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gstreamer/gst/rtsp/gstrtspsrc.c
Aleix Conchillo Flaqué 62f5a27416 rtspsrc: add tls-database property 
Add support for a new property: tls-database. If the property is set,
the certificate database will be given to the rtsp connection if TLS
protocol is being used. If the server certificate can't be verified with
the default database, this additional database will be used.

https://bugzilla.gnome.org/show_bug.cgi?id=724396
2014-02-20 20:03:40 +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>
*
* 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>
*
* Last reviewed on 2006-08-18 (0.10.5)
*/
#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/rtp/gstrtppayloads.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,
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;
}
#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
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_LAST
};
#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;
}
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 void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
GstCaps * caps);
static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
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);
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);
/* 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));
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 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",
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_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",
DEFAULT_USE_PIPELINE_CLOCK,
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));
/**
* 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::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);
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_pad_template (gstelement_class,
gst_static_pad_template_get (&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;
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;
/* 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;
GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
}
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);
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);
/* free locks */
g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
g_rec_mutex_clear (&rtspsrc->state_rec_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)
gst_object_ref (clock);
return clock;
}
/* 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:
if (rtspsrc->prop_proxy_id)
g_free (rtspsrc->prop_proxy_id);
rtspsrc->prop_proxy_id = g_value_dup_string (value);
break;
case PROP_PROXY_PW:
if (rtspsrc->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:
if (rtspsrc->user_id)
g_free (rtspsrc->user_id);
rtspsrc->user_id = g_value_dup_string (value);
break;
case PROP_USER_PW:
if (rtspsrc->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 (str) {
sscanf (str, "%u-%u",
&rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
} else {
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);
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;
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 = 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;
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)
{
if (stream->channel[0] == *channel || stream->channel[1] == *channel)
return 0;
return -1;
}
static gint
find_stream_by_pt (GstRTSPStream * stream, gint * pt)
{
if (stream->pt == *pt)
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)
{
/* check qualified setup_url */
if (!strcmp (stream->conninfo.location, (gchar *) a))
return 0;
/* 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 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 GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
{
GstRTSPStream *stream;
const gchar *control_url;
const gchar *payload;
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->disabled = FALSE;
stream->id = src->numstreams++;
stream->eos = FALSE;
stream->discont = TRUE;
stream->seqbase = -1;
stream->timebase = -1;
/* 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);
/* we must have a payload. No payload means we cannot create caps */
/* FIXME, handle multiple formats. The problem here is that we just want to
* take the first available format that we can handle but in order to do that
* we need to scan for depayloader plugins. Scanning for payloader plugins is
* also suboptimal because the user maybe just wants to save the raw stream
* and then we don't care. */
if ((payload = gst_sdp_media_get_format (media, 0))) {
stream->pt = atoi (payload);
/* convert caps */
stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
GST_DEBUG ("mapping sdp session level attributes to caps");
gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, stream->caps);
GST_DEBUG ("mapping sdp media level attributes to caps");
gst_rtspsrc_sdp_attributes_to_caps (media->attributes, stream->caps);
if (stream->pt >= 96) {
/* If we have a dynamic payload type, see if we have a stream with the
* same payload number. If there is one, they are part of the same
* container and we only need to add one pad. */
if (find_stream (src, &stream->pt, (gpointer) find_stream_by_pt)) {
stream->container = TRUE;
GST_DEBUG ("found another stream with pt %d, marking as container",
stream->pt);
}
}
}
/* 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, " pt: %d", stream->pt);
GST_DEBUG_OBJECT (src, " port: %d", stream->port);
GST_DEBUG_OBJECT (src, " container: %d", stream->container);
GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (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);
if (stream->caps)
gst_caps_unref (stream->caps);
g_free (stream->destination);
g_free (stream->control_url);
g_free (stream->conninfo.location);
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]);
stream->udpsrc[i] = NULL;
}
if (stream->channelpad[i]) {
gst_object_unref (stream->channelpad[i]);
stream->channelpad[i] = NULL;
}
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]);
stream->udpsink[i] = NULL;
}
}
if (stream->fakesrc) {
gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
gst_object_unref (stream->fakesrc);
stream->fakesrc = NULL;
}
if (stream->srcpad) {
gst_pad_set_active (stream->srcpad, FALSE);
if (stream->added) {
gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
stream->added = FALSE;
}
stream->srcpad = NULL;
}
if (stream->rtcppad) {
gst_object_unref (stream->rtcppad);
stream->rtcppad = NULL;
}
if (stream->session) {
g_object_unref (stream->session);
stream->session = NULL;
}
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;
}
src->numstreams = 0;
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;
}
if (src->start_segment) {
gst_event_unref (src->start_segment);
src->start_segment = NULL;
}
if (src->provided_clock) {
gst_object_unref (src->provided_clock);
src->provided_clock = NULL;
}
}
#define PARSE_INT(p, del, res) \
G_STMT_START { \
gchar *t = p; \
p = strstr (p, del); \
if (p == NULL) \
res = -1; \
else { \
*p = '\0'; \
p++; \
res = atoi (t); \
} \
} G_STMT_END
#define PARSE_STRING(p, del, res) \
G_STMT_START { \
gchar *t = p; \
p = strstr (p, del); \
if (p == NULL) { \
res = NULL; \
p = t; \
} \
else { \
*p = '\0'; \
p++; \
res = t; \
} \
} G_STMT_END
#define SKIP_SPACES(p) \
while (*p && g_ascii_isspace (*p)) \
p++;
/* rtpmap contains:
*
* <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
*/
static gboolean
gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
gint * rate, gchar ** params)
{
gchar *p, *t;
p = (gchar *) rtpmap;
PARSE_INT (p, " ", *payload);
if (*payload == -1)
return FALSE;
SKIP_SPACES (p);
if (*p == '\0')
return FALSE;
PARSE_STRING (p, "/", *name);
if (*name == NULL) {
GST_DEBUG ("no rate, name %s", p);
/* no rate, assume -1 then, this is not supposed to happen but RealMedia
* streams seem to omit the rate. */
*name = p;
*rate = -1;
return TRUE;
}
t = p;
p = strstr (p, "/");
if (p == NULL) {
*rate = atoi (t);
return TRUE;
}
*p = '\0';
p++;
*rate = atoi (t);
t = p;
if (*p == '\0')
return TRUE;
*params = t;
return TRUE;
}
/*
* Mapping SDP attributes to caps
*
* prepend 'a-' to IANA registered sdp attributes names
* (ie: not prefixed with 'x-') in order to avoid
* collision with gstreamer standard caps properties names
*/
static void
gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
{
if (attributes->len > 0) {
GstStructure *s;
guint i;
s = gst_caps_get_structure (caps, 0);
for (i = 0; i < attributes->len; i++) {
GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
gchar *tofree, *key;
key = attr->key;
/* skip some of the attribute we already handle */
if (!strcmp (key, "fmtp"))
continue;
if (!strcmp (key, "rtpmap"))
continue;
if (!strcmp (key, "control"))
continue;
if (!strcmp (key, "range"))
continue;
/* string must be valid UTF8 */
if (!g_utf8_validate (attr->value, -1, NULL))
continue;
if (!g_str_has_prefix (key, "x-"))
tofree = key = g_strdup_printf ("a-%s", key);
else
tofree = NULL;
GST_DEBUG ("adding caps: %s=%s", key, attr->value);
gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
g_free (tofree);
}
}
}
/*
* Mapping of caps to and from SDP fields:
*
* m=<media> <UDP port> RTP/AVP <payload>
* a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
* a=fmtp:<payload> <param>[=<value>];...
*/
static GstCaps *
gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
{
GstCaps *caps;
const gchar *rtpmap;
const gchar *fmtp;
gchar *name = NULL;
gint rate = -1;
gchar *params = NULL;
gchar *tmp;
GstStructure *s;
gint payload = 0;
gboolean ret;
/* get and parse rtpmap */
if ((rtpmap = gst_sdp_media_get_attribute_val (media, "rtpmap"))) {
ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, &params);
if (ret) {
if (payload != pt) {
/* we ignore the rtpmap if the payload type is different. */
g_warning ("rtpmap of wrong payload type, ignoring");
name = NULL;
rate = -1;
params = NULL;
}
} else {
/* if we failed to parse the rtpmap for a dynamic payload type, we have an
* error */
if (pt >= 96)
goto no_rtpmap;
/* else we can ignore */
g_warning ("error parsing rtpmap, ignoring");
}
} else {
/* dynamic payloads need rtpmap or we fail */
if (pt >= 96)
goto no_rtpmap;
}
/* check if we have a rate, if not, we need to look up the rate from the
* default rates based on the payload types. */
if (rate == -1) {
const GstRTPPayloadInfo *info;
if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
/* dynamic types, use media and encoding_name */
tmp = g_ascii_strdown (media->media, -1);
info = gst_rtp_payload_info_for_name (tmp, name);
g_free (tmp);
} else {
/* static types, use payload type */
info = gst_rtp_payload_info_for_pt (pt);
}
if (info) {
if ((rate = info->clock_rate) == 0)
rate = -1;
}
/* we fail if we cannot find one */
if (rate == -1)
goto no_rate;
}
tmp = g_ascii_strdown (media->media, -1);
caps = gst_caps_new_simple ("application/x-unknown",
"media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
g_free (tmp);
s = gst_caps_get_structure (caps, 0);
gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
/* encoding name must be upper case */
if (name != NULL) {
tmp = g_ascii_strup (name, -1);
gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
g_free (tmp);
}
/* params must be lower case */
if (params != NULL) {
tmp = g_ascii_strdown (params, -1);
gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
g_free (tmp);
}
/* parse optional fmtp: field */
if ((fmtp = gst_sdp_media_get_attribute_val (media, "fmtp"))) {
gchar *p;
gint payload = 0;
p = (gchar *) fmtp;
/* p is now of the format <payload> <param>[=<value>];... */
PARSE_INT (p, " ", payload);
if (payload != -1 && payload == pt) {
gchar **pairs;
gint i;
/* <param>[=<value>] are separated with ';' */
pairs = g_strsplit (p, ";", 0);
for (i = 0; pairs[i]; i++) {
gchar *valpos;
const gchar *val, *key;
/* the key may not have a '=', the value can have other '='s */
valpos = strstr (pairs[i], "=");
if (valpos) {
/* we have a '=' and thus a value, remove the '=' with \0 */
*valpos = '\0';
/* value is everything between '=' and ';'. We split the pairs at ;
* boundaries so we can take the remainder of the value. Some servers
* put spaces around the value which we strip off here. Alternatively
* we could strip those spaces in the depayloaders should these spaces
* actually carry any meaning in the future. */
val = g_strstrip (valpos + 1);
} else {
/* simple <param>;.. is translated into <param>=1;... */
val = "1";
}
/* strip the key of spaces, convert key to lowercase but not the value. */
key = g_strstrip (pairs[i]);
if (strlen (key) > 1) {
tmp = g_ascii_strdown (key, -1);
gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
g_free (tmp);
}
}
g_strfreev (pairs);
}
}
return caps;
/* ERRORS */
no_rtpmap:
{
g_warning ("rtpmap type not given for dynamic payload %d", pt);
return NULL;
}
no_rate:
{
g_warning ("rate unknown for payload type %d", pt);
return 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_PAUSED);
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_PAUSED);
/* 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, GstRTSPConnection * conn,
GstRTSPMessage * message, GTimeVal * timeout)
{
GstRTSPResult ret;
if (conn)
ret = gst_rtsp_connection_send (conn, message, timeout);
else
ret = GST_RTSP_ERROR;
return ret;
}
static GstRTSPResult
gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
GstRTSPMessage * message, GTimeVal * timeout)
{
GstRTSPResult ret;
if (conn)
ret = gst_rtsp_connection_receive (conn, message, timeout);
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;
return;
}
}
}
src->last_pos = 0;
}
static gboolean
gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
{
src->state = GST_RTSP_STATE_SEEKING;
/* PLAY will add the range header now. */
src->need_range = TRUE;
return TRUE;
}
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;
if (event) {
GST_DEBUG_OBJECT (src, "doing seek with 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;
} 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");
/* 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;
playing = (src->state == GST_RTSP_STATE_PLAYING);
/* if we were playing, pause first */
if (playing) {
/* obtain current position in case seek fails */
gst_rtspsrc_get_position (src);
gst_rtspsrc_pause (src, FALSE);
}
src->skip = skip;
gst_rtspsrc_do_seek (src, &seeksegment);
/* and continue playing */
if (playing)
gst_rtspsrc_play (src, &seeksegment, FALSE);
/* 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;
}
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;
}
}
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;
}
/* 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;
/* seeking without duration is unlikely */
seekable = seekable && src->seekable && src->segment.duration &&
GST_CLOCK_TIME_IS_VALID (src->segment.duration);
/* FIXME ?? should we have 0 and segment.duration here; see demuxers */
gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
src->segment.start, src->segment.stop);
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 };
GstRTSPConnection *conn;
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)
conn = stream->conninfo.connection;
else
conn = src->conninfo.connection;
GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
ret = gst_rtspsrc_connection_send (src, conn, &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, &map);
gst_buffer_unref (buffer);
return res;
}
static GstPadProbeReturn
pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
{
GstRTSPSrc *src = user_data;
GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
GST_DEBUG_PAD_NAME (pad));
/* activate the streams */
GST_OBJECT_LOCK (src);
if (!src->need_activate)
goto was_ok;
src->need_activate = FALSE;
GST_OBJECT_UNLOCK (src);
gst_rtspsrc_activate_streams (src);
return GST_PAD_PROBE_OK;
was_ok:
{
GST_OBJECT_UNLOCK (src);
return GST_PAD_PROBE_OK;
}
}
/* this callback is called when the session manager generated a new src pad with
* payloaded RTP packets. We simply ghost the pad here. */
static void
new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
{
gchar *name;
GstPadTemplate *template;
gint id, ssrc, pt;
GList *ostreams;
GstRTSPStream *stream;
gboolean all_added;
GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
GST_RTSP_STATE_LOCK (src);
/* find stream */
name = gst_object_get_name (GST_OBJECT_CAST (pad));
if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
goto unknown_stream;
GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
stream = find_stream (src, &id, (gpointer) find_stream_by_id);
if (stream == NULL)
goto unknown_stream;
/* save SSRC */
stream->ssrc = ssrc;
/* we'll add it later see below */
stream->added = TRUE;
/* check if we added all streams */
all_added = TRUE;
for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
GST_DEBUG_OBJECT (src, "stream %p, container %d, disabled %d, added %d",
ostream, ostream->container, ostream->disabled, ostream->added);
/* a container stream only needs one pad added. Also disabled streams don't
* count */
if (!ostream->container && !ostream->disabled && !ostream->added) {
all_added = FALSE;
break;
}
}
GST_RTSP_STATE_UNLOCK (src);
/* create a new pad we will use to stream to */
template = gst_static_pad_template_get (&rtptemplate);
stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
gst_object_unref (template);
g_free (name);
gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
gst_pad_set_active (stream->srcpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
if (all_added) {
GST_DEBUG_OBJECT (src, "We added all streams");
/* when we get here, all stream are added and we can fire the no-more-pads
* signal. */
gst_element_no_more_pads (GST_ELEMENT_CAST (src));
}
return;
/* ERRORS */
unknown_stream:
{
GST_DEBUG_OBJECT (src, "ignoring unknown stream");
GST_RTSP_STATE_UNLOCK (src);
g_free (name);
return;
}
}
static GstCaps *
request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
{
GstRTSPStream *stream;
GstCaps *caps;
GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
GST_RTSP_STATE_LOCK (src);
stream = find_stream (src, &session, (gpointer) find_stream_by_id);
if (!stream)
goto unknown_stream;
caps = stream->caps;
if (caps)
gst_caps_ref (caps);
GST_RTSP_STATE_UNLOCK (src);
return caps;
unknown_stream:
{
GST_DEBUG_OBJECT (src, "unknown stream %d", session);
GST_RTSP_STATE_UNLOCK (src);
return NULL;
}
}
static void
gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
{
GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
if (stream->eos)
goto was_eos;
stream->eos = TRUE;
gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
return;
/* ERRORS */
was_eos:
{
GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
return;
}
}
static void
on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPSrc *src = stream->parent;
guint ssrc;
g_object_get (source, "ssrc", &ssrc, NULL);
GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
ssrc, stream->ssrc, stream->id);
if (ssrc == stream->ssrc)
gst_rtspsrc_do_stream_eos (src, stream);
}
static void
on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPSrc *src = stream->parent;
guint ssrc;
g_object_get (source, "ssrc", &ssrc, NULL);
GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
ssrc, stream->ssrc, stream->id);
if (ssrc == stream->ssrc)
gst_rtspsrc_do_stream_eos (src, stream);
}
static void
on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
{
GstRTSPStream *stream;
GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
/* get stream for session */
stream = find_stream (src, &session, (gpointer) find_stream_by_id);
if (stream) {
gst_rtspsrc_do_stream_eos (src, stream);
}
}
static void
on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
{
GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
stream->id);
}
static void
set_manager_buffer_mode (GstRTSPSrc * src)
{
GObjectClass *klass;
if (src->manager == NULL)
return;
klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
if (!g_object_class_find_property (klass, "buffer-mode"))
return;
if (src->buffer_mode != BUFFER_MODE_AUTO) {
g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
return;
}
GST_DEBUG_OBJECT (src,
"auto buffering mode, have clock %" GST_PTR_FORMAT, src->provided_clock);
if (src->provided_clock) {
GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
if (clock == src->provided_clock) {
GST_DEBUG_OBJECT (src, "selected synced");
g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SYNCED, NULL);
if (clock)
gst_object_unref (clock);
return;
}
/* Otherwise fall-through and use another buffer mode */
if (clock)
gst_object_unref (clock);
}
GST_DEBUG_OBJECT (src, "auto buffering mode");
if (src->use_buffering) {
GST_DEBUG_OBJECT (src, "selected buffer");
g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL);
} else {
GST_DEBUG_OBJECT (src, "selected slave");
g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
}
}
/* try to get and configure a manager */
static gboolean
gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
GstRTSPTransport * transport)
{
const gchar *manager;
gchar *name;
GstStateChangeReturn ret;
/* find a manager */
if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
goto no_manager;
if (manager) {
GST_DEBUG_OBJECT (src, "using manager %s", manager);
/* configure the manager */
if (src->manager == NULL) {
GObjectClass *klass;
GstStructure *s;
const gchar *encoding;
gboolean need_slave;
if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
/* fallback */
if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
goto no_manager;
if (!manager)
goto use_no_manager;
if (!(src->manager = gst_element_factory_make (manager, "manager")))
goto manager_failed;
}
/* we manage this element */
gst_element_set_locked_state (src->manager, TRUE);
gst_bin_add (GST_BIN_CAST (src), src->manager);
ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_manager_failure;
g_object_set (src->manager, "latency", src->latency, NULL);
klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
if (g_object_class_find_property (klass, "ntp-sync")) {
g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
}
if (g_object_class_find_property (klass, "use-pipeline-clock")) {
g_object_set (src->manager, "use-pipeline-clock",
src->use_pipeline_clock, NULL);
}
if (src->sdes && g_object_class_find_property (klass, "sdes")) {
g_object_set (src->manager, "sdes", src->sdes, NULL);
}
if (g_object_class_find_property (klass, "drop-on-latency")) {
g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
NULL);
}
/* buffer mode pauses are handled by adding offsets to buffer times,
* but some depayloaders may have a hard time syncing output times
* with such input times, e.g. container ones, most notably ASF */
/* TODO alternatives are having an event that indicates these shifts,
* or having rtsp extensions provide suggestion on buffer mode */
need_slave = stream->container;
if (stream->caps && (s = gst_caps_get_structure (stream->caps, 0)) &&
(encoding = gst_structure_get_string (s, "encoding-name")))
need_slave = need_slave || (strcmp (encoding, "X-ASF-PF") == 0);
/* valid duration implies not likely live pipeline,
* so slaving in jitterbuffer does not make much sense
* (and might mess things up due to bursts) */
if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
src->segment.duration && !need_slave) {
src->use_buffering = TRUE;
} else {
src->use_buffering = FALSE;
}
set_manager_buffer_mode (src);
/* connect to signals */
GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
stream);
src->manager_sig_id =
g_signal_connect (src->manager, "pad-added",
(GCallback) new_manager_pad, src);
src->manager_ptmap_id =
g_signal_connect (src->manager, "request-pt-map",
(GCallback) request_pt_map, src);
g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
src);
g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_NEW_MANAGER], 0,
src->manager);
}
/* we stream directly to the manager, get some pads. Each RTSP stream goes
* into a separate RTP session. */
name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
g_free (name);
name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
g_free (name);
/* now configure the bandwidth in the manager */
if (g_signal_lookup ("get-internal-session",
G_OBJECT_TYPE (src->manager)) != 0) {
GObject *rtpsession;
g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
&rtpsession);
if (rtpsession) {
GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
stream->session = rtpsession;
if (stream->as_bandwidth != -1) {
GST_INFO_OBJECT (src, "setting AS: %f",
(gdouble) (stream->as_bandwidth * 1000));
g_object_set (rtpsession, "bandwidth",
(gdouble) (stream->as_bandwidth * 1000), NULL);
}
if (stream->rr_bandwidth != -1) {
GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
NULL);
}
if (stream->rs_bandwidth != -1) {
GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
NULL);
}
g_object_set (rtpsession, "probation", src->probation, NULL);
g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
stream);
g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
stream);
g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
stream);
g_signal_connect (rtpsession, "on-ssrc-active",
(GCallback) on_ssrc_active, stream);
}
}
}
use_no_manager:
return TRUE;
/* ERRORS */
no_manager:
{
GST_DEBUG_OBJECT (src, "cannot get a session manager");
return FALSE;
}
manager_failed:
{
GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
return FALSE;
}
start_manager_failure:
{
GST_DEBUG_OBJECT (src, "could not start session manager");
return FALSE;
}
}
/* free the UDP sources allocated when negotiating a transport.
* This function is called when the server negotiated to a transport where the
* UDP sources are not needed anymore, such as TCP or multicast. */
static void
gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
{
gint i;
for (i = 0; i < 2; i++) {
if (stream->udpsrc[i]) {
GST_DEBUG ("free UDP source %d for stream %p", i, stream);
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
gst_object_unref (stream->udpsrc[i]);
stream->udpsrc[i] = NULL;
}
}
}
/* for TCP, create pads to send and receive data to and from the manager and to
* intercept various events and queries
*/
static gboolean
gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
GstRTSPTransport * transport, GstPad ** outpad)
{
gchar *name;
GstPadTemplate *template;
GstPad *pad0, *pad1;
/* configure for interleaved delivery, nothing needs to be done
* here, the loop function will call the chain functions of the
* session manager. */
stream->channel[0] = transport->interleaved.min;
stream->channel[1] = transport->interleaved.max;
GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
stream->channel[0], stream->channel[1]);
/* we can remove the allocated UDP ports now */
gst_rtspsrc_stream_free_udp (stream);
/* no session manager, send data to srcpad directly */
if (!stream->channelpad[0]) {
GST_DEBUG_OBJECT (src, "no manager, creating pad");
/* create a new pad we will use to stream to */
name = g_strdup_printf ("stream_%u", stream->id);
template = gst_static_pad_template_get (&rtptemplate);
stream->channelpad[0] = gst_pad_new_from_template (template, name);
gst_object_unref (template);
g_free (name);
/* set caps and activate */
gst_pad_use_fixed_caps (stream->channelpad[0]);
gst_pad_set_active (stream->channelpad[0], TRUE);
*outpad = gst_object_ref (stream->channelpad[0]);
} else {
GST_DEBUG_OBJECT (src, "using manager source pad");
template = gst_static_pad_template_get (&anysrctemplate);
/* allocate pads for sending the channel data into the manager */
pad0 = gst_pad_new_from_template (template, "internalsrc_0");
gst_pad_link_full (pad0, stream->channelpad[0], GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (stream->channelpad[0]);
stream->channelpad[0] = pad0;
gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
gst_pad_set_element_private (pad0, src);
gst_pad_set_active (pad0, TRUE);
if (stream->channelpad[1]) {
/* if we have a sinkpad for the other channel, create a pad and link to the
* manager. */
pad1 = gst_pad_new_from_template (template, "internalsrc_1");
gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
gst_pad_link_full (pad1, stream->channelpad[1],
GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (stream->channelpad[1]);
stream->channelpad[1] = pad1;
gst_pad_set_active (pad1, TRUE);
}
gst_object_unref (template);
}
/* setup RTCP transport back to the server if we have to. */
if (src->manager && src->do_rtcp) {
GstPad *pad;
template = gst_static_pad_template_get (&anysinktemplate);
stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
gst_pad_set_element_private (stream->rtcppad, stream);
gst_pad_set_active (stream->rtcppad, TRUE);
/* get session RTCP pad */
name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
pad = gst_element_get_request_pad (src->manager, name);
g_free (name);
/* and link */
if (pad) {
gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (pad);
}
gst_object_unref (template);
}
return TRUE;
}
static void
gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
GstRTSPTransport * transport, const gchar ** destination, gint * min,
gint * max, guint * ttl)
{
if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
if (destination) {
if (!(*destination = transport->destination))
*destination = stream->destination;
}
if (min && max) {
/* transport first */
*min = transport->port.min;
*max = transport->port.max;
if (*min == -1 && *max == -1) {
/* then try from SDP */
if (stream->port != 0) {
*min = stream->port;
*max = stream->port + 1;
}
}
}
if (ttl) {
if (!(*ttl = transport->ttl))
*ttl = stream->ttl;
}
} else {
if (destination) {
/* first take the source, then the endpoint to figure out where to send
* the RTCP. */
if (!(*destination = transport->source)) {
if (src->conninfo.connection)
*destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
else if (stream->conninfo.connection)
*destination =
gst_rtsp_connection_get_ip (stream->conninfo.connection);
}
}
if (min && max) {
/* for unicast we only expect the ports here */
*min = transport->server_port.min;
*max = transport->server_port.max;
}
}
}
/* For multicast create UDP sources and join the multicast group. */
static gboolean
gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
GstRTSPTransport * transport, GstPad ** outpad)
{
gchar *uri;
const gchar *destination;
gint min, max;
GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
/* we can remove the allocated UDP ports now */
gst_rtspsrc_stream_free_udp (stream);
gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
&max, NULL);
/* we need a destination now */
if (destination == NULL)
goto no_destination;
/* we really need ports now or we won't be able to receive anything at all */
if (min == -1 && max == -1)
goto no_ports;
GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
destination, min, max);
/* creating UDP source for RTP */
if (min != -1) {
uri = g_strdup_printf ("udp://%s:%d", destination, min);
stream->udpsrc[0] =
gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
g_free (uri);
if (stream->udpsrc[0] == NULL)
goto no_element;
/* take ownership */
gst_object_ref_sink (stream->udpsrc[0]);
if (src->udp_buffer_size != 0)
g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
src->udp_buffer_size, NULL);
if (src->multi_iface != NULL)
g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
src->multi_iface, NULL);
/* change state */
gst_element_set_locked_state (stream->udpsrc[0], TRUE);
gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
}
/* creating another UDP source for RTCP */
if (max != -1) {
GstCaps *caps;
uri = g_strdup_printf ("udp://%s:%d", destination, max);
stream->udpsrc[1] =
gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
g_free (uri);
if (stream->udpsrc[1] == NULL)
goto no_element;
caps = gst_caps_new_empty_simple ("application/x-rtcp");
g_object_set (stream->udpsrc[1], "caps", caps, NULL);
gst_caps_unref (caps);
/* take ownership */
gst_object_ref_sink (stream->udpsrc[1]);
if (src->multi_iface != NULL)
g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
src->multi_iface, NULL);
gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
}
return TRUE;
/* ERRORS */
no_element:
{
GST_DEBUG_OBJECT (src, "no UDP source element found");
return FALSE;
}
no_destination:
{
GST_DEBUG_OBJECT (src, "no destination found");
return FALSE;
}
no_ports:
{
GST_DEBUG_OBJECT (src, "no ports found");
return FALSE;
}
}
/* configure the remainder of the UDP ports */
static gboolean
gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
GstRTSPTransport * transport, GstPad ** outpad)
{
/* we manage the UDP elements now. For unicast, the UDP sources where
* allocated in the stream when we suggested a transport. */
if (stream->udpsrc[0]) {
gst_element_set_locked_state (stream->udpsrc[0], TRUE);
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
GST_DEBUG_OBJECT (src, "setting up UDP source");
/* configure a timeout on the UDP port. When the timeout message is
* posted, we assume UDP transport is not possible. We reconnect using TCP
* if we can. */
g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
src->udp_timeout * 1000, NULL);
/* get output pad of the UDP source. */
*outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
/* save it so we can unblock */
stream->blockedpad = *outpad;
/* configure pad block on the pad. As soon as there is dataflow on the
* UDP source, we know that UDP is not blocked by a firewall and we can
* configure all the streams to let the application autoplug decoders. */
stream->blockid =
gst_pad_add_probe (stream->blockedpad,
GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, pad_blocked, src, NULL);
if (stream->channelpad[0]) {
GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
/* configure for UDP delivery, we need to connect the UDP pads to
* the session plugin. */
gst_pad_link_full (*outpad, stream->channelpad[0],
GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (*outpad);
*outpad = NULL;
/* we connected to pad-added signal to get pads from the manager */
} else {
GST_DEBUG_OBJECT (src, "using UDP src pad as output");
}
}
/* RTCP port */
if (stream->udpsrc[1]) {
GstCaps *caps;
gst_element_set_locked_state (stream->udpsrc[1], TRUE);
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
caps = gst_caps_new_empty_simple ("application/x-rtcp");
g_object_set (stream->udpsrc[1], "caps", caps, NULL);
gst_caps_unref (caps);
if (stream->channelpad[1]) {
GstPad *pad;
GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
gst_pad_link_full (pad, stream->channelpad[1],
GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (pad);
} else {
/* leave unlinked */
}
}
return TRUE;
}
/* configure the UDP sink back to the server for status reports */
static gboolean
gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
GstRTSPStream * stream, GstRTSPTransport * transport)
{
GstPad *pad;
gint rtp_port, rtcp_port;
gboolean do_rtp, do_rtcp;
const gchar *destination;
gchar *uri, *name;
guint ttl = 0;
GSocket *socket;
/* get transport info */
gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
&rtp_port, &rtcp_port, &ttl);
/* see what we need to do */
do_rtp = (rtp_port != -1);
/* it's possible that the server does not want us to send RTCP in which case
* the port is -1 */
do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
/* we need a destination when we have RTP or RTCP ports */
if (destination == NULL && (do_rtp || do_rtcp))
goto no_destination;
/* try to construct the fakesrc to the RTP port of the server to open up any
* NAT firewalls */
if (do_rtp) {
GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
rtp_port);
uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
stream->udpsink[0] =
gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
g_free (uri);
if (stream->udpsink[0] == NULL)
goto no_sink_element;
/* don't join multicast group, we will have the source socket do that */
/* no sync or async state changes needed */
g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
"loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
if (ttl > 0)
g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
if (stream->udpsrc[0]) {
/* configure socket, we give it the same UDP socket as the udpsrc for RTP
* so that NAT firewalls will open a hole for us */
g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
/* configure socket and make sure udpsink does not close it when shutting
* down, it belongs to udpsrc after all. */
g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
"close-socket", FALSE, NULL);
g_object_unref (socket);
}
/* the source for the dummy packets to open up NAT */
stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
if (stream->fakesrc == NULL)
goto no_fakesrc_element;
/* random data in 5 buffers, a size of 200 bytes should be fine */
g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
"sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
/* we don't want to consider this a sink */
GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
/* keep everything locked */
gst_element_set_locked_state (stream->udpsink[0], TRUE);
gst_element_set_locked_state (stream->fakesrc, TRUE);
gst_object_ref (stream->udpsink[0]);
gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
gst_object_ref (stream->fakesrc);
gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
"sink", GST_PAD_LINK_CHECK_NOTHING);
}
if (do_rtcp) {
GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
rtcp_port);
uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
stream->udpsink[1] =
gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
g_free (uri);
if (stream->udpsink[1] == NULL)
goto no_sink_element;
/* don't join multicast group, we will have the source socket do that */
/* no sync or async state changes needed */
g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
"loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
if (ttl > 0)
g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
if (stream->udpsrc[1]) {
/* configure socket, we give it the same UDP socket as the udpsrc for RTCP
* because some servers check the port number of where it sends RTCP to identify
* the RTCP packets it receives */
g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
/* configure socket and make sure udpsink does not close it when shutting
* down, it belongs to udpsrc after all. */
g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
"close-socket", FALSE, NULL);
g_object_unref (socket);
}
/* we don't want to consider this a sink */
GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
/* we keep this playing always */
gst_element_set_locked_state (stream->udpsink[1], TRUE);
gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
gst_object_ref (stream->udpsink[1]);
gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
/* get session RTCP pad */
name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
pad = gst_element_get_request_pad (src->manager, name);
g_free (name);
/* and link */
if (pad) {
gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (pad);
}
}
return TRUE;
/* ERRORS */
no_destination:
{
GST_DEBUG_OBJECT (src, "no destination address specified");
return FALSE;
}
no_sink_element:
{
GST_DEBUG_OBJECT (src, "no UDP sink element found");
return FALSE;
}
no_fakesrc_element:
{
GST_DEBUG_OBJECT (src, "no fakesrc element found");
return FALSE;
}
}
/* sets up all elements needed for streaming over the specified transport.
* Does not yet expose the element pads, this will be done when there is actuall
* dataflow detected, which might never happen when UDP is blocked in a
* firewall, for example.
*/
static gboolean
gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
GstRTSPTransport * transport)
{
GstRTSPSrc *src;
GstPad *outpad = NULL;
GstPadTemplate *template;
gchar *name;
GstStructure *s;
const gchar *media_type;
src = stream->parent;
GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
s = gst_caps_get_structure (stream->caps, 0);
/* get the proper media type for this stream now */
if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
goto unknown_transport;
if (!media_type)
goto unknown_transport;
/* configure the final media type */
GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
gst_structure_set_name (s, media_type);
/* try to get and configure a manager, channelpad[0-1] will be configured with
* the pads for the manager, or NULL when no manager is needed. */
if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
goto no_manager;
switch (transport->lower_transport) {
case GST_RTSP_LOWER_TRANS_TCP:
if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
goto transport_failed;
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
goto transport_failed;
/* fallthrough, the rest is the same for UDP and MCAST */
case GST_RTSP_LOWER_TRANS_UDP:
if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
goto transport_failed;
/* configure udpsinks back to the server for RTCP messages and for the
* dummy RTP messages to open NAT. */
if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
goto transport_failed;
break;
default:
goto unknown_transport;
}
if (outpad) {
GST_DEBUG_OBJECT (src, "creating ghostpad");
gst_pad_use_fixed_caps (outpad);
/* create ghostpad, don't add just yet, this will be done when we activate
* the stream. */
name = g_strdup_printf ("stream_%u", stream->id);
template = gst_static_pad_template_get (&rtptemplate);
stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
gst_object_unref (template);
g_free (name);
gst_object_unref (outpad);
}
/* mark pad as ok */
stream->last_ret = GST_FLOW_OK;
return TRUE;
/* ERRORS */
transport_failed:
{
GST_DEBUG_OBJECT (src, "failed to configure transport");
return FALSE;
}
unknown_transport:
{
GST_DEBUG_OBJECT (src, "unknown transport");
return FALSE;
}
no_manager:
{
GST_DEBUG_OBJECT (src, "cannot get a session manager");
return FALSE;
}
}
/* send a couple of dummy random packets on the receiver RTP port to the server,
* this should make a firewall think we initiated the data transfer and
* hopefully allow packets to go from the sender port to our RTP receiver port */
static gboolean
gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
{
GList *walk;
if (src->nat_method != GST_RTSP_NAT_DUMMY)
return TRUE;
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
if (stream->fakesrc && stream->udpsink[0]) {
GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
}
}
return TRUE;
}
/* Adds the source pads of all configured streams to the element.
* This code is performed when we detected dataflow.
*
* We detect dataflow from either the _loop function or with pad probes on the
* udp sources.
*/
static gboolean
gst_rtspsrc_activate_streams (GstRTSPSrc * src)
{
GList *walk;
GST_DEBUG_OBJECT (src, "activating streams");
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
if (stream->udpsrc[0]) {
/* remove timeout, we are streaming now and timeouts will be handled by
* the session manager and jitter buffer */
g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
}
if (stream->srcpad) {
GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
gst_pad_set_active (stream->srcpad, TRUE);
/* if we don't have a session manager, set the caps now. If we have a
* session, we will get a notification of the pad and the caps. */
if (!src->manager) {
GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
gst_pad_set_caps (stream->srcpad, stream->caps);
}
/* add the pad */
if (!stream->added) {
GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
stream->added = TRUE;
}
}
}
/* unblock all pads */
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
if (stream->blockid) {
GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
gst_pad_remove_probe (stream->blockedpad, stream->blockid);
stream->blockid = 0;
}
}
return TRUE;
}
static void
gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
gboolean reset_manager)
{
GList *walk;
guint64 start, stop;
gdouble play_speed, play_scale;
GST_DEBUG_OBJECT (src, "configuring stream caps");
start = segment->position;
stop = segment->duration;
play_speed = segment->rate;
play_scale = segment->applied_rate;
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
GstCaps *caps;
if ((caps = stream->caps)) {
caps = gst_caps_make_writable (caps);
/* update caps */
if (stream->timebase != -1)
gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
(guint) stream->timebase, NULL);
if (stream->seqbase != -1)
gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
(guint) stream->seqbase, NULL);
gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
if (stop != -1)
gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
stream->caps = caps;
}
GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
}
if (reset_manager && src->manager) {
GST_DEBUG_OBJECT (src, "clear session");
g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
}
}
static GstFlowReturn
gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
GstFlowReturn ret)
{
GList *streams;
/* store the value */
stream->last_ret = ret;
/* if it's success we can return the value right away */
if (ret == GST_FLOW_OK)
goto done;
/* any other error that is not-linked can be returned right
* away */
if (ret != GST_FLOW_NOT_LINKED)
goto done;
/* only return NOT_LINKED if all other pads returned NOT_LINKED */
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
ret = ostream->last_ret;
/* some other return value (must be SUCCESS but we can return
* other values as well) */
if (ret != GST_FLOW_NOT_LINKED)
goto done;
}
/* if we get here, all other pads were unlinked and we return
* NOT_LINKED then */
done:
return ret;
}
static gboolean
gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
GstEvent * event)
{
gboolean res = TRUE;
/* only streams that have a connection to the outside world */
if (stream->container || stream->disabled)
goto done;
if (stream->udpsrc[0]) {
gst_event_ref (event);
res = gst_element_send_event (stream->udpsrc[0], event);
} else if (stream->channelpad[0]) {
gst_event_ref (event);
if (GST_PAD_IS_SRC (stream->channelpad[0]))
res = gst_pad_push_event (stream->channelpad[0], event);
else
res = gst_pad_send_event (stream->channelpad[0], event);
}
if (stream->udpsrc[1]) {
gst_event_ref (event);
res &= gst_element_send_event (stream->udpsrc[1], event);
} else if (stream->channelpad[1]) {
gst_event_ref (event);
if (GST_PAD_IS_SRC (stream->channelpad[1]))
res &= gst_pad_push_event (stream->channelpad[1], event);
else
res &= gst_pad_send_event (stream->channelpad[1], event);
}
done:
gst_event_unref (event);
return res;
}
static gboolean
gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
{
GList *streams;
gboolean res = TRUE;
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
gst_event_ref (event);
res &= gst_rtspsrc_stream_push_event (src, ostream, event);
}
gst_event_unref (event);
return res;
}
static GstRTSPResult
gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
gboolean async)
{
GstRTSPResult res;
if (info->connection == NULL) {
if (info->url == NULL) {
GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
goto parse_error;
}
/* create connection */
GST_DEBUG_OBJECT (src, "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 (src, "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,
src->tls_validation_flags))
GST_WARNING_OBJECT (src, "Unable to set TLS validation flags");
if (src->tls_database)
gst_rtsp_connection_set_tls_database (info->connection,
src->tls_database);
}
if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
gst_rtsp_connection_set_tunneled (info->connection, TRUE);
if (src->proxy_host) {
GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
src->proxy_port);
gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
src->proxy_port);
}
}
if (!info->connected) {
/* connect */
if (async)
GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
("Connecting to %s", info->location));
GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
if ((res =
gst_rtsp_connection_connect (info->connection,
src->ptcp_timeout)) < 0)
goto could_not_connect;
info->connected = TRUE;
}
return GST_RTSP_OK;
/* ERRORS */
parse_error:
{
GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
return res;
}
could_not_create:
{
gchar *str = gst_rtsp_strresult (res);
GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
g_free (str);
return res;
}
could_not_connect:
{
gchar *str = gst_rtsp_strresult (res);
GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
g_free (str);
return res;
}
}
static GstRTSPResult
gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
gboolean free)
{
GST_RTSP_STATE_LOCK (src);
if (info->connected) {
GST_DEBUG_OBJECT (src, "closing connection...");
gst_rtsp_connection_close (info->connection);
info->connected = FALSE;
}
if (free && info->connection) {
/* free connection */
GST_DEBUG_OBJECT (src, "freeing connection...");
gst_rtsp_connection_free (info->connection);
info->connection = NULL;
}
GST_RTSP_STATE_UNLOCK (src);
return GST_RTSP_OK;
}
static GstRTSPResult
gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
gboolean async)
{
GstRTSPResult res;
GST_DEBUG_OBJECT (src, "reconnecting connection...");
gst_rtsp_conninfo_close (src, info, FALSE);
res = gst_rtsp_conninfo_connect (src, info, async);
return res;
}
static void
gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
{
GList *walk;
GST_DEBUG_OBJECT (src, "set flushing %d", flush);
GST_RTSP_STATE_LOCK (src);
if (src->conninfo.connection && src->conninfo.flushing != flush) {
GST_DEBUG_OBJECT (src, "connection flush");
gst_rtsp_connection_flush (src->conninfo.connection, flush);
src->conninfo.flushing = flush;
}
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
GST_DEBUG_OBJECT (src, "stream %p flush", stream);
gst_rtsp_connection_flush (stream->conninfo.connection, flush);
stream->conninfo.flushing = flush;
}
}
GST_RTSP_STATE_UNLOCK (src);
}
/* FIXME, handle server request, reply with OK, for now */
static GstRTSPResult
gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
GstRTSPMessage * request)
{
GstRTSPMessage response = { 0 };
GstRTSPResult res;
GST_DEBUG_OBJECT (src, "got server request message");
if (src->debug)
gst_rtsp_message_dump (request);
res = gst_rtsp_ext_list_receive_request (src->extensions, request);
if (res == GST_RTSP_ENOTIMPL) {
/* default implementation, send OK */
GST_DEBUG_OBJECT (src, "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 (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
0, request, &response);
if (src->debug)
gst_rtsp_message_dump (&response);
res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
if (res < 0)
goto send_error;
gst_rtsp_message_unset (&response);
} else if (res == GST_RTSP_EEOF)
return res;
return GST_RTSP_OK;
/* ERRORS */
send_error:
{
gst_rtsp_message_unset (&response);
return res;
}
}
/* send server keep-alive */
static GstRTSPResult
gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
{
GstRTSPMessage request = { 0 };
GstRTSPResult res;
GstRTSPMethod method;
const gchar *control;
if (src->do_rtsp_keep_alive == FALSE) {
GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
gst_rtsp_connection_reset_timeout (src->conninfo.connection);
return GST_RTSP_OK;
}
GST_DEBUG_OBJECT (src, "creating server keep-alive");
/* find a method to use for keep-alive */
if (src->methods & GST_RTSP_GET_PARAMETER)
method = GST_RTSP_GET_PARAMETER;
else
method = GST_RTSP_OPTIONS;
control = get_aggregate_control (src);
if (control == NULL)
goto no_control;
res = gst_rtsp_message_init_request (&request, method, control);
if (res < 0)
goto send_error;
if (src->debug)
gst_rtsp_message_dump (&request);
res =
gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
NULL);
if (res < 0)
goto send_error;
gst_rtsp_connection_reset_timeout (src->conninfo.connection);
gst_rtsp_message_unset (&request);
return GST_RTSP_OK;
/* ERRORS */
no_control:
{
GST_WARNING_OBJECT (src, "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 (src, RESOURCE, WRITE, (NULL),
("Could not send keep-alive. (%s)", str));
g_free (str);
return res;
}
}
static GstFlowReturn
gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
{
GstFlowReturn ret = GST_FLOW_OK;
gint channel;
GstRTSPStream *stream;
GstPad *outpad = NULL;
guint8 *data;
guint size;
GstBuffer *buf;
gboolean is_rtcp;
GstEvent *event;
channel = message->type_data.data.channel;
stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
if (!stream)
goto unknown_stream;
if (channel == stream->channel[0]) {
outpad = stream->channelpad[0];
is_rtcp = FALSE;
} else if (channel == stream->channel[1]) {
outpad = stream->channelpad[1];
is_rtcp = TRUE;
} else {
is_rtcp = FALSE;
}
/* take a look at the body to figure out what we have */
gst_rtsp_message_get_body (message, &data, &size);
if (size < 2)
goto invalid_length;
/* channels are not correct on some servers, do extra check */
if (data[1] >= 200 && data[1] <= 204) {
/* hmm RTCP message switch to the RTCP pad of the same stream. */
outpad = stream->channelpad[1];
is_rtcp = TRUE;
}
/* we have no clue what this is, just ignore then. */
if (outpad == NULL)
goto unknown_stream;
/* take the message body for further processing */
gst_rtsp_message_steal_body (message, &data, &size);
/* strip the trailing \0 */
size -= 1;
buf = gst_buffer_new ();
gst_buffer_append_memory (buf,
gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
/* don't need message anymore */
gst_rtsp_message_unset (message);
GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
channel);
if (src->need_activate) {
gchar *stream_id;
GstEvent *event;
GChecksum *cs;
gchar *uri;
GList *streams;
guint group_id = gst_util_group_id_next ();
/* generate an SHA256 sum of the URI */
cs = g_checksum_new (G_CHECKSUM_SHA256);
uri = src->conninfo.location;
g_checksum_update (cs, (const guchar *) uri, strlen (uri));
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
stream_id =
g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
event = gst_event_new_stream_start (stream_id);
gst_event_set_group_id (event, group_id);
g_free (stream_id);
gst_rtspsrc_stream_push_event (src, ostream, event);
}
g_checksum_free (cs);
gst_rtspsrc_activate_streams (src);
src->need_activate = FALSE;
}
if ((event = src->start_segment) != NULL) {
src->start_segment = NULL;
gst_rtspsrc_push_event (src, event);
}
if (src->base_time == -1) {
/* Take current running_time. This timestamp will be put on
* the first buffer of each stream because we are a live source and so we
* timestamp with the running_time. When we are dealing with TCP, we also
* only timestamp the first buffer (using the DISCONT flag) because a server
* typically bursts data, for which we don't want to compensate by speeding
* up the media. The other timestamps will be interpollated from this one
* using the RTP timestamps. */
GST_OBJECT_LOCK (src);
if (GST_ELEMENT_CLOCK (src)) {
GstClockTime now;
GstClockTime base_time;
now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
base_time = GST_ELEMENT_CAST (src)->base_time;
src->base_time = now - base_time;
GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
}
GST_OBJECT_UNLOCK (src);
}
if (stream->discont && !is_rtcp) {
/* mark first RTP buffer as discont */
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
stream->discont = FALSE;
/* first buffer gets the timestamp, other buffers are not timestamped and
* their presentation time will be interpollated from the rtp timestamps. */
GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (src->base_time));
GST_BUFFER_TIMESTAMP (buf) = src->base_time;
}
/* chain to the peer pad */
if (GST_PAD_IS_SINK (outpad))
ret = gst_pad_chain (outpad, buf);
else
ret = gst_pad_push (outpad, buf);
if (!is_rtcp) {
/* combine all stream flows for the data transport */
ret = gst_rtspsrc_combine_flows (src, stream, ret);
}
return ret;
/* ERRORS */
unknown_stream:
{
GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
gst_rtsp_message_unset (message);
return GST_FLOW_OK;
}
invalid_length:
{
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Short message received, ignoring."));
gst_rtsp_message_unset (message);
return GST_FLOW_OK;
}
}
static GstFlowReturn
gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
{
GstRTSPMessage message = { 0 };
GstRTSPResult res;
GstFlowReturn ret = GST_FLOW_OK;
GTimeVal tv_timeout;
while (TRUE) {
/* get the next timeout interval */
gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
/* see if the timeout period expired */
if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
/* send keep-alive, only act on interrupt, a warning will be posted for
* other errors. */
if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
goto interrupt;
/* get new timeout */
gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
}
GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
tv_timeout.tv_sec, tv_timeout.tv_usec);
/* protect the connection with the connection lock so that we can see when
* we are finished doing server communication */
res =
gst_rtspsrc_connection_receive (src, src->conninfo.connection,
&message, src->ptcp_timeout);
switch (res) {
case GST_RTSP_OK:
GST_DEBUG_OBJECT (src, "we received a server message");
break;
case GST_RTSP_EINTR:
/* we got interrupted this means we need to stop */
goto interrupt;
case GST_RTSP_ETIMEOUT:
/* no reply, send keep alive */
GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
goto interrupt;
continue;
case GST_RTSP_EEOF:
/* go EOS when the server closed the connection */
goto server_eof;
default:
goto receive_error;
}
switch (message.type) {
case GST_RTSP_MESSAGE_REQUEST:
/* server sends us a request message, handle it */
res =
gst_rtspsrc_handle_request (src, src->conninfo.connection,
&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 messages */
GST_DEBUG_OBJECT (src, "ignoring response message");
if (src->debug)
gst_rtsp_message_dump (&message);
break;
case GST_RTSP_MESSAGE_DATA:
GST_DEBUG_OBJECT (src, "got data message");
ret = gst_rtspsrc_handle_data (src, &message);
if (ret != GST_FLOW_OK)
goto handle_data_failed;
break;
default:
GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
message.type);
break;
}
}
g_assert_not_reached ();
/* ERRORS */
server_eof:
{
GST_DEBUG_OBJECT (src, "we got an eof from the server");
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("The server closed the connection."));
src->conninfo.connected = FALSE;
gst_rtsp_message_unset (&message);
return GST_FLOW_EOS;
}
interrupt:
{
gst_rtsp_message_unset (&message);
GST_DEBUG_OBJECT (src, "got interrupted");
return GST_FLOW_FLUSHING;
}
receive_error:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
g_free (str);
gst_rtsp_message_unset (&message);
return GST_FLOW_ERROR;
}
handle_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not handle server message. (%s)", str));
g_free (str);
gst_rtsp_message_unset (&message);
return GST_FLOW_ERROR;
}
handle_data_failed:
{
GST_DEBUG_OBJECT (src, "could no handle data message");
return ret;
}
}
static GstFlowReturn
gst_rtspsrc_loop_udp (GstRTSPSrc * src)
{
GstRTSPResult res;
GstRTSPMessage message = { 0 };
gint retry = 0;
while (TRUE) {
GTimeVal tv_timeout;
/* get the next timeout interval */
gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
GST_DEBUG_OBJECT (src, "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_rtspsrc_connection_receive (src, src->conninfo.connection,
&message, &tv_timeout);
switch (res) {
case GST_RTSP_OK:
GST_DEBUG_OBJECT (src, "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 (src, "timeout, sending keep-alive");
if ((res = gst_rtspsrc_send_keep_alive (src)) == 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 (src, RESOURCE, READ, (NULL),
("The server closed the connection."));
if (src->udp_reconnect) {
if ((res =
gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
goto connect_error;
} else {
goto server_eof;
}
continue;
case GST_RTSP_ENET:
GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
default:
GST_ELEMENT_WARNING (src, 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_rtspsrc_handle_request (src, src->conninfo.connection,
&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 (src, "ignoring response message");
if (src->debug)
gst_rtsp_message_dump (&message);
if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
GST_DEBUG_OBJECT (src, "so retrying keep-alive");
if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
goto interrupt;
}
} else {
retry = 0;
}
break;
case GST_RTSP_MESSAGE_DATA:
/* we ignore response and data messages */
GST_DEBUG_OBJECT (src, "ignoring data message");
break;
default:
GST_WARNING_OBJECT (src, "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 (src, "got interrupted");
return GST_FLOW_FLUSHING;
}
connect_error:
{
gchar *str = gst_rtsp_strresult (res);
GstFlowReturn ret;
src->conninfo.connected = FALSE;
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (src, 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 (src, 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 (src, 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 (src, "we got an eof from the server");
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("The server closed the connection."));
src->conninfo.connected = FALSE;
gst_rtsp_message_unset (&message);
return GST_FLOW_EOS;
}
}
static GstRTSPResult
gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
{
GstRTSPResult res = GST_RTSP_OK;
gboolean restart;
GST_DEBUG_OBJECT (src, "doing reconnect");
GST_OBJECT_LOCK (src);
/* only restart when the pads were not yet activated, else we were
* streaming over UDP */
restart = src->need_activate;
GST_OBJECT_UNLOCK (src);
/* no need to restart, we're done */
if (!restart)
goto done;
/* we can try only TCP now */
src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
/* close and cleanup our state */
if ((res = gst_rtspsrc_close (src, 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 (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->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 (src, 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 (src->udp_timeout / 1000000.0)));
/* open new connection using tcp */
if (gst_rtspsrc_open (src, async) < 0)
goto open_failed;
/* start playback */
if (gst_rtspsrc_play (src, &src->segment, async) < 0)
goto play_failed;
done:
return res;
/* ERRORS */
no_protocols:
{
src->cur_protocols = 0;
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (src, 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 (src->udp_timeout / 1000000.0)));
return GST_RTSP_ERROR;
}
open_failed:
{
GST_DEBUG_OBJECT (src, "open failed");
return GST_RTSP_OK;
}
play_failed:
{
GST_DEBUG_OBJECT (src, "play failed");
return GST_RTSP_OK;
}
}
static void
gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
break;
case CMD_PLAY:
GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
break;
default:
break;
}
}
static void
gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
break;
case CMD_PLAY:
GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
break;
default:
break;
}
}
static void
gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
break;
case CMD_PLAY:
GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
break;
default:
break;
}
}
static void
gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
{
switch (cmd) {
case CMD_OPEN:
GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
break;
case CMD_PLAY:
GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
break;
case CMD_PAUSE:
GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
break;
case CMD_CLOSE:
GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
break;
default:
break;
}
}
static void
gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
{
if (ret == GST_RTSP_OK)
gst_rtspsrc_loop_complete_cmd (src, cmd);
else if (ret == GST_RTSP_EINTR)
gst_rtspsrc_loop_cancel_cmd (src, cmd);
else
gst_rtspsrc_loop_error_cmd (src, cmd);
}
static gboolean
gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
{
gint old;
gboolean flushed = FALSE;
/* start new request */
gst_rtspsrc_loop_start_cmd (src, cmd);
GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
GST_OBJECT_LOCK (src);
old = src->pending_cmd;
if (old == CMD_RECONNECT) {
GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
cmd = CMD_RECONNECT;
}
if (old != CMD_WAIT) {
src->pending_cmd = CMD_WAIT;
GST_OBJECT_UNLOCK (src);
/* cancel previous request */
GST_DEBUG_OBJECT (src, "cancel previous request %d", old);
gst_rtspsrc_loop_cancel_cmd (src, old);
GST_OBJECT_LOCK (src);
}
src->pending_cmd = cmd;
/* interrupt if allowed */
if (src->busy_cmd & mask) {
GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
gst_rtspsrc_connection_flush (src, TRUE);
flushed = TRUE;
} else {
GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
}
if (src->task)
gst_task_start (src->task);
GST_OBJECT_UNLOCK (src);
return flushed;
}
static gboolean
gst_rtspsrc_loop (GstRTSPSrc * src)
{
GstFlowReturn ret;
if (!src->conninfo.connection || !src->conninfo.connected)
goto no_connection;
if (src->interleaved)
ret = gst_rtspsrc_loop_interleaved (src);
else
ret = gst_rtspsrc_loop_udp (src);
if (ret != GST_FLOW_OK)
goto pause;
return TRUE;
/* ERRORS */
no_connection:
{
GST_WARNING_OBJECT (src, "we are not connected");
ret = GST_FLOW_FLUSHING;
goto pause;
}
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
src->running = FALSE;
if (ret == GST_FLOW_EOS) {
/* perform EOS logic */
if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (src),
gst_message_new_segment_done (GST_OBJECT_CAST (src),
src->segment.format, src->segment.position));
gst_rtspsrc_push_event (src,
gst_event_new_segment_done (src->segment.format,
src->segment.position));
} else {
gst_rtspsrc_push_event (src, gst_event_new_eos ());
}
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
/* for fatal errors we post an error message, post the error before the
* EOS so the app knows about the error first. */
GST_ELEMENT_ERROR (src, STREAM, FAILED,
("Internal data flow error."),
("streaming task paused, reason %s (%d)", reason, ret));
gst_rtspsrc_push_event (src, gst_event_new_eos ());
}
gst_rtspsrc_loop_send_cmd (src, 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
static const gchar *
gst_rtspsrc_skip_lws (const gchar * s)
{
while (g_ascii_isspace (*s))
s++;
return s;
}
static const gchar *
gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
{
while (s > start && g_ascii_isspace (*(s - 1)))
s--;
return s;
}
static const gchar *
gst_rtspsrc_skip_commas (const gchar * s)
{
/* The grammar allows for multiple commas */
while (g_ascii_isspace (*s) || *s == ',')
s++;
return s;
}
static const gchar *
gst_rtspsrc_skip_item (const gchar * s)
{
gboolean quoted = FALSE;
const gchar *start = s;
/* A list item ends at the last non-whitespace character
* before a comma which is not inside a quoted-string. Or at
* the end of the string.
*/
while (*s) {
if (*s == '"')
quoted = !quoted;
else if (quoted) {
if (*s == '\\' && *(s + 1))
s++;
} else {
if (*s == ',')
break;
}
s++;
}
return gst_rtspsrc_unskip_lws (s, start);
}
static void
gst_rtsp_decode_quoted_string (gchar * quoted_string)
{
gchar *src, *dst;
src = quoted_string + 1;
dst = quoted_string;
while (*src && *src != '"') {
if (*src == '\\' && *(src + 1))
src++;
*dst++ = *src++;
}
*dst = '\0';
}
/* Extract the authentication tokens that the server provided for each method
* into an array of structures and give those to the connection object.
*/
static void
gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
const gchar * header, gboolean * stale)
{
GSList *list = NULL, *iter;
const gchar *end;
gchar *item, *eq, *name_end, *value;
g_return_if_fail (stale != NULL);
gst_rtsp_connection_clear_auth_params (conn);
*stale = FALSE;
/* Parse a header whose content is described by RFC2616 as
* "#something", where "something" does not itself contain commas,
* except as part of quoted-strings, into a list of allocated strings.
*/
header = gst_rtspsrc_skip_commas (header);
while (*header) {
end = gst_rtspsrc_skip_item (header);
list = g_slist_prepend (list, g_strndup (header, end - header));
header = gst_rtspsrc_skip_commas (end);
}
if (!list)
return;
list = g_slist_reverse (list);
for (iter = list; iter; iter = iter->next) {
item = iter->data;
eq = strchr (item, '=');
if (eq) {
name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
if (name_end == item) {
/* That's no good... */
g_free (item);
continue;
}
*name_end = '\0';
value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
if (*value == '"')
gst_rtsp_decode_quoted_string (value);
} else
value = NULL;
if (item && (strcmp (item, "stale") == 0) &&
value && (strcmp (value, "TRUE") == 0))
*stale = TRUE;
gst_rtsp_connection_set_auth_param (conn, item, value);
g_free (item);
}
g_slist_free (list);
}
/* 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_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
GstRTSPConnection * conn, gboolean * stale)
{
gchar *start;
g_return_if_fail (hdr != NULL);
g_return_if_fail (methods != NULL);
g_return_if_fail (stale != NULL);
/* Skip whitespace at the start of the string */
for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
if (g_ascii_strncasecmp (start, "basic", 5) == 0)
*methods |= GST_RTSP_AUTH_BASIC;
else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
*methods |= GST_RTSP_AUTH_DIGEST;
gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
}
}
/**
* gst_rtspsrc_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_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
{
gchar *user = NULL;
gchar *pass = NULL;
GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
GstRTSPAuthMethod method;
GstRTSPResult auth_result;
GstRTSPUrl *url;
GstRTSPConnection *conn;
gchar *hdr;
gboolean stale = FALSE;
conn = src->conninfo.connection;
/* Identify the available auth methods and see if any are supported */
if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
&hdr, 0) == GST_RTSP_OK) {
gst_rtspsrc_parse_auth_hdr (hdr, &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)
src->tried_url_auth = FALSE;
url = gst_rtsp_connection_get_url (conn);
/* Do we have username and password available? */
if (url != NULL && !src->tried_url_auth && url->user != NULL
&& url->passwd != NULL) {
user = url->user;
pass = url->passwd;
src->tried_url_auth = TRUE;
GST_DEBUG_OBJECT (src,
"Attempting authentication using credentials from the URL");
} else {
user = src->user_id;
pass = src->user_pw;
GST_DEBUG_OBJECT (src,
"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 (src, "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 (src, 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_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
GstRTSPMessage * request, GstRTSPMessage * response,
GstRTSPStatusCode * code)
{
GstRTSPResult res;
GstRTSPStatusCode thecode;
gchar *content_base = NULL;
gint try = 0;
again:
if (!src->short_header)
gst_rtsp_ext_list_before_send (src->extensions, request);
GST_DEBUG_OBJECT (src, "sending message");
if (src->debug)
gst_rtsp_message_dump (request);
res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
if (res < 0)
goto send_error;
gst_rtsp_connection_reset_timeout (conn);
next:
res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
if (res < 0)
goto receive_error;
if (src->debug)
gst_rtsp_message_dump (response);
switch (response->type) {
case GST_RTSP_MESSAGE_REQUEST:
res = gst_rtspsrc_handle_request (src, conn, response);
if (res == GST_RTSP_EEOF)
goto server_eof;
else if (res < 0)
goto handle_request_failed;
goto next;
case GST_RTSP_MESSAGE_RESPONSE:
/* ok, a response is good */
GST_DEBUG_OBJECT (src, "received response message");
break;
case GST_RTSP_MESSAGE_DATA:
/* get next response */
GST_DEBUG_OBJECT (src, "handle data response message");
gst_rtspsrc_handle_data (src, response);
goto next;
default:
GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
response->type);
goto next;
}
thecode = response->type_data.response.code;
GST_DEBUG_OBJECT (src, "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 (src->content_base);
src->content_base = g_strdup (content_base);
}
gst_rtsp_ext_list_after_send (src->extensions, request, response);
return GST_RTSP_OK;
/* ERRORS */
send_error:
{
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (src, "send interrupted");
}
g_free (str);
return res;
}
receive_error:
{
switch (res) {
case GST_RTSP_EEOF:
GST_WARNING_OBJECT (src, "server closed connection");
if ((try == 0) && !src->interleaved && src->udp_reconnect) {
try++;
/* if reconnect succeeds, try again */
if ((res =
gst_rtsp_conninfo_reconnect (src, &src->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 (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
} else {
GST_WARNING_OBJECT (src, "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 (src, "we got an eof from the server");
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("The server closed the connection."));
gst_rtsp_message_unset (response);
return res;
}
}
/**
* gst_rtspsrc_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_rtspsrc_setup_auth and retry
* the request.
*
* Returns: #GST_RTSP_OK if the processing was successful.
*/
static GstRTSPResult
gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
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_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
goto error;
switch (int_code) {
case GST_RTSP_STS_UNAUTHORIZED:
if (gst_rtspsrc_setup_auth (src, 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 (src, "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 (src, RESOURCE, NOT_FOUND, (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 (src, "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 (src, "redirection to %s", new_location);
/* save current transports */
if (src->conninfo.url)
transports = src->conninfo.url->transports;
else
transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
/* set old transports */
if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
src->conninfo.url->transports = transports;
src->need_redirect = TRUE;
src->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 (src, "got NOT IMPLEMENTED, disable method %s",
gst_rtsp_method_as_text (method));
src->methods &= ~method;
res = GST_RTSP_OK;
break;
default:
GST_ELEMENT_ERROR (src, 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;
}
}
static GstRTSPResult
gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
GstRTSPMessage * response, GstRTSPSrc * src)
{
return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
NULL);
}
/* 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_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
{
GstRTSPHeaderField field;
gchar *respoptions;
gint indx = 0;
/* reset supported methods */
src->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;
src->methods |= gst_rtsp_options_from_text (respoptions);
indx++;
}
if (src->methods == 0) {
/* neither Allow nor Public are required, assume the server supports
* at least DESCRIBE, SETUP, we always assume it supports PLAY as
* well. */
GST_DEBUG_OBJECT (src, "could not get OPTIONS");
src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
}
/* always assume PLAY, FIXME, extensions should be able to override
* this */
src->methods |= GST_RTSP_PLAY;
/* also assume it will support Range */
src->seekable = TRUE;
/* we need describe and setup */
if (!(src->methods & GST_RTSP_DESCRIBE))
goto no_describe;
if (!(src->methods & GST_RTSP_SETUP))
goto no_setup;
return TRUE;
/* ERRORS */
no_describe:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Server does not support DESCRIBE."));
return FALSE;
}
no_setup:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Server does not support SETUP."));
return FALSE;
}
}
/* masks to be kept in sync with the hardcoded protocol order of preference
* in code below */
static guint protocol_masks[] = {
GST_RTSP_LOWER_TRANS_UDP,
GST_RTSP_LOWER_TRANS_UDP_MCAST,
GST_RTSP_LOWER_TRANS_TCP,
0
};
static GstRTSPResult
gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
GstRTSPLowerTrans protocols, gchar ** transports)
{
GstRTSPResult res;
GString *result;
gboolean add_udp_str;
*transports = NULL;
res =
gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
if (res < 0)
goto failed;
GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
/* extension listed transports, use those */
if (*transports != NULL)
return GST_RTSP_OK;
/* it's the default */
add_udp_str = FALSE;
/* the default RTSP transports */
result = g_string_new ("");
if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
GST_DEBUG_OBJECT (src, "adding UDP unicast");
g_string_append (result, "RTP/AVP");
if (add_udp_str)
g_string_append (result, "/UDP");
g_string_append (result, ";unicast;client_port=%%u1-%%u2");
} else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
GST_DEBUG_OBJECT (src, "adding UDP multicast");
/* we don't have to allocate any UDP ports yet, if the selected transport
* turns out to be multicast we can create them and join the multicast
* group indicated in the transport reply */
if (result->len > 0)
g_string_append (result, ",");
g_string_append (result, "RTP/AVP");
if (add_udp_str)
g_string_append (result, "/UDP");
g_string_append (result, ";multicast");
if (src->next_port_num != 0) {
if (src->client_port_range.max > 0 &&
src->next_port_num >= src->client_port_range.max)
goto no_ports;
g_string_append_printf (result, ";client_port=%d-%d",
src->next_port_num, src->next_port_num + 1);
}
} else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
GST_DEBUG_OBJECT (src, "adding TCP");
if (result->len > 0)
g_string_append (result, ",");
g_string_append (result, "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2");
}
*transports = g_string_free (result, FALSE);
GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
return GST_RTSP_OK;
/* ERRORS */
failed:
{
GST_ERROR ("extension gave error %d", res);
return res;
}
no_ports:
{
GST_ERROR ("no more ports available");
return GST_RTSP_ERROR;
}
}
static GstRTSPResult
gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
gint orig_rtpport, gint orig_rtcpport)
{
GstRTSPSrc *src;
gint nr_udp, nr_int;
gchar *next, *p;
gint rtpport = 0, rtcpport = 0;
GString *str;
src = stream->parent;
/* find number of placeholders first */
if (strstr (*transports, "%%i2"))
nr_int = 2;
else if (strstr (*transports, "%%i1"))
nr_int = 1;
else
nr_int = 0;
if (strstr (*transports, "%%u2"))
nr_udp = 2;
else if (strstr (*transports, "%%u1"))
nr_udp = 1;
else
nr_udp = 0;
if (nr_udp == 0 && nr_int == 0)
goto done;
if (nr_udp > 0) {
if (!orig_rtpport || !orig_rtcpport) {
if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
goto failed;
} else {
rtpport = orig_rtpport;
rtcpport = orig_rtcpport;
}
}
str = g_string_new ("");
p = *transports;
while ((next = strstr (p, "%%"))) {
g_string_append_len (str, p, next - p);
if (next[2] == 'u') {
if (next[3] == '1')
g_string_append_printf (str, "%d", rtpport);
else if (next[3] == '2')
g_string_append_printf (str, "%d", rtcpport);
}
if (next[2] == 'i') {
if (next[3] == '1')
g_string_append_printf (str, "%d", src->free_channel);
else if (next[3] == '2')
g_string_append_printf (str, "%d", src->free_channel + 1);
}
p = next + 4;
}
/* append final part */
g_string_append (str, p);
g_free (*transports);
*transports = g_string_free (str, FALSE);
done:
return GST_RTSP_OK;
/* ERRORS */
failed:
{
GST_ERROR ("failed to allocate udp ports");
return GST_RTSP_ERROR;
}
}
static gboolean
gst_rtspsrc_stream_is_real_media (GstRTSPStream * stream)
{
gboolean res = FALSE;
if (stream->caps) {
GstStructure *s;
const gchar *enc = NULL;
s = gst_caps_get_structure (stream->caps, 0);
if ((enc = gst_structure_get_string (s, "encoding-name"))) {
res = (strstr (enc, "-REAL") != NULL);
}
}
return res;
}
/* Perform the SETUP request for all the streams.
*
* We ask the server for a specific transport, which initially includes all the
* ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
* two local UDP ports that we send to the server.
*
* Once the server replied with a transport, we configure the other streams
* with the same transport.
*
* This function will also configure the stream for the selected transport,
* which basically means creating the pipeline.
*/
static GstRTSPResult
gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
{
GList *walk;
GstRTSPResult res = GST_RTSP_ERROR;
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPStream *stream = NULL;
GstRTSPLowerTrans protocols;
GstRTSPStatusCode code;
gboolean unsupported_real = FALSE;
gint rtpport, rtcpport;
GstRTSPUrl *url;
gchar *hval;
if (src->conninfo.connection) {
url = gst_rtsp_connection_get_url (src->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 & src->cur_protocols;
} else {
url = NULL;
protocols = src->cur_protocols;
}
if (protocols == 0)
goto no_protocols;
/* reset some state */
src->free_channel = 0;
src->interleaved = FALSE;
src->need_activate = FALSE;
/* keep track of next port number, 0 is random */
src->next_port_num = src->client_port_range.min;
rtpport = rtcpport = 0;
if (G_UNLIKELY (src->streams == NULL))
goto no_streams;
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPConnection *conn;
gchar *transports;
gint retry = 0;
guint mask = 0;
gboolean selected;
stream = (GstRTSPStream *) walk->data;
/* see if we need to configure this stream */
if (!gst_rtsp_ext_list_configure_stream (src->extensions, stream->caps)) {
GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
stream);
stream->disabled = TRUE;
continue;
}
g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
stream->id, stream->caps, &selected);
if (!selected) {
GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
stream->disabled = TRUE;
continue;
}
stream->disabled = FALSE;
/* merge/overwrite global caps */
if (stream->caps) {
guint j, num;
GstStructure *s;
s = gst_caps_get_structure (stream->caps, 0);
num = gst_structure_n_fields (src->props);
for (j = 0; j < num; j++) {
const gchar *name;
const GValue *val;
name = gst_structure_nth_field_name (src->props, j);
val = gst_structure_get_value (src->props, name);
gst_structure_set_value (s, name, val);
GST_DEBUG_OBJECT (src, "copied %s", name);
}
}
/* skip setup if we have no URL for it */
if (stream->conninfo.location == NULL) {
GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
continue;
}
if (src->conninfo.connection == NULL) {
if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
continue;
}
conn = stream->conninfo.connection;
} else {
conn = src->conninfo.connection;
}
GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
stream->conninfo.location);
/* if we have a multicast connection, only suggest multicast from now on */
if (stream->is_multicast)
protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
next_protocol:
/* first selectable protocol */
while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
mask++;
if (!protocol_masks[mask])
goto no_protocols;
retry:
GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
protocol_masks[mask]);
/* create a string with first transport in line */
transports = NULL;
res = gst_rtspsrc_create_transports_string (src,
protocols & protocol_masks[mask], &transports);
if (res < 0 || transports == NULL)
goto setup_transport_failed;
if (strlen (transports) == 0) {
g_free (transports);
GST_DEBUG_OBJECT (src, "no transports found");
mask++;
goto next_protocol;
}
GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
/* replace placeholders with real values, this function will optionally
* allocate UDP ports and other info needed to execute the setup request */
res = gst_rtspsrc_prepare_transports (stream, &transports,
retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
if (res < 0) {
g_free (transports);
goto setup_transport_failed;
}
GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
/* create SETUP request */
res =
gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
stream->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);
/* if the user wants a non default RTP packet size we add the blocksize
* parameter */
if (src->rtp_blocksize > 0) {
hval = g_strdup_printf ("%d", src->rtp_blocksize);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
}
if (async)
GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
stream->id));
/* handle the code ourselves */
if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 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);
/* cleanup of leftover transport */
gst_rtspsrc_stream_free_udp (stream);
/* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
* we might be in this case */
if (stream->container && rtpport && rtcpport && !retry) {
GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
rtpport, rtcpport);
retry++;
goto retry;
}
/* this transport did not go down well, but we may have others to try
* that we did not send yet, try those and only give up then
* but not without checking for lost cause/extension so we can
* post a nicer/more useful error message later */
if (!unsupported_real)
unsupported_real = gst_rtspsrc_stream_is_real_media (stream);
/* select next available protocol, give up on this stream if none */
mask++;
while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
mask++;
if (!protocol_masks[mask] || unsupported_real)
continue;
else
goto retry;
default:
/* cleanup of leftover transport and move to the next stream */
gst_rtspsrc_stream_free_udp (stream);
goto response_error;
}
/* parse response transport */
{
gchar *resptrans = NULL;
GstRTSPTransport transport = { 0 };
gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
&resptrans, 0);
if (!resptrans) {
gst_rtspsrc_stream_free_udp (stream);
goto no_transport;
}
/* parse transport, go to next stream on parse error */
if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
GST_WARNING_OBJECT (src, "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 (src, "stream %p as TCP interleaved", stream);
protocols = GST_RTSP_LOWER_TRANS_TCP;
src->interleaved = TRUE;
/* update free channels */
src->free_channel =
MAX (transport.interleaved.min, src->free_channel);
src->free_channel =
MAX (transport.interleaved.max, src->free_channel);
src->free_channel++;
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
/* only allow multicast for other streams */
GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
/* if the server selected our ports, increment our counters so that
* we select a new port later */
if (src->next_port_num == transport.port.min &&
src->next_port_num + 1 == transport.port.max) {
src->next_port_num += 2;
}
break;
case GST_RTSP_LOWER_TRANS_UDP:
/* only allow unicast for other streams */
GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
protocols = GST_RTSP_LOWER_TRANS_UDP;
break;
default:
GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
transport.lower_transport);
break;
}
if (!stream->container || (!src->interleaved && !retry)) {
/* now configure the stream with the selected transport */
if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
GST_DEBUG_OBJECT (src,
"could not configure stream %p transport, skipping stream",
stream);
goto next;
} else if (stream->udpsrc[0] && stream->udpsrc[1]) {
/* retain the first allocated UDP port pair */
g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
}
}
/* we need to activate at least one streams when we detect activity */
src->need_activate = TRUE;
next:
/* clean up our transport struct */
gst_rtsp_transport_init (&transport);
/* clean up used RTSP messages */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
}
}
/* store the transport protocol that was configured */
src->cur_protocols = protocols;
gst_rtsp_ext_list_stream_select (src->extensions, url);
/* if there is nothing to activate, error out */
if (!src->need_activate)
goto nothing_to_activate;
return res;
/* ERRORS */
no_protocols:
{
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not connect to server, no protocols left"));
return GST_RTSP_ERROR;
}
no_streams:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("SDP contains no streams"));
return GST_RTSP_ERROR;
}
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto cleanup_error;
}
setup_transport_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Could not setup transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
response_error:
{
const gchar *str = gst_rtsp_status_as_text (code);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Error (%d): %s", code, GST_STR_NULL (str)));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
send_error:
{
gchar *str = gst_rtsp_strresult (res);
if (res != GST_RTSP_EINTR) {
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (src, "send interrupted");
}
g_free (str);
goto cleanup_error;
}
no_transport:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server did not select transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
nothing_to_activate:
{
/* none of the available error codes is really right .. */
if (unsupported_real) {
GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
(_("No supported stream was found. You might need to install a "
"GStreamer RTSP extension plugin for Real media streams.")),
(NULL));
} else {
GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
(_("No supported stream was found. You might need to allow "
"more transport protocols or may otherwise be missing "
"the right GStreamer RTSP extension plugin.")), (NULL));
}
return GST_RTSP_ERROR;
}
cleanup_error:
{
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
return res;
}
}
static gboolean
gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
GstSegment * segment)
{
gint64 seconds;
GstRTSPTimeRange *therange;
if (src->range)
gst_rtsp_range_free (src->range);
if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
GST_DEBUG_OBJECT (src, "parsed range %s", range);
src->range = therange;
} else {
GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
src->range = NULL;
gst_segment_init (segment, GST_FORMAT_TIME);
return FALSE;
}
GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
therange->min.type, therange->min.seconds, therange->max.type,
therange->max.seconds);
if (therange->min.type == GST_RTSP_TIME_NOW)
seconds = 0;
else if (therange->min.type == GST_RTSP_TIME_END)
seconds = 0;
else
seconds = therange->min.seconds * GST_SECOND;
GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
GST_TIME_ARGS (seconds));
/* we need to start playback without clipping from the position reported by
* the server */
segment->start = seconds;
segment->position = seconds;
if (therange->max.type == GST_RTSP_TIME_NOW)
seconds = -1;
else if (therange->max.type == GST_RTSP_TIME_END)
seconds = -1;
else
seconds = therange->max.seconds * GST_SECOND;
GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
GST_TIME_ARGS (seconds));
/* live (WMS) server might send overflowed large max as its idea of infinity,
* compensate to prevent problems later on */
if (seconds != -1 && seconds < 0) {
seconds = -1;
GST_DEBUG_OBJECT (src, "insane range, set to NONE");
}
/* live (WMS) might send min == max, which is not worth recording */
if (segment->duration == -1 && seconds == segment->start)
seconds = -1;
/* don't change duration with unknown value, we might have a valid value
* there that we want to keep. */
if (seconds != -1)
segment->duration = seconds;
return TRUE;
}
/* Parse clock profived by the server with following syntax:
*
* "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
*/
static gboolean
gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
{
gboolean res = FALSE;
if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
gchar **fields = NULL, **parts = NULL;
gchar *remote_ip, *str;
gint port;
GstClockTime base_time;
GstClock *netclock;
fields = g_strsplit (gstclock, " ", 0);
/* wrapped clock, not very interesting for now */
if (fields[1] == NULL)
goto cleanup;
/* remote IP address and port */
if ((str = fields[2]) == NULL)
goto cleanup;
parts = g_strsplit (str, ":", 0);
if ((remote_ip = parts[0]) == NULL)
goto cleanup;
if ((str = parts[1]) == NULL)
goto cleanup;
port = atoi (str);
if (port == 0)
goto cleanup;
/* base-time */
if ((str = fields[3]) == NULL)
goto cleanup;
base_time = g_ascii_strtoull (str, NULL, 10);
netclock =
gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
base_time);
if (src->provided_clock)
gst_object_unref (src->provided_clock);
src->provided_clock = netclock;
gst_element_post_message (GST_ELEMENT_CAST (src),
gst_message_new_clock_provide (GST_OBJECT_CAST (src),
src->provided_clock, TRUE));
res = TRUE;
cleanup:
g_strfreev (fields);
g_strfreev (parts);
}
return res;
}
/* must be called with the RTSP state lock */
static GstRTSPResult
gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
gboolean async)
{
GstRTSPResult res;
gint i, n_streams;
/* prepare global stream caps properties */
if (src->props)
gst_structure_remove_all_fields (src->props);
else
src->props = gst_structure_new_empty ("RTSPProperties");
if (src->debug)
gst_sdp_message_dump (sdp);
gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
/* let the app inspect and change the SDP */
g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_ON_SDP], 0, sdp);
gst_segment_init (&src->segment, GST_FORMAT_TIME);
/* parse range for duration reporting. */
{
const gchar *range;
for (i = 0;; i++) {
range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
if (range == NULL)
break;
/* keep track of the range and configure it in the segment */
if (gst_rtspsrc_parse_range (src, range, &src->segment))
break;
}
}
/* parse clock information. This is GStreamer specific, a server can tell the
* client what clock it is using and wrap that in a network clock. The
* advantage of that is that we can slave to it. */
{
const gchar *gstclock;
for (i = 0;; i++) {
gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
if (gstclock == NULL)
break;
/* parse the clock and expose it in the provide_clock method */
if (gst_rtspsrc_parse_gst_clock (src, gstclock))
break;
}
}
/* try to find a global control attribute. Note that a '*' means that we should
* do aggregate control with the current url (so we don't do anything and
* leave the current connection as is) */
{
const gchar *control;
for (i = 0;; i++) {
control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
if (control == NULL)
break;
/* only take fully qualified urls */
if (g_str_has_prefix (control, "rtsp://"))
break;
}
if (control) {
g_free (src->conninfo.location);
src->conninfo.location = g_strdup (control);
/* make a connection for this, if there was a connection already, nothing
* happens. */
if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
GST_ERROR_OBJECT (src, "could not connect");
}
}
/* we need to keep the control url separate from the connection url because
* the rules for constructing the media control url need it */
g_free (src->control);
src->control = g_strdup (control);
}
/* create streams */
n_streams = gst_sdp_message_medias_len (sdp);
for (i = 0; i < n_streams; i++) {
gst_rtspsrc_create_stream (src, sdp, i);
}
src->state = GST_RTSP_STATE_INIT;
/* setup streams */
if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
goto setup_failed;
/* reset our state */
src->need_range = TRUE;
src->skip = FALSE;
src->state = GST_RTSP_STATE_READY;
return res;
/* ERRORS */
setup_failed:
{
GST_ERROR_OBJECT (src, "setup failed");
gst_rtspsrc_cleanup (src);
return res;
}
}
static GstRTSPResult
gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
gboolean async)
{
GstRTSPResult res;
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
guint8 *data;
guint size;
gchar *respcont = NULL;
restart:
src->need_redirect = FALSE;
/* can't continue without a valid url */
if (G_UNLIKELY (src->conninfo.url == NULL)) {
res = GST_RTSP_EINVAL;
goto no_url;
}
src->tried_url_auth = FALSE;
if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
goto connect_failed;
/* create OPTIONS */
GST_DEBUG_OBJECT (src, "create options...");
res =
gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
src->conninfo.url_str);
if (res < 0)
goto create_request_failed;
/* send OPTIONS */
GST_DEBUG_OBJECT (src, "send options...");
if (async)
GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
if ((res =
gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
NULL)) < 0)
goto send_error;
/* parse OPTIONS */
if (!gst_rtspsrc_parse_methods (src, &response))
goto methods_error;
/* create DESCRIBE */
GST_DEBUG_OBJECT (src, "create describe...");
res =
gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
src->conninfo.url_str);
if (res < 0)
goto create_request_failed;
/* we only accept SDP for now */
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
"application/sdp");
/* send DESCRIBE */
GST_DEBUG_OBJECT (src, "send describe...");
if (async)
GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
if ((res =
gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
NULL)) < 0)
goto send_error;
/* we only perform redirect for the describe, currently */
if (src->need_redirect) {
/* close connection, we don't have to send a TEARDOWN yet, ignore the
* result. */
gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
/* and now retry */
goto restart;
}
/* it could be that the DESCRIBE method was not implemented */
if (!src->methods & GST_RTSP_DESCRIBE)
goto no_describe;
/* check if reply is SDP */
gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
0);
/* could not be set but since the request returned OK, we assume it
* was SDP, else check it. */
if (respcont) {
if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
goto wrong_content_type;
}
/* get message body and parse as SDP */
gst_rtsp_message_get_body (&response, &data, &size);
if (data == NULL || size == 0)
goto no_describe;
GST_DEBUG_OBJECT (src, "parse SDP...");
gst_sdp_message_new (sdp);
gst_sdp_message_parse_buffer (data, size, *sdp);
/* clean up any messages */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
return res;
/* ERRORS */
no_url:
{
GST_ELEMENT_ERROR (src, 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 (src, RESOURCE, OPEN_READ_WRITE, (NULL),
("Failed to connect. (%s)", str));
} else {
GST_WARNING_OBJECT (src, "connect interrupted");
}
g_free (str);
goto cleanup_error;
}
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (src, 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;
}
wrong_content_type:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server does not support SDP, got %s.", respcont));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
no_describe:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server can not provide an SDP."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
cleanup_error:
{
if (src->conninfo.connection) {
GST_DEBUG_OBJECT (src, "free connection");
gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
}
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
return res;
}
}
static GstRTSPResult
gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
{
GstRTSPResult ret;
src->methods =
GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
if (src->sdp == NULL) {
if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
goto no_sdp;
}
if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
goto open_failed;
done:
if (async)
gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
return ret;
/* ERRORS */
no_sdp:
{
GST_WARNING_OBJECT (src, "can't get sdp");
src->open_error = TRUE;
goto done;
}
open_failed:
{
GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
src->open_error = TRUE;
goto done;
}
}
static GstRTSPResult
gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
{
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPResult res = GST_RTSP_OK;
GList *walk;
const gchar *control;
GST_DEBUG_OBJECT (src, "TEARDOWN...");
gst_rtspsrc_set_state (src, GST_STATE_READY);
if (src->state < GST_RTSP_STATE_READY) {
GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
goto close;
}
if (only_close)
goto close;
/* construct a control url */
control = get_aggregate_control (src);
if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
goto not_supported;
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
const gchar *setup_url;
GstRTSPConnInfo *info;
/* 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 (src->conninfo.connection) {
info = &src->conninfo;
} else if (stream->conninfo.connection) {
info = &stream->conninfo;
} else {
continue;
}
if (!info->connected)
goto next;
/* do TEARDOWN */
res =
gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
if (res < 0)
goto create_request_failed;
if (async)
GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
if ((res =
gst_rtspsrc_send (src, info->connection, &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 (src, "closing connection...");
gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
}
/* cleanup */
gst_rtspsrc_cleanup (src);
src->state = GST_RTSP_STATE_INVALID;
if (async)
gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
return res;
/* ERRORS */
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (src, 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 (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
}
g_free (str);
goto close;
}
not_supported:
{
GST_DEBUG_OBJECT (src,
"TEARDOWN and PLAY not supported, can't do TEARDOWN");
goto close;
}
}
/* RTP-Info is of the format:
*
* url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
*
* rtptime corresponds to the timestamp for the NPT time given in the header
* seqbase corresponds to the next sequence number we received. This number
* indicates the first seqnum after the seek and should be used to discard
* packets that are from before the seek.
*/
static gboolean
gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
{
gchar **infos;
gint i, j;
GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
infos = g_strsplit (rtpinfo, ",", 0);
for (i = 0; infos[i]; i++) {
gchar **fields;
GstRTSPStream *stream;
gint32 seqbase;
gint64 timebase;
GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
/* init values, types of seqbase and timebase are bigger than needed so we
* can store -1 as uninitialized values */
stream = NULL;
seqbase = -1;
timebase = -1;
/* parse url, find stream for url.
* parse seq and rtptime. The seq number should be configured in the rtp
* depayloader or session manager to detect gaps. Same for the rtptime, it
* should be used to create an initial time newsegment. */
fields = g_strsplit (infos[i], ";", 0);
for (j = 0; fields[j]; j++) {
GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
/* remove leading whitespace */
fields[j] = g_strchug (fields[j]);
if (g_str_has_prefix (fields[j], "url=")) {
/* get the url and the stream */
stream =
find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
} else if (g_str_has_prefix (fields[j], "seq=")) {
seqbase = atoi (fields[j] + 4);
} else if (g_str_has_prefix (fields[j], "rtptime=")) {
timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
}
}
g_strfreev (fields);
/* now we need to store the values for the caps of the stream */
if (stream != NULL) {
GST_DEBUG_OBJECT (src,
"found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
stream, seqbase, timebase);
/* we have a stream, configure detected params */
stream->seqbase = seqbase;
stream->timebase = timebase;
}
}
g_strfreev (infos);
return TRUE;
}
static void
gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
{
guint64 interval;
GList *walk;
interval = strtoul (rtcp, NULL, 10);
GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
if (!interval)
return;
interval *= GST_MSECOND;
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
/* already (optionally) retrieved this when configuring manager */
if (stream->session) {
GObject *rtpsession = stream->session;
GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
rtpsession);
g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
}
}
/* now it happens that (Xenon) server sending this may also provide bogus
* RTCP SR sync data (i.e. with quite some jitter), so never mind those
* and just use RTP-Info to sync */
if (src->manager) {
GObjectClass *klass;
klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
if (g_object_class_find_property (klass, "rtcp-sync")) {
GST_DEBUG_OBJECT (src, "configuring rtp sync method");
g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
}
}
}
static gdouble
gst_rtspsrc_get_float (const gchar * dstr)
{
gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
/* canonicalise floating point string so we can handle float strings
* in the form "24.930" or "24,930" irrespective of the current locale */
g_strlcpy (s, dstr, sizeof (s));
g_strdelimit (s, ",", '.');
return g_ascii_strtod (s, NULL);
}
static gchar *
gen_range_header (GstRTSPSrc * src, GstSegment * segment)
{
gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
g_strlcpy (val_str, "now", sizeof (val_str));
} else {
if (segment->position == 0) {
g_strlcpy (val_str, "0", sizeof (val_str));
} else {
g_ascii_dtostr (val_str, sizeof (val_str),
((gdouble) segment->position) / GST_SECOND);
}
}
return g_strdup_printf ("npt=%s-", val_str);
}
static void
clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
{
stream->timebase = -1;
stream->seqbase = -1;
if (stream->caps) {
GstStructure *s;
stream->caps = gst_caps_make_writable (stream->caps);
s = gst_caps_get_structure (stream->caps, 0);
gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
}
}
static GstRTSPResult
gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
{
GstRTSPResult res = GST_RTSP_OK;
if (src->state < GST_RTSP_STATE_READY) {
res = GST_RTSP_ERROR;
if (src->open_error) {
GST_DEBUG_OBJECT (src, "the stream was in error");
goto done;
}
if (async)
gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
if ((res = gst_rtspsrc_open (src, async)) < 0) {
GST_DEBUG_OBJECT (src, "failed to open stream");
goto done;
}
}
done:
return res;
}
static GstRTSPResult
gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
{
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPResult res = GST_RTSP_OK;
GList *walk;
gchar *hval;
gint hval_idx;
const gchar *control;
GST_DEBUG_OBJECT (src, "PLAY...");
if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
goto open_failed;
if (!(src->methods & GST_RTSP_PLAY))
goto not_supported;
if (src->state == GST_RTSP_STATE_PLAYING)
goto was_playing;
if (!src->conninfo.connection || !src->conninfo.connected)
goto done;
/* send some dummy packets before we activate the receive in the
* udp sources */
gst_rtspsrc_send_dummy_packets (src);
/* require new SR packets */
if (src->manager)
g_signal_emit_by_name (src->manager, "reset-sync", NULL);
gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
/* construct a control url */
control = get_aggregate_control (src);
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
const gchar *setup_url;
GstRTSPConnection *conn;
/* 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 (src->conninfo.connection) {
conn = src->conninfo.connection;
} else if (stream->conninfo.connection) {
conn = stream->conninfo.connection;
} else {
continue;
}
/* do play */
res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
if (res < 0)
goto create_request_failed;
if (src->need_range) {
hval = gen_range_header (src, segment);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
/* store the newsegment event so it can be sent from the streaming thread. */
if (src->start_segment)
gst_event_unref (src->start_segment);
src->start_segment = gst_event_new_segment (&src->segment);
}
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);
}
if (async)
GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
goto send_error;
/* seek may have silently failed as it is not supported */
if (!(src->methods & GST_RTSP_PLAY)) {
GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
/* obviously it is supported as we made it here */
src->methods |= GST_RTSP_PLAY;
src->seekable = FALSE;
/* but there is nothing to parse in the response,
* so convey we have no idea and not to expect anything particular */
clear_rtp_base (src, stream);
if (control) {
GList *run;
/* need to do for all streams */
for (run = src->streams; run; run = g_list_next (run))
clear_rtp_base (src, (GstRTSPStream *) run->data);
}
/* NOTE the above also disables npt based eos detection */
/* and below forces position to 0,
* which is visible feedback we lost the plot */
segment->start = segment->position = src->last_pos;
}
gst_rtsp_message_unset (&request);
/* parse RTP npt field. This is the current position in the stream (Normal
* Play Time) and should be put in the NEWSEGMENT position field. */
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
0) == GST_RTSP_OK)
gst_rtspsrc_parse_range (src, hval, segment);
/* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
segment->rate = 1.0;
/* parse Speed header. This is the intended playback rate of the stream
* and should be put in the NEWSEGMENT rate field. */
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
0) == GST_RTSP_OK) {
segment->rate = gst_rtspsrc_get_float (hval);
} else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
&hval, 0) == GST_RTSP_OK) {
segment->rate = gst_rtspsrc_get_float (hval);
}
/* 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);
gst_rtsp_message_unset (&response);
/* early exit when we did aggregate control */
if (control)
break;
}
/* configure the caps of the streams after we parsed all headers. Only reset
* the manager object when we set a new Range header (we did a seek) */
gst_rtspsrc_configure_caps (src, segment, src->need_range);
/* set again when needed */
src->need_range = FALSE;
src->running = TRUE;
src->base_time = -1;
src->state = GST_RTSP_STATE_PLAYING;
/* 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;
}
done:
if (async)
gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
return res;
/* ERRORS */
open_failed:
{
GST_DEBUG_OBJECT (src, "failed to open stream");
goto done;
}
not_supported:
{
GST_DEBUG_OBJECT (src, "PLAY is not supported");
goto done;
}
was_playing:
{
GST_DEBUG_OBJECT (src, "we were already PLAYING");
goto done;
}
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (src, 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 (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (src, "PLAY interrupted");
}
g_free (str);
goto done;
}
}
static GstRTSPResult
gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
{
GstRTSPResult res = GST_RTSP_OK;
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GList *walk;
const gchar *control;
GST_DEBUG_OBJECT (src, "PAUSE...");
if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
goto open_failed;
if (!(src->methods & GST_RTSP_PAUSE))
goto not_supported;
if (src->state == GST_RTSP_STATE_READY)
goto was_paused;
if (!src->conninfo.connection || !src->conninfo.connected)
goto no_connection;
/* construct a control url */
control = get_aggregate_control (src);
/* loop over the streams. We might exit the loop early when we could do an
* aggregate control */
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
GstRTSPConnection *conn;
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 (src->conninfo.connection) {
conn = src->conninfo.connection;
} else if (stream->conninfo.connection) {
conn = stream->conninfo.connection;
} else {
continue;
}
if (async)
GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
("Sending PAUSE request"));
if ((res =
gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
setup_url)) < 0)
goto create_request_failed;
if ((res = gst_rtspsrc_send (src, conn, &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_rtspsrc_set_state (src, GST_STATE_PAUSED);
no_connection:
src->state = GST_RTSP_STATE_READY;
done:
if (async)
gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
return res;
/* ERRORS */
open_failed:
{
GST_DEBUG_OBJECT (src, "failed to open stream");
goto done;
}
not_supported:
{
GST_DEBUG_OBJECT (src, "PAUSE is not supported");
goto done;
}
was_paused:
{
GST_DEBUG_OBJECT (src, "we were already PAUSED");
goto done;
}
create_request_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ELEMENT_ERROR (src, 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 (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
} else {
GST_WARNING_OBJECT (src, "PAUSE interrupted");
}
g_free (str);
goto done;
}
}
static void
gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
{
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (bin);
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_EOS:
gst_message_unref (message);
break;
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 (rtspsrc);
ignore_timeout = rtspsrc->ignore_timeout;
rtspsrc->ignore_timeout = TRUE;
GST_OBJECT_UNLOCK (rtspsrc);
/* we only act on the first udp timeout message, others are irrelevant
* and can be ignored. */
if (!ignore_timeout)
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
/* eat and free */
gst_message_unref (message);
return;
}
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
case GST_MESSAGE_ERROR:
{
GstObject *udpsrc;
GstRTSPStream *stream;
GstFlowReturn ret;
udpsrc = GST_MESSAGE_SRC (message);
GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
GST_ELEMENT_NAME (udpsrc));
stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
if (!stream)
goto forward;
/* we ignore the RTCP udpsrc */
if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
goto done;
/* if we get error messages from the udp sources, that's not a problem as
* long as not all of them error out. We also don't really know what the
* problem is, the message does not give enough detail... */
ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
if (ret != GST_FLOW_OK)
goto forward;
done:
gst_message_unref (message);
break;
forward:
/* fatal but not our message, forward */
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
default:
{
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
}
}
/* the thread where everything happens */
static void
gst_rtspsrc_thread (GstRTSPSrc * src)
{
gint cmd;
GST_OBJECT_LOCK (src);
cmd = src->pending_cmd;
if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
|| cmd == CMD_LOOP || cmd == CMD_OPEN)
src->pending_cmd = CMD_LOOP;
else
src->pending_cmd = CMD_WAIT;
GST_DEBUG_OBJECT (src, "got command %d", cmd);
/* we got the message command, so ensure communication is possible again */
gst_rtspsrc_connection_flush (src, FALSE);
src->busy_cmd = cmd;
GST_OBJECT_UNLOCK (src);
switch (cmd) {
case CMD_OPEN:
gst_rtspsrc_open (src, TRUE);
break;
case CMD_PLAY:
gst_rtspsrc_play (src, &src->segment, TRUE);
break;
case CMD_PAUSE:
gst_rtspsrc_pause (src, TRUE);
break;
case CMD_CLOSE:
gst_rtspsrc_close (src, TRUE, FALSE);
break;
case CMD_LOOP:
gst_rtspsrc_loop (src);
break;
case CMD_RECONNECT:
gst_rtspsrc_reconnect (src, FALSE);
break;
default:
break;
}
GST_OBJECT_LOCK (src);
/* and go back to sleep */
if (src->pending_cmd == CMD_WAIT) {
if (src->task)
gst_task_pause (src->task);
}
/* reset waiting */
src->busy_cmd = CMD_WAIT;
GST_OBJECT_UNLOCK (src);
}
static gboolean
gst_rtspsrc_start (GstRTSPSrc * src)
{
GST_DEBUG_OBJECT (src, "starting");
GST_OBJECT_LOCK (src);
src->pending_cmd = CMD_WAIT;
if (src->task == NULL) {
src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
if (src->task == NULL)
goto task_error;
gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
}
GST_OBJECT_UNLOCK (src);
return TRUE;
/* ERRORS */
task_error:
{
GST_ERROR_OBJECT (src, "failed to create task");
return FALSE;
}
}
static gboolean
gst_rtspsrc_stop (GstRTSPSrc * src)
{
GstTask *task;
GST_DEBUG_OBJECT (src, "stopping");
/* also cancels pending task */
gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
GST_OBJECT_LOCK (src);
if ((task = src->task)) {
src->task = NULL;
GST_OBJECT_UNLOCK (src);
gst_task_stop (task);
/* make sure it is not running */
GST_RTSP_STREAM_LOCK (src);
GST_RTSP_STREAM_UNLOCK (src);
/* now wait for the task to finish */
gst_task_join (task);
/* and free the task */
gst_object_unref (GST_OBJECT (task));
GST_OBJECT_LOCK (src);
}
GST_OBJECT_UNLOCK (src);
/* ensure synchronously all is closed and clean */
gst_rtspsrc_close (src, FALSE, TRUE);
return TRUE;
}
static GstStateChangeReturn
gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
{
GstRTSPSrc *rtspsrc;
GstStateChangeReturn ret;
rtspsrc = GST_RTSPSRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!gst_rtspsrc_start (rtspsrc))
goto start_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* init some state */
rtspsrc->cur_protocols = rtspsrc->protocols;
/* first attempt, don't ignore timeouts */
rtspsrc->ignore_timeout = FALSE;
rtspsrc->open_error = FALSE;
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
set_manager_buffer_mode (rtspsrc);
/* fall-through */
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* unblock the tcp tasks and make the loop waiting */
if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
/* make sure it is waiting before we send PAUSE or PLAY below */
GST_RTSP_STREAM_LOCK (rtspsrc);
GST_RTSP_STREAM_UNLOCK (rtspsrc);
}
break;
case GST_STATE_CHANGE_PAUSED_TO_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:
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* send pause request and keep the idle task around */
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_rtspsrc_stop (rtspsrc);
ret = GST_STATE_CHANGE_SUCCESS;
break;
default:
break;
}
done:
return ret;
start_failed:
{
GST_DEBUG_OBJECT (rtspsrc, "start failed");
return GST_STATE_CHANGE_FAILURE;
}
}
static gboolean
gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
{
gboolean res;
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (element);
if (GST_EVENT_IS_DOWNSTREAM (event)) {
res = gst_rtspsrc_push_event (rtspsrc, event);
} else {
res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
}
return res;
}
/*** GSTURIHANDLER INTERFACE *************************************************/
static GstURIType
gst_rtspsrc_uri_get_type (GType type)
{
return GST_URI_SRC;
}
static const gchar *const *
gst_rtspsrc_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_rtspsrc_uri_get_uri (GstURIHandler * handler)
{
GstRTSPSrc *src = GST_RTSPSRC (handler);
/* FIXME: make thread-safe */
return g_strdup (src->conninfo.location);
}
static gboolean
gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
GError ** error)
{
GstRTSPSrc *src;
GstRTSPResult res;
GstRTSPUrl *newurl = NULL;
GstSDPMessage *sdp = NULL;
src = GST_RTSPSRC (handler);
/* same URI, we're fine */
if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
goto was_ok;
if (g_str_has_prefix (uri, "rtsp-sdp://")) {
if ((res = gst_sdp_message_new (&sdp) < 0))
goto sdp_failed;
GST_DEBUG_OBJECT (src, "parsing SDP message");
if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
goto invalid_sdp;
} else {
/* try to parse */
GST_DEBUG_OBJECT (src, "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 (src, "configuring URI");
g_free (src->conninfo.location);
src->conninfo.location = g_strdup (uri);
gst_rtsp_url_free (src->conninfo.url);
src->conninfo.url = newurl;
g_free (src->conninfo.url_str);
if (newurl)
src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
else
src->conninfo.url_str = NULL;
if (src->sdp)
gst_sdp_message_free (src->sdp);
src->sdp = sdp;
src->from_sdp = sdp != NULL;
GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
GST_DEBUG_OBJECT (src, "request uri is: %s",
GST_STR_NULL (src->conninfo.url_str));
return TRUE;
/* Special cases */
was_ok:
{
GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
return TRUE;
}
sdp_failed:
{
GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
"Could not create SDP");
return FALSE;
}
invalid_sdp:
{
GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
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 (src, "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_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_rtspsrc_uri_get_type;
iface->get_protocols = gst_rtspsrc_uri_get_protocols;
iface->get_uri = gst_rtspsrc_uri_get_uri;
iface->set_uri = gst_rtspsrc_uri_set_uri;
}
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