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gstreamer/gst/rtsp-server/rtsp-stream.c
Jan Schmidt b6ca057c72 rtsp-stream: Add functions for using rtsp-stream from the client 
Add a boolean to indicate that the rtsp-stream is running on the
'client' side of an RTSP connection, for sending streams via
RECORD. In that case, the roles of the client/server ports
in transport setup are swapped.

https://bugzilla.gnome.org/show_bug.cgi?id=758180
2016-01-29 01:44:26 +11:00

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/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
* Copyright (C) 2015 Centricular Ltd
* Author: Sebastian Dröge <sebastian@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:rtsp-stream
* @short_description: A media stream
* @see_also: #GstRTSPMedia
*
* The #GstRTSPStream object manages the data transport for one stream. It
* is created from a payloader element and a source pad that produce the RTP
* packets for the stream.
*
* With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
* and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
*
* The #GstRTSPStream will use the configured addresspool, as set with
* gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
* stream. With gst_rtsp_stream_get_multicast_address() you can get the
* configured address.
*
* With gst_rtsp_stream_get_server_port () you can get the port that the server
* will use to receive RTCP. This is the part that the clients will use to send
* RTCP to.
*
* With gst_rtsp_stream_add_transport() destinations can be added where the
* stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
* the destination again.
*
* Last reviewed on 2013-07-16 (1.0.0)
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <gio/gio.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "rtsp-stream.h"
#define GST_RTSP_STREAM_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
typedef struct
{
GstRTSPStreamTransport *transport;
/* RTP and RTCP source */
GstElement *udpsrc[2];
GstPad *selpad[2];
} GstRTSPMulticastTransportSource;
struct _GstRTSPStreamPrivate
{
GMutex lock;
guint idx;
/* Only one pad is ever set */
GstPad *srcpad, *sinkpad;
GstElement *payloader;
guint buffer_size;
gboolean is_joined;
/* TRUE if this stream is running on
* the client side of an RTSP link (for RECORD) */
gboolean client_side;
gchar *control;
GstRTSPProfile profiles;
GstRTSPLowerTrans protocols;
/* pads on the rtpbin */
GstPad *send_rtp_sink;
GstPad *recv_rtp_src;
GstPad *recv_sink[2];
GstPad *send_src[2];
/* the RTPSession object */
GObject *session;
/* SRTP encoder/decoder */
GstElement *srtpenc;
GstElement *srtpdec;
GHashTable *keys;
/* sinks used for sending and receiving RTP and RTCP over ipv4, they share
* sockets */
GstElement *udpsrc_v4[2];
/* sinks used for sending and receiving RTP and RTCP over ipv6, they share
* sockets */
GstElement *udpsrc_v6[2];
GstElement *udpqueue[2];
GstElement *udpsink[2];
/* for TCP transport */
GstElement *appsrc[2];
GstClockTime appsrc_base_time[2];
GstElement *appqueue[2];
GstElement *appsink[2];
GstElement *tee[2];
GstElement *funnel[2];
/* retransmission */
GstElement *rtxsend;
guint rtx_pt;
GstClockTime rtx_time;
/* server ports for sending/receiving over ipv4 */
GstRTSPRange server_port_v4;
GstRTSPAddress *server_addr_v4;
gboolean have_ipv4;
/* server ports for sending/receiving over ipv6 */
GstRTSPRange server_port_v6;
GstRTSPAddress *server_addr_v6;
gboolean have_ipv6;
/* multicast addresses */
GstRTSPAddressPool *pool;
GstRTSPAddress *addr_v4;
GstRTSPAddress *addr_v6;
/* the caps of the stream */
gulong caps_sig;
GstCaps *caps;
/* transports we stream to */
guint n_active;
GList *transports;
guint transports_cookie;
GList *tr_cache_rtp;
GList *tr_cache_rtcp;
guint tr_cache_cookie_rtp;
guint tr_cache_cookie_rtcp;
/* UDP sources for UDP multicast transports */
GList *transport_sources;
gint dscp_qos;
/* stream blocking */
gulong blocked_id;
gboolean blocking;
/* pt->caps map for RECORD streams */
GHashTable *ptmap;
};
#define DEFAULT_CONTROL NULL
#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
GST_RTSP_LOWER_TRANS_TCP
enum
{
PROP_0,
PROP_CONTROL,
PROP_PROFILES,
PROP_PROTOCOLS,
PROP_LAST
};
enum
{
SIGNAL_NEW_RTP_ENCODER,
SIGNAL_NEW_RTCP_ENCODER,
SIGNAL_LAST
};
GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
#define GST_CAT_DEFAULT rtsp_stream_debug
static GQuark ssrc_stream_map_key;
static void gst_rtsp_stream_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec);
static void gst_rtsp_stream_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec);
static void gst_rtsp_stream_finalize (GObject * obj);
static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
static void
gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
{
GObjectClass *gobject_class;
g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->get_property = gst_rtsp_stream_get_property;
gobject_class->set_property = gst_rtsp_stream_set_property;
gobject_class->finalize = gst_rtsp_stream_finalize;
g_object_class_install_property (gobject_class, PROP_CONTROL,
g_param_spec_string ("control", "Control",
"The control string for this stream", DEFAULT_CONTROL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PROFILES,
g_param_spec_flags ("profiles", "Profiles",
"Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
DEFAULT_PROFILES, 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));
gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
}
static void
gst_rtsp_stream_init (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
GST_DEBUG ("new stream %p", stream);
stream->priv = priv;
priv->dscp_qos = -1;
priv->control = g_strdup (DEFAULT_CONTROL);
priv->profiles = DEFAULT_PROFILES;
priv->protocols = DEFAULT_PROTOCOLS;
g_mutex_init (&priv->lock);
priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
NULL, (GDestroyNotify) gst_caps_unref);
priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
(GDestroyNotify) gst_caps_unref);
}
static void
gst_rtsp_stream_finalize (GObject * obj)
{
GstRTSPStream *stream;
GstRTSPStreamPrivate *priv;
stream = GST_RTSP_STREAM (obj);
priv = stream->priv;
GST_DEBUG ("finalize stream %p", stream);
/* we really need to be unjoined now */
g_return_if_fail (!priv->is_joined);
if (priv->addr_v4)
gst_rtsp_address_free (priv->addr_v4);
if (priv->addr_v6)
gst_rtsp_address_free (priv->addr_v6);
if (priv->server_addr_v4)
gst_rtsp_address_free (priv->server_addr_v4);
if (priv->server_addr_v6)
gst_rtsp_address_free (priv->server_addr_v6);
if (priv->pool)
g_object_unref (priv->pool);
if (priv->rtxsend)
g_object_unref (priv->rtxsend);
gst_object_unref (priv->payloader);
if (priv->srcpad)
gst_object_unref (priv->srcpad);
if (priv->sinkpad)
gst_object_unref (priv->sinkpad);
g_free (priv->control);
g_mutex_clear (&priv->lock);
g_hash_table_unref (priv->keys);
g_hash_table_destroy (priv->ptmap);
G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
}
static void
gst_rtsp_stream_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec)
{
GstRTSPStream *stream = GST_RTSP_STREAM (object);
switch (propid) {
case PROP_CONTROL:
g_value_take_string (value, gst_rtsp_stream_get_control (stream));
break;
case PROP_PROFILES:
g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
break;
case PROP_PROTOCOLS:
g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
gst_rtsp_stream_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec)
{
GstRTSPStream *stream = GST_RTSP_STREAM (object);
switch (propid) {
case PROP_CONTROL:
gst_rtsp_stream_set_control (stream, g_value_get_string (value));
break;
case PROP_PROFILES:
gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
break;
case PROP_PROTOCOLS:
gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
/**
* gst_rtsp_stream_new:
* @idx: an index
* @pad: a #GstPad
* @payloader: a #GstElement
*
* Create a new media stream with index @idx that handles RTP data on
* @pad and has a payloader element @payloader if @pad is a source pad
* or a depayloader element @payloader if @pad is a sink pad.
*
* Returns: (transfer full): a new #GstRTSPStream
*/
GstRTSPStream *
gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * pad)
{
GstRTSPStreamPrivate *priv;
GstRTSPStream *stream;
g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
g_return_val_if_fail (GST_IS_PAD (pad), NULL);
stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
priv = stream->priv;
priv->idx = idx;
priv->payloader = gst_object_ref (payloader);
if (GST_PAD_IS_SRC (pad))
priv->srcpad = gst_object_ref (pad);
else
priv->sinkpad = gst_object_ref (pad);
return stream;
}
/**
* gst_rtsp_stream_get_index:
* @stream: a #GstRTSPStream
*
* Get the stream index.
*
* Return: the stream index.
*/
guint
gst_rtsp_stream_get_index (GstRTSPStream * stream)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
return stream->priv->idx;
}
/**
* gst_rtsp_stream_get_pt:
* @stream: a #GstRTSPStream
*
* Get the stream payload type.
*
* Return: the stream payload type.
*/
guint
gst_rtsp_stream_get_pt (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
guint pt;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
priv = stream->priv;
g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
return pt;
}
/**
* gst_rtsp_stream_get_srcpad:
* @stream: a #GstRTSPStream
*
* Get the srcpad associated with @stream.
*
* Returns: (transfer full): the srcpad. Unref after usage.
*/
GstPad *
gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
if (!stream->priv->srcpad)
return NULL;
return gst_object_ref (stream->priv->srcpad);
}
/**
* gst_rtsp_stream_get_sinkpad:
* @stream: a #GstRTSPStream
*
* Get the sinkpad associated with @stream.
*
* Returns: (transfer full): the sinkpad. Unref after usage.
*/
GstPad *
gst_rtsp_stream_get_sinkpad (GstRTSPStream * stream)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
if (!stream->priv->sinkpad)
return NULL;
return gst_object_ref (stream->priv->sinkpad);
}
/**
* gst_rtsp_stream_get_control:
* @stream: a #GstRTSPStream
*
* Get the control string to identify this stream.
*
* Returns: (transfer full): the control string. g_free() after usage.
*/
gchar *
gst_rtsp_stream_get_control (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
gchar *result;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if ((result = g_strdup (priv->control)) == NULL)
result = g_strdup_printf ("stream=%u", priv->idx);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_stream_set_control:
* @stream: a #GstRTSPStream
* @control: a control string
*
* Set the control string in @stream.
*/
void
gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
{
GstRTSPStreamPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
g_mutex_lock (&priv->lock);
g_free (priv->control);
priv->control = g_strdup (control);
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_stream_has_control:
* @stream: a #GstRTSPStream
* @control: a control string
*
* Check if @stream has the control string @control.
*
* Returns: %TRUE is @stream has @control as the control string
*/
gboolean
gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
{
GstRTSPStreamPrivate *priv;
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if (priv->control)
res = (g_strcmp0 (priv->control, control) == 0);
else {
guint streamid;
if (sscanf (control, "stream=%u", &streamid) > 0)
res = (streamid == priv->idx);
else
res = FALSE;
}
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_stream_set_mtu:
* @stream: a #GstRTSPStream
* @mtu: a new MTU
*
* Configure the mtu in the payloader of @stream to @mtu.
*/
void
gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
{
GstRTSPStreamPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
GST_LOG_OBJECT (stream, "set MTU %u", mtu);
g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
}
/**
* gst_rtsp_stream_get_mtu:
* @stream: a #GstRTSPStream
*
* Get the configured MTU in the payloader of @stream.
*
* Returns: the MTU of the payloader.
*/
guint
gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
guint mtu;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
priv = stream->priv;
g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
return mtu;
}
/* Update the dscp qos property on the udp sinks */
static void
update_dscp_qos (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
if (priv->udpsink[0]) {
g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
NULL);
}
if (priv->udpsink[1]) {
g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
NULL);
}
}
/**
* gst_rtsp_stream_set_dscp_qos:
* @stream: a #GstRTSPStream
* @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
*
* Configure the dscp qos of the outgoing sockets to @dscp_qos.
*/
void
gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
{
GstRTSPStreamPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
if (dscp_qos < -1 || dscp_qos > 63) {
GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
return;
}
priv->dscp_qos = dscp_qos;
update_dscp_qos (stream);
}
/**
* gst_rtsp_stream_get_dscp_qos:
* @stream: a #GstRTSPStream
*
* Get the configured DSCP QoS in of the outgoing sockets.
*
* Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
*/
gint
gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
priv = stream->priv;
return priv->dscp_qos;
}
/**
* gst_rtsp_stream_is_transport_supported:
* @stream: a #GstRTSPStream
* @transport: (transfer none): a #GstRTSPTransport
*
* Check if @transport can be handled by stream
*
* Returns: %TRUE if @transport can be handled by @stream.
*/
gboolean
gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
GstRTSPTransport * transport)
{
GstRTSPStreamPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if (transport->trans != GST_RTSP_TRANS_RTP)
goto unsupported_transmode;
if (!(transport->profile & priv->profiles))
goto unsupported_profile;
if (!(transport->lower_transport & priv->protocols))
goto unsupported_ltrans;
g_mutex_unlock (&priv->lock);
return TRUE;
/* ERRORS */
unsupported_transmode:
{
GST_DEBUG ("unsupported transport mode %d", transport->trans);
g_mutex_unlock (&priv->lock);
return FALSE;
}
unsupported_profile:
{
GST_DEBUG ("unsupported profile %d", transport->profile);
g_mutex_unlock (&priv->lock);
return FALSE;
}
unsupported_ltrans:
{
GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
g_mutex_unlock (&priv->lock);
return FALSE;
}
}
/**
* gst_rtsp_stream_set_profiles:
* @stream: a #GstRTSPStream
* @profiles: the new profiles
*
* Configure the allowed profiles for @stream.
*/
void
gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
{
GstRTSPStreamPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
g_mutex_lock (&priv->lock);
priv->profiles = profiles;
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_stream_get_profiles:
* @stream: a #GstRTSPStream
*
* Get the allowed profiles of @stream.
*
* Returns: a #GstRTSPProfile
*/
GstRTSPProfile
gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
GstRTSPProfile res;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
priv = stream->priv;
g_mutex_lock (&priv->lock);
res = priv->profiles;
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_stream_set_protocols:
* @stream: a #GstRTSPStream
* @protocols: the new flags
*
* Configure the allowed lower transport for @stream.
*/
void
gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
GstRTSPLowerTrans protocols)
{
GstRTSPStreamPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
g_mutex_lock (&priv->lock);
priv->protocols = protocols;
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_stream_get_protocols:
* @stream: a #GstRTSPStream
*
* Get the allowed protocols of @stream.
*
* Returns: a #GstRTSPLowerTrans
*/
GstRTSPLowerTrans
gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
GstRTSPLowerTrans res;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
GST_RTSP_LOWER_TRANS_UNKNOWN);
priv = stream->priv;
g_mutex_lock (&priv->lock);
res = priv->protocols;
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_stream_set_address_pool:
* @stream: a #GstRTSPStream
* @pool: (transfer none): a #GstRTSPAddressPool
*
* configure @pool to be used as the address pool of @stream.
*/
void
gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
GstRTSPAddressPool * pool)
{
GstRTSPStreamPrivate *priv;
GstRTSPAddressPool *old;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
GST_LOG_OBJECT (stream, "set address pool %p", pool);
g_mutex_lock (&priv->lock);
if ((old = priv->pool) != pool)
priv->pool = pool ? g_object_ref (pool) : NULL;
else
old = NULL;
g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_stream_get_address_pool:
* @stream: a #GstRTSPStream
*
* Get the #GstRTSPAddressPool used as the address pool of @stream.
*
* Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
* usage.
*/
GstRTSPAddressPool *
gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
GstRTSPAddressPool *result;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->pool))
g_object_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_stream_get_multicast_address:
* @stream: a #GstRTSPStream
* @family: the #GSocketFamily
*
* Get the multicast address of @stream for @family.
*
* Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
* or %NULL when no address could be allocated. gst_rtsp_address_free()
* after usage.
*/
GstRTSPAddress *
gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
GSocketFamily family)
{
GstRTSPStreamPrivate *priv;
GstRTSPAddress *result;
GstRTSPAddress **addrp;
GstRTSPAddressFlags flags;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
priv = stream->priv;
if (family == G_SOCKET_FAMILY_IPV6) {
flags = GST_RTSP_ADDRESS_FLAG_IPV6;
addrp = &priv->addr_v6;
} else {
flags = GST_RTSP_ADDRESS_FLAG_IPV4;
addrp = &priv->addr_v4;
}
g_mutex_lock (&priv->lock);
if (*addrp == NULL) {
if (priv->pool == NULL)
goto no_pool;
flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
*addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
if (*addrp == NULL)
goto no_address;
}
result = gst_rtsp_address_copy (*addrp);
g_mutex_unlock (&priv->lock);
return result;
/* ERRORS */
no_pool:
{
GST_ERROR_OBJECT (stream, "no address pool specified");
g_mutex_unlock (&priv->lock);
return NULL;
}
no_address:
{
GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
g_mutex_unlock (&priv->lock);
return NULL;
}
}
/**
* gst_rtsp_stream_reserve_address:
* @stream: a #GstRTSPStream
* @address: an address
* @port: a port
* @n_ports: n_ports
* @ttl: a TTL
*
* Reserve @address and @port as the address and port of @stream.
*
* Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
* the address could be reserved. gst_rtsp_address_free() after usage.
*/
GstRTSPAddress *
gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
const gchar * address, guint port, guint n_ports, guint ttl)
{
GstRTSPStreamPrivate *priv;
GstRTSPAddress *result;
GInetAddress *addr;
GSocketFamily family;
GstRTSPAddress **addrp;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
g_return_val_if_fail (address != NULL, NULL);
g_return_val_if_fail (port > 0, NULL);
g_return_val_if_fail (n_ports > 0, NULL);
g_return_val_if_fail (ttl > 0, NULL);
priv = stream->priv;
addr = g_inet_address_new_from_string (address);
if (!addr) {
GST_ERROR ("failed to get inet addr from %s", address);
family = G_SOCKET_FAMILY_IPV4;
} else {
family = g_inet_address_get_family (addr);
g_object_unref (addr);
}
if (family == G_SOCKET_FAMILY_IPV6)
addrp = &priv->addr_v6;
else
addrp = &priv->addr_v4;
g_mutex_lock (&priv->lock);
if (*addrp == NULL) {
GstRTSPAddressPoolResult res;
if (priv->pool == NULL)
goto no_pool;
res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
port, n_ports, ttl, addrp);
if (res != GST_RTSP_ADDRESS_POOL_OK)
goto no_address;
} else {
if (strcmp ((*addrp)->address, address) ||
(*addrp)->port != port || (*addrp)->n_ports != n_ports ||
(*addrp)->ttl != ttl)
goto different_address;
}
result = gst_rtsp_address_copy (*addrp);
g_mutex_unlock (&priv->lock);
return result;
/* ERRORS */
no_pool:
{
GST_ERROR_OBJECT (stream, "no address pool specified");
g_mutex_unlock (&priv->lock);
return NULL;
}
no_address:
{
GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
address);
g_mutex_unlock (&priv->lock);
return NULL;
}
different_address:
{
GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
" reserved", address);
g_mutex_unlock (&priv->lock);
return NULL;
}
}
static gboolean
alloc_ports_one_family (GstRTSPStream * stream, GstRTSPAddressPool * pool,
gint buffer_size, GSocketFamily family, GstElement * udpsrc_out[2],
GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
GstRTSPAddress ** server_addr_out)
{
GstRTSPStreamPrivate *priv = stream->priv;
GstStateChangeReturn ret;
GstElement *udpsrc0, *udpsrc1;
GstElement *udpsink0, *udpsink1;
GSocket *rtp_socket = NULL;
GSocket *rtcp_socket;
gint tmp_rtp, tmp_rtcp;
guint count;
gint rtpport, rtcpport;
GList *rejected_addresses = NULL;
GstRTSPAddress *addr = NULL;
GInetAddress *inetaddr = NULL;
GSocketAddress *rtp_sockaddr = NULL;
GSocketAddress *rtcp_sockaddr = NULL;
const gchar *multisink_socket;
if (family == G_SOCKET_FAMILY_IPV6)
multisink_socket = "socket-v6";
else
multisink_socket = "socket";
udpsrc0 = NULL;
udpsrc1 = NULL;
udpsink0 = NULL;
udpsink1 = NULL;
count = 0;
/* Start with random port */
tmp_rtp = 0;
rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
G_SOCKET_PROTOCOL_UDP, NULL);
if (!rtcp_socket)
goto no_udp_protocol;
if (*server_addr_out)
gst_rtsp_address_free (*server_addr_out);
/* 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 (rtp_socket == NULL) {
rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
G_SOCKET_PROTOCOL_UDP, NULL);
if (!rtp_socket)
goto no_udp_protocol;
}
if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
GstRTSPAddressFlags flags;
if (addr)
rejected_addresses = g_list_prepend (rejected_addresses, addr);
flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
if (family == G_SOCKET_FAMILY_IPV6)
flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
else
flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
if (addr == NULL)
goto no_ports;
tmp_rtp = addr->port;
g_clear_object (&inetaddr);
inetaddr = g_inet_address_new_from_string (addr->address);
} else {
if (tmp_rtp != 0) {
tmp_rtp += 2;
if (++count > 20)
goto no_ports;
}
if (inetaddr == NULL)
inetaddr = g_inet_address_new_any (family);
}
rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
g_object_unref (rtp_sockaddr);
goto again;
}
g_object_unref (rtp_sockaddr);
rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
g_clear_object (&rtp_sockaddr);
goto socket_error;
}
tmp_rtp =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
g_object_unref (rtp_sockaddr);
/* check if port is even */
if ((tmp_rtp & 1) != 0) {
/* port not even, close and allocate another */
tmp_rtp++;
g_clear_object (&rtp_socket);
goto again;
}
/* set port */
tmp_rtcp = tmp_rtp + 1;
rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
g_object_unref (rtcp_sockaddr);
g_clear_object (&rtp_socket);
goto again;
}
g_object_unref (rtcp_sockaddr);
g_clear_object (&inetaddr);
udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
if (udpsrc0 == NULL || udpsrc1 == NULL)
goto no_udp_protocol;
g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
if (ret == GST_STATE_CHANGE_FAILURE)
goto element_error;
ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
if (ret == GST_STATE_CHANGE_FAILURE)
goto element_error;
/* 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;
if (udpsink_out[0])
udpsink0 = udpsink_out[0];
else
udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
if (!udpsink0)
goto no_udp_protocol;
g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
if (udpsink_out[1])
udpsink1 = udpsink_out[1];
else
udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
if (!udpsink1)
goto no_udp_protocol;
g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
/* Needs to be async for RECORD streams, otherwise we will never go to
* PLAYING because the sinks will wait for data while the udpsrc can't
* provide data with timestamps in PAUSED. */
if (priv->sinkpad)
g_object_set (G_OBJECT (udpsink0), "async", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
/* we keep these elements, we will further configure them when the
* client told us to really use the UDP ports. */
udpsrc_out[0] = udpsrc0;
udpsrc_out[1] = udpsrc1;
udpsink_out[0] = udpsink0;
udpsink_out[1] = udpsink1;
server_port_out->min = rtpport;
server_port_out->max = rtcpport;
*server_addr_out = addr;
g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
g_object_unref (rtp_socket);
g_object_unref (rtcp_socket);
return TRUE;
/* ERRORS */
no_udp_protocol:
{
goto cleanup;
}
no_ports:
{
goto cleanup;
}
port_error:
{
goto cleanup;
}
socket_error:
{
goto cleanup;
}
element_error:
{
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);
}
if (udpsink0) {
gst_element_set_state (udpsink0, GST_STATE_NULL);
gst_object_unref (udpsink0);
}
if (inetaddr)
g_object_unref (inetaddr);
g_list_free_full (rejected_addresses,
(GDestroyNotify) gst_rtsp_address_free);
if (addr)
gst_rtsp_address_free (addr);
if (rtp_socket)
g_object_unref (rtp_socket);
if (rtcp_socket)
g_object_unref (rtcp_socket);
return FALSE;
}
}
/* must be called with lock */
static gboolean
alloc_ports (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = stream->priv;
priv->have_ipv4 =
alloc_ports_one_family (stream, priv->pool, priv->buffer_size,
G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
&priv->server_port_v4, &priv->server_addr_v4);
priv->have_ipv6 =
alloc_ports_one_family (stream, priv->pool, priv->buffer_size,
G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
&priv->server_port_v6, &priv->server_addr_v6);
return priv->have_ipv4 || priv->have_ipv6;
}
/**
* gst_rtsp_stream_set_client_side:
* @stream: a #GstRTSPStream
* @client_side: TRUE if this #GstRTSPStream is running on the 'client' side of
* an RTSP connection.
*
* Sets the #GstRTSPStream as a 'client side' stream - used for sending
* streams to an RTSP server via RECORD. This has the practical effect
* of changing which UDP port numbers are used when setting up the local
* side of the stream sending to be either the 'server' or 'client' pair
* of a configured UDP transport.
*/
void
gst_rtsp_stream_set_client_side (GstRTSPStream * stream, gboolean client_side)
{
GstRTSPStreamPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
g_mutex_lock (&priv->lock);
priv->client_side = client_side;
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_stream_set_client_side:
* @stream: a #GstRTSPStream
*
* See gst_rtsp_stream_set_client_side()
*
* Returns: TRUE if this #GstRTSPStream is client-side.
*/
gboolean
gst_rtsp_stream_is_client_side (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
gboolean ret;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
ret = priv->client_side;
g_mutex_unlock (&priv->lock);
return ret;
}
/**
* gst_rtsp_stream_get_server_port:
* @stream: a #GstRTSPStream
* @server_port: (out): result server port
* @family: the port family to get
*
* Fill @server_port with the port pair used by the server. This function can
* only be called when @stream has been joined.
*/
void
gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
GstRTSPRange * server_port, GSocketFamily family)
{
GstRTSPStreamPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
g_return_if_fail (priv->is_joined);
g_mutex_lock (&priv->lock);
if (family == G_SOCKET_FAMILY_IPV4) {
if (server_port)
*server_port = priv->server_port_v4;
} else {
if (server_port)
*server_port = priv->server_port_v6;
}
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_stream_get_rtpsession:
* @stream: a #GstRTSPStream
*
* Get the RTP session of this stream.
*
* Returns: (transfer full): The RTP session of this stream. Unref after usage.
*/
GObject *
gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
GObject *session;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if ((session = priv->session))
g_object_ref (session);
g_mutex_unlock (&priv->lock);
return session;
}
/**
* gst_rtsp_stream_get_ssrc:
* @stream: a #GstRTSPStream
* @ssrc: (out): result ssrc
*
* Get the SSRC used by the RTP session of this stream. This function can only
* be called when @stream has been joined.
*/
void
gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
{
GstRTSPStreamPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
g_return_if_fail (priv->is_joined);
g_mutex_lock (&priv->lock);
if (ssrc && priv->session)
g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_stream_set_retransmission_time:
* @stream: a #GstRTSPStream
* @time: a #GstClockTime
*
* Set the amount of time to store retransmission packets.
*/
void
gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
GstClockTime time)
{
GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
g_mutex_lock (&stream->priv->lock);
stream->priv->rtx_time = time;
if (stream->priv->rtxsend)
g_object_set (stream->priv->rtxsend, "max-size-time",
GST_TIME_AS_MSECONDS (time), NULL);
g_mutex_unlock (&stream->priv->lock);
}
/**
* gst_rtsp_stream_get_retransmission_time:
* @stream: a #GstRTSPStream
*
* Get the amount of time to store retransmission data.
*
* Returns: the amount of time to store retransmission data.
*/
GstClockTime
gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
{
GstClockTime ret;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
g_mutex_lock (&stream->priv->lock);
ret = stream->priv->rtx_time;
g_mutex_unlock (&stream->priv->lock);
return ret;
}
/**
* gst_rtsp_stream_set_retransmission_pt:
* @stream: a #GstRTSPStream
* @rtx_pt: a #guint
*
* Set the payload type (pt) for retransmission of this stream.
*/
void
gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
{
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
g_mutex_lock (&stream->priv->lock);
stream->priv->rtx_pt = rtx_pt;
if (stream->priv->rtxsend) {
guint pt = gst_rtsp_stream_get_pt (stream);
gchar *pt_s = g_strdup_printf ("%d", pt);
GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
pt_s, G_TYPE_UINT, rtx_pt, NULL);
g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
g_free (pt_s);
gst_structure_free (rtx_pt_map);
}
g_mutex_unlock (&stream->priv->lock);
}
/**
* gst_rtsp_stream_get_retransmission_pt:
* @stream: a #GstRTSPStream
*
* Get the payload-type used for retransmission of this stream
*
* Returns: The retransmission PT.
*/
guint
gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
{
guint rtx_pt;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
g_mutex_lock (&stream->priv->lock);
rtx_pt = stream->priv->rtx_pt;
g_mutex_unlock (&stream->priv->lock);
return rtx_pt;
}
/**
* gst_rtsp_stream_set_buffer_size:
* @stream: a #GstRTSPStream
* @size: the buffer size
*
* Set the size of the UDP transmission buffer (in bytes)
* Needs to be set before the stream is joined to a bin.
*
* Since: 1.6
*/
void
gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
{
g_mutex_lock (&stream->priv->lock);
stream->priv->buffer_size = size;
g_mutex_unlock (&stream->priv->lock);
}
/**
* gst_rtsp_stream_get_buffer_size:
* @stream: a #GstRTSPStream
*
* Get the size of the UDP transmission buffer (in bytes)
*
* Returns: the size of the UDP TX buffer
*
* Since: 1.6
*/
guint
gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
{
guint buffer_size;
g_mutex_lock (&stream->priv->lock);
buffer_size = stream->priv->buffer_size;
g_mutex_unlock (&stream->priv->lock);
return buffer_size;
}
/* executed from streaming thread */
static void
caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = stream->priv;
GstCaps *newcaps, *oldcaps;
newcaps = gst_pad_get_current_caps (pad);
GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
newcaps);
g_mutex_lock (&priv->lock);
oldcaps = priv->caps;
priv->caps = newcaps;
g_mutex_unlock (&priv->lock);
if (oldcaps)
gst_caps_unref (oldcaps);
}
static void
dump_structure (const GstStructure * s)
{
gchar *sstr;
sstr = gst_structure_to_string (s);
GST_INFO ("structure: %s", sstr);
g_free (sstr);
}
static GstRTSPStreamTransport *
find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
{
GstRTSPStreamPrivate *priv = stream->priv;
GList *walk;
GstRTSPStreamTransport *result = NULL;
const gchar *tmp;
gchar *dest;
guint port;
if (rtcp_from == NULL)
return NULL;
tmp = g_strrstr (rtcp_from, ":");
if (tmp == NULL)
return NULL;
port = atoi (tmp + 1);
dest = g_strndup (rtcp_from, tmp - rtcp_from);
g_mutex_lock (&priv->lock);
GST_INFO ("finding %s:%d in %d transports", dest, port,
g_list_length (priv->transports));
for (walk = priv->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *trans = walk->data;
const GstRTSPTransport *tr;
gint min, max;
tr = gst_rtsp_stream_transport_get_transport (trans);
if (priv->client_side) {
/* In client side mode the 'destination' is the RTSP server, so send
* to those ports */
min = tr->server_port.min;
max = tr->server_port.max;
} else {
min = tr->client_port.min;
max = tr->client_port.max;
}
if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
result = trans;
break;
}
}
if (result)
g_object_ref (result);
g_mutex_unlock (&priv->lock);
g_free (dest);
return result;
}
static GstRTSPStreamTransport *
check_transport (GObject * source, GstRTSPStream * stream)
{
GstStructure *stats;
GstRTSPStreamTransport *trans;
/* see if we have a stream to match with the origin of the RTCP packet */
trans = g_object_get_qdata (source, ssrc_stream_map_key);
if (trans == NULL) {
g_object_get (source, "stats", &stats, NULL);
if (stats) {
const gchar *rtcp_from;
dump_structure (stats);
rtcp_from = gst_structure_get_string (stats, "rtcp-from");
if ((trans = find_transport (stream, rtcp_from))) {
GST_INFO ("%p: found transport %p for source %p", stream, trans,
source);
g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
g_object_unref);
}
gst_structure_free (stats);
}
}
return trans;
}
static void
on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPStreamTransport *trans;
GST_INFO ("%p: new source %p", stream, source);
trans = check_transport (source, stream);
if (trans)
GST_INFO ("%p: source %p for transport %p", stream, source, trans);
}
static void
on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
{
GST_INFO ("%p: new SDES %p", stream, source);
}
static void
on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPStreamTransport *trans;
trans = check_transport (source, stream);
if (trans) {
GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
gst_rtsp_stream_transport_keep_alive (trans);
}
#ifdef DUMP_STATS
{
GstStructure *stats;
g_object_get (source, "stats", &stats, NULL);
if (stats) {
dump_structure (stats);
gst_structure_free (stats);
}
}
#endif
}
static void
on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
{
GST_INFO ("%p: source %p bye", stream, source);
}
static void
on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPStreamTransport *trans;
GST_INFO ("%p: source %p bye timeout", stream, source);
if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
g_object_set_qdata (source, ssrc_stream_map_key, NULL);
}
}
static void
on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPStreamTransport *trans;
GST_INFO ("%p: source %p timeout", stream, source);
if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
g_object_set_qdata (source, ssrc_stream_map_key, NULL);
}
}
static void
on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
{
GST_INFO ("%p: new sender source %p", stream, source);
#ifndef DUMP_STATS
{
GstStructure *stats;
g_object_get (source, "stats", &stats, NULL);
if (stats) {
dump_structure (stats);
gst_structure_free (stats);
}
}
#endif
}
static void
on_sender_ssrc_active (GObject * session, GObject * source,
GstRTSPStream * stream)
{
#ifndef DUMP_STATS
{
GstStructure *stats;
g_object_get (source, "stats", &stats, NULL);
if (stats) {
dump_structure (stats);
gst_structure_free (stats);
}
}
#endif
}
static void
clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
{
if (is_rtp) {
g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
g_list_free (priv->tr_cache_rtp);
priv->tr_cache_rtp = NULL;
} else {
g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
g_list_free (priv->tr_cache_rtcp);
priv->tr_cache_rtcp = NULL;
}
}
static GstFlowReturn
handle_new_sample (GstAppSink * sink, gpointer user_data)
{
GstRTSPStreamPrivate *priv;
GList *walk;
GstSample *sample;
GstBuffer *buffer;
GstRTSPStream *stream;
gboolean is_rtp;
sample = gst_app_sink_pull_sample (sink);
if (!sample)
return GST_FLOW_OK;
stream = (GstRTSPStream *) user_data;
priv = stream->priv;
buffer = gst_sample_get_buffer (sample);
is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
g_mutex_lock (&priv->lock);
if (is_rtp) {
if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
clear_tr_cache (priv, is_rtp);
for (walk = priv->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
priv->tr_cache_rtp =
g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
}
priv->tr_cache_cookie_rtp = priv->transports_cookie;
}
} else {
if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
clear_tr_cache (priv, is_rtp);
for (walk = priv->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
priv->tr_cache_rtcp =
g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
}
priv->tr_cache_cookie_rtcp = priv->transports_cookie;
}
}
g_mutex_unlock (&priv->lock);
if (is_rtp) {
for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
gst_rtsp_stream_transport_send_rtp (tr, buffer);
}
} else {
for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
gst_rtsp_stream_transport_send_rtcp (tr, buffer);
}
}
gst_sample_unref (sample);
return GST_FLOW_OK;
}
static GstAppSinkCallbacks sink_cb = {
NULL, /* not interested in EOS */
NULL, /* not interested in preroll samples */
handle_new_sample,
};
static GstElement *
get_rtp_encoder (GstRTSPStream * stream, guint session)
{
GstRTSPStreamPrivate *priv = stream->priv;
if (priv->srtpenc == NULL) {
gchar *name;
name = g_strdup_printf ("srtpenc_%u", session);
priv->srtpenc = gst_element_factory_make ("srtpenc", name);
g_free (name);
g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
}
return gst_object_ref (priv->srtpenc);
}
static GstElement *
request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = stream->priv;
GstElement *oldenc, *enc;
GstPad *pad;
gchar *name;
if (priv->idx != session)
return NULL;
GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
oldenc = priv->srtpenc;
enc = get_rtp_encoder (stream, session);
name = g_strdup_printf ("rtp_sink_%d", session);
pad = gst_element_get_request_pad (enc, name);
g_free (name);
gst_object_unref (pad);
if (oldenc == NULL)
g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
enc);
return enc;
}
static GstElement *
request_rtcp_encoder (GstElement * rtpbin, guint session,
GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = stream->priv;
GstElement *oldenc, *enc;
GstPad *pad;
gchar *name;
if (priv->idx != session)
return NULL;
GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
oldenc = priv->srtpenc;
enc = get_rtp_encoder (stream, session);
name = g_strdup_printf ("rtcp_sink_%d", session);
pad = gst_element_get_request_pad (enc, name);
g_free (name);
gst_object_unref (pad);
if (oldenc == NULL)
g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
enc);
return enc;
}
static GstCaps *
request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = stream->priv;
GstCaps *caps;
GST_DEBUG ("request key %08x", ssrc);
g_mutex_lock (&priv->lock);
if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
gst_caps_ref (caps);
g_mutex_unlock (&priv->lock);
return caps;
}
static GstElement *
request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = stream->priv;
if (priv->idx != session)
return NULL;
if (priv->srtpdec == NULL) {
gchar *name;
name = g_strdup_printf ("srtpdec_%u", session);
priv->srtpdec = gst_element_factory_make ("srtpdec", name);
g_free (name);
g_signal_connect (priv->srtpdec, "request-key",
(GCallback) request_key, stream);
}
return gst_object_ref (priv->srtpdec);
}
/**
* gst_rtsp_stream_request_aux_sender:
* @stream: a #GstRTSPStream
* @sessid: the session id
*
* Creating a rtxsend bin
*
* Returns: (transfer full): a #GstElement.
*
* Since: 1.6
*/
GstElement *
gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
{
GstElement *bin;
GstPad *pad;
GstStructure *pt_map;
gchar *name;
guint pt, rtx_pt;
gchar *pt_s;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
pt = gst_rtsp_stream_get_pt (stream);
pt_s = g_strdup_printf ("%u", pt);
rtx_pt = stream->priv->rtx_pt;
GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
bin = gst_bin_new (NULL);
stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
pt_map = gst_structure_new ("application/x-rtp-pt-map",
pt_s, G_TYPE_UINT, rtx_pt, NULL);
g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
"max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
g_free (pt_s);
gst_structure_free (pt_map);
gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
name = g_strdup_printf ("src_%u", sessid);
gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
g_free (name);
gst_object_unref (pad);
pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
name = g_strdup_printf ("sink_%u", sessid);
gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
g_free (name);
gst_object_unref (pad);
return bin;
}
/**
* gst_rtsp_stream_set_pt_map:
* @stream: a #GstRTSPStream
* @pt: the pt
* @caps: a #GstCaps
*
* Configure a pt map between @pt and @caps.
*/
void
gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
{
GstRTSPStreamPrivate *priv = stream->priv;
g_mutex_lock (&priv->lock);
g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
g_mutex_unlock (&priv->lock);
}
static GstCaps *
request_pt_map (GstElement * rtpbin, guint session, guint pt,
GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = stream->priv;
GstCaps *caps = NULL;
g_mutex_lock (&priv->lock);
if (priv->idx == session) {
caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
if (caps) {
GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
gst_caps_ref (caps);
} else {
GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
}
}
g_mutex_unlock (&priv->lock);
return caps;
}
static void
pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = stream->priv;
gchar *name;
GstPadLinkReturn ret;
guint sessid;
GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
name = gst_pad_get_name (pad);
if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
g_free (name);
return;
}
g_free (name);
if (priv->idx != sessid)
return;
if (gst_pad_is_linked (priv->sinkpad)) {
GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
GST_DEBUG_PAD_NAME (priv->sinkpad));
return;
}
/* link the RTP pad to the session manager, it should not really fail unless
* this is not really an RTP pad */
ret = gst_pad_link (pad, priv->sinkpad);
if (ret != GST_PAD_LINK_OK)
goto link_failed;
priv->recv_rtp_src = gst_object_ref (pad);
return;
/* ERRORS */
link_failed:
{
GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
}
}
static void
on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
GstRTSPStream * stream)
{
/* TODO: What to do here other than this? */
GST_DEBUG ("Stream %p: Got EOS", stream);
gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
}
/**
* gst_rtsp_stream_join_bin:
* @stream: a #GstRTSPStream
* @bin: (transfer none): a #GstBin to join
* @rtpbin: (transfer none): a rtpbin element in @bin
* @state: the target state of the new elements
*
* Join the #GstBin @bin that contains the element @rtpbin.
*
* @stream will link to @rtpbin, which must be inside @bin. The elements
* added to @bin will be set to the state given in @state.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
GstElement * rtpbin, GstState state)
{
GstRTSPStreamPrivate *priv;
gint i;
guint idx;
gchar *name;
GstPad *pad, *sinkpad = NULL, *selpad;
GstPadLinkReturn ret;
gboolean is_tcp = FALSE, is_udp = FALSE;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if (priv->is_joined)
goto was_joined;
/* create a session with the same index as the stream */
idx = priv->idx;
GST_INFO ("stream %p joining bin as session %u", stream, idx);
is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
(priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
if (is_udp && !alloc_ports (stream))
goto no_ports;
/* update the dscp qos field in the sinks */
update_dscp_qos (stream);
if (priv->profiles & GST_RTSP_PROFILE_SAVP
|| priv->profiles & GST_RTSP_PROFILE_SAVPF) {
/* For SRTP */
g_signal_connect (rtpbin, "request-rtp-encoder",
(GCallback) request_rtp_encoder, stream);
g_signal_connect (rtpbin, "request-rtcp-encoder",
(GCallback) request_rtcp_encoder, stream);
g_signal_connect (rtpbin, "request-rtp-decoder",
(GCallback) request_rtp_rtcp_decoder, stream);
g_signal_connect (rtpbin, "request-rtcp-decoder",
(GCallback) request_rtp_rtcp_decoder, stream);
}
if (priv->sinkpad) {
g_signal_connect (rtpbin, "request-pt-map",
(GCallback) request_pt_map, stream);
}
/* get pads from the RTP session element for sending and receiving
* RTP/RTCP*/
if (priv->srcpad) {
/* get a pad for sending RTP */
name = g_strdup_printf ("send_rtp_sink_%u", idx);
priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
g_free (name);
/* link the RTP pad to the session manager, it should not really fail unless
* this is not really an RTP pad */
ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
if (ret != GST_PAD_LINK_OK)
goto link_failed;
name = g_strdup_printf ("send_rtp_src_%u", idx);
priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
g_free (name);
} else {
/* Need to connect our sinkpad from here */
g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
/* EOS */
g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
name = g_strdup_printf ("recv_rtp_sink_%u", idx);
priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
g_free (name);
}
name = g_strdup_printf ("send_rtcp_src_%u", idx);
priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
g_free (name);
name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
g_free (name);
/* get the session */
g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
stream);
g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
stream);
g_signal_connect (priv->session, "on-ssrc-active",
(GCallback) on_ssrc_active, stream);
g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
stream);
g_signal_connect (priv->session, "on-bye-timeout",
(GCallback) on_bye_timeout, stream);
g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
stream);
/* signal for sender ssrc */
g_signal_connect (priv->session, "on-new-sender-ssrc",
(GCallback) on_new_sender_ssrc, stream);
g_signal_connect (priv->session, "on-sender-ssrc-active",
(GCallback) on_sender_ssrc_active, stream);
for (i = 0; i < 2; i++) {
GstPad *teepad, *queuepad;
/* For the sender we create this bit of pipeline for both
* RTP and RTCP. Sync and preroll are enabled on udpsink so
* we need to add a queue before appsink and udpsink to make
* the pipeline not block. For the TCP case, we want to pump
* client as fast as possible anyway. This pipeline is used
* when both TCP and UDP are present.
*
* .--------. .-----. .---------. .---------.
* | rtpbin | | tee | | queue | | udpsink |
* | send->sink src->sink src->sink |
* '--------' | | '---------' '---------'
* | | .---------. .---------.
* | | | queue | | appsink |
* | src->sink src->sink |
* '-----' '---------' '---------'
*
* When only UDP or only TCP is allowed, we skip the tee and queue
* and link the udpsink (for UDP) or appsink (for TCP) directly to
* the session.
*/
/* Only link the RTP send src if we're going to send RTP, link
* the RTCP send src always */
if (priv->srcpad || i == 1) {
if (is_udp) {
/* add udpsink */
gst_bin_add (bin, priv->udpsink[i]);
sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
}
if (is_tcp) {
/* make appsink */
priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
gst_bin_add (bin, priv->appsink[i]);
gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
&sink_cb, stream, NULL);
}
if (is_udp && is_tcp) {
g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
/* make tee for RTP/RTCP */
priv->tee[i] = gst_element_factory_make ("tee", NULL);
gst_bin_add (bin, priv->tee[i]);
/* and link to rtpbin send pad */
pad = gst_element_get_static_pad (priv->tee[i], "sink");
gst_pad_link (priv->send_src[i], pad);
gst_object_unref (pad);
priv->udpqueue[i] = gst_element_factory_make ("queue", NULL);
g_object_set (priv->udpqueue[i], "max-size-buffers",
1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
NULL);
gst_bin_add (bin, priv->udpqueue[i]);
/* link tee to udpqueue */
teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
pad = gst_element_get_static_pad (priv->udpqueue[i], "sink");
gst_pad_link (teepad, pad);
gst_object_unref (pad);
gst_object_unref (teepad);
/* link udpqueue to udpsink */
queuepad = gst_element_get_static_pad (priv->udpqueue[i], "src");
gst_pad_link (queuepad, sinkpad);
gst_object_unref (queuepad);
gst_object_unref (sinkpad);
/* make appqueue */
priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
g_object_set (priv->appqueue[i], "max-size-buffers",
1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
NULL);
gst_bin_add (bin, priv->appqueue[i]);
/* and link tee to appqueue */
teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
gst_pad_link (teepad, pad);
gst_object_unref (pad);
gst_object_unref (teepad);
/* and link appqueue to appsink */
queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
pad = gst_element_get_static_pad (priv->appsink[i], "sink");
gst_pad_link (queuepad, pad);
gst_object_unref (pad);
gst_object_unref (queuepad);
} else if (is_tcp) {
/* only appsink needed, link it to the session */
pad = gst_element_get_static_pad (priv->appsink[i], "sink");
gst_pad_link (priv->send_src[i], pad);
gst_object_unref (pad);
/* when its only TCP, we need to set sync and preroll to FALSE
* for the sink to avoid deadlock. And this is only needed for
* sink used for RTCP data, not the RTP data. */
if (i == 1)
g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
} else {
/* else only udpsink needed, link it to the session */
gst_pad_link (priv->send_src[i], sinkpad);
gst_object_unref (sinkpad);
}
}
/* Only connect recv RTP sink if we expect to receive RTP. Connect recv
* RTCP sink always */
if (priv->sinkpad || i == 1) {
/* For the receiver we create this bit of pipeline for both
* RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
* and it is all funneled into the rtpbin receive pad.
*
* .--------. .--------. .--------.
* | udpsrc | | funnel | | rtpbin |
* | src->sink src->sink |
* '--------' | | '--------'
* .--------. | |
* | appsrc | | |
* | src->sink |
* '--------' '--------'
*/
/* make funnel for the RTP/RTCP receivers */
priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
gst_bin_add (bin, priv->funnel[i]);
pad = gst_element_get_static_pad (priv->funnel[i], "src");
gst_pad_link (pad, priv->recv_sink[i]);
gst_object_unref (pad);
if (priv->udpsrc_v4[i]) {
if (priv->srcpad) {
/* we set and keep these to playing so that they don't cause NO_PREROLL return
* values. This is only relevant for PLAY pipelines */
gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
}
/* add udpsrc */
gst_bin_add (bin, priv->udpsrc_v4[i]);
/* and link to the funnel v4 */
selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
gst_pad_link (pad, selpad);
gst_object_unref (pad);
gst_object_unref (selpad);
}
if (priv->udpsrc_v6[i]) {
if (priv->srcpad) {
gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
}
gst_bin_add (bin, priv->udpsrc_v6[i]);
/* and link to the funnel v6 */
selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
gst_pad_link (pad, selpad);
gst_object_unref (pad);
gst_object_unref (selpad);
}
if (is_tcp) {
/* make and add appsrc */
priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
priv->appsrc_base_time[i] = -1;
g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, NULL);
gst_bin_add (bin, priv->appsrc[i]);
/* and link to the funnel */
selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
pad = gst_element_get_static_pad (priv->appsrc[i], "src");
gst_pad_link (pad, selpad);
gst_object_unref (pad);
gst_object_unref (selpad);
}
}
/* check if we need to set to a special state */
if (state != GST_STATE_NULL) {
if (priv->udpsink[i] && (priv->srcpad || i == 1))
gst_element_set_state (priv->udpsink[i], state);
if (priv->appsink[i] && (priv->srcpad || i == 1))
gst_element_set_state (priv->appsink[i], state);
if (priv->appqueue[i] && (priv->srcpad || i == 1))
gst_element_set_state (priv->appqueue[i], state);
if (priv->udpqueue[i] && (priv->srcpad || i == 1))
gst_element_set_state (priv->udpqueue[i], state);
if (priv->tee[i] && (priv->srcpad || i == 1))
gst_element_set_state (priv->tee[i], state);
if (priv->funnel[i] && (priv->sinkpad || i == 1))
gst_element_set_state (priv->funnel[i], state);
if (priv->appsrc[i] && (priv->sinkpad || i == 1))
gst_element_set_state (priv->appsrc[i], state);
}
}
if (priv->srcpad) {
/* be notified of caps changes */
priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
(GCallback) caps_notify, stream);
}
priv->is_joined = TRUE;
g_mutex_unlock (&priv->lock);
return TRUE;
/* ERRORS */
was_joined:
{
g_mutex_unlock (&priv->lock);
return TRUE;
}
no_ports:
{
g_mutex_unlock (&priv->lock);
GST_WARNING ("failed to allocate ports %u", idx);
return FALSE;
}
link_failed:
{
GST_WARNING ("failed to link stream %u", idx);
gst_object_unref (priv->send_rtp_sink);
priv->send_rtp_sink = NULL;
g_mutex_unlock (&priv->lock);
return FALSE;
}
}
/**
* gst_rtsp_stream_leave_bin:
* @stream: a #GstRTSPStream
* @bin: (transfer none): a #GstBin
* @rtpbin: (transfer none): a rtpbin #GstElement
*
* Remove the elements of @stream from @bin.
*
* Return: %TRUE on success.
*/
gboolean
gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
GstElement * rtpbin)
{
GstRTSPStreamPrivate *priv;
gint i;
GList *l;
gboolean is_tcp, is_udp;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if (!priv->is_joined)
goto was_not_joined;
/* all transports must be removed by now */
if (priv->transports != NULL)
goto transports_not_removed;
clear_tr_cache (priv, TRUE);
clear_tr_cache (priv, FALSE);
GST_INFO ("stream %p leaving bin", stream);
if (priv->srcpad) {
gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
gst_object_unref (priv->send_rtp_sink);
priv->send_rtp_sink = NULL;
} else if (priv->recv_rtp_src) {
gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
gst_object_unref (priv->recv_rtp_src);
priv->recv_rtp_src = NULL;
}
is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
(priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
for (i = 0; i < 2; i++) {
if (priv->udpsink[i])
gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
if (priv->appsink[i])
gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
if (priv->appqueue[i])
gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
if (priv->udpqueue[i])
gst_element_set_state (priv->udpqueue[i], GST_STATE_NULL);
if (priv->tee[i])
gst_element_set_state (priv->tee[i], GST_STATE_NULL);
if (priv->funnel[i])
gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
if (priv->appsrc[i])
gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
if (priv->udpsrc_v4[i]) {
if (priv->sinkpad || i == 1) {
/* and set udpsrc to NULL now before removing */
gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
/* removing them should also nicely release the request
* pads when they finalize */
gst_bin_remove (bin, priv->udpsrc_v4[i]);
} else {
/* we need to set the state to NULL before unref */
gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
gst_object_unref (priv->udpsrc_v4[i]);
}
}
if (priv->udpsrc_v6[i]) {
if (priv->sinkpad || i == 1) {
gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
gst_bin_remove (bin, priv->udpsrc_v6[i]);
} else {
gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
gst_object_unref (priv->udpsrc_v6[i]);
}
}
for (l = priv->transport_sources; l; l = l->next) {
GstRTSPMulticastTransportSource *s = l->data;
if (!s->udpsrc[i])
continue;
gst_element_set_locked_state (s->udpsrc[i], FALSE);
gst_element_set_state (s->udpsrc[i], GST_STATE_NULL);
gst_bin_remove (bin, s->udpsrc[i]);
}
if (priv->udpsink[i] && is_udp && (priv->srcpad || i == 1))
gst_bin_remove (bin, priv->udpsink[i]);
if (priv->appsrc[i] && (priv->sinkpad || i == 1))
gst_bin_remove (bin, priv->appsrc[i]);
if (priv->appsink[i] && is_tcp && (priv->srcpad || i == 1))
gst_bin_remove (bin, priv->appsink[i]);
if (priv->appqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
gst_bin_remove (bin, priv->appqueue[i]);
if (priv->udpqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
gst_bin_remove (bin, priv->udpqueue[i]);
if (priv->tee[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
gst_bin_remove (bin, priv->tee[i]);
if (priv->funnel[i] && (priv->sinkpad || i == 1))
gst_bin_remove (bin, priv->funnel[i]);
if (priv->sinkpad || i == 1) {
gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
gst_object_unref (priv->recv_sink[i]);
priv->recv_sink[i] = NULL;
}
priv->udpsrc_v4[i] = NULL;
priv->udpsrc_v6[i] = NULL;
priv->udpsink[i] = NULL;
priv->appsrc[i] = NULL;
priv->appsink[i] = NULL;
priv->appqueue[i] = NULL;
priv->udpqueue[i] = NULL;
priv->tee[i] = NULL;
priv->funnel[i] = NULL;
}
for (l = priv->transport_sources; l; l = l->next) {
GstRTSPMulticastTransportSource *s = l->data;
g_slice_free (GstRTSPMulticastTransportSource, s);
}
g_list_free (priv->transport_sources);
priv->transport_sources = NULL;
if (priv->srcpad) {
gst_object_unref (priv->send_src[0]);
priv->send_src[0] = NULL;
}
gst_element_release_request_pad (rtpbin, priv->send_src[1]);
gst_object_unref (priv->send_src[1]);
priv->send_src[1] = NULL;
g_object_unref (priv->session);
priv->session = NULL;
if (priv->caps)
gst_caps_unref (priv->caps);
priv->caps = NULL;
if (priv->srtpenc)
gst_object_unref (priv->srtpenc);
if (priv->srtpdec)
gst_object_unref (priv->srtpdec);
priv->is_joined = FALSE;
g_mutex_unlock (&priv->lock);
return TRUE;
was_not_joined:
{
g_mutex_unlock (&priv->lock);
return TRUE;
}
transports_not_removed:
{
GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
g_mutex_unlock (&priv->lock);
return FALSE;
}
}
/**
* gst_rtsp_stream_get_rtpinfo:
* @stream: a #GstRTSPStream
* @rtptime: (allow-none): result RTP timestamp
* @seq: (allow-none): result RTP seqnum
* @clock_rate: (allow-none): the clock rate
* @running_time: (allow-none): result running-time
*
* Retrieve the current rtptime, seq and running-time. This is used to
* construct a RTPInfo reply header.
*
* Returns: %TRUE when rtptime, seq and running-time could be determined.
*/
gboolean
gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
guint * rtptime, guint * seq, guint * clock_rate,
GstClockTime * running_time)
{
GstRTSPStreamPrivate *priv;
GstStructure *stats;
GObjectClass *payobjclass;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
g_mutex_lock (&priv->lock);
/* First try to extract the information from the last buffer on the sinks.
* This will have a more accurate sequence number and timestamp, as between
* the payloader and the sink there can be some queues
*/
if (priv->udpsink[0] || priv->appsink[0]) {
GstSample *last_sample;
if (priv->udpsink[0])
g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
else
g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
if (last_sample) {
GstCaps *caps;
GstBuffer *buffer;
GstSegment *segment;
GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
caps = gst_sample_get_caps (last_sample);
buffer = gst_sample_get_buffer (last_sample);
segment = gst_sample_get_segment (last_sample);
if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
if (seq) {
*seq = gst_rtp_buffer_get_seq (&rtp_buffer);
}
if (rtptime) {
*rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
}
gst_rtp_buffer_unmap (&rtp_buffer);
if (running_time) {
*running_time =
gst_segment_to_running_time (segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (buffer));
}
if (clock_rate) {
GstStructure *s = gst_caps_get_structure (caps, 0);
gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
if (*clock_rate == 0 && running_time)
*running_time = GST_CLOCK_TIME_NONE;
}
gst_sample_unref (last_sample);
goto done;
} else {
gst_sample_unref (last_sample);
}
}
}
if (g_object_class_find_property (payobjclass, "stats")) {
g_object_get (priv->payloader, "stats", &stats, NULL);
if (stats == NULL)
goto no_stats;
if (seq)
gst_structure_get_uint (stats, "seqnum", seq);
if (rtptime)
gst_structure_get_uint (stats, "timestamp", rtptime);
if (running_time)
gst_structure_get_clock_time (stats, "running-time", running_time);
if (clock_rate) {
gst_structure_get_uint (stats, "clock-rate", clock_rate);
if (*clock_rate == 0 && running_time)
*running_time = GST_CLOCK_TIME_NONE;
}
gst_structure_free (stats);
} else {
if (!g_object_class_find_property (payobjclass, "seqnum") ||
!g_object_class_find_property (payobjclass, "timestamp"))
goto no_stats;
if (seq)
g_object_get (priv->payloader, "seqnum", seq, NULL);
if (rtptime)
g_object_get (priv->payloader, "timestamp", rtptime, NULL);
if (running_time)
*running_time = GST_CLOCK_TIME_NONE;
}
done:
g_mutex_unlock (&priv->lock);
return TRUE;
/* ERRORS */
no_stats:
{
GST_WARNING ("Could not get payloader stats");
g_mutex_unlock (&priv->lock);
return FALSE;
}
}
/**
* gst_rtsp_stream_get_caps:
* @stream: a #GstRTSPStream
*
* Retrieve the current caps of @stream.
*
* Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
* after usage.
*/
GstCaps *
gst_rtsp_stream_get_caps (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
GstCaps *result;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->caps))
gst_caps_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_stream_recv_rtp:
* @stream: a #GstRTSPStream
* @buffer: (transfer full): a #GstBuffer
*
* Handle an RTP buffer for the stream. This method is usually called when a
* message has been received from a client using the TCP transport.
*
* This function takes ownership of @buffer.
*
* Returns: a GstFlowReturn.
*/
GstFlowReturn
gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
{
GstRTSPStreamPrivate *priv;
GstFlowReturn ret;
GstElement *element;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
priv = stream->priv;
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
g_return_val_if_fail (priv->is_joined, FALSE);
g_mutex_lock (&priv->lock);
if (priv->appsrc[0])
element = gst_object_ref (priv->appsrc[0]);
else
element = NULL;
g_mutex_unlock (&priv->lock);
if (element) {
if (priv->appsrc_base_time[0] == -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 (element);
if (GST_ELEMENT_CLOCK (element)) {
GstClockTime now;
GstClockTime base_time;
now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
base_time = GST_ELEMENT_CAST (element)->base_time;
priv->appsrc_base_time[0] = now - base_time;
GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
GST_TIME_ARGS (base_time));
}
GST_OBJECT_UNLOCK (element);
}
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
gst_object_unref (element);
} else {
ret = GST_FLOW_OK;
}
return ret;
}
/**
* gst_rtsp_stream_recv_rtcp:
* @stream: a #GstRTSPStream
* @buffer: (transfer full): a #GstBuffer
*
* Handle an RTCP buffer for the stream. This method is usually called when a
* message has been received from a client using the TCP transport.
*
* This function takes ownership of @buffer.
*
* Returns: a GstFlowReturn.
*/
GstFlowReturn
gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
{
GstRTSPStreamPrivate *priv;
GstFlowReturn ret;
GstElement *element;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
priv = stream->priv;
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
if (!priv->is_joined) {
gst_buffer_unref (buffer);
return GST_FLOW_NOT_LINKED;
}
g_mutex_lock (&priv->lock);
if (priv->appsrc[1])
element = gst_object_ref (priv->appsrc[1]);
else
element = NULL;
g_mutex_unlock (&priv->lock);
if (element) {
if (priv->appsrc_base_time[1] == -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 (element);
if (GST_ELEMENT_CLOCK (element)) {
GstClockTime now;
GstClockTime base_time;
now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
base_time = GST_ELEMENT_CAST (element)->base_time;
priv->appsrc_base_time[1] = now - base_time;
GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
GST_TIME_ARGS (base_time));
}
GST_OBJECT_UNLOCK (element);
}
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
gst_object_unref (element);
} else {
ret = GST_FLOW_OK;
gst_buffer_unref (buffer);
}
return ret;
}
/* must be called with lock */
static gboolean
update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
gboolean add)
{
GstRTSPStreamPrivate *priv = stream->priv;
const GstRTSPTransport *tr;
tr = gst_rtsp_stream_transport_get_transport (trans);
switch (tr->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
{
GstRTSPMulticastTransportSource *source;
GstBin *bin;
bin = GST_BIN (gst_object_get_parent (GST_OBJECT (priv->funnel[1])));
if (add) {
gchar *host;
gint i;
GstPad *selpad, *pad;
source = g_slice_new0 (GstRTSPMulticastTransportSource);
source->transport = trans;
for (i = 0; i < 2; i++) {
host =
g_strdup_printf ("udp://%s:%d", tr->destination,
(i == 0) ? tr->port.min : tr->port.max);
source->udpsrc[i] =
gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
g_free (host);
g_object_set (source->udpsrc[i], "loop", FALSE, NULL);
if (priv->srcpad) {
/* we set and keep these to playing so that they don't cause NO_PREROLL return
* values. This is only relevant for PLAY pipelines */
gst_element_set_state (source->udpsrc[i], GST_STATE_PLAYING);
gst_element_set_locked_state (source->udpsrc[i], TRUE);
}
/* add udpsrc */
gst_bin_add (bin, source->udpsrc[i]);
/* and link to the funnel v4 */
if (priv->sinkpad || i == 1) {
source->selpad[i] = selpad =
gst_element_get_request_pad (priv->funnel[i], "sink_%u");
pad = gst_element_get_static_pad (source->udpsrc[i], "src");
gst_pad_link (pad, selpad);
gst_object_unref (pad);
gst_object_unref (selpad);
}
}
priv->transport_sources =
g_list_prepend (priv->transport_sources, source);
} else {
GList *l;
for (l = priv->transport_sources; l; l = l->next) {
source = l->data;
if (source->transport == trans) {
priv->transport_sources =
g_list_delete_link (priv->transport_sources, l);
break;
}
}
if (l != NULL) {
gint i;
for (i = 0; i < 2; i++) {
/* Will automatically unlink everything */
gst_bin_remove (bin,
GST_ELEMENT (gst_object_ref (source->udpsrc[i])));
gst_element_set_state (source->udpsrc[i], GST_STATE_NULL);
gst_object_unref (source->udpsrc[i]);
if (priv->sinkpad || i == 1) {
gst_element_release_request_pad (priv->funnel[i],
source->selpad[i]);
}
}
g_slice_free (GstRTSPMulticastTransportSource, source);
}
}
gst_object_unref (bin);
/* fall through for the generic case */
}
case GST_RTSP_LOWER_TRANS_UDP:
{
gchar *dest;
gint min, max;
guint ttl = 0;
dest = tr->destination;
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
min = tr->port.min;
max = tr->port.max;
ttl = tr->ttl;
} else if (priv->client_side) {
/* In client side mode the 'destination' is the RTSP server, so send
* to those ports */
min = tr->server_port.min;
max = tr->server_port.max;
} else {
min = tr->client_port.min;
max = tr->client_port.max;
}
if (add) {
if (ttl > 0) {
GST_INFO ("setting ttl-mc %d", ttl);
g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
}
GST_INFO ("adding %s:%d-%d", dest, min, max);
g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
priv->transports = g_list_prepend (priv->transports, trans);
} else {
GST_INFO ("removing %s:%d-%d", dest, min, max);
g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
priv->transports = g_list_remove (priv->transports, trans);
}
priv->transports_cookie++;
break;
}
case GST_RTSP_LOWER_TRANS_TCP:
if (add) {
GST_INFO ("adding TCP %s", tr->destination);
priv->transports = g_list_prepend (priv->transports, trans);
} else {
GST_INFO ("removing TCP %s", tr->destination);
priv->transports = g_list_remove (priv->transports, trans);
}
priv->transports_cookie++;
break;
default:
goto unknown_transport;
}
return TRUE;
/* ERRORS */
unknown_transport:
{
GST_INFO ("Unknown transport %d", tr->lower_transport);
return FALSE;
}
}
/**
* gst_rtsp_stream_add_transport:
* @stream: a #GstRTSPStream
* @trans: (transfer none): a #GstRTSPStreamTransport
*
* Add the transport in @trans to @stream. The media of @stream will
* then also be send to the values configured in @trans.
*
* @stream must be joined to a bin.
*
* @trans must contain a valid #GstRTSPTransport.
*
* Returns: %TRUE if @trans was added
*/
gboolean
gst_rtsp_stream_add_transport (GstRTSPStream * stream,
GstRTSPStreamTransport * trans)
{
GstRTSPStreamPrivate *priv;
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
g_return_val_if_fail (priv->is_joined, FALSE);
g_mutex_lock (&priv->lock);
res = update_transport (stream, trans, TRUE);
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_stream_remove_transport:
* @stream: a #GstRTSPStream
* @trans: (transfer none): a #GstRTSPStreamTransport
*
* Remove the transport in @trans from @stream. The media of @stream will
* not be sent to the values configured in @trans.
*
* @stream must be joined to a bin.
*
* @trans must contain a valid #GstRTSPTransport.
*
* Returns: %TRUE if @trans was removed
*/
gboolean
gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
GstRTSPStreamTransport * trans)
{
GstRTSPStreamPrivate *priv;
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
g_return_val_if_fail (priv->is_joined, FALSE);
g_mutex_lock (&priv->lock);
res = update_transport (stream, trans, FALSE);
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_stream_update_crypto:
* @stream: a #GstRTSPStream
* @ssrc: the SSRC
* @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
*
* Update the new crypto information for @ssrc in @stream. If information
* for @ssrc did not exist, it will be added. If information
* for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
* be removed from @stream.
*
* Returns: %TRUE if @crypto could be updated
*/
gboolean
gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
guint ssrc, GstCaps * crypto)
{
GstRTSPStreamPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
priv = stream->priv;
GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
g_mutex_lock (&priv->lock);
if (crypto)
g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
gst_caps_ref (crypto));
else
g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
g_mutex_unlock (&priv->lock);
return TRUE;
}
/**
* gst_rtsp_stream_get_rtp_socket:
* @stream: a #GstRTSPStream
* @family: the socket family
*
* Get the RTP socket from @stream for a @family.
*
* @stream must be joined to a bin.
*
* Returns: (transfer full) (nullable): the RTP socket or %NULL if no
* socket could be allocated for @family. Unref after usage
*/
GSocket *
gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
{
GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
GSocket *socket;
const gchar *name;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
family == G_SOCKET_FAMILY_IPV6, NULL);
g_return_val_if_fail (priv->udpsink[0], NULL);
if (family == G_SOCKET_FAMILY_IPV6)
name = "socket-v6";
else
name = "socket";
g_object_get (priv->udpsink[0], name, &socket, NULL);
return socket;
}
/**
* gst_rtsp_stream_get_rtcp_socket:
* @stream: a #GstRTSPStream
* @family: the socket family
*
* Get the RTCP socket from @stream for a @family.
*
* @stream must be joined to a bin.
*
* Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
* socket could be allocated for @family. Unref after usage
*/
GSocket *
gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
{
GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
GSocket *socket;
const gchar *name;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
family == G_SOCKET_FAMILY_IPV6, NULL);
g_return_val_if_fail (priv->udpsink[1], NULL);
if (family == G_SOCKET_FAMILY_IPV6)
name = "socket-v6";
else
name = "socket";
g_object_get (priv->udpsink[1], name, &socket, NULL);
return socket;
}
/**
* gst_rtsp_stream_set_seqnum:
* @stream: a #GstRTSPStream
* @seqnum: a new sequence number
*
* Configure the sequence number in the payloader of @stream to @seqnum.
*/
void
gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
{
GstRTSPStreamPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
}
/**
* gst_rtsp_stream_get_seqnum:
* @stream: a #GstRTSPStream
*
* Get the configured sequence number in the payloader of @stream.
*
* Returns: the sequence number of the payloader.
*/
guint16
gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
guint seqnum;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
priv = stream->priv;
g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
return seqnum;
}
/**
* gst_rtsp_stream_transport_filter:
* @stream: a #GstRTSPStream
* @func: (scope call) (allow-none): a callback
* @user_data: (closure): user data passed to @func
*
* Call @func for each transport managed by @stream. The result value of @func
* determines what happens to the transport. @func will be called with @stream
* locked so no further actions on @stream can be performed from @func.
*
* If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
* @stream.
*
* If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
*
* If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
* will also be added with an additional ref to the result #GList of this
* function..
*
* When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
*
* Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
* transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
* element in the #GList should be unreffed before the list is freed.
*/
GList *
gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
GstRTSPStreamTransportFilterFunc func, gpointer user_data)
{
GstRTSPStreamPrivate *priv;
GList *result, *walk, *next;
GHashTable *visited = NULL;
guint cookie;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
priv = stream->priv;
result = NULL;
if (func)
visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
g_mutex_lock (&priv->lock);
restart:
cookie = priv->transports_cookie;
for (walk = priv->transports; walk; walk = next) {
GstRTSPStreamTransport *trans = walk->data;
GstRTSPFilterResult res;
gboolean changed;
next = g_list_next (walk);
if (func) {
/* only visit each transport once */
if (g_hash_table_contains (visited, trans))
continue;
g_hash_table_add (visited, g_object_ref (trans));
g_mutex_unlock (&priv->lock);
res = func (stream, trans, user_data);
g_mutex_lock (&priv->lock);
} else
res = GST_RTSP_FILTER_REF;
changed = (cookie != priv->transports_cookie);
switch (res) {
case GST_RTSP_FILTER_REMOVE:
update_transport (stream, trans, FALSE);
break;
case GST_RTSP_FILTER_REF:
result = g_list_prepend (result, g_object_ref (trans));
break;
case GST_RTSP_FILTER_KEEP:
default:
break;
}
if (changed)
goto restart;
}
g_mutex_unlock (&priv->lock);
if (func)
g_hash_table_unref (visited);
return result;
}
static GstPadProbeReturn
pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
{
GstRTSPStreamPrivate *priv;
GstRTSPStream *stream;
stream = user_data;
priv = stream->priv;
GST_DEBUG_OBJECT (pad, "now blocking");
g_mutex_lock (&priv->lock);
priv->blocking = TRUE;
g_mutex_unlock (&priv->lock);
gst_element_post_message (priv->payloader,
gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
gst_structure_new_empty ("GstRTSPStreamBlocking")));
return GST_PAD_PROBE_OK;
}
/**
* gst_rtsp_stream_set_blocked:
* @stream: a #GstRTSPStream
* @blocked: boolean indicating we should block or unblock
*
* Blocks or unblocks the dataflow on @stream.
*
* Returns: %TRUE on success
*/
gboolean
gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
{
GstRTSPStreamPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if (blocked) {
priv->blocking = FALSE;
if (priv->blocked_id == 0) {
priv->blocked_id = gst_pad_add_probe (priv->srcpad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
g_object_ref (stream), g_object_unref);
}
} else {
if (priv->blocked_id != 0) {
gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
priv->blocked_id = 0;
priv->blocking = FALSE;
}
}
g_mutex_unlock (&priv->lock);
return TRUE;
}
/**
* gst_rtsp_stream_is_blocking:
* @stream: a #GstRTSPStream
*
* Check if @stream is blocking on a #GstBuffer.
*
* Returns: %TRUE if @stream is blocking
*/
gboolean
gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
gboolean result;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
result = priv->blocking;
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_stream_query_position:
* @stream: a #GstRTSPStream
*
* Query the position of the stream in %GST_FORMAT_TIME. This only considers
* the RTP parts of the pipeline and not the RTCP parts.
*
* Returns: %TRUE if the position could be queried
*/
gboolean
gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
{
GstRTSPStreamPrivate *priv;
GstElement *sink;
gboolean ret;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
/* depending on the transport type, it should query corresponding sink */
if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
(priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
sink = priv->udpsink[0];
else
sink = priv->appsink[0];
if (sink)
gst_object_ref (sink);
g_mutex_unlock (&priv->lock);
if (!sink)
return FALSE;
ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
gst_object_unref (sink);
return ret;
}
/**
* gst_rtsp_stream_query_stop:
* @stream: a #GstRTSPStream
*
* Query the stop of the stream in %GST_FORMAT_TIME. This only considers
* the RTP parts of the pipeline and not the RTCP parts.
*
* Returns: %TRUE if the stop could be queried
*/
gboolean
gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
{
GstRTSPStreamPrivate *priv;
GstElement *sink;
GstQuery *query;
gboolean ret;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
/* depending on the transport type, it should query corresponding sink */
if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
(priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
sink = priv->udpsink[0];
else
sink = priv->appsink[0];
if (sink)
gst_object_ref (sink);
g_mutex_unlock (&priv->lock);
if (!sink)
return FALSE;
query = gst_query_new_segment (GST_FORMAT_TIME);
if ((ret = gst_element_query (sink, query))) {
GstFormat format;
gst_query_parse_segment (query, NULL, &format, NULL, stop);
if (format != GST_FORMAT_TIME)
*stop = -1;
}
gst_query_unref (query);
gst_object_unref (sink);
return ret;
}
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