEL_NONE,
BACKCHANNEL_ONVIF
};
#define GST_TYPE_RTSP_BACKCHANNEL (gst_rtsp_backchannel_get_type())
static GType
gst_rtsp_backchannel_get_type (void)
{
static GType backchannel_type = 0;
static const GEnumValue backchannel_values[] = {
{BACKCHANNEL_NONE, "No backchannel", "none"},
{BACKCHANNEL_ONVIF, "ONVIF audio backchannel", "onvif"},
{0, NULL, NULL},
};
if (G_UNLIKELY (backchannel_type == 0)) {
backchannel_type =
g_enum_register_static ("GstRTSPBackchannel", backchannel_values);
}
return backchannel_type;
}
#define BACKCHANNEL_ONVIF_HDR_REQUIRE_VAL "www.onvif.org/ver20/backchannel"
#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_DROP_ON_LATENCY FALSE
#define DEFAULT_CONNECTION_SPEED 0
#define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
#define DEFAULT_DO_RTCP TRUE
#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_BUFFER_MODE BUFFER_MODE_AUTO
#define DEFAULT_PORT_RANGE NULL
#define DEFAULT_SHORT_HEADER FALSE
#define DEFAULT_PROBATION 2
#define DEFAULT_UDP_RECONNECT TRUE
#define DEFAULT_MULTICAST_IFACE NULL
#define DEFAULT_NTP_SYNC FALSE
#define DEFAULT_USE_PIPELINE_CLOCK FALSE
#define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL
#define DEFAULT_TLS_DATABASE NULL
#define DEFAULT_TLS_INTERACTION NULL
#define DEFAULT_DO_RETRANSMISSION TRUE
#define DEFAULT_NTP_TIME_SOURCE NTP_TIME_SOURCE_NTP
#define DEFAULT_USER_AGENT "GStreamer/" PACKAGE_VERSION
#define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
#define DEFAULT_RFC7273_SYNC FALSE
#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT G_GUINT64_CONSTANT(0)
#define DEFAULT_MAX_TS_OFFSET G_GINT64_CONSTANT(3000000000)
#define DEFAULT_VERSION GST_RTSP_VERSION_1_0
#define DEFAULT_BACKCHANNEL GST_RTSP_BACKCHANNEL_NONE
enum
{
PROP_0,
PROP_LOCATION,
PROP_PROTOCOLS,
PROP_DEBUG,
PROP_RETRY,
PROP_TIMEOUT,
PROP_TCP_TIMEOUT,
PROP_LATENCY,
PROP_DROP_ON_LATENCY,
PROP_CONNECTION_SPEED,
PROP_NAT_METHOD,
PROP_DO_RTCP,
PROP_DO_RTSP_KEEP_ALIVE,
PROP_PROXY,
PROP_PROXY_ID,
PROP_PROXY_PW,
PROP_RTP_BLOCKSIZE,
PROP_USER_ID,
PROP_USER_PW,
PROP_BUFFER_MODE,
PROP_PORT_RANGE,
PROP_UDP_BUFFER_SIZE,
PROP_SHORT_HEADER,
PROP_PROBATION,
PROP_UDP_RECONNECT,
PROP_MULTICAST_IFACE,
PROP_NTP_SYNC,
PROP_USE_PIPELINE_CLOCK,
PROP_SDES,
PROP_TLS_VALIDATION_FLAGS,
PROP_TLS_DATABASE,
PROP_TLS_INTERACTION,
PROP_DO_RETRANSMISSION,
PROP_NTP_TIME_SOURCE,
PROP_USER_AGENT,
PROP_MAX_RTCP_RTP_TIME_DIFF,
PROP_RFC7273_SYNC,
PROP_MAX_TS_OFFSET_ADJUSTMENT,
PROP_MAX_TS_OFFSET,
PROP_DEFAULT_VERSION,
PROP_BACKCHANNEL,
};
#define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
static GType
gst_rtsp_nat_method_get_type (void)
{
static GType rtsp_nat_method_type = 0;
static const GEnumValue rtsp_nat_method[] = {
{GST_RTSP_NAT_NONE, "None", "none"},
{GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
{0, NULL, NULL},
};
if (!rtsp_nat_method_type) {
rtsp_nat_method_type =
g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
}
return rtsp_nat_method_type;
}
#define RTSP_SRC_RESPONSE_ERROR(src, response_msg, err_cat, err_code, error_message) \
do { \
GST_ELEMENT_ERROR_WITH_DETAILS((src), err_cat, err_code, ("%s", error_message), \
("%s (%d)", (response_msg)->type_data.response.reason, (response_msg)->type_data.response.code), \
("rtsp-status-code", G_TYPE_UINT, (response_msg)->type_data.response.code, \
"rtsp-status-reason", G_TYPE_STRING, GST_STR_NULL((response_msg)->type_data.response.reason), NULL)); \
} while (0)
static void gst_rtspsrc_finalize (GObject * object);
static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstClock *gst_rtspsrc_provide_clock (GstElement * element);
static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
gpointer iface_data);
static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
GstRTSPMessage * response);
static gboolean gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
gint mask);
static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
gboolean async, const gchar * seek_style);
static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
gboolean only_close);
static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
const gchar * uri, GError ** error);
static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
GstRTSPStream * stream, GstEvent * event);
static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
static void gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush);
static GstRTSPResult gst_rtsp_conninfo_close (GstRTSPSrc * src,
GstRTSPConnInfo * info, gboolean free);
static void
gst_rtspsrc_print_rtsp_message (GstRTSPSrc * src, const GstRTSPMessage * msg);
static void
gst_rtspsrc_print_sdp_message (GstRTSPSrc * src, const GstSDPMessage * msg);
static GstFlowReturn gst_rtspsrc_push_backchannel_buffer (GstRTSPSrc * src,
guint id, GstSample * sample);
typedef struct
{
guint8 pt;
GstCaps *caps;
} PtMapItem;
/* commands we send to out loop to notify it of events */
#define CMD_OPEN (1 << 0)
#define CMD_PLAY (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_rtspsrc_signals[LAST_SIGNAL] = { 0 };
#define gst_rtspsrc_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
#ifndef GST_DISABLE_GST_DEBUG
static inline const char *
cmd_to_string (guint cmd)
{
switch (cmd) {
case CMD_OPEN:
return "OPEN";
case CMD_PLAY:
return "PLAY";
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 gboolean
default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
{
GST_DEBUG_OBJECT (src, "default handler");
return TRUE;
}
static gboolean
select_stream_accum (GSignalInvocationHint * ihint,
GValue * return_accu, const GValue * handler_return, gpointer data)
{
gboolean myboolean;
myboolean = g_value_get_boolean (handler_return);
GST_DEBUG ("accum %d", myboolean);
g_value_set_boolean (return_accu, myboolean);
/* stop emission if FALSE */
return myboolean;
}
static gboolean
default_before_send (GstRTSPSrc * src, GstRTSPMessage * msg)
{
GST_DEBUG_OBJECT (src, "default handler");
return TRUE;
}
static gboolean
before_send_accum (GSignalInvocationHint * ihint,
GValue * return_accu, const GValue * handler_return, gpointer data)
{
gboolean myboolean;
myboolean = g_value_get_boolean (handler_return);
g_value_set_boolean (return_accu, myboolean);
/* prevent send if FALSE */
return myboolean;
}
static void
gst_rtspsrc_class_init (GstRTSPSrcClass * 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 (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
gobject_class->set_property = gst_rtspsrc_set_property;
gobject_class->get_property = gst_rtspsrc_get_property;
gobject_class->finalize = gst_rtspsrc_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_DEBUG,
g_param_spec_boolean ("debug", "Debug",
"Dump request and response messages to stdout"
"(DEPRECATED: Printed all RTSP message to gstreamer log as 'log' level)",
DEFAULT_DEBUG,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
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_DROP_ON_LATENCY,
g_param_spec_boolean ("drop-on-latency",
"Drop buffers when maximum latency is reached",
"Tells the jitterbuffer to never exceed the given latency in size",
DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
g_param_spec_uint64 ("connection-speed", "Connection Speed",
"Network connection speed in kbps (0 = unknown)",
0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
g_param_spec_enum ("nat-method", "NAT Method",
"Method to use for traversing firewalls and NAT",
GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc:do-rtcp:
*
* Enable RTCP support. Some old server don't like RTCP and then this property
* needs to be set to FALSE.
*/
g_object_class_install_property (gobject_class, PROP_DO_RTCP,
g_param_spec_boolean ("do-rtcp", "Do RTCP",
"Send RTCP packets, disable for old incompatible server.",
DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc: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));
/**
* GstRTSPSrc: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));
/**
* GstRTSPSrc: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.
*
* Since: 1.2
*/
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));
/**
* GstRTSPSrc: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.
*
* Since: 1.2
*/
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));
/**
* GstRTSPSrc: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));
/**
* GstRTSPSrc:buffer-mode:
*
* Control the buffering and timestamping mode used by the jitterbuffer.
*/
g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
g_param_spec_enum ("buffer-mode", "Buffer Mode",
"Control the buffering algorithm in use",
GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc:port-range:
*
* Configure the client port numbers that can be used to recieve RTP and
* 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 RTP and RTCP data, "
"eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc: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));
/**
* GstRTSPSrc:short-header:
*
* Only send the basic RTSP headers for broken encoders.
*/
g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
g_param_spec_boolean ("short-header", "Short Header",
"Only send the basic RTSP headers for broken encoders",
DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PROBATION,
g_param_spec_uint ("probation", "Number of probations",
"Consecutive packet sequence numbers to accept the source",
0, G_MAXUINT, DEFAULT_PROBATION,
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_NTP_SYNC,
g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
"Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
"Use the pipeline running-time to set the NTP time in the RTCP SR messages"
"(DEPRECATED: Use ntp-time-source property)",
DEFAULT_USE_PIPELINE_CLOCK,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
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));
/**
* GstRTSPSrc::tls-validation-flags:
*
* TLS certificate validation flags used to validate server
* certificate.
*
* Since: 1.2.1
*/
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));
/**
* GstRTSPSrc::tls-database:
*
* TLS database with anchor certificate authorities used to validate
* the server certificate.
*
* Since: 1.4
*/
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));
/**
* GstRTSPSrc::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.
*
* Since: 1.6
*/
g_object_class_install_property (gobject_class, PROP_TLS_INTERACTION,
g_param_spec_object ("tls-interaction", "TLS interaction",
"A GTlsInteraction object to promt the user for password or certificate",
G_TYPE_TLS_INTERACTION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc::do-retransmission:
*
* Attempt to ask the server to retransmit lost packets according to RFC4588.
*
* Note: currently only works with SSRC-multiplexed retransmission streams
*
* Since: 1.6
*/
g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
g_param_spec_boolean ("do-retransmission", "Retransmission",
"Ask the server to retransmit lost packets",
DEFAULT_DO_RETRANSMISSION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc::ntp-time-source:
*
* allows to select the time source that should be used
* for the NTP time in RTCP packets
*
* Since: 1.6
*/
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_SRC_NTP_TIME_SOURCE, DEFAULT_NTP_TIME_SOURCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc::user-agent:
*
* The string to set in the User-Agent header.
*
* Since: 1.6
*/
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));
g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
"Maximum amount of time in ms that the RTP time in RTCP SRs "
"is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
"Synchronize received streams to the RFC7273 clock "
"(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc:default-rtsp-version:
*
* The preferred RTSP version to use while negotiating the version with the server.
*
* Since: 1.14
*/
g_object_class_install_property (gobject_class, PROP_DEFAULT_VERSION,
g_param_spec_enum ("default-rtsp-version",
"The RTSP version to try first",
"The RTSP version that should be tried first when negotiating version.",
GST_TYPE_RTSP_VERSION, DEFAULT_VERSION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc:max-ts-offset-adjustment:
*
* Syncing time stamps to NTP time adds a time offset. This parameter
* specifies the maximum number of nanoseconds per frame that this time offset
* may be adjusted with. This is used to avoid sudden large changes to time
* stamps.
*/
g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
g_param_spec_uint64 ("max-ts-offset-adjustment",
"Max Timestamp Offset Adjustment",
"The maximum number of nanoseconds per frame that time stamp offsets "
"may be adjusted (0 = no limit).", 0, G_MAXUINT64,
DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
/**
* GstRtpBin:max-ts-offset:
*
* Used to set an upper limit of how large a time offset may be. This
* is used to protect against unrealistic values as a result of either
* client,server or clock issues.
*/
g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET,
g_param_spec_int64 ("max-ts-offset", "Max TS Offset",
"The maximum absolute value of the time offset in (nanoseconds). "
"Note, if the ntp-sync parameter is set the default value is "
"changed to 0 (no limit)", 0, G_MAXINT64, DEFAULT_MAX_TS_OFFSET,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSrc:backchannel
*
* Select a type of backchannel to setup with the RTSP server.
* Default value is "none". Allowed values are "none" and "onvif".
*
* Since: 1.14
*/
g_object_class_install_property (gobject_class, PROP_BACKCHANNEL,
g_param_spec_enum ("backchannel", "Backchannel type",
"The type of backchannel to setup. Default is 'none'.",
GST_TYPE_RTSP_BACKCHANNEL, BACKCHANNEL_NONE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc::handle-request:
* @rtspsrc: a #GstRTSPSrc
* @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 @rtspsrc 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.
*
* Since: 1.2
*/
gst_rtspsrc_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);
/**
* GstRTSPSrc::on-sdp:
* @rtspsrc: a #GstRTSPSrc
* @sdp: a #GstSDPMessage
*
* Emited when the client has retrieved the SDP and before it configures the
* streams in the SDP. @sdp can be inspected and modified.
*
* This signal is called from the streaming thread, you should therefore not
* do any state changes on @rtspsrc 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.
*
* Since: 1.2
*/
gst_rtspsrc_signals[SIGNAL_ON_SDP] =
g_signal_new ("on-sdp", G_TYPE_FROM_CLASS (klass), 0,
0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
/**
* GstRTSPSrc::select-stream:
* @rtspsrc: a #GstRTSPSrc
* @num: the stream number
* @caps: the stream caps
*
* Emited before the client decides to configure the stream @num with
* @caps.
*
* Returns: %TRUE when the stream should be selected, %FALSE when the stream
* is to be ignored.
*
* Since: 1.2
*/
gst_rtspsrc_signals[SIGNAL_SELECT_STREAM] =
g_signal_new_class_handler ("select-stream", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
(GCallback) default_select_stream, select_stream_accum, NULL,
g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2, G_TYPE_UINT,
GST_TYPE_CAPS);
/**
* GstRTSPSrc::new-manager:
* @rtspsrc: a #GstRTSPSrc
* @manager: a #GstElement
*
* Emited after a new manager (like rtpbin) was created and the default
* properties were configured.
*
* Since: 1.4
*/
gst_rtspsrc_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);
/**
* GstRTSPSrc::request-rtcp-key:
* @rtspsrc: a #GstRTSPSrc
* @num: the stream number
*
* Signal emited 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.
*
* Since: 1.4
*/
gst_rtspsrc_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);
/**
* GstRTSPSrc::accept-certificate:
* @rtspsrc: a #GstRTSPSrc
* @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_rtspsrc_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);
/*
* GstRTSPSrc::before-send
* @rtspsrc: a #GstRTSPSrc
* @num: the stream number
*
* Emitted before each RTSP request is sent, in order to allow
* the application to modify send parameters or to skip the message entirely.
* This can be used, for example, to work with ONVIF Profile G servers,
* which need a different/additional range, rate-control, and intra/x
* parameters.
*
* Returns: %TRUE when the command should be sent, %FALSE when the
* command should be dropped.
*
* Since: 1.14
*/
gst_rtspsrc_signals[SIGNAL_BEFORE_SEND] =
g_signal_new_class_handler ("before-send", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
(GCallback) default_before_send, before_send_accum, NULL,
g_cclosure_marshal_generic, G_TYPE_BOOLEAN,
1, GST_TYPE_RTSP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
/**
* GstRTSPSrc::push-backchannel-buffer:
* @rtspsrc: a #GstRTSPSrc
* @buffer: RTP buffer to send back
*
*
*/
gst_rtspsrc_signals[SIGNAL_PUSH_BACKCHANNEL_BUFFER] =
g_signal_new ("push-backchannel-buffer", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRTSPSrcClass,
push_backchannel_buffer), NULL, NULL, NULL, GST_TYPE_FLOW_RETURN, 2,
G_TYPE_UINT, GST_TYPE_BUFFER);
gstelement_class->send_event = gst_rtspsrc_send_event;
gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
gstelement_class->change_state = gst_rtspsrc_change_state;
gst_element_class_add_static_pad_template (gstelement_class, &rtptemplate);
gst_element_class_set_static_metadata (gstelement_class,
"RTSP packet receiver", "Source/Network",
"Receive data over the network via RTSP (RFC 2326)",
"Wim Taymans <wim@fluendo.com>, "
"Thijs Vermeir <thijs.vermeir@barco.com>, "
"Lutz Mueller <lutz@topfrose.de>");
gstbin_class->handle_message = gst_rtspsrc_handle_message;
klass->push_backchannel_buffer = gst_rtspsrc_push_backchannel_buffer;
gst_rtsp_ext_list_init ();
}
static void
gst_rtspsrc_init (GstRTSPSrc * src)
{
src->conninfo.location = g_strdup (DEFAULT_LOCATION);
src->protocols = DEFAULT_PROTOCOLS;
src->debug = DEFAULT_DEBUG;
src->retry = DEFAULT_RETRY;
src->udp_timeout = DEFAULT_TIMEOUT;
gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
src->latency = DEFAULT_LATENCY_MS;
src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
src->connection_speed = DEFAULT_CONNECTION_SPEED;
src->nat_method = DEFAULT_NAT_METHOD;
src->do_rtcp = DEFAULT_DO_RTCP;
src->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
src->user_id = g_strdup (DEFAULT_USER_ID);
src->user_pw = g_strdup (DEFAULT_USER_PW);
src->buffer_mode = DEFAULT_BUFFER_MODE;
src->client_port_range.min = 0;
src->client_port_range.max = 0;
src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
src->short_header = DEFAULT_SHORT_HEADER;
src->probation = DEFAULT_PROBATION;
src->udp_reconnect = DEFAULT_UDP_RECONNECT;
src->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
src->ntp_sync = DEFAULT_NTP_SYNC;
src->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
src->sdes = NULL;
src->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
src->tls_database = DEFAULT_TLS_DATABASE;
src->tls_interaction = DEFAULT_TLS_INTERACTION;
src->do_retransmission = DEFAULT_DO_RETRANSMISSION;
src->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
src->user_agent = g_strdup (DEFAULT_USER_AGENT);
src->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
src->rfc7273_sync = DEFAULT_RFC7273_SYNC;
src->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
src->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
src->max_ts_offset_is_set = FALSE;
src->default_version = DEFAULT_VERSION;
src->version = GST_RTSP_VERSION_INVALID;
/* get a list of all extensions */
src->extensions = gst_rtsp_ext_list_get ();
/* connect to send signal */
gst_rtsp_ext_list_connect (src->extensions, "send",
(GCallback) gst_rtspsrc_send_cb, src);
/* protects the streaming thread in interleaved mode or the polling
* thread in UDP mode. */
g_rec_mutex_init (&src->stream_rec_lock);
/* protects our state changes from multiple invocations */
g_rec_mutex_init (&src->state_rec_lock);
src->state = GST_RTSP_STATE_INVALID;
g_mutex_init (&src->conninfo.send_lock);
g_mutex_init (&src->conninfo.recv_lock);
GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
gst_bin_set_suppressed_flags (GST_BIN (src),
GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
}
static void
gst_rtspsrc_finalize (GObject * object)
{
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (object);
gst_rtsp_ext_list_free (rtspsrc->extensions);
g_free (rtspsrc->conninfo.location);
gst_rtsp_url_free (rtspsrc->conninfo.url);
g_free (rtspsrc->conninfo.url_str);
g_free (rtspsrc->user_id);
g_free (rtspsrc->user_pw);
g_free (rtspsrc->multi_iface);
g_free (rtspsrc->user_agent);
if (rtspsrc->sdp) {
gst_sdp_message_free (rtspsrc->sdp);
rtspsrc->sdp = NULL;
}
if (rtspsrc->provided_clock)
gst_object_unref (rtspsrc->provided_clock);
if (rtspsrc->sdes)
gst_structure_free (rtspsrc->sdes);
if (rtspsrc->tls_database)
g_object_unref (rtspsrc->tls_database);
if (rtspsrc->tls_interaction)
g_object_unref (rtspsrc->tls_interaction);
/* free locks */
g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
g_rec_mutex_clear (&rtspsrc->state_rec_lock);
g_mutex_clear (&rtspsrc->conninfo.send_lock);
g_mutex_clear (&rtspsrc->conninfo.recv_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstClock *
gst_rtspsrc_provide_clock (GstElement * element)
{
GstRTSPSrc *src = GST_RTSPSRC (element);
GstClock *clock;
if ((clock = src->provided_clock) != NULL)
return gst_object_ref (clock);
return GST_ELEMENT_CLASS (parent_class)->provide_clock (element);
}
/* a proxy string of the format [user:passwd@]host[:port] */
static gboolean
gst_rtspsrc_set_proxy (GstRTSPSrc * 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_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
{
rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
if (timeout != 0)
rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
else
rtspsrc->ptcp_timeout = NULL;
}
static void
gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (object);
switch (prop_id) {
case PROP_LOCATION:
gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
g_value_get_string (value), NULL);
break;
case PROP_PROTOCOLS:
rtspsrc->protocols = g_value_get_flags (value);
break;
case PROP_DEBUG:
rtspsrc->debug = g_value_get_boolean (value);
break;
case PROP_RETRY:
rtspsrc->retry = g_value_get_uint (value);
break;
case PROP_TIMEOUT:
rtspsrc->udp_timeout = g_value_get_uint64 (value);
break;
case PROP_TCP_TIMEOUT:
gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
break;
case PROP_LATENCY:
rtspsrc->latency = g_value_get_uint (value);
break;
case PROP_DROP_ON_LATENCY:
rtspsrc->drop_on_latency = g_value_get_boolean (value);
break;
case PROP_CONNECTION_SPEED:
rtspsrc->connection_speed = g_value_get_uint64 (value);
break;
case PROP_NAT_METHOD:
rtspsrc->nat_method = g_value_get_enum (value);
break;
case PROP_DO_RTCP:
rtspsrc->do_rtcp = g_value_get_boolean (value);
break;
case PROP_DO_RTSP_KEEP_ALIVE:
rtspsrc->do_rtsp_keep_alive = g_value_get_boolean (value);
break;
case PROP_PROXY:
gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
break;
case PROP_PROXY_ID:
g_free (rtspsrc->prop_proxy_id);
rtspsrc->prop_proxy_id = g_value_dup_string (value);
break;
case PROP_PROXY_PW:
g_free (rtspsrc->prop_proxy_pw);
rtspsrc->prop_proxy_pw = g_value_dup_string (value);
break;
case PROP_RTP_BLOCKSIZE:
rtspsrc->rtp_blocksize = g_value_get_uint (value);
break;
case PROP_USER_ID:
g_free (rtspsrc->user_id);
rtspsrc->user_id = g_value_dup_string (value);
break;
case PROP_USER_PW:
g_free (rtspsrc->user_pw);
rtspsrc->user_pw = g_value_dup_string (value);
break;
case PROP_BUFFER_MODE:
rtspsrc->buffer_mode = g_value_get_enum (value);
break;
case PROP_PORT_RANGE:
{
const gchar *str;
str = g_value_get_string (value);
if (sscanf (str, "%u-%u", &rtspsrc->client_port_range.min,
&rtspsrc->client_port_range.max) != 2) {
rtspsrc->client_port_range.min = 0;
rtspsrc->client_port_range.max = 0;
}
break;
}
case PROP_UDP_BUFFER_SIZE:
rtspsrc->udp_buffer_size = g_value_get_int (value);
break;
case PROP_SHORT_HEADER:
rtspsrc->short_header = g_value_get_boolean (value);
break;
case PROP_PROBATION:
rtspsrc->probation = g_value_get_uint (value);
break;
case PROP_UDP_RECONNECT:
rtspsrc->udp_reconnect = g_value_get_boolean (value);
break;
case PROP_MULTICAST_IFACE:
g_free (rtspsrc->multi_iface);
if (g_value_get_string (value) == NULL)
rtspsrc->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
else
rtspsrc->multi_iface = g_value_dup_string (value);
break;
case PROP_NTP_SYNC:
rtspsrc->ntp_sync = g_value_get_boolean (value);
/* The default value of max_ts_offset depends on ntp_sync. If user
* hasn't set it then change default value */
if (!rtspsrc->max_ts_offset_is_set) {
if (rtspsrc->ntp_sync) {
rtspsrc->max_ts_offset = 0;
} else {
rtspsrc->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
}
}
break;
case PROP_USE_PIPELINE_CLOCK:
rtspsrc->use_pipeline_clock = g_value_get_boolean (value);
break;
case PROP_SDES:
rtspsrc->sdes = g_value_dup_boxed (value);
break;
case PROP_TLS_VALIDATION_FLAGS:
rtspsrc->tls_validation_flags = g_value_get_flags (value);
break;
case PROP_TLS_DATABASE:
g_clear_object (&rtspsrc->tls_database);
rtspsrc->tls_database = g_value_dup_object (value);
break;
case PROP_TLS_INTERACTION:
g_clear_object (&rtspsrc->tls_interaction);
rtspsrc->tls_interaction = g_value_dup_object (value);
break;
case PROP_DO_RETRANSMISSION:
rtspsrc->do_retransmission = g_value_get_boolean (value);
break;
case PROP_NTP_TIME_SOURCE:
rtspsrc->ntp_time_source = g_value_get_enum (value);
break;
case PROP_USER_AGENT:
g_free (rtspsrc->user_agent);
rtspsrc->user_agent = g_value_dup_string (value);
break;
case PROP_MAX_RTCP_RTP_TIME_DIFF:
rtspsrc->max_rtcp_rtp_time_diff = g_value_get_int (value);
break;
case PROP_RFC7273_SYNC:
rtspsrc->rfc7273_sync = g_value_get_boolean (value);
break;
case PROP_MAX_TS_OFFSET_ADJUSTMENT:
rtspsrc->max_ts_offset_adjustment = g_value_get_uint64 (value);
break;
case PROP_MAX_TS_OFFSET:
rtspsrc->max_ts_offset = g_value_get_int64 (value);
rtspsrc->max_ts_offset_is_set = TRUE;
break;
case PROP_DEFAULT_VERSION:
rtspsrc->default_version = g_value_get_enum (value);
break;
case PROP_BACKCHANNEL:
rtspsrc->backchannel = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (object);
switch (prop_id) {
case PROP_LOCATION:
g_value_set_string (value, rtspsrc->conninfo.location);
break;
case PROP_PROTOCOLS:
g_value_set_flags (value, rtspsrc->protocols);
break;
case PROP_DEBUG:
g_value_set_boolean (value, rtspsrc->debug);
break;
case PROP_RETRY:
g_value_set_uint (value, rtspsrc->retry);
break;
case PROP_TIMEOUT:
g_value_set_uint64 (value, rtspsrc->udp_timeout);
break;
case PROP_TCP_TIMEOUT:
{
guint64 timeout;
timeout = ((guint64) rtspsrc->tcp_timeout.tv_sec) * G_USEC_PER_SEC +
rtspsrc->tcp_timeout.tv_usec;
g_value_set_uint64 (value, timeout);
break;
}
case PROP_LATENCY:
g_value_set_uint (value, rtspsrc->latency);
break;
case PROP_DROP_ON_LATENCY:
g_value_set_boolean (value, rtspsrc->drop_on_latency);
break;
case PROP_CONNECTION_SPEED:
g_value_set_uint64 (value, rtspsrc->connection_speed);
break;
case PROP_NAT_METHOD:
g_value_set_enum (value, rtspsrc->nat_method);
break;
case PROP_DO_RTCP:
g_value_set_boolean (value, rtspsrc->do_rtcp);
break;
case PROP_DO_RTSP_KEEP_ALIVE:
g_value_set_boolean (value, rtspsrc->do_rtsp_keep_alive);
break;
case PROP_PROXY:
{
gchar *str;
if (rtspsrc->proxy_host) {
str =
g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
} else {
str = NULL;
}
g_value_take_string (value, str);
break;
}
case PROP_PROXY_ID:
g_value_set_string (value, rtspsrc->prop_proxy_id);
break;
case PROP_PROXY_PW:
g_value_set_string (value, rtspsrc->prop_proxy_pw);
break;
case PROP_RTP_BLOCKSIZE:
g_value_set_uint (value, rtspsrc->rtp_blocksize);
break;
case PROP_USER_ID:
g_value_set_string (value, rtspsrc->user_id);
break;
case PROP_USER_PW:
g_value_set_string (value, rtspsrc->user_pw);
break;
case PROP_BUFFER_MODE:
g_value_set_enum (value, rtspsrc->buffer_mode);
break;
case PROP_PORT_RANGE:
{
gchar *str;
if (rtspsrc->client_port_range.min != 0) {
str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
rtspsrc->client_port_range.max);
} else {
str = NULL;
}
g_value_take_string (value, str);
break;
}
case PROP_UDP_BUFFER_SIZE:
g_value_set_int (value, rtspsrc->udp_buffer_size);
break;
case PROP_SHORT_HEADER:
g_value_set_boolean (value, rtspsrc->short_header);
break;
case PROP_PROBATION:
g_value_set_uint (value, rtspsrc->probation);
break;
case PROP_UDP_RECONNECT:
g_value_set_boolean (value, rtspsrc->udp_reconnect);
break;
case PROP_MULTICAST_IFACE:
g_value_set_string (value, rtspsrc->multi_iface);
break;
case PROP_NTP_SYNC:
g_value_set_boolean (value, rtspsrc->ntp_sync);
break;
case PROP_USE_PIPELINE_CLOCK:
g_value_set_boolean (value, rtspsrc->use_pipeline_clock);
break;
case PROP_SDES:
g_value_set_boxed (value, rtspsrc->sdes);
break;
case PROP_TLS_VALIDATION_FLAGS:
g_value_set_flags (value, rtspsrc->tls_validation_flags);
break;
case PROP_TLS_DATABASE:
g_value_set_object (value, rtspsrc->tls_database);
break;
case PROP_TLS_INTERACTION:
g_value_set_object (value, rtspsrc->tls_interaction);
break;
case PROP_DO_RETRANSMISSION:
g_value_set_boolean (value, rtspsrc->do_retransmission);
break;
case PROP_NTP_TIME_SOURCE:
g_value_set_enum (value, rtspsrc->ntp_time_source);
break;
case PROP_USER_AGENT:
g_value_set_string (value, rtspsrc->user_agent);
break;
case PROP_MAX_RTCP_RTP_TIME_DIFF:
g_value_set_int (value, rtspsrc->max_rtcp_rtp_time_diff);
break;
case PROP_RFC7273_SYNC:
g_value_set_boolean (value, rtspsrc->rfc7273_sync);
break;
case PROP_MAX_TS_OFFSET_ADJUSTMENT:
g_value_set_uint64 (value, rtspsrc->max_ts_offset_adjustment);
break;
case PROP_MAX_TS_OFFSET:
g_value_set_int64 (value, rtspsrc->max_ts_offset);
break;
case PROP_DEFAULT_VERSION:
g_value_set_enum (value, rtspsrc->default_version);
break;
case PROP_BACKCHANNEL:
g_value_set_enum (value, rtspsrc->backchannel);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gint
find_stream_by_id (GstRTSPStream * stream, gint * id)
{
if (stream->id == *id)
return 0;
return -1;
}
static gint
find_stream_by_channel (GstRTSPStream * stream, gint * channel)
{
/* ignore unconfigured channels here (e.g., those that
* were explicitly skipped during SETUP) */
if ((stream->channelpad[0] != NULL) &&
(stream->channel[0] == *channel || stream->channel[1] == *channel))
return 0;
return -1;
}
static gint
find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
{
GstElement *src = (GstElement *) a;
if (stream->udpsrc[0] == src)
return 0;
if (stream->udpsrc[1] == src)
return 0;
return -1;
}
static gint
find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
{
if (stream->conninfo.location) {
/* check qualified setup_url */
if (!strcmp (stream->conninfo.location, (gchar *) a))
return 0;
}
if (stream->control_url) {
/* check original control_url */
if (!strcmp (stream->control_url, (gchar *) a))
return 0;
/* check if qualified setup_url ends with string */
if (g_str_has_suffix (stream->control_url, (gchar *) a))
return 0;
}
return -1;
}
static GstRTSPStream *
find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
{
GList *lstream;
/* find and get stream */
if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
return (GstRTSPStream *) lstream->data;
return NULL;
}
static const GstSDPBandwidth *
gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
const GstSDPMedia * media, const gchar * type)
{
guint i, len;
/* first look in the media specific section */
len = gst_sdp_media_bandwidths_len (media);
for (i = 0; i < len; i++) {
const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
if (strcmp (bw->bwtype, type) == 0)
return bw;
}
/* then look in the message specific section */
len = gst_sdp_message_bandwidths_len (sdp);
for (i = 0; i < len; i++) {
const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
if (strcmp (bw->bwtype, type) == 0)
return bw;
}
return NULL;
}
static void
gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
const GstSDPMedia * media, GstRTSPStream * stream)
{
const GstSDPBandwidth *bw;
if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
stream->as_bandwidth = bw->bandwidth;
else
stream->as_bandwidth = -1;
if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
stream->rr_bandwidth = bw->bandwidth;
else
stream->rr_bandwidth = -1;
if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
stream->rs_bandwidth = bw->bandwidth;
else
stream->rs_bandwidth = -1;
}
static void
gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
const GstSDPConnection * conn)
{
if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
return;
if (conn->addrtype == NULL)
return;
/* check for IPV6 */
if (strcmp (conn->addrtype, "IP4") == 0)
stream->is_ipv6 = FALSE;
else if (strcmp (conn->addrtype, "IP6") == 0)
stream->is_ipv6 = TRUE;
else
return;
/* save address */
g_free (stream->destination);
stream->destination = g_strdup (conn->address);
/* check for multicast */
stream->is_multicast =
gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
conn->address);
stream->ttl = conn->ttl;
}
/* Go over the connections for a stream.
* - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
* receiving.
* - If we are dealing with a localhost address, we disable multicast
*/
static void
gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
const GstSDPMedia * media, GstRTSPStream * stream)
{
const GstSDPConnection *conn;
guint i, len;
/* first look in the media specific section */
len = gst_sdp_media_connections_len (media);
for (i = 0; i < len; i++) {
conn = gst_sdp_media_get_connection (media, i);
gst_rtspsrc_do_stream_connection (src, stream, conn);
}
/* then look in the message specific section */
if ((conn = gst_sdp_message_get_connection (sdp))) {
gst_rtspsrc_do_stream_connection (src, stream, conn);
}
}
static gchar *
make_stream_id (GstRTSPStream * stream, const GstSDPMedia * media)
{
gchar *stream_id =
g_strdup_printf ("%s:%d:%d:%s:%d", media->media, media->port,
media->num_ports, media->proto, stream->default_pt);
g_strcanon (stream_id, G_CSET_a_2_z G_CSET_A_2_Z G_CSET_DIGITS, ':');
return stream_id;
}
/* m=<media> <UDP port> RTP/AVP <payload>
*/
static void
gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
const GstSDPMedia * media, GstRTSPStream * stream)
{
guint i, len;
const gchar *proto;
GstCaps *global_caps;
/* get proto */
proto = gst_sdp_media_get_proto (media);
if (proto == NULL)
goto no_proto;
if (g_str_equal (proto, "RTP/AVP"))
stream->profile = GST_RTSP_PROFILE_AVP;
else if (g_str_equal (proto, "RTP/SAVP"))
stream->profile = GST_RTSP_PROFILE_SAVP;
else if (g_str_equal (proto, "RTP/AVPF"))
stream->profile = GST_RTSP_PROFILE_AVPF;
else if (g_str_equal (proto, "RTP/SAVPF"))
stream->profile = GST_RTSP_PROFILE_SAVPF;
else
goto unknown_proto;
if (gst_sdp_media_get_attribute_val (media, "recvonly") != NULL &&
/* We want to setup caps for streams configured as backchannel */
!stream->is_backchannel && src->backchannel != BACKCHANNEL_NONE)
goto recvonly_media;
/* Parse global SDP attributes once */
global_caps = gst_caps_new_empty_simple ("application/x-unknown");
GST_DEBUG ("mapping sdp session level attributes to caps");
gst_sdp_message_attributes_to_caps (sdp, global_caps);
GST_DEBUG ("mapping sdp media level attributes to caps");
gst_sdp_media_attributes_to_caps (media, global_caps);
/* Keep a copy of the SDP key management */
gst_sdp_media_parse_keymgmt (media, &stream->mikey);
if (stream->mikey == NULL)
gst_sdp_message_parse_keymgmt (sdp, &stream->mikey);
len = gst_sdp_media_formats_len (media);
for (i = 0; i < len; i++) {
gint pt;
GstCaps *caps, *outcaps;
GstStructure *s;
const gchar *enc;
PtMapItem item;
pt = atoi (gst_sdp_media_get_format (media, i));
GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);
/* convert caps */
caps = gst_sdp_media_get_caps_from_media (media, pt);
if (caps == NULL) {
GST_WARNING_OBJECT (src, " skipping pt %d without caps", pt);
continue;
}
/* do some tweaks */
s = gst_caps_get_structure (caps, 0);
if ((enc = gst_structure_get_string (s, "encoding-name"))) {
stream->is_real = (strstr (enc, "-REAL") != NULL);
if (strcmp (enc, "X-ASF-PF") == 0)
stream->container = TRUE;
}
/* Merge in global caps */
/* Intersect will merge in missing fields to the current caps */
outcaps = gst_caps_intersect (caps, global_caps);
gst_caps_unref (caps);
/* the first pt will be the default */
if (stream->ptmap->len == 0)
stream->default_pt = pt;
item.pt = pt;
item.caps = outcaps;
g_array_append_val (stream->ptmap, item);
}
stream->stream_id = make_stream_id (stream, media);
gst_caps_unref (global_caps);
return;
no_proto:
{
GST_ERROR_OBJECT (src, "can't find proto in media");
return;
}
unknown_proto:
{
GST_ERROR_OBJECT (src, "unknown proto in media: '%s'", proto);
return;
}
recvonly_media:
{
GST_WARNING_OBJECT (src, "recvonly media ignored, no backchannel");
return;
}
}
static const gchar *
get_aggregate_control (GstRTSPSrc * src)
{
const gchar *base;
if (src->control)
base = src->control;
else if (src->content_base)
base = src->content_base;
else if (src->conninfo.url_str)
base = src->conninfo.url_str;
else
base = "/";
return base;
}
static void
clear_ptmap_item (PtMapItem * item)
{
if (item->caps)
gst_caps_unref (item->caps);
}
static GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx,
gint n_streams)
{
GstRTSPStream *stream;
const gchar *control_url;
const GstSDPMedia *media;
/* get media, should not return NULL */
media = gst_sdp_message_get_media (sdp, idx);
if (media == NULL)
return NULL;
stream = g_new0 (GstRTSPStream, 1);
stream->parent = src;
/* we mark the pad as not linked, we will mark it as OK when we add the pad to
* the element. */
stream->last_ret = GST_FLOW_NOT_LINKED;
stream->added = FALSE;
stream->setup = FALSE;
stream->skipped = FALSE;
stream->id = idx;
stream->eos = FALSE;
stream->discont = TRUE;
stream->seqbase = -1;
stream->timebase = -1;
stream->send_ssrc = g_random_int ();
stream->profile = GST_RTSP_PROFILE_AVP;
stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
stream->mikey = NULL;
stream->stream_id = NULL;
stream->is_backchannel = FALSE;
g_mutex_init (&stream->conninfo.send_lock);
g_mutex_init (&stream->conninfo.recv_lock);
g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
/* stream is recvonly and onvif backchannel is requested */
if (gst_sdp_media_get_attribute_val (media, "recvonly") != NULL &&
src->backchannel != BACKCHANNEL_NONE)
stream->is_backchannel = TRUE;
/* collect bandwidth information for this steam. FIXME, configure in the RTP
* session manager to scale RTCP. */
gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
/* collect connection info */
gst_rtspsrc_collect_connections (src, sdp, media, stream);
/* make the payload type map */
gst_rtspsrc_collect_payloads (src, sdp, media, stream);
/* collect port number */
stream->port = gst_sdp_media_get_port (media);
/* get control url to construct the setup url. The setup url is used to
* configure the transport of the stream and is used to identity the stream in
* the RTP-Info header field returned from PLAY. */
control_url = gst_sdp_media_get_attribute_val (media, "control");
if (control_url == NULL)
control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
GST_DEBUG_OBJECT (src, " port: %d", stream->port);
GST_DEBUG_OBJECT (src, " container: %d", stream->container);
GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
/* RFC 2326, C.3: missing control_url permitted in case of a single stream */
if (control_url == NULL && n_streams == 1) {
control_url = "";
}
if (control_url != NULL) {
stream->control_url = g_strdup (control_url);
/* Build a fully qualified url using the content_base if any or by prefixing
* the original request.
* If the control_url starts with a '/' or a non rtsp: protocol we will most
* likely build a URL that the server will fail to understand, this is ok,
* we will fail then. */
if (g_str_has_prefix (control_url, "rtsp://"))
stream->conninfo.location = g_strdup (control_url);
else {
const gchar *base;
gboolean has_slash;
if (g_strcmp0 (control_url, "*") == 0)
control_url = "";
base = get_aggregate_control (src);
/* check if the base ends or control starts with / */
has_slash = g_str_has_prefix (control_url, "/");
has_slash = has_slash || g_str_has_suffix (base, "/");
/* concatenate the two strings, insert / when not present */
stream->conninfo.location =
g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
}
}
GST_DEBUG_OBJECT (src, " setup: %s",
GST_STR_NULL (stream->conninfo.location));
/* we keep track of all streams */
src->streams = g_list_append (src->streams, stream);
return stream;
/* ERRORS */
}
static void
gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
{
gint i;
GST_DEBUG_OBJECT (src, "free stream %p", stream);
g_array_free (stream->ptmap, TRUE);
g_free (stream->destination);
g_free (stream->control_url);
g_free (stream->conninfo.location);
g_free (stream->stream_id);
for (i = 0; i < 2; i++) {
if (stream->udpsrc[i]) {
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
gst_object_unref (stream->udpsrc[i]);
}
if (stream->channelpad[i])
gst_object_unref (stream->channelpad[i]);
if (stream->udpsink[i]) {
gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
gst_object_unref (stream->udpsink[i]);
}
}
if (stream->rtpsrc) {
gst_element_set_state (stream->rtpsrc, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), stream->rtpsrc);
gst_object_unref (stream->rtpsrc);
}
if (stream->srcpad) {
gst_pad_set_active (stream->srcpad, FALSE);
if (stream->added)
gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
}
if (stream->srtpenc)
gst_object_unref (stream->srtpenc);
if (stream->srtpdec)
gst_object_unref (stream->srtpdec);
if (stream->srtcpparams)
gst_caps_unref (stream->srtcpparams);
if (stream->mikey)
gst_mikey_message_unref (stream->mikey);
if (stream->rtcppad)
gst_object_unref (stream->rtcppad);
if (stream->session)
g_object_unref (stream->session);
if (stream->rtx_pt_map)
gst_structure_free (stream->rtx_pt_map);
g_mutex_clear (&stream->conninfo.send_lock);
g_mutex_clear (&stream->conninfo.recv_lock);
g_free (stream);
}
static void
gst_rtspsrc_cleanup (GstRTSPSrc * src)
{
GList *walk;
GST_DEBUG_OBJECT (src, "cleanup");
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
gst_rtspsrc_stream_free (src, stream);
}
g_list_free (src->streams);
src->streams = NULL;
if (src->manager) {
if (src->manager_sig_id) {
g_signal_handler_disconnect (src->manager, src->manager_sig_id);
src->manager_sig_id = 0;
}
gst_element_set_state (src->manager, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), src->manager);
src->manager = NULL;
}
if (src->props)
gst_structure_free (src->props);
src->props = NULL;
g_free (src->content_base);
src->content_base = NULL;
g_free (src->control);
src->control = NULL;
if (src->range)
gst_rtsp_range_free (src->range);
src->range = NULL;
/* don't clear the SDP when it was used in the url */
if (src->sdp && !src->from_sdp) {
gst_sdp_message_free (src->sdp);
src->sdp = NULL;
}
src->need_segment = FALSE;
if (src->provided_clock) {
gst_object_unref (src->provided_clock);
src->provided_clock = NULL;
}
}
static gboolean
gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
gint * rtpport, gint * rtcpport)
{
GstRTSPSrc *src;
GstStateChangeReturn ret;
GstElement *udpsrc0, *udpsrc1;
gint tmp_rtp, tmp_rtcp;
guint count;
const gchar *host;
src = stream->parent;
udpsrc0 = NULL;
udpsrc1 = NULL;
count = 0;
/* Start at next port */
tmp_rtp = src->next_port_num;
if (stream->is_ipv6)
host = "udp://[::0]";
else
host = "udp://0.0.0.0";
/* try to allocate 2 UDP ports, the RTP port should be an even
* number and the RTCP port should be the next (uneven) port */
again:
if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
tmp_rtp >= src->client_port_range.max)
goto no_ports;
udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
if (udpsrc0 == NULL)
goto no_udp_protocol;
g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
if (src->udp_buffer_size != 0)
g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
NULL);
ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
if (ret == GST_STATE_CHANGE_FAILURE) {
if (tmp_rtp != 0) {
GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
tmp_rtp += 2;
if (++count > src->retry)
goto no_ports;
GST_DEBUG_OBJECT (src, "free RTP udpsrc");
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
udpsrc0 = NULL;
GST_DEBUG_OBJECT (src, "retry %d", count);
goto again;
}
goto no_udp_protocol;
}
g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
/* check if port is even */
if ((tmp_rtp & 0x01) != 0) {
/* port not even, close and allocate another */
if (++count > src->retry)
goto no_ports;
GST_DEBUG_OBJECT (src, "RTP port not even");
GST_DEBUG_OBJECT (src, "free RTP udpsrc");
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
udpsrc0 = NULL;
GST_DEBUG_OBJECT (src, "retry %d", count);
tmp_rtp++;
goto again;
}
/* allocate port+1 for RTCP now */
udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
if (udpsrc1 == NULL)
goto no_udp_rtcp_protocol;
/* set port */
tmp_rtcp = tmp_rtp + 1;
if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
goto no_ports;
g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
/* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
if (ret == GST_STATE_CHANGE_FAILURE) {
GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
if (++count > src->retry)
goto no_ports;
GST_DEBUG_OBJECT (src, "free RTP udpsrc");
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
udpsrc0 = NULL;
GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
gst_element_set_state (udpsrc1, GST_STATE_NULL);
gst_object_unref (udpsrc1);
udpsrc1 = NULL;
tmp_rtp += 2;
GST_DEBUG_OBJECT (src, "retry %d", count);
goto again;
}
/* all fine, do port check */
g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
/* this should not happen... */
if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
goto port_error;
/* we keep these elements, we configure all in configure_transport when the
* server told us to really use the UDP ports. */
stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
gst_element_set_locked_state (stream->udpsrc[0], TRUE);
gst_element_set_locked_state (stream->udpsrc[1], TRUE);
/* keep track of next available port number when we have a range
* configured */
if (src->next_port_num != 0)
src->next_port_num = tmp_rtcp + 1;
return TRUE;
/* ERRORS */
no_udp_protocol:
{
GST_DEBUG_OBJECT (src, "could not get UDP source");
goto cleanup;
}
no_ports:
{
GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
count);
goto cleanup;
}
no_udp_rtcp_protocol:
{
GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
goto cleanup;
}
port_error:
{
GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
goto cleanup;
}
cleanup:
{
if (udpsrc0) {
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
}
if (udpsrc1) {
gst_element_set_state (udpsrc1, GST_STATE_NULL);
gst_object_unref (udpsrc1);
}
return FALSE;
}
}
static void
gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
{
GList *walk;
if (src->manager)
gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
gint i;
for (i = 0; i < 2; i++) {
if (stream->udpsrc[i])
gst_element_set_state (stream->udpsrc[i], state);
}
}
}
static void
gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
{
GstEvent *event;
gint cmd;
GstState state;
if (flush) {
event = gst_event_new_flush_start ();
GST_DEBUG_OBJECT (src, "start flush");
cmd = CMD_WAIT;
state = GST_STATE_PAUSED;
} else {
event = gst_event_new_flush_stop (FALSE);
GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
cmd = CMD_LOOP;
if (playing)
state = GST_STATE_PLAYING;
else
state = GST_STATE_PAUSED;
}
gst_rtspsrc_push_event (src, event);
gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
gst_rtspsrc_set_state (src, state);
}
static GstRTSPResult
gst_rtspsrc_connection_send (GstRTSPSrc * src, 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_rtspsrc_connection_receive (GstRTSPSrc * src, 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 void
gst_rtspsrc_get_position (GstRTSPSrc * src)
{
GstQuery *query;
GList *walk;
query = gst_query_new_position (GST_FORMAT_TIME);
/* should be known somewhere down the stream (e.g. jitterbuffer) */
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
GstFormat fmt;
gint64 pos;
if (stream->srcpad) {
if (gst_pad_query (stream->srcpad, query)) {
gst_query_parse_position (query, &fmt, &pos);
GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
GST_TIME_ARGS (pos));
src->last_pos = pos;
goto out;
}
}
}
src->last_pos = 0;
out:
gst_query_unref (query);
}
static gboolean
gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
{
gdouble rate;
GstFormat format;
GstSeekFlags flags;
GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
gint64 cur, stop;
gboolean flush, skip;
gboolean update;
gboolean playing;
GstSegment seeksegment = { 0, };
GList *walk;
const gchar *seek_style = NULL;
if (event) {
GST_DEBUG_OBJECT (src, "doing seek with event %" GST_PTR_FORMAT, event);
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
/* no negative rates yet */
if (rate < 0.0)
goto negative_rate;
/* we need TIME format */
if (format != src->segment.format)
goto no_format;
/* Check if we are not at all seekable */
if (src->seekable == -1.0)
goto not_seekable;
/* Additional seeking-to-beginning-only check */
if (src->seekable == 0.0 && cur != 0)
goto not_seekable;
} else {
GST_DEBUG_OBJECT (src, "doing seek without event");
flags = 0;
cur_type = GST_SEEK_TYPE_SET;
stop_type = GST_SEEK_TYPE_SET;
}
/* get flush flag */
flush = flags & GST_SEEK_FLAG_FLUSH;
skip = flags & GST_SEEK_FLAG_SKIP;
/* now we need to make sure the streaming thread is stopped. We do this by
* either sending a FLUSH_START event downstream which will cause the
* streaming thread to stop with a WRONG_STATE.
* For a non-flushing seek we simply pause the task, which will happen as soon
* as it completes one iteration (and thus might block when the sink is
* blocking in preroll). */
if (flush) {
GST_DEBUG_OBJECT (src, "starting flush");
gst_rtspsrc_flush (src, TRUE, FALSE);
} else {
if (src->task) {
gst_task_pause (src->task);
}
}
/* we should now be able to grab the streaming thread because we stopped it
* with the above flush/pause code */
GST_RTSP_STREAM_LOCK (src);
GST_DEBUG_OBJECT (src, "stopped streaming");
/* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
gst_rtspsrc_connection_flush (src, FALSE);
/* copy segment, we need this because we still need the old
* segment when we close the current segment. */
memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
/* configure the seek parameters in the seeksegment. We will then have the
* right values in the segment to perform the seek */
if (event) {
GST_DEBUG_OBJECT (src, "configuring seek");
gst_segment_do_seek (&seeksegment, rate, format, flags,
cur_type, cur, stop_type, stop, &update);
}
/* figure out the last position we need to play. If it's configured (stop !=
* -1), use that, else we play until the total duration of the file */
if ((stop = seeksegment.stop) == -1)
stop = seeksegment.duration;
/* if we were playing, pause first */
playing = (src->state == GST_RTSP_STATE_PLAYING);
if (playing) {
/* obtain current position in case seek fails */
gst_rtspsrc_get_position (src);
gst_rtspsrc_pause (src, FALSE);
}
src->skip = skip;
src->state = GST_RTSP_STATE_SEEKING;
/* PLAY will add the range header now. */
src->need_range = TRUE;
/* prepare for streaming again */
if (flush) {
/* if we started flush, we stop now */
GST_DEBUG_OBJECT (src, "stopping flush");
gst_rtspsrc_flush (src, FALSE, playing);
}
/* now we did the seek and can activate the new segment values */
memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
/* if we're doing a segment seek, post a SEGMENT_START message */
if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (src),
gst_message_new_segment_start (GST_OBJECT_CAST (src),
src->segment.format, src->segment.position));
}
/* now create the newsegment */
GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, src->segment.position, stop);
/* mark discont */
GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
stream->discont = TRUE;
}
/* and continue playing if needed */
GST_OBJECT_LOCK (src);
playing = (GST_STATE_PENDING (src) == GST_STATE_VOID_PENDING
&& GST_STATE (src) == GST_STATE_PLAYING)
|| (GST_STATE_PENDING (src) == GST_STATE_PLAYING);
GST_OBJECT_UNLOCK (src);
if (src->version >= GST_RTSP_VERSION_2_0) {
if (flags & GST_SEEK_FLAG_ACCURATE)
seek_style = "RAP";
else if (flags & GST_SEEK_FLAG_KEY_UNIT)
seek_style = "CoRAP";
else if (flags & GST_SEEK_FLAG_KEY_UNIT
&& flags & GST_SEEK_FLAG_SNAP_BEFORE)
seek_style = "First-Prior";
else if (flags & GST_SEEK_FLAG_KEY_UNIT && flags & GST_SEEK_FLAG_SNAP_AFTER)
seek_style = "Next";
}
if (playing)
gst_rtspsrc_play (src, &seeksegment, FALSE, seek_style);
GST_RTSP_STREAM_UNLOCK (src);
return TRUE;
/* ERRORS */
negative_rate:
{
GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
return FALSE;
}
no_format:
{
GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
return FALSE;
}
not_seekable:
{
GST_DEBUG_OBJECT (src, "stream is not seekable");
return FALSE;
}
}
static gboolean
gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRTSPSrc *src;
gboolean res = TRUE;
gboolean forward;
src = GST_RTSPSRC_CAST (parent);
GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
res = gst_rtspsrc_perform_seek (src, event);
forward = FALSE;
break;
case GST_EVENT_QOS:
case GST_EVENT_NAVIGATION:
case GST_EVENT_LATENCY:
default:
forward = TRUE;
break;
}
if (forward) {
GstPad *target;
if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
res = gst_pad_send_event (target, event);
gst_object_unref (target);
} else {
gst_event_unref (event);
}
} else {
gst_event_unref (event);
}
return res;
}
static gboolean
gst_rtspsrc_handle_src_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRTSPStream *stream;
stream = gst_pad_get_element_private (pad);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_STREAM_START:{
const gchar *upstream_id;
gchar *stream_id;
gst_event_parse_stream_start (event, &upstream_id);
stream_id = g_strdup_printf ("%s/%s", upstream_id, stream->stream_id);
gst_event_unref (event);
event = gst_event_new_stream_start (stream_id);
g_free (stream_id);
break;
}
default:
break;
}
return gst_pad_push_event (stream->srcpad, event);
}
/* this is the final event function we receive on the internal source pad when
* we deal with TCP connections */
static gboolean
gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
gboolean res;
GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
case GST_EVENT_QOS:
case GST_EVENT_NAVIGATION:
case GST_EVENT_LATENCY:
default:
gst_event_unref (event);
res = TRUE;
break;
}
return res;
}
/* this is the final query function we receive on the internal source pad when
* we deal with TCP connections */
static gboolean
gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
GstQuery * query)
{
GstRTSPSrc *src;
gboolean res = TRUE;
src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
/* no idea */
break;
}
case GST_QUERY_DURATION:
{
GstFormat format;
gst_query_parse_duration (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
gst_query_set_duration (query, format, src->segment.duration);
break;
default:
res = FALSE;
break;
}
break;
}
case GST_QUERY_LATENCY:
{
/* we are live with a min latency of 0 and unlimited max latency, this
* result will be updated by the session manager if there is any. */
gst_query_set_latency (query, TRUE, 0, -1);
break;
}
default:
break;
}
return res;
}
/* this query is executed on the ghost source pad exposed on rtspsrc. */
static gboolean
gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
GstQuery * query)
{
GstRTSPSrc *src;
gboolean res = FALSE;
src = GST_RTSPSRC_CAST (parent);
GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_DURATION:
{
GstFormat format;
gst_query_parse_duration (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
gst_query_set_duration (query, format, src->segment.duration);
res = TRUE;
break;
default:
break;
}
break;
}
case GST_QUERY_SEEKING:
{
GstFormat format;
gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
if (format == GST_FORMAT_TIME) {
gboolean seekable =
src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
GstClockTime start = 0, duration = src->segment.duration;
/* seeking without duration is unlikely */
seekable = seekable && src->seekable >= 0.0 && src->segment.duration &&
GST_CLOCK_TIME_IS_VALID (src->segment.duration);
if (seekable) {
if (src->seekable > 0.0) {
start = src->last_pos - src->seekable * GST_SECOND;
} else {
/* src->seekable == 0 means that we can only seek to 0 */
start = 0;
duration = 0;
}
}
GST_LOG_OBJECT (src, "seekable : %d", seekable);
gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, start,
duration);
res = TRUE;
}
break;
}
case GST_QUERY_URI:
{
gchar *uri;
uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
if (uri != NULL) {
gst_query_set_uri (query, uri);
g_free (uri);
res = TRUE;
}
break;
}
default:
{
GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
/* forward the query to the proxy target pad */
if (target) {
res = gst_pad_query (target, query);
gst_object_unref (target);
}
break;
}
}
return res;
}
/* callback for RTCP messages to be sent to the server when operating in TCP
* mode. */
static GstFlowReturn
gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstRTSPSrc *src;
GstRTSPStream *stream;
GstFlowReturn res = GST_FLOW_OK;
GstMapInfo map;
guint8 *data;
guint size;
GstRTSPResult ret;
GstRTSPMessage message = { 0 };
GstRTSPConnInfo *conninfo;
stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
src = stream->parent;
gst_buffer_map (buffer, &map, GST_MAP_READ);
size = map.size;
data = map.data;
gst_rtsp_message_init_data (&message, stream->channel[1]);
/* lend the body data to the message */
gst_rtsp_message_take_body (&message, data, size);
if (stream->conninfo.connection)
conninfo = &stream->conninfo;
else
conninfo = &src->conninfo;
GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
ret = gst_rtspsrc_connection_send (src, conninfo, &message, NULL);
GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
/* and steal it away again because we will free it when unreffing the
* buffer */
gst_rtsp_message_steal_body (&message, &data, &size);
gst_rtsp_message_unset (&message);
gst_buffer_unmap (buffer<