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gstreamer/gst/rtsp-server/rtsp-stream.c

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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.
*
* Each #GstRTSPStreamTransport spawns one queue that will serve as a backlog of a
* controllable maximum size when the reflux from the TCP connection's backpressure
* starts spilling all over.
*
* Unlike the backlog in rtspconnection, which we have decided should only contain
* at most one RTP and one RTCP data message in order to allow control messages to
* go through unobstructed, this backlog only consists of data messages, allowing
* us to fill it up without concern.
*
* When multiple TCP transports exist, for example in the context of a shared media,
* we only pop samples from our appsinks when at least one of the transports doesn't
* experience back pressure: this allows us to pace our sample popping to the speed
* of the fastest client.
*
* When a sample is popped, it is either sent directly on transports that don't
* experience backpressure, or queued on the transport's backlog otherwise. Samples
* are then popped from that backlog when the transport reports it has sent the message.
*
* Once the backlog reaches an overly large duration, the transport is dropped as
* the client was deemed too slow.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#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"
#include "rtsp-server-internal.h"
struct _GstRTSPStreamPrivate
{
GMutex lock;
guint idx;
/* Only one pad is ever set */
GstPad *srcpad, *sinkpad;
GstElement *payloader;
guint buffer_size;
GstBin *joined_bin;
/* TRUE if this stream is running on
* the client side of an RTSP link (for RECORD) */
gboolean client_side;
gchar *control;
/* TRUE if stream is complete. This means that the receiver and the sender
* parts are present in the stream. */
gboolean is_complete;
GstRTSPProfile profiles;
GstRTSPLowerTrans allowed_protocols;
GstRTSPLowerTrans configured_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;
/* for UDP unicast */
GstElement *udpsrc_v4[2];
GstElement *udpsrc_v6[2];
GstElement *udpqueue[2];
GstElement *udpsink[2];
GSocket *socket_v4[2];
GSocket *socket_v6[2];
/* for UDP multicast */
GstElement *mcast_udpsrc_v4[2];
GstElement *mcast_udpsrc_v6[2];
GstElement *mcast_udpqueue[2];
GstElement *mcast_udpsink[2];
GSocket *mcast_socket_v4[2];
GSocket *mcast_socket_v6[2];
GList *mcast_clients;
/* 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;
GstElement *rtxreceive;
guint rtx_pt;
GstClockTime rtx_time;
/* rate control */
gboolean do_rate_control;
/* Forward Error Correction with RFC 5109 */
GstElement *ulpfec_decoder;
GstElement *ulpfec_encoder;
guint ulpfec_pt;
gboolean ulpfec_enabled;
guint ulpfec_percentage;
/* pool used to manage unicast and multicast addresses */
GstRTSPAddressPool *pool;
/* unicast server addr/port */
GstRTSPAddress *server_addr_v4;
GstRTSPAddress *server_addr_v6;
/* multicast addresses */
GstRTSPAddress *mcast_addr_v4;
GstRTSPAddress *mcast_addr_v6;
gchar *multicast_iface;
guint max_mcast_ttl;
gboolean bind_mcast_address;
/* the caps of the stream */
gulong caps_sig;
GstCaps *caps;
/* transports we stream to */
guint n_active;
GList *transports;
guint transports_cookie;
GPtrArray *tr_cache;
guint tr_cache_cookie;
guint n_tcp_transports;
gboolean have_buffer[2];
gint dscp_qos;
/* Sending logic for TCP */
GThread *send_thread;
GCond send_cond;
GMutex send_lock;
/* @send_lock is released when pushing data out, we use
* a cookie to decide whether we should wait on @send_cond
* before checking the transports' backlogs again
*/
guint send_cookie;
/* Used to control shutdown of @send_thread */
gboolean continue_sending;
/* stream blocking */
gulong blocked_id[2];
gboolean blocking;
/* current stream postion */
GstClockTime position;
/* pt->caps map for RECORD streams */
GHashTable *ptmap;
GstRTSPPublishClockMode publish_clock_mode;
GThreadPool *send_pool;
guint32 last_seqnum;
};
#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
#define DEFAULT_MAX_MCAST_TTL 255
#define DEFAULT_BIND_MCAST_ADDRESS FALSE
#define DEFAULT_DO_RATE_CONTROL TRUE
enum
{
PROP_0,
PROP_CONTROL,
PROP_PROFILES,
PROP_PROTOCOLS,
PROP_LAST
};
enum
{
SIGNAL_NEW_RTP_ENCODER,
SIGNAL_NEW_RTCP_ENCODER,
SIGNAL_NEW_RTP_RTCP_DECODER,
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 gboolean
update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
gboolean add);
static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
static void
gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
{
GObjectClass *gobject_class;
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, NULL, 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, NULL, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
gst_rtsp_stream_signals[SIGNAL_NEW_RTP_RTCP_DECODER] =
g_signal_new ("new-rtp-rtcp-decoder", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, 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_instance_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->allowed_protocols = DEFAULT_PROTOCOLS;
priv->configured_protocols = 0;
priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
g_mutex_init (&priv->lock);
priv->continue_sending = TRUE;
priv->send_cookie = 0;
g_cond_init (&priv->send_cond);
g_mutex_init (&priv->send_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);
priv->send_pool = NULL;
}
typedef struct _UdpClientAddrInfo UdpClientAddrInfo;
struct _UdpClientAddrInfo
{
gchar *address;
guint rtp_port;
guint add_count; /* how often this address has been added */
};
static void
free_mcast_client (gpointer data)
{
UdpClientAddrInfo *client = data;
g_free (client->address);
g_free (client);
}
static void
gst_rtsp_stream_finalize (GObject * obj)
{
GstRTSPStream *stream;
GstRTSPStreamPrivate *priv;
guint i;
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->joined_bin == NULL);
if (priv->send_pool)
g_thread_pool_free (priv->send_pool, TRUE, TRUE);
if (priv->mcast_addr_v4)
gst_rtsp_address_free (priv->mcast_addr_v4);
if (priv->mcast_addr_v6)
gst_rtsp_address_free (priv->mcast_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);
if (priv->rtxreceive)
g_object_unref (priv->rtxreceive);
if (priv->ulpfec_encoder)
gst_object_unref (priv->ulpfec_encoder);
if (priv->ulpfec_decoder)
gst_object_unref (priv->ulpfec_decoder);
for (i = 0; i < 2; i++) {
if (priv->socket_v4[i])
g_object_unref (priv->socket_v4[i]);
if (priv->socket_v6[i])
g_object_unref (priv->socket_v6[i]);
if (priv->mcast_socket_v4[i])
g_object_unref (priv->mcast_socket_v4[i]);
if (priv->mcast_socket_v6[i])
g_object_unref (priv->mcast_socket_v6[i]);
}
g_free (priv->multicast_iface);
g_list_free_full (priv->mcast_clients, (GDestroyNotify) free_mcast_client);
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_mutex_clear (&priv->send_lock);
g_cond_clear (&priv->send_cond);
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) (nullable): 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) (nullable): 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) (nullable): 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: (nullable): 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: (nullable): 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, GstElement ** udpsink)
{
GstRTSPStreamPrivate *priv;
priv = stream->priv;
if (*udpsink) {
g_object_set (G_OBJECT (*udpsink), "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, priv->udpsink);
}
/**
* 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);
g_return_val_if_fail (transport != NULL, 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->allowed_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->allowed_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->allowed_protocols;
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_stream_set_address_pool:
* @stream: a #GstRTSPStream
* @pool: (transfer none) (nullable): 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) (nullable): 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_set_multicast_iface:
* @stream: a #GstRTSPStream
* @multicast_iface: (transfer none) (nullable): a multicast interface name
*
* configure @multicast_iface to be used for @stream.
*/
void
gst_rtsp_stream_set_multicast_iface (GstRTSPStream * stream,
const gchar * multicast_iface)
{
GstRTSPStreamPrivate *priv;
gchar *old;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
GST_LOG_OBJECT (stream, "set multicast iface %s",
GST_STR_NULL (multicast_iface));
g_mutex_lock (&priv->lock);
if ((old = priv->multicast_iface) != multicast_iface)
priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
else
old = NULL;
g_mutex_unlock (&priv->lock);
if (old)
g_free (old);
}
/**
* gst_rtsp_stream_get_multicast_iface:
* @stream: a #GstRTSPStream
*
* Get the multicast interface used for @stream.
*
* Returns: (transfer full) (nullable): the multicast interface for @stream.
* g_free() after usage.
*/
gchar *
gst_rtsp_stream_get_multicast_iface (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 = priv->multicast_iface))
result = g_strdup (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. The original
* #GstRTSPAddress is cached and copy is returned, so freeing the return value
* won't release the address from the pool.
*
* 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;
g_mutex_lock (&stream->priv->lock);
if (family == G_SOCKET_FAMILY_IPV6) {
flags = GST_RTSP_ADDRESS_FLAG_IPV6;
addrp = &priv->mcast_addr_v6;
} else {
flags = GST_RTSP_ADDRESS_FLAG_IPV4;
addrp = &priv->mcast_addr_v4;
}
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;
/* FIXME: Also reserve the same port with unicast ANY address, since that's
* where we are going to bind our socket. Probably loop until we find a port
* available in both mcast and unicast pools. Maybe GstRTSPAddressPool
* should do it for us when both GST_RTSP_ADDRESS_FLAG_MULTICAST and
* GST_RTSP_ADDRESS_FLAG_UNICAST are givent. */
}
result = gst_rtsp_address_copy (*addrp);
g_mutex_unlock (&stream->priv->lock);
return result;
/* ERRORS */
no_pool:
{
GST_ERROR_OBJECT (stream, "no address pool specified");
g_mutex_unlock (&stream->priv->lock);
return NULL;
}
no_address:
{
GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
g_mutex_unlock (&stream->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. The original
* #GstRTSPAddress is cached and copy is returned, so freeing the return value
* won't release the address from the pool.
*
* Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
* the address could not 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->mcast_addr_v6;
else
addrp = &priv->mcast_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;
/* FIXME: Also reserve the same port with unicast ANY address, since that's
* where we are going to bind our socket. */
} else {
if (g_ascii_strcasecmp ((*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 as %s that was already reserved",
address, (*addrp)->address);
g_mutex_unlock (&priv->lock);
return NULL;
}
}
/* must be called with lock */
static void
set_socket_for_udpsink (GstElement * udpsink, GSocket * socket,
GSocketFamily family)
{
const gchar *multisink_socket;
if (family == G_SOCKET_FAMILY_IPV6)
multisink_socket = "socket-v6";
else
multisink_socket = "socket";
g_object_set (G_OBJECT (udpsink), multisink_socket, socket, NULL);
}
/* must be called with lock */
static void
set_multicast_socket_for_udpsink (GstElement * udpsink, GSocket * socket,
GSocketFamily family, const gchar * multicast_iface,
const gchar * addr_str, gint port, gint mcast_ttl)
{
set_socket_for_udpsink (udpsink, socket, family);
if (multicast_iface) {
GST_INFO ("setting multicast-iface %s", multicast_iface);
g_object_set (G_OBJECT (udpsink), "multicast-iface", multicast_iface, NULL);
}
if (mcast_ttl > 0) {
GST_INFO ("setting ttl-mc %d", mcast_ttl);
g_object_set (G_OBJECT (udpsink), "ttl-mc", mcast_ttl, NULL);
}
}
/* must be called with lock */
static void
set_unicast_socket_for_udpsink (GstElement * udpsink, GSocket * socket,
GSocketFamily family)
{
set_socket_for_udpsink (udpsink, socket, family);
}
static guint16
get_port_from_socket (GSocket * socket)
{
guint16 port;
GSocketAddress *sockaddr;
GError *err;
GST_DEBUG ("socket: %p", socket);
sockaddr = g_socket_get_local_address (socket, &err);
if (sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (sockaddr)) {
g_clear_object (&sockaddr);
GST_ERROR ("failed to get sockaddr: %s", err->message);
g_error_free (err);
return 0;
}
port = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
g_object_unref (sockaddr);
return port;
}
static gboolean
create_and_configure_udpsink (GstRTSPStream * stream, GstElement ** udpsink,
GSocket * socket_v4, GSocket * socket_v6, gboolean multicast,
gboolean is_rtp, gint mcast_ttl)
{
GstRTSPStreamPrivate *priv = stream->priv;
*udpsink = gst_element_factory_make ("multiudpsink", NULL);
if (!*udpsink)
goto no_udp_protocol;
/* configure sinks */
g_object_set (G_OBJECT (*udpsink), "close-socket", FALSE, NULL);
g_object_set (G_OBJECT (*udpsink), "send-duplicates", FALSE, NULL);
if (is_rtp)
g_object_set (G_OBJECT (*udpsink), "buffer-size", priv->buffer_size, NULL);
else
g_object_set (G_OBJECT (*udpsink), "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 (!is_rtp || priv->sinkpad)
g_object_set (G_OBJECT (*udpsink), "async", FALSE, NULL);
if (multicast) {
/* join multicast group when adding clients, so we'll start receiving from it.
* We cannot rely on the udpsrc to join the group since its socket is always a
* local unicast one. */
g_object_set (G_OBJECT (*udpsink), "auto-multicast", TRUE, NULL);
g_object_set (G_OBJECT (*udpsink), "loop", FALSE, NULL);
}
/* update the dscp qos field in the sinks */
update_dscp_qos (stream, udpsink);
if (priv->server_addr_v4) {
GST_DEBUG_OBJECT (stream, "udp IPv4, configure udpsinks");
set_unicast_socket_for_udpsink (*udpsink, socket_v4, G_SOCKET_FAMILY_IPV4);
}
if (priv->server_addr_v6) {
GST_DEBUG_OBJECT (stream, "udp IPv6, configure udpsinks");
set_unicast_socket_for_udpsink (*udpsink, socket_v6, G_SOCKET_FAMILY_IPV6);
}
if (multicast) {
gint port;
if (priv->mcast_addr_v4) {
GST_DEBUG_OBJECT (stream, "mcast IPv4, configure udpsinks");
port = get_port_from_socket (socket_v4);
if (!port)
goto get_port_failed;
set_multicast_socket_for_udpsink (*udpsink, socket_v4,
G_SOCKET_FAMILY_IPV4, priv->multicast_iface,
priv->mcast_addr_v4->address, port, mcast_ttl);
}
if (priv->mcast_addr_v6) {
GST_DEBUG_OBJECT (stream, "mcast IPv6, configure udpsinks");
port = get_port_from_socket (socket_v6);
if (!port)
goto get_port_failed;
set_multicast_socket_for_udpsink (*udpsink, socket_v6,
G_SOCKET_FAMILY_IPV6, priv->multicast_iface,
priv->mcast_addr_v6->address, port, mcast_ttl);
}
}
return TRUE;
/* ERRORS */
no_udp_protocol:
{
GST_ERROR_OBJECT (stream, "failed to create udpsink element");
return FALSE;
}
get_port_failed:
{
GST_ERROR_OBJECT (stream, "failed to get udp port");
return FALSE;
}
}
/* must be called with lock */
static gboolean
create_and_configure_udpsource (GstElement ** udpsrc, GSocket * socket)
{
GstStateChangeReturn ret;
g_assert (socket != NULL);
*udpsrc = gst_element_factory_make ("udpsrc", NULL);
if (*udpsrc == NULL)
goto error;
g_object_set (G_OBJECT (*udpsrc), "socket", socket, NULL);
/* The udpsrc cannot do the join because its socket is always a local unicast
* one. The udpsink sharing the same socket will do it for us. */
g_object_set (G_OBJECT (*udpsrc), "auto-multicast", FALSE, NULL);
g_object_set (G_OBJECT (*udpsrc), "loop", FALSE, NULL);
g_object_set (G_OBJECT (*udpsrc), "close-socket", FALSE, NULL);
ret = gst_element_set_state (*udpsrc, GST_STATE_READY);
if (ret == GST_STATE_CHANGE_FAILURE)
goto error;
return TRUE;
/* ERRORS */
error:
{
if (*udpsrc) {
gst_element_set_state (*udpsrc, GST_STATE_NULL);
g_clear_object (udpsrc);
}
return FALSE;
}
}
static gboolean
alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
GSocket * socket_out[2], GstRTSPAddress ** server_addr_out,
gboolean multicast, GstRTSPTransport * ct, gboolean use_transport_settings)
{
GstRTSPStreamPrivate *priv = stream->priv;
GSocket *rtp_socket = NULL;
GSocket *rtcp_socket = NULL;
gint tmp_rtp, tmp_rtcp;
guint count;
GList *rejected_addresses = NULL;
GstRTSPAddress *addr = NULL;
GInetAddress *inetaddr = NULL;
GSocketAddress *rtp_sockaddr = NULL;
GSocketAddress *rtcp_sockaddr = NULL;
GstRTSPAddressPool *pool;
gboolean transport_settings_defined = FALSE;
pool = priv->pool;
count = 0;
/* Start with random port */
tmp_rtp = 0;
if (use_transport_settings) {
if (!multicast)
goto no_mcast;
if (ct == NULL)
goto no_transport;
/* multicast and transport specific case */
if (ct->destination != NULL) {
tmp_rtp = ct->port.min;
tmp_rtcp = ct->port.max;
/* check if the provided address is a multicast address */
inetaddr = g_inet_address_new_from_string (ct->destination);
if (inetaddr == NULL)
goto destination_error;
if (!g_inet_address_get_is_multicast (inetaddr))
goto destination_no_mcast;
if (!priv->bind_mcast_address) {
g_clear_object (&inetaddr);
inetaddr = g_inet_address_new_any (family);
}
GST_DEBUG_OBJECT (stream, "use transport settings");
transport_settings_defined = TRUE;
}
}
rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
G_SOCKET_PROTOCOL_UDP, NULL);
if (!rtcp_socket)
goto no_udp_protocol;
g_socket_set_multicast_loopback (rtcp_socket, FALSE);
/* 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;
g_socket_set_multicast_loopback (rtp_socket, FALSE);
}
if (!transport_settings_defined) {
if ((pool && gst_rtsp_address_pool_has_unicast_addresses (pool))
|| multicast) {
GstRTSPAddressFlags flags;
if (addr)
rejected_addresses = g_list_prepend (rejected_addresses, addr);
if (!pool)
goto no_pool;
flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT;
if (multicast)
flags |= GST_RTSP_ADDRESS_FLAG_MULTICAST;
else
flags |= GST_RTSP_ADDRESS_FLAG_UNICAST;
if (family == G_SOCKET_FAMILY_IPV6)
flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
else
flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
if (*server_addr_out)
addr = *server_addr_out;
else
addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
if (addr == NULL)
goto no_address;
tmp_rtp = addr->port;
g_clear_object (&inetaddr);
/* FIXME: Does it really work with the IP_MULTICAST_ALL socket option and
* socket control message set in udpsrc? */
if (priv->bind_mcast_address || !multicast)
inetaddr = g_inet_address_new_from_string (addr->address);
else
inetaddr = g_inet_address_new_any (family);
} 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)) {
GST_DEBUG_OBJECT (stream, "rtp bind() failed, will try again");
g_object_unref (rtp_sockaddr);
if (transport_settings_defined)
goto transport_settings_error;
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;
}
if (!transport_settings_defined) {
tmp_rtp =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
/* check if port is even. RFC 3550 encorages the use of an even/odd port
* pair, however it's not a strict requirement so this check is not done
* for the client selected ports. */
if ((tmp_rtp & 1) != 0) {
/* port not even, close and allocate another */
tmp_rtp++;
g_object_unref (rtp_sockaddr);
g_clear_object (&rtp_socket);
goto again;
}
}
g_object_unref (rtp_sockaddr);
/* 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)) {
GST_DEBUG_OBJECT (stream, "rctp bind() failed, will try again");
g_object_unref (rtcp_sockaddr);
g_clear_object (&rtp_socket);
if (transport_settings_defined)
goto transport_settings_error;
goto again;
}
g_object_unref (rtcp_sockaddr);
if (!addr) {
addr = g_slice_new0 (GstRTSPAddress);
addr->port = tmp_rtp;
addr->n_ports = 2;
if (transport_settings_defined)
addr->address = g_strdup (ct->destination);
else
addr->address = g_inet_address_to_string (inetaddr);
addr->ttl = ct->ttl;
}
g_clear_object (&inetaddr);
if (multicast && (ct->ttl > 0) && (ct->ttl <= priv->max_mcast_ttl)) {
GST_DEBUG ("setting mcast ttl to %d", ct->ttl);
g_socket_set_multicast_ttl (rtp_socket, ct->ttl);
g_socket_set_multicast_ttl (rtcp_socket, ct->ttl);
}
socket_out[0] = rtp_socket;
socket_out[1] = rtcp_socket;
*server_addr_out = addr;
GST_DEBUG_OBJECT (stream, "allocated address: %s and ports: %d, %d",
addr->address, tmp_rtp, tmp_rtcp);
g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
return TRUE;
/* ERRORS */
no_mcast:
{
GST_ERROR_OBJECT (stream, "failed to allocate UDP ports: wrong transport");
goto cleanup;
}
no_transport:
{
GST_ERROR_OBJECT (stream, "failed to allocate UDP ports: no transport");
goto cleanup;
}
destination_error:
{
GST_ERROR_OBJECT (stream,
"failed to allocate UDP ports: destination error");
goto cleanup;
}
destination_no_mcast:
{
GST_ERROR_OBJECT (stream,
"failed to allocate UDP ports: destination not multicast address");
goto cleanup;
}
no_udp_protocol:
{
GST_WARNING_OBJECT (stream, "failed to allocate UDP ports: protocol error");
goto cleanup;
}
no_pool:
{
GST_WARNING_OBJECT (stream,
"failed to allocate UDP ports: no address pool specified");
goto cleanup;
}
no_address:
{
GST_WARNING_OBJECT (stream, "failed to acquire address from pool");
goto cleanup;
}
no_ports:
{
GST_WARNING_OBJECT (stream, "failed to allocate UDP ports: no ports");
goto cleanup;
}
transport_settings_error:
{
GST_ERROR_OBJECT (stream,
"failed to allocate UDP ports with requested transport settings");
goto cleanup;
}
socket_error:
{
GST_WARNING_OBJECT (stream, "failed to allocate UDP ports: socket error");
goto cleanup;
}
cleanup:
{
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
add_mcast_client_addr (GstRTSPStream * stream, const gchar * destination,
guint rtp_port, guint rtcp_port)
{
GstRTSPStreamPrivate *priv;
GList *walk;
UdpClientAddrInfo *client;
GInetAddress *inet;
priv = stream->priv;
if (destination == NULL)
return FALSE;
inet = g_inet_address_new_from_string (destination);
if (inet == NULL)
goto invalid_address;
if (!g_inet_address_get_is_multicast (inet)) {
g_object_unref (inet);
goto invalid_address;
}
g_object_unref (inet);
for (walk = priv->mcast_clients; walk; walk = g_list_next (walk)) {
UdpClientAddrInfo *cli = walk->data;
if ((g_strcmp0 (cli->address, destination) == 0) &&
(cli->rtp_port == rtp_port)) {
GST_DEBUG ("requested destination already exists: %s:%u-%u",
destination, rtp_port, rtcp_port);
cli->add_count++;
return TRUE;
}
}
client = g_new0 (UdpClientAddrInfo, 1);
client->address = g_strdup (destination);
client->rtp_port = rtp_port;
client->add_count = 1;
priv->mcast_clients = g_list_prepend (priv->mcast_clients, client);
GST_DEBUG ("added mcast client %s:%u-%u", destination, rtp_port, rtcp_port);
return TRUE;
invalid_address:
{
GST_WARNING_OBJECT (stream, "Multicast address is invalid: %s",
destination);
return FALSE;
}
}
/* must be called with lock */
static gboolean
remove_mcast_client_addr (GstRTSPStream * stream, const gchar * destination,
guint rtp_port, guint rtcp_port)
{
GstRTSPStreamPrivate *priv;
GList *walk;
priv = stream->priv;
if (destination == NULL)
goto no_destination;
for (walk = priv->mcast_clients; walk; walk = g_list_next (walk)) {
UdpClientAddrInfo *cli = walk->data;
if ((g_strcmp0 (cli->address, destination) == 0) &&
(cli->rtp_port == rtp_port)) {
cli->add_count--;
if (!cli->add_count) {
priv->mcast_clients = g_list_remove (priv->mcast_clients, cli);
free_mcast_client (cli);
}
return TRUE;
}
}
GST_WARNING_OBJECT (stream, "Address not found");
return FALSE;
no_destination:
{
GST_WARNING_OBJECT (stream, "No destination has been provided");
return FALSE;
}
}
/**
* gst_rtsp_stream_allocate_udp_sockets:
* @stream: a #GstRTSPStream
* @family: protocol family
* @transport: transport method
* @use_client_settings: Whether to use client settings or not
*
* Allocates RTP and RTCP ports.
*
* Returns: %TRUE if the RTP and RTCP sockets have been succeccully allocated.
*/
gboolean
gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream,
GSocketFamily family, GstRTSPTransport * ct,
gboolean use_transport_settings)
{
GstRTSPStreamPrivate *priv;
gboolean ret = FALSE;
GstRTSPLowerTrans transport;
gboolean allocated = FALSE;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (ct != NULL, FALSE);
priv = stream->priv;
transport = ct->lower_transport;
g_mutex_lock (&priv->lock);
if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
if (family == G_SOCKET_FAMILY_IPV4 && priv->mcast_socket_v4[0])
allocated = TRUE;
else if (family == G_SOCKET_FAMILY_IPV6 && priv->mcast_socket_v6[0])
allocated = TRUE;
} else if (transport == GST_RTSP_LOWER_TRANS_UDP) {
if (family == G_SOCKET_FAMILY_IPV4 && priv->socket_v4[0])
allocated = TRUE;
else if (family == G_SOCKET_FAMILY_IPV6 && priv->socket_v6[0])
allocated = TRUE;
}
if (allocated) {
GST_DEBUG_OBJECT (stream, "Allocated already");
g_mutex_unlock (&priv->lock);
return TRUE;
}
if (family == G_SOCKET_FAMILY_IPV4) {
/* IPv4 */
if (transport == GST_RTSP_LOWER_TRANS_UDP) {
/* UDP unicast */
GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_UDP, ipv4");
ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
priv->socket_v4, &priv->server_addr_v4, FALSE, ct, FALSE);
} else {
/* multicast */
GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_MCAST_UDP, ipv4");
ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
priv->mcast_socket_v4, &priv->mcast_addr_v4, TRUE, ct,
use_transport_settings);
}
} else {
/* IPv6 */
if (transport == GST_RTSP_LOWER_TRANS_UDP) {
/* unicast */
GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_UDP, ipv6");
ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
priv->socket_v6, &priv->server_addr_v6, FALSE, ct, FALSE);
} else {
/* multicast */
GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_MCAST_UDP, ipv6");
ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
priv->mcast_socket_v6, &priv->mcast_addr_v6, TRUE, ct,
use_transport_settings);
}
}
g_mutex_unlock (&priv->lock);
return ret;
}
/**
* 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_is_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->joined_bin != NULL);
if (server_port) {
server_port->min = 0;
server_port->max = 0;
}
g_mutex_lock (&priv->lock);
if (family == G_SOCKET_FAMILY_IPV4) {
if (server_port && priv->server_addr_v4) {
server_port->min = priv->server_addr_v4->port;
server_port->max =
priv->server_addr_v4->port + priv->server_addr_v4->n_ports - 1;
}
} else {
if (server_port && priv->server_addr_v6) {
server_port->min = priv->server_addr_v6->port;
server_port->max =
priv->server_addr_v6->port + priv->server_addr_v6->n_ports - 1;
}
}
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_srtp_encoder:
* @stream: a #GstRTSPStream
*
* Get the SRTP encoder for this stream.
*
* Returns: (transfer full): The SRTP encoder for this stream. Unref after usage.
*/
GstElement *
gst_rtsp_stream_get_srtp_encoder (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
GstElement *encoder;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if ((encoder = priv->srtpenc))
g_object_ref (encoder);
g_mutex_unlock (&priv->lock);
return encoder;
}
/**
* 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->joined_bin != NULL);
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;
}
/**
* gst_rtsp_stream_set_max_mcast_ttl:
* @stream: a #GstRTSPStream
* @ttl: the new multicast ttl value
*
* Set the maximum time-to-live value of outgoing multicast packets.
*
* Returns: %TRUE if the requested ttl has been set successfully.
*
* Since: 1.16
*/
gboolean
gst_rtsp_stream_set_max_mcast_ttl (GstRTSPStream * stream, guint ttl)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_mutex_lock (&stream->priv->lock);
if (ttl == 0 || ttl > DEFAULT_MAX_MCAST_TTL) {
GST_WARNING_OBJECT (stream, "The reqested mcast TTL value is not valid.");
g_mutex_unlock (&stream->priv->lock);
return FALSE;
}
stream->priv->max_mcast_ttl = ttl;
g_mutex_unlock (&stream->priv->lock);
return TRUE;
}
/**
* gst_rtsp_stream_get_max_mcast_ttl:
* @stream: a #GstRTSPStream
*
* Get the the maximum time-to-live value of outgoing multicast packets.
*
* Returns: the maximum time-to-live value of outgoing multicast packets.
*
* Since: 1.16
*/
guint
gst_rtsp_stream_get_max_mcast_ttl (GstRTSPStream * stream)
{
guint ttl;
g_mutex_lock (&stream->priv->lock);
ttl = stream->priv->max_mcast_ttl;
g_mutex_unlock (&stream->priv->lock);
return ttl;
}
/**
* gst_rtsp_stream_verify_mcast_ttl:
* @stream: a #GstRTSPStream
* @ttl: a requested multicast ttl
*
* Check if the requested multicast ttl value is allowed.
*
* Returns: TRUE if the requested ttl value is allowed.
*
* Since: 1.16
*/
gboolean
gst_rtsp_stream_verify_mcast_ttl (GstRTSPStream * stream, guint ttl)
{
gboolean res = FALSE;
g_mutex_lock (&stream->priv->lock);
if ((ttl > 0) && (ttl <= stream->priv->max_mcast_ttl))
res = TRUE;
g_mutex_unlock (&stream->priv->lock);
return res;
}
/**
* gst_rtsp_stream_set_bind_mcast_address:
* @stream: a #GstRTSPStream,
* @bind_mcast_addr: the new value
*
* Decide whether the multicast socket should be bound to a multicast address or
* INADDR_ANY.
*
* Since: 1.16
*/
void
gst_rtsp_stream_set_bind_mcast_address (GstRTSPStream * stream,
gboolean bind_mcast_addr)
{
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
g_mutex_lock (&stream->priv->lock);
stream->priv->bind_mcast_address = bind_mcast_addr;
g_mutex_unlock (&stream->priv->lock);
}
/**
* gst_rtsp_stream_is_bind_mcast_address:
* @stream: a #GstRTSPStream
*
* Check if multicast sockets are configured to be bound to multicast addresses.
*
* Returns: %TRUE if multicast sockets are configured to be bound to multicast addresses.
*
* Since: 1.16
*/
gboolean
gst_rtsp_stream_is_bind_mcast_address (GstRTSPStream * stream)
{
gboolean result;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_mutex_lock (&stream->priv->lock);
result = stream->priv->bind_mcast_address;
g_mutex_unlock (&stream->priv->lock);
return result;
}
/* 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 ((g_ascii_strcasecmp (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)
{
if (priv->tr_cache)
g_ptr_array_unref (priv->tr_cache);
priv->tr_cache = NULL;
}
/* With lock taken */
static gboolean
any_transport_ready (GstRTSPStream * stream, gboolean is_rtp)
{
gboolean ret = TRUE;
GstRTSPStreamPrivate *priv = stream->priv;
GPtrArray *transports;
gint index;
transports = priv->tr_cache;
if (!transports)
goto done;
for (index = 0; index < transports->len; index++) {
GstRTSPStreamTransport *tr = g_ptr_array_index (transports, index);
if (!gst_rtsp_stream_transport_check_back_pressure (tr, is_rtp)) {
ret = TRUE;
break;
} else {
ret = FALSE;
}
}
done:
return ret;
}
/* Must be called *without* priv->lock */
static gboolean
push_data (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
GstBuffer * buffer, GstBufferList * buffer_list, gboolean is_rtp)
{
gboolean send_ret = TRUE;
if (is_rtp) {
if (buffer)
send_ret = gst_rtsp_stream_transport_send_rtp (trans, buffer);
if (buffer_list)
send_ret = gst_rtsp_stream_transport_send_rtp_list (trans, buffer_list);
} else {
if (buffer)
send_ret = gst_rtsp_stream_transport_send_rtcp (trans, buffer);
if (buffer_list)
send_ret = gst_rtsp_stream_transport_send_rtcp_list (trans, buffer_list);
}
return send_ret;
}
/* With priv->lock */
static void
ensure_cached_transports (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = stream->priv;
GList *walk;
if (priv->tr_cache_cookie != priv->transports_cookie) {
clear_tr_cache (priv);
priv->tr_cache =
g_ptr_array_new_full (priv->n_tcp_transports, g_object_unref);
for (walk = priv->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
const GstRTSPTransport *t = gst_rtsp_stream_transport_get_transport (tr);
if (t->lower_transport != GST_RTSP_LOWER_TRANS_TCP)
continue;
g_ptr_array_add (priv->tr_cache, g_object_ref (tr));
}
priv->tr_cache_cookie = priv->transports_cookie;
}
}
/* Must be called *without* priv->lock */
static void
check_transport_backlog (GstRTSPStream * stream, GstRTSPStreamTransport * trans)
{
GstRTSPStreamPrivate *priv = stream->priv;
gboolean send_ret = TRUE;
gst_rtsp_stream_transport_lock_backlog (trans);
if (!gst_rtsp_stream_transport_backlog_is_empty (trans)) {
GstBuffer *buffer;
GstBufferList *buffer_list;
gboolean is_rtp;
gboolean popped;
popped =
gst_rtsp_stream_transport_backlog_pop (trans, &buffer, &buffer_list,
&is_rtp);
g_assert (popped == TRUE);
send_ret = push_data (stream, trans, buffer, buffer_list, is_rtp);
gst_clear_buffer (&buffer);
gst_clear_buffer_list (&buffer_list);
}
gst_rtsp_stream_transport_unlock_backlog (trans);
if (!send_ret) {
/* remove transport on send error */
g_mutex_lock (&priv->lock);
update_transport (stream, trans, FALSE);
g_mutex_unlock (&priv->lock);
}
}
/* Must be called with priv->lock */
static void
send_tcp_message (GstRTSPStream * stream, gint idx)
{
GstRTSPStreamPrivate *priv = stream->priv;
GstAppSink *sink;
GstSample *sample;
GstBuffer *buffer;
GstBufferList *buffer_list;
guint n_messages = 0;
gboolean is_rtp;
GPtrArray *transports;
if (!priv->have_buffer[idx])
return;
ensure_cached_transports (stream);
is_rtp = (idx == 0);
if (!any_transport_ready (stream, is_rtp))
return;
priv->have_buffer[idx] = FALSE;
if (priv->appsink[idx] == NULL) {
/* session expired */
return;
}
sink = GST_APP_SINK (priv->appsink[idx]);
sample = gst_app_sink_pull_sample (sink);
if (!sample) {
return;
}
buffer = gst_sample_get_buffer (sample);
buffer_list = gst_sample_get_buffer_list (sample);
/* We will get one message-sent notification per buffer or
* complete buffer-list. We handle each buffer-list as a unit */
if (buffer)
n_messages += 1;
if (buffer_list)
n_messages += 1;
transports = priv->tr_cache;
if (transports)
g_ptr_array_ref (transports);
if (transports) {
gint index;
for (index = 0; index < transports->len; index++) {
GstRTSPStreamTransport *tr = g_ptr_array_index (transports, index);
GstBuffer *buf_ref = NULL;
GstBufferList *buflist_ref = NULL;
gst_rtsp_stream_transport_lock_backlog (tr);
if (buffer)
buf_ref = gst_buffer_ref (buffer);
if (buffer_list)
buflist_ref = gst_buffer_list_ref (buffer_list);
if (!gst_rtsp_stream_transport_backlog_push (tr,
buf_ref, buflist_ref, is_rtp)) {
GST_ERROR_OBJECT (stream,
"Dropping slow transport %" GST_PTR_FORMAT, tr);
update_transport (stream, tr, FALSE);
}
gst_rtsp_stream_transport_unlock_backlog (tr);
}
}
gst_sample_unref (sample);
g_mutex_unlock (&priv->lock);
if (transports) {
gint index;
for (index = 0; index < transports->len; index++) {
GstRTSPStreamTransport *tr = g_ptr_array_index (transports, index);
check_transport_backlog (stream, tr);
}
g_ptr_array_unref (transports);
}
g_mutex_lock (&priv->lock);
}
static gpointer
send_func (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = stream->priv;
gboolean cont = TRUE;
g_mutex_lock (&priv->send_lock);
while (cont) {
int i;
int idx = -1;
guint cookie;
cookie = priv->send_cookie;
g_mutex_unlock (&priv->send_lock);
g_mutex_lock (&priv->lock);
/* iterate from 1 and down, so we prioritize RTCP over RTP */
for (i = 1; i >= 0; i--) {
if (priv->have_buffer[i]) {
/* send message */
idx = i;
break;
}
}
if (idx != -1) {
send_tcp_message (stream, idx);
}
g_mutex_unlock (&priv->lock);
g_mutex_lock (&priv->send_lock);
while (cookie == priv->send_cookie) {
g_cond_wait (&priv->send_cond, &priv->send_lock);
}
cont = priv->continue_sending;
}
g_mutex_unlock (&priv->send_lock);
return NULL;
}
static GstFlowReturn
handle_new_sample (GstAppSink * sink, gpointer user_data)
{
GstRTSPStream *stream = user_data;
GstRTSPStreamPrivate *priv = stream->priv;
int i;
g_mutex_lock (&priv->lock);
for (i = 0; i < 2; i++) {
if (GST_ELEMENT_CAST (sink) == priv->appsink[i]) {
priv->have_buffer[i] = TRUE;
break;
}
}
if (priv->send_thread == NULL) {
priv->send_thread = g_thread_new (NULL, (GThreadFunc) send_func, user_data);
}
g_mutex_unlock (&priv->lock);
g_mutex_lock (&priv->send_lock);
priv->send_cookie++;
g_cond_signal (&priv->send_cond);
g_mutex_unlock (&priv->send_lock);
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);
g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_RTCP_DECODER],
0, priv->srtpdec);
}
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) (nullable): 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;
}
static void
add_rtx_pt (gpointer key, GstCaps * caps, GstStructure * pt_map)
{
guint pt = GPOINTER_TO_INT (key);
const GstStructure *s = gst_caps_get_structure (caps, 0);
const gchar *apt;
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "RTX") &&
(apt = gst_structure_get_string (s, "apt"))) {
gst_structure_set (pt_map, apt, G_TYPE_UINT, pt, NULL);
}
}
/* Call with priv->lock taken */
static void
update_rtx_receive_pt_map (GstRTSPStream * stream)
{
GstStructure *pt_map;
if (!stream->priv->rtxreceive)
goto done;
pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
g_hash_table_foreach (stream->priv->ptmap, (GHFunc) add_rtx_pt, pt_map);
g_object_set (stream->priv->rtxreceive, "payload-type-map", pt_map, NULL);
gst_structure_free (pt_map);
done:
return;
}
static void
retrieve_ulpfec_pt (gpointer key, GstCaps * caps, GstElement * ulpfec_decoder)
{
guint pt = GPOINTER_TO_INT (key);
const GstStructure *s = gst_caps_get_structure (caps, 0);
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
g_object_set (ulpfec_decoder, "pt", pt, NULL);
}
static void
update_ulpfec_decoder_pt (GstRTSPStream * stream)
{
if (!stream->priv->ulpfec_decoder)
goto done;
g_hash_table_foreach (stream->priv->ptmap, (GHFunc) retrieve_ulpfec_pt,
stream->priv->ulpfec_decoder);
done:
return;
}
/**
* gst_rtsp_stream_request_aux_receiver:
* @stream: a #GstRTSPStream
* @sessid: the session id
*
* Creating a rtxreceive bin
*
* Returns: (transfer full) (nullable): a #GstElement.
*
* Since: 1.16
*/
GstElement *
gst_rtsp_stream_request_aux_receiver (GstRTSPStream * stream, guint sessid)
{
GstElement *bin;
GstPad *pad;
gchar *name;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
bin = gst_bin_new (NULL);
stream->priv->rtxreceive = gst_element_factory_make ("rtprtxreceive", NULL);
update_rtx_receive_pt_map (stream);
update_ulpfec_decoder_pt (stream);
gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxreceive));
pad = gst_element_get_static_pad (stream->priv->rtxreceive, "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->rtxreceive, "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;
if (!GST_IS_CAPS (caps))
return;
g_mutex_lock (&priv->lock);
g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
update_rtx_receive_pt_map (stream);
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_stream_set_publish_clock_mode:
* @stream: a #GstRTSPStream
* @mode: the clock publish mode
*
* Sets if and how the stream clock should be published according to RFC7273.
*
* Since: 1.8
*/
void
gst_rtsp_stream_set_publish_clock_mode (GstRTSPStream * stream,
GstRTSPPublishClockMode mode)
{
GstRTSPStreamPrivate *priv;
priv = stream->priv;
g_mutex_lock (&priv->lock);
priv->publish_clock_mode = mode;
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_stream_get_publish_clock_mode:
* @stream: a #GstRTSPStream
*
* Gets if and how the stream clock should be published according to RFC7273.
*
* Returns: The GstRTSPPublishClockMode
*
* Since: 1.8
*/
GstRTSPPublishClockMode
gst_rtsp_stream_get_publish_clock_mode (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
GstRTSPPublishClockMode ret;
priv = stream->priv;
g_mutex_lock (&priv->lock);
ret = priv->publish_clock_mode;
g_mutex_unlock (&priv->lock);
return ret;
}
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 ());
}
typedef struct _ProbeData ProbeData;
struct _ProbeData
{
GstRTSPStream *stream;
/* existing sink, already linked to tee */
GstElement *sink1;
/* new sink, about to be linked */
GstElement *sink2;
/* new queue element, that will be linked to tee and sink1 */
GstElement **queue1;
/* new queue element, that will be linked to tee and sink2 */
GstElement **queue2;
GstPad *sink_pad;
GstPad *tee_pad;
guint index;
};
static void
free_cb_data (gpointer user_data)
{
ProbeData *data = user_data;
gst_object_unref (data->stream);
gst_object_unref (data->sink1);
gst_object_unref (data->sink2);
gst_object_unref (data->sink_pad);
gst_object_unref (data->tee_pad);
g_free (data);
}
static void
create_and_plug_queue_to_unlinked_stream (GstRTSPStream * stream,
GstElement * tee, GstElement * sink, GstElement ** queue)
{
GstRTSPStreamPrivate *priv = stream->priv;
GstPad *tee_pad;
GstPad *queue_pad;
GstPad *sink_pad;
/* create queue for the new stream */
*queue = gst_element_factory_make ("queue", NULL);
g_object_set (*queue, "max-size-buffers", 1, "max-size-bytes", 0,
"max-size-time", G_GINT64_CONSTANT (0), NULL);
gst_bin_add (priv->joined_bin, *queue);
/* link tee to queue */
tee_pad = gst_element_get_request_pad (tee, "src_%u");
queue_pad = gst_element_get_static_pad (*queue, "sink");
gst_pad_link (tee_pad, queue_pad);
gst_object_unref (queue_pad);
gst_object_unref (tee_pad);
/* link queue to sink */
queue_pad = gst_element_get_static_pad (*queue, "src");
sink_pad = gst_element_get_static_pad (sink, "sink");
gst_pad_link (queue_pad, sink_pad);
gst_object_unref (queue_pad);
gst_object_unref (sink_pad);
gst_element_sync_state_with_parent (sink);
gst_element_sync_state_with_parent (*queue);
}
static GstPadProbeReturn
create_and_plug_queue_to_linked_stream_probe_cb (GstPad * inpad,
GstPadProbeInfo * info, gpointer user_data)
{
GstRTSPStreamPrivate *priv;
ProbeData *data = user_data;
GstRTSPStream *stream;
GstElement **queue1;
GstElement **queue2;
GstPad *sink_pad;
GstPad *tee_pad;
GstPad *queue_pad;
guint index;
stream = data->stream;
priv = stream->priv;
queue1 = data->queue1;
queue2 = data->queue2;
sink_pad = data->sink_pad;
tee_pad = data->tee_pad;
index = data->index;
/* unlink tee and the existing sink:
* .-----. .---------.
* | tee | | sink1 |
* sink src->sink |
* '-----' '---------'
*/
g_assert (gst_pad_unlink (tee_pad, sink_pad));
/* add queue to the already existing stream */
*queue1 = gst_element_factory_make ("queue", NULL);
g_object_set (*queue1, "max-size-buffers", 1, "max-size-bytes", 0,
"max-size-time", G_GINT64_CONSTANT (0), NULL);
gst_bin_add (priv->joined_bin, *queue1);
/* link tee, queue and sink:
* .-----. .---------. .---------.
* | tee | | queue1 | | sink1 |
* sink src->sink src->sink |
* '-----' '---------' '---------'
*/
queue_pad = gst_element_get_static_pad (*queue1, "sink");
gst_pad_link (tee_pad, queue_pad);
gst_object_unref (queue_pad);
queue_pad = gst_element_get_static_pad (*queue1, "src");
gst_pad_link (queue_pad, sink_pad);
gst_object_unref (queue_pad);
gst_element_sync_state_with_parent (*queue1);
/* create queue and link it to tee and the new sink */
create_and_plug_queue_to_unlinked_stream (stream,
priv->tee[index], data->sink2, queue2);
/* the final stream:
*
* .-----. .---------. .---------.
* | tee | | queue1 | | sink1 |
* sink src->sink src->sink |
* | | '---------' '---------'
* | | .---------. .---------.
* | | | queue2 | | sink2 |
* | src->sink src->sink |
* '-----' '---------' '---------'
*/
return GST_PAD_PROBE_REMOVE;
}
static void
create_and_plug_queue_to_linked_stream (GstRTSPStream * stream,
GstElement * sink1, GstElement * sink2, guint index, GstElement ** queue1,
GstElement ** queue2)
{
ProbeData *data;
data = g_new0 (ProbeData, 1);
data->stream = gst_object_ref (stream);
data->sink1 = gst_object_ref (sink1);
data->sink2 = gst_object_ref (sink2);
data->queue1 = queue1;
data->queue2 = queue2;
data->index = index;
data->sink_pad = gst_element_get_static_pad (sink1, "sink");
g_assert (data->sink_pad);
data->tee_pad = gst_pad_get_peer (data->sink_pad);
g_assert (data->tee_pad);
gst_pad_add_probe (data->tee_pad, GST_PAD_PROBE_TYPE_IDLE,
create_and_plug_queue_to_linked_stream_probe_cb, data, free_cb_data);
}
static void
plug_udp_sink (GstRTSPStream * stream, GstElement * sink_to_plug,
GstElement ** queue_to_plug, guint index, gboolean is_mcast)
{
GstRTSPStreamPrivate *priv = stream->priv;
GstElement *existing_sink;
if (is_mcast)
existing_sink = priv->udpsink[index];
else
existing_sink = priv->mcast_udpsink[index];
GST_DEBUG_OBJECT (stream, "plug %s sink", is_mcast ? "mcast" : "udp");
/* add sink to the bin */
gst_bin_add (priv->joined_bin, sink_to_plug);
if (priv->appsink[index] && existing_sink) {
/* queues are already added for the existing stream, add one for
the newly added udp stream */
create_and_plug_queue_to_unlinked_stream (stream, priv->tee[index],
sink_to_plug, queue_to_plug);
} else if (priv->appsink[index] || existing_sink) {
GstElement **queue;
GstElement *element;
/* add queue to the already existing stream plus the newly created udp
stream */
if (priv->appsink[index]) {
element = priv->appsink[index];
queue = &priv->appqueue[index];
} else {
element = existing_sink;
if (is_mcast)
queue = &priv->udpqueue[index];
else
queue = &priv->mcast_udpqueue[index];
}
create_and_plug_queue_to_linked_stream (stream, element, sink_to_plug,
index, queue, queue_to_plug);
} else {
GstPad *tee_pad;
GstPad *sink_pad;
GST_DEBUG_OBJECT (stream, "creating first stream");
/* no need to add queues */
tee_pad = gst_element_get_request_pad (priv->tee[index], "src_%u");
sink_pad = gst_element_get_static_pad (sink_to_plug, "sink");
gst_pad_link (tee_pad, sink_pad);
gst_object_unref (tee_pad);
gst_object_unref (sink_pad);
}
gst_element_sync_state_with_parent (sink_to_plug);
}
static void
plug_tcp_sink (GstRTSPStream * stream, guint index)
{
GstRTSPStreamPrivate *priv = stream->priv;
GST_DEBUG_OBJECT (stream, "plug tcp sink");
/* add sink to the bin */
gst_bin_add (priv->joined_bin, priv->appsink[index]);
if (priv->mcast_udpsink[index] && priv->udpsink[index]) {
/* queues are already added for the existing stream, add one for
the newly added tcp stream */
create_and_plug_queue_to_unlinked_stream (stream,
priv->tee[index], priv->appsink[index], &priv->appqueue[index]);
} else if (priv->mcast_udpsink[index] || priv->udpsink[index]) {
GstElement **queue;
GstElement *element;
/* add queue to the already existing stream plus the newly created tcp
stream */
if (priv->mcast_udpsink[index]) {
element = priv->mcast_udpsink[index];
queue = &priv->mcast_udpqueue[index];
} else {
element = priv->udpsink[index];
queue = &priv->udpqueue[index];
}
create_and_plug_queue_to_linked_stream (stream, element,
priv->appsink[index], index, queue, &priv->appqueue[index]);
} else {
GstPad *tee_pad;
GstPad *sink_pad;
/* no need to add queues */
tee_pad = gst_element_get_request_pad (priv->tee[index], "src_%u");
sink_pad = gst_element_get_static_pad (priv->appsink[index], "sink");
gst_pad_link (tee_pad, sink_pad);
gst_object_unref (tee_pad);
gst_object_unref (sink_pad);
}
gst_element_sync_state_with_parent (priv->appsink[index]);
}
static void
plug_sink (GstRTSPStream * stream, const GstRTSPTransport * transport,
guint index)
{
GstRTSPStreamPrivate *priv;
gboolean is_tcp, is_udp, is_mcast;
priv = stream->priv;
is_tcp = transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP;
is_udp = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP;
is_mcast = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST;
if (is_udp)
plug_udp_sink (stream, priv->udpsink[index],
&priv->udpqueue[index], index, FALSE);
else if (is_mcast)
plug_udp_sink (stream, priv->mcast_udpsink[index],
&priv->mcast_udpqueue[index], index, TRUE);
else if (is_tcp)
plug_tcp_sink (stream, index);
}
/* must be called with lock */
static gboolean
create_sender_part (GstRTSPStream * stream, const GstRTSPTransport * transport)
{
GstRTSPStreamPrivate *priv;
GstPad *pad;
GstBin *bin;
gboolean is_tcp, is_udp, is_mcast;
gint mcast_ttl = 0;
gint i;
GST_DEBUG_OBJECT (stream, "create sender part");
priv = stream->priv;
bin = priv->joined_bin;
is_tcp = transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP;
is_udp = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP;
is_mcast = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST;
if (is_mcast)
mcast_ttl = transport->ttl;
GST_DEBUG_OBJECT (stream, "tcp: %d, udp: %d, mcast: %d (ttl: %d)", is_tcp,
is_udp, is_mcast, mcast_ttl);
if (is_udp && !priv->server_addr_v4 && !priv->server_addr_v6) {
GST_WARNING_OBJECT (stream, "no sockets assigned for UDP");
return FALSE;
}
if (is_mcast && !priv->mcast_addr_v4 && !priv->mcast_addr_v6) {
GST_WARNING_OBJECT (stream, "no sockets assigned for UDP multicast");
return FALSE;
}
if (g_object_class_find_property (G_OBJECT_GET_CLASS (priv->payloader),
"onvif-no-rate-control"))
g_object_set (priv->payloader, "onvif-no-rate-control",
!priv->do_rate_control, NULL);
for (i = 0; i < 2; i++) {
gboolean link_tee = FALSE;
/* For the sender we create this bit of pipeline for both
* RTP and RTCP.
* Initially there will be only one active transport for
* the stream, so the pipeline will look like this:
*
* .--------. .-----. .---------.
* | rtpbin | | tee | | sink |
* | send->sink src->sink |
* '--------' '-----' '---------'
*
* For each new transport, the already existing branch will
* be reconfigured by adding a queue element:
*
* .--------. .-----. .---------. .---------.
* | rtpbin | | tee | | queue | | udpsink |
* | send->sink src->sink src->sink |
* '--------' | | '---------' '---------'
* | | .---------. .---------.
* | | | queue | | udpsink |
* | src->sink src->sink |
* | | '---------' '---------'
* | | .---------. .---------.
* | | | queue | | appsink |
* | src->sink src->sink |
* '-----' '---------' '---------'
*/
/* Only link the RTP send src if we're going to send RTP, link
* the RTCP send src always */
if (!priv->srcpad && i == 0)
continue;
if (!priv->tee[i]) {
/* make tee for RTP/RTCP */
priv->tee[i] = gst_element_factory_make ("tee", NULL);
gst_bin_add (bin, priv->tee[i]);
link_tee = TRUE;
}
if (is_udp && !priv->udpsink[i]) {
/* we create only one pair of udpsinks for IPv4 and IPv6 */
create_and_configure_udpsink (stream, &priv->udpsink[i],
priv->socket_v4[i], priv->socket_v6[i], FALSE, (i == 0), mcast_ttl);
plug_sink (stream, transport, i);
} else if (is_mcast && !priv->mcast_udpsink[i]) {
/* we create only one pair of mcast-udpsinks for IPv4 and IPv6 */
create_and_configure_udpsink (stream, &priv->mcast_udpsink[i],
priv->mcast_socket_v4[i], priv->mcast_socket_v6[i], TRUE, (i == 0),
mcast_ttl);
plug_sink (stream, transport, i);
} else if (is_tcp && !priv->appsink[i]) {
/* make appsink */
priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
g_object_set (priv->appsink[i], "emit-signals", FALSE, "buffer-list",
TRUE, "max-buffers", 1, NULL);
if (i == 0)
g_object_set (priv->appsink[i], "sync", priv->do_rate_control, NULL);
/* we need to set sync and preroll to FALSE for the sink to avoid
* deadlock. This is only needed for sink sending RTCP data. */
if (i == 1)
g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
&sink_cb, stream, NULL);
plug_sink (stream, transport, i);
}
if (link_tee) {
/* and link to rtpbin send pad */
gst_element_sync_state_with_parent (priv->tee[i]);
pad = gst_element_get_static_pad (priv->tee[i], "sink");
gst_pad_link (priv->send_src[i], pad);
gst_object_unref (pad);
}
}
return TRUE;
}
/* must be called with lock */
static void
plug_src (GstRTSPStream * stream, GstBin * bin, GstElement * src,
GstElement * funnel)
{
GstRTSPStreamPrivate *priv;
GstPad *pad, *selpad;
gulong id = 0;
priv = stream->priv;
/* add src */
gst_bin_add (bin, src);
pad = gst_element_get_static_pad (src, "src");
if (priv->srcpad) {
/* block pad so src can't push data while it's not yet linked */
id = gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BLOCK |
GST_PAD_PROBE_TYPE_BUFFER, NULL, NULL, NULL);
/* 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 (src, GST_STATE_PLAYING);
gst_element_set_locked_state (src, TRUE);
}
/* and link to the funnel */
selpad = gst_element_get_request_pad (funnel, "sink_%u");
gst_pad_link (pad, selpad);
if (id != 0)
gst_pad_remove_probe (pad, id);
gst_object_unref (pad);
gst_object_unref (selpad);
}
/* must be called with lock */
static gboolean
create_receiver_part (GstRTSPStream * stream, const GstRTSPTransport *
transport)
{
GstRTSPStreamPrivate *priv;
GstPad *pad;
GstBin *bin;
gboolean tcp;
gboolean udp;
gboolean mcast;
gint i;
GST_DEBUG_OBJECT (stream, "create receiver part");
priv = stream->priv;
bin = priv->joined_bin;
tcp = transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP;
udp = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP;
mcast = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST;
for (i = 0; i < 2; i++) {
/* 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 |
* | RTP src->sink src->sink |
* '--------' | | | |
* .--------. | | | |
* | appsrc | | | | |
* | RTP src->sink | | |
* '--------' '--------' | |
* | |
* .--------. .--------. | |
* | udpsrc | | funnel | | |
* | RTCP src->sink src->sink |
* '--------' | | '--------'
* .--------. | |
* | appsrc | | |
* | RTCP src->sink |
* '--------' '--------'
*/
if (!priv->sinkpad && i == 0) {
/* Only connect recv RTP sink if we expect to receive RTP. Connect recv
* RTCP sink always */
continue;
}
/* make funnel for the RTP/RTCP receivers */
if (!priv->funnel[i]) {
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 (udp && !priv->udpsrc_v4[i] && priv->server_addr_v4) {
GST_DEBUG_OBJECT (stream, "udp IPv4, create and configure udpsources");
if (!create_and_configure_udpsource (&priv->udpsrc_v4[i],
priv->socket_v4[i]))
goto udpsrc_error;
plug_src (stream, bin, priv->udpsrc_v4[i], priv->funnel[i]);
}
if (udp && !priv->udpsrc_v6[i] && priv->server_addr_v6) {
GST_DEBUG_OBJECT (stream, "udp IPv6, create and configure udpsources");
if (!create_and_configure_udpsource (&priv->udpsrc_v6[i],
priv->socket_v6[i]))
goto udpsrc_error;
plug_src (stream, bin, priv->udpsrc_v6[i], priv->funnel[i]);
}
if (mcast && !priv->mcast_udpsrc_v4[i] && priv->mcast_addr_v4) {
GST_DEBUG_OBJECT (stream, "mcast IPv4, create and configure udpsources");
if (!create_and_configure_udpsource (&priv->mcast_udpsrc_v4[i],
priv->mcast_socket_v4[i]))
goto mcast_udpsrc_error;
plug_src (stream, bin, priv->mcast_udpsrc_v4[i], priv->funnel[i]);
}
if (mcast && !priv->mcast_udpsrc_v6[i] && priv->mcast_addr_v6) {
GST_DEBUG_OBJECT (stream, "mcast IPv6, create and configure udpsources");
if (!create_and_configure_udpsource (&priv->mcast_udpsrc_v6[i],
priv->mcast_socket_v6[i]))
goto mcast_udpsrc_error;
plug_src (stream, bin, priv->mcast_udpsrc_v6[i], priv->funnel[i]);
}
if (tcp && !priv->appsrc[i]) {
/* 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, "is-live",
TRUE, NULL);
plug_src (stream, bin, priv->appsrc[i], priv->funnel[i]);
}
gst_element_sync_state_with_parent (priv->funnel[i]);
}
return TRUE;
mcast_udpsrc_error:
udpsrc_error:
return FALSE;
}
static gboolean
check_mcast_client_addr (GstRTSPStream * stream, const GstRTSPTransport * tr)
{
GstRTSPStreamPrivate *priv = stream->priv;
GList *walk;
if (priv->mcast_clients == NULL)
goto no_addr;
if (tr == NULL)
goto no_transport;
if (tr->destination == NULL)
goto no_destination;
for (walk = priv->mcast_clients; walk; walk = g_list_next (walk)) {
UdpClientAddrInfo *cli = walk->data;
if ((g_strcmp0 (cli->address, tr->destination) == 0) &&
(cli->rtp_port == tr->port.min))
return TRUE;
}
return FALSE;
no_addr:
{
GST_WARNING_OBJECT (stream, "Adding mcast transport, but no mcast address "
"has been reserved");
return FALSE;
}
no_transport:
{
GST_WARNING_OBJECT (stream, "Adding mcast transport, but no transport "
"has been provided");
return FALSE;
}
no_destination:
{
GST_WARNING_OBJECT (stream, "Adding mcast transport, but it doesn't match "
"the reserved address");
return FALSE;
}
}
/**
* 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;
guint idx;
gchar *name;
GstPadLinkReturn ret;
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->joined_bin != NULL)
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);
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 {
/* RECORD case: 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);
g_object_set (priv->session, "disable-sr-timestamp", !priv->do_rate_control,
NULL);
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->caps = gst_pad_get_current_caps (priv->send_src[0]);
}
priv->joined_bin = bin;
GST_DEBUG_OBJECT (stream, "successfully joined bin");
g_mutex_unlock (&priv->lock);
return TRUE;
/* ERRORS */
was_joined:
{
g_mutex_unlock (&priv->lock);
return TRUE;
}
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;
}
}
static void
clear_element (GstBin * bin, GstElement ** elementptr)
{
if (*elementptr) {
gst_element_set_locked_state (*elementptr, FALSE);
gst_element_set_state (*elementptr, GST_STATE_NULL);
if (GST_ELEMENT_PARENT (*elementptr))
gst_bin_remove (bin, *elementptr);
else
gst_object_unref (*elementptr);
*elementptr = NULL;
}
}
/**
* 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;
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->send_lock);
priv->continue_sending = FALSE;
priv->send_cookie++;
g_cond_signal (&priv->send_cond);
g_mutex_unlock (&priv->send_lock);
if (priv->send_thread) {
g_thread_join (priv->send_thread);
}
g_mutex_lock (&priv->lock);
if (priv->joined_bin == NULL)
goto was_not_joined;
if (priv->joined_bin != bin)
goto wrong_bin;
priv->joined_bin = NULL;
/* all transports must be removed by now */
if (priv->transports != NULL)
goto transports_not_removed;
if (priv->send_pool) {
GThreadPool *slask;
slask = priv->send_pool;
priv->send_pool = NULL;
g_mutex_unlock (&priv->lock);
g_thread_pool_free (slask, TRUE, TRUE);
g_mutex_lock (&priv->lock);
}
clear_tr_cache (priv);
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;
}
for (i = 0; i < 2; i++) {
clear_element (bin, &priv->udpsrc_v4[i]);
clear_element (bin, &priv->udpsrc_v6[i]);
clear_element (bin, &priv->udpqueue[i]);
clear_element (bin, &priv->udpsink[i]);
clear_element (bin, &priv->mcast_udpsrc_v4[i]);
clear_element (bin, &priv->mcast_udpsrc_v6[i]);
clear_element (bin, &priv->mcast_udpqueue[i]);
clear_element (bin, &priv->mcast_udpsink[i]);
clear_element (bin, &priv->appsrc[i]);
clear_element (bin, &priv->appqueue[i]);
clear_element (bin, &priv->appsink[i]);
clear_element (bin, &priv->tee[i]);
clear_element (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;
}
}
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);
if (priv->mcast_addr_v4)
gst_rtsp_address_free (priv->mcast_addr_v4);
priv->mcast_addr_v4 = NULL;
if (priv->mcast_addr_v6)
gst_rtsp_address_free (priv->mcast_addr_v6);
priv->mcast_addr_v6 = NULL;
if (priv->server_addr_v4)
gst_rtsp_address_free (priv->server_addr_v4);
priv->server_addr_v4 = NULL;
if (priv->server_addr_v6)
gst_rtsp_address_free (priv->server_addr_v6);
priv->server_addr_v6 = NULL;
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;
}
wrong_bin:
{
GST_ERROR_OBJECT (stream, "leaving the wrong bin");
g_mutex_unlock (&priv->lock);
return FALSE;
}
}
/**
* gst_rtsp_stream_get_joined_bin:
* @stream: a #GstRTSPStream
*
* Get the previous joined bin with gst_rtsp_stream_join_bin() or NULL.
*
* Return: (transfer full) (nullable): the joined bin or NULL.
*/
GstBin *
gst_rtsp_stream_get_joined_bin (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
GstBin *bin = NULL;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
bin = priv->joined_bin ? gst_object_ref (priv->joined_bin) : NULL;
g_mutex_unlock (&priv->lock);
return bin;
}
/**
* gst_rtsp_stream_get_rtpinfo:
* @stream: a #GstRTSPStream
* @rtptime: (allow-none) (out caller-allocates): result RTP timestamp
* @seq: (allow-none) (out caller-allocates): result RTP seqnum
* @clock_rate: (allow-none) (out caller-allocates): the clock rate
* @running_time: (out caller-allocates): 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->mcast_udpsink[0] || priv->appsink[0]) {
GstSample *last_sample;
if (priv->udpsink[0])
g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
else if (priv->mcast_udpsink[0])
g_object_get (priv->mcast_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;
GstStructure *s;
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);
s = gst_caps_get_structure (caps, 0);
if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
guint ssrc_buf = gst_rtp_buffer_get_ssrc (&rtp_buffer);
guint ssrc_stream = 0;
if (gst_structure_has_field_typed (s, "ssrc", G_TYPE_UINT) &&
gst_structure_get_uint (s, "ssrc", &ssrc_stream) &&
ssrc_buf != ssrc_stream) {
/* Skip buffers from auxiliary streams. */
GST_DEBUG_OBJECT (stream,
"not a buffer from the payloader, SSRC: %08x", ssrc_buf);
gst_rtp_buffer_unmap (&rtp_buffer);
gst_sample_unref (last_sample);
goto stats;
}
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) {
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);
}
}
}
stats:
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-offset", 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_rates:
* @stream: a #GstRTSPStream
* @rate: (optional) (out caller-allocates): the configured rate
* @applied_rate: (optional) (out caller-allocates): the configured applied_rate
*
* Retrieve the current rate and/or applied_rate.
*
* Returns: %TRUE if rate and/or applied_rate could be determined.
* Since: 1.18
*/
gboolean
gst_rtsp_stream_get_rates (GstRTSPStream * stream, gdouble * rate,
gdouble * applied_rate)
{
GstRTSPStreamPrivate *priv;
GstEvent *event;
const GstSegment *segment;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
if (!rate && !applied_rate) {
GST_WARNING_OBJECT (stream, "rate and applied_rate are both NULL");
return FALSE;
}
priv = stream->priv;
g_mutex_lock (&priv->lock);
if (!priv->send_rtp_sink)
goto no_rtp_sink_pad;
event = gst_pad_get_sticky_event (priv->send_rtp_sink, GST_EVENT_SEGMENT, 0);
if (!event)
goto no_sticky_event;
gst_event_parse_segment (event, &segment);
if (rate)
*rate = segment->rate;
if (applied_rate)
*applied_rate = segment->applied_rate;
gst_event_unref (event);
g_mutex_unlock (&priv->lock);
return TRUE;
/* ERRORS */
no_rtp_sink_pad:
{
GST_WARNING_OBJECT (stream, "no send_rtp_sink pad yet");
g_mutex_unlock (&priv->lock);
return FALSE;
}
no_sticky_event:
{
GST_WARNING_OBJECT (stream, "no segment event on send_rtp_sink pad");
g_mutex_unlock (&priv->lock);
return FALSE;
}
}
/**
* gst_rtsp_stream_get_caps:
* @stream: a #GstRTSPStream
*
* Retrieve the current caps of @stream.
*
* Returns: (transfer full) (nullable): 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->joined_bin != NULL, 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->joined_bin == NULL) {
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 inline void
add_client (GstElement * rtp_sink, GstElement * rtcp_sink, const gchar * host,
gint rtp_port, gint rtcp_port)
{
if (rtp_sink != NULL)
g_signal_emit_by_name (rtp_sink, "add", host, rtp_port, NULL);
if (rtcp_sink != NULL)
g_signal_emit_by_name (rtcp_sink, "add", host, rtcp_port, NULL);
}
/* must be called with lock */
static void
remove_client (GstElement * rtp_sink, GstElement * rtcp_sink,
const gchar * host, gint rtp_port, gint rtcp_port)
{
if (rtp_sink != NULL)
g_signal_emit_by_name (rtp_sink, "remove", host, rtp_port, NULL);
if (rtcp_sink != NULL)
g_signal_emit_by_name (rtcp_sink, "remove", host, rtcp_port, NULL);
}
/* must be called with lock */
static gboolean
update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
gboolean add)
{
GstRTSPStreamPrivate *priv = stream->priv;
const GstRTSPTransport *tr;
gchar *dest;
gint min, max;
GList *tr_element;
tr = gst_rtsp_stream_transport_get_transport (trans);
dest = tr->destination;
tr_element = g_list_find (priv->transports, trans);
if (add && tr_element)
return TRUE;
else if (!add && !tr_element)
return FALSE;
switch (tr->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
{
min = tr->port.min;
max = tr->port.max;
if (add) {
GST_INFO ("adding %s:%d-%d", dest, min, max);
if (!check_mcast_client_addr (stream, tr))
goto mcast_error;
add_client (priv->mcast_udpsink[0], priv->mcast_udpsink[1], dest, min,
max);
if (tr->ttl > 0) {
GST_INFO ("setting ttl-mc %d", tr->ttl);
if (priv->mcast_udpsink[0])
g_object_set (G_OBJECT (priv->mcast_udpsink[0]), "ttl-mc", tr->ttl,
NULL);
if (priv->mcast_udpsink[1])
g_object_set (G_OBJECT (priv->mcast_udpsink[1]), "ttl-mc", tr->ttl,
NULL);
}
priv->transports = g_list_prepend (priv->transports, trans);
} else {
GST_INFO ("removing %s:%d-%d", dest, min, max);
if (!remove_mcast_client_addr (stream, dest, min, max))
GST_WARNING_OBJECT (stream,
"Failed to remove multicast address: %s:%d-%d", dest, min, max);
priv->transports = g_list_delete_link (priv->transports, tr_element);
remove_client (priv->mcast_udpsink[0], priv->mcast_udpsink[1], dest,
min, max);
}
break;
}
case GST_RTSP_LOWER_TRANS_UDP:
{
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) {
GST_INFO ("adding %s:%d-%d", dest, min, max);
add_client (priv->udpsink[0], priv->udpsink[1], dest, min, max);
priv->transports = g_list_prepend (priv->transports, trans);
} else {
GST_INFO ("removing %s:%d-%d", dest, min, max);
priv->transports = g_list_delete_link (priv->transports, tr_element);
remove_client (priv->udpsink[0], priv->udpsink[1], dest, min, max);
}
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);
priv->n_tcp_transports++;
} else {
GST_INFO ("removing TCP %s", tr->destination);
priv->transports = g_list_delete_link (priv->transports, tr_element);
gst_rtsp_stream_transport_lock_backlog (trans);
gst_rtsp_stream_transport_clear_backlog (trans);
gst_rtsp_stream_transport_unlock_backlog (trans);
priv->n_tcp_transports--;
}
priv->transports_cookie++;
break;
default:
goto unknown_transport;
}
return TRUE;
/* ERRORS */
unknown_transport:
{
GST_INFO ("Unknown transport %d", tr->lower_transport);
return FALSE;
}
mcast_error:
{
return FALSE;
}
}
static void
on_message_sent (GstRTSPStreamTransport * trans, gpointer user_data)
{
GstRTSPStream *stream = GST_RTSP_STREAM (user_data);
GstRTSPStreamPrivate *priv = stream->priv;
GST_DEBUG_OBJECT (stream, "message send complete");
check_transport_backlog (stream, trans);
g_mutex_lock (&priv->send_lock);
priv->send_cookie++;
g_cond_signal (&priv->send_cond);
g_mutex_unlock (&priv->send_lock);
}
/**
* 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. Adding the
* same transport twice will not add it a second time.
*
* @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->joined_bin != NULL, FALSE);
g_mutex_lock (&priv->lock);
res = update_transport (stream, trans, TRUE);
if (res)
gst_rtsp_stream_transport_set_message_sent_full (trans, on_message_sent,
stream, NULL);
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->joined_bin != NULL, 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 = stream->priv;
GSocket *socket;
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_mutex_lock (&priv->lock);
if (family == G_SOCKET_FAMILY_IPV6)
socket = priv->socket_v6[0];
else
socket = priv->socket_v4[0];
if (socket != NULL)
socket = g_object_ref (socket);
g_mutex_unlock (&priv->lock);
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 = stream->priv;
GSocket *socket;
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_mutex_lock (&priv->lock);
if (family == G_SOCKET_FAMILY_IPV6)
socket = priv->socket_v6[1];
else
socket = priv->socket_v4[1];
if (socket != NULL)
socket = g_object_ref (socket);
g_mutex_unlock (&priv->lock);
return socket;
}
/**
* gst_rtsp_stream_get_rtp_multicast_socket:
* @stream: a #GstRTSPStream
* @family: the socket family
*
* Get the multicast RTP socket from @stream for a @family.
*
* Returns: (transfer full) (nullable): the multicast RTP socket or %NULL if no
*
* socket could be allocated for @family. Unref after usage
*/
GSocket *
gst_rtsp_stream_get_rtp_multicast_socket (GstRTSPStream * stream,
GSocketFamily family)
{
GstRTSPStreamPrivate *priv = stream->priv;
GSocket *socket;
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_mutex_lock (&priv->lock);
if (family == G_SOCKET_FAMILY_IPV6)
socket = priv->mcast_socket_v6[0];
else
socket = priv->mcast_socket_v4[0];
if (socket != NULL)
socket = g_object_ref (socket);
g_mutex_unlock (&priv->lock);
return socket;
}
/**
* gst_rtsp_stream_get_rtcp_multicast_socket:
* @stream: a #GstRTSPStream
* @family: the socket family
*
* Get the multicast RTCP socket from @stream for a @family.
*
* Returns: (transfer full) (nullable): the multicast RTCP socket or %NULL if no
* socket could be allocated for @family. Unref after usage
*
* Since: 1.14
*/
GSocket *
gst_rtsp_stream_get_rtcp_multicast_socket (GstRTSPStream * stream,
GSocketFamily family)
{
GstRTSPStreamPrivate *priv = stream->priv;
GSocket *socket;
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_mutex_lock (&priv->lock);
if (family == G_SOCKET_FAMILY_IPV6)
socket = priv->mcast_socket_v6[1];
else
socket = priv->mcast_socket_v4[1];
if (socket != NULL)
socket = g_object_ref (socket);
g_mutex_unlock (&priv->lock);
return socket;
}
/**
* gst_rtsp_stream_add_multicast_client_address:
* @stream: a #GstRTSPStream
* @destination: (transfer none): a multicast address to add
* @rtp_port: RTP port
* @rtcp_port: RTCP port
* @family: socket family
*
* Add multicast client address to stream. At this point, the sockets that
* will stream RTP and RTCP data to @destination are supposed to be
* allocated.
*
* Returns: %TRUE if @destination can be addedd and handled by @stream.
*
* Since: 1.16
*/
gboolean
gst_rtsp_stream_add_multicast_client_address (GstRTSPStream * stream,
const gchar * destination, guint rtp_port, guint rtcp_port,
GSocketFamily family)
{
GstRTSPStreamPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (destination != NULL, FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if ((family == G_SOCKET_FAMILY_IPV4) && (priv->mcast_socket_v4[0] == NULL))
goto socket_error;
else if ((family == G_SOCKET_FAMILY_IPV6) &&
(priv->mcast_socket_v6[0] == NULL))
goto socket_error;
if (!add_mcast_client_addr (stream, destination, rtp_port, rtcp_port))
goto add_addr_error;
g_mutex_unlock (&priv->lock);
return TRUE;
socket_error:
{
GST_WARNING_OBJECT (stream,
"Failed to add multicast address: no udp socket");
g_mutex_unlock (&priv->lock);
return FALSE;
}
add_addr_error:
{
GST_WARNING_OBJECT (stream,
"Failed to add multicast address: invalid address");
g_mutex_unlock (&priv->lock);
return FALSE;
}
}
/**
* gst_rtsp_stream_get_multicast_client_addresses
* @stream: a #GstRTSPStream
*
* Get all multicast client addresses that RTP data will be sent to
*
* Returns: A comma separated list of host:port pairs with destinations
*
* Since: 1.16
*/
gchar *
gst_rtsp_stream_get_multicast_client_addresses (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
GString *str;
GList *clients;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
priv = stream->priv;
str = g_string_new ("");
g_mutex_lock (&priv->lock);
clients = priv->mcast_clients;
while (clients != NULL) {
UdpClientAddrInfo *client;
client = (UdpClientAddrInfo *) clients->data;
clients = g_list_next (clients);
g_string_append_printf (str, "%s:%d%s", client->address, client->rtp_port,
(clients != NULL ? "," : ""));
}
g_mutex_unlock (&priv->lock);
return g_string_free (str, FALSE);
}
/**
* 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;
GstBuffer *buffer = NULL;
stream = user_data;
priv = stream->priv;
GST_DEBUG_OBJECT (pad, "now blocking");
g_mutex_lock (&priv->lock);
priv->blocking = TRUE;
if ((info->type & GST_PAD_PROBE_TYPE_BUFFER)) {
buffer = gst_pad_probe_info_get_buffer (info);
} else if ((info->type & GST_PAD_PROBE_TYPE_BUFFER_LIST)) {
GstBufferList *list = gst_pad_probe_info_get_buffer_list (info);
buffer = gst_buffer_list_get (list, 0);
} else {
g_assert_not_reached ();
}
g_assert (buffer);
priv->position = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (stream, "buffer position: %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
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;
}
static void
set_blocked (GstRTSPStream * stream, gboolean blocked)
{
GstRTSPStreamPrivate *priv;
int i;
GST_DEBUG_OBJECT (stream, "blocked: %d", blocked);
priv = stream->priv;
if (blocked) {
/* if receiver */
if (priv->sinkpad) {
priv->blocking = TRUE;
return;
}
for (i = 0; i < 2; i++) {
if (priv->blocked_id[i] != 0)
continue;
if (priv->send_src[i]) {
priv->blocking = FALSE;
priv->blocked_id[i] = gst_pad_add_probe (priv->send_src[i],
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 {
for (i = 0; i < 2; i++) {
if (priv->blocked_id[i] != 0) {
gst_pad_remove_probe (priv->send_src[i], priv->blocked_id[i]);
priv->blocked_id[i] = 0;
}
}
priv->blocking = FALSE;
}
}
/**
* 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);
set_blocked (stream, blocked);
g_mutex_unlock (&priv->lock);
return TRUE;
}
/**
* gst_rtsp_stream_ublock_linked:
* @stream: a #GstRTSPStream
*
* Unblocks the dataflow on @stream if it is linked.
*
* Returns: %TRUE on success
*
* Since: 1.14
*/
gboolean
gst_rtsp_stream_unblock_linked (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
if (priv->send_src[0] && gst_pad_is_linked (priv->send_src[0]))
set_blocked (stream, 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
* @position: (out): current position of 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;
GstPad *pad = NULL;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
/* query position: if no sinks have been added yet,
* we obtain the position from the pad otherwise we query the sinks */
priv = stream->priv;
g_mutex_lock (&priv->lock);
/* depending on the transport type, it should query corresponding sink */
if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP)
sink = priv->udpsink[0];
else if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
sink = priv->mcast_udpsink[0];
else
sink = priv->appsink[0];
if (sink) {
gst_object_ref (sink);
} else if (priv->send_src[0]) {
pad = gst_object_ref (priv->send_src[0]);
} else {
g_mutex_unlock (&priv->lock);
GST_WARNING_OBJECT (stream, "Couldn't obtain postion: erroneous pipeline");
return FALSE;
}
g_mutex_unlock (&priv->lock);
if (sink) {
if (!gst_element_query_position (sink, GST_FORMAT_TIME, position)) {
GST_WARNING_OBJECT (stream,
"Couldn't obtain postion: position query failed");
gst_object_unref (sink);
return FALSE;
}
gst_object_unref (sink);
} else if (pad) {
GstEvent *event;
const GstSegment *segment;
event = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0);
if (!event) {
GST_WARNING_OBJECT (stream, "Couldn't obtain postion: no segment event");
gst_object_unref (pad);
return FALSE;
}
gst_event_parse_segment (event, &segment);
if (segment->format != GST_FORMAT_TIME) {
*position = -1;
} else {
g_mutex_lock (&priv->lock);
*position = priv->position;
g_mutex_unlock (&priv->lock);
*position =
gst_segment_to_stream_time (segment, GST_FORMAT_TIME, *position);
}
gst_event_unref (event);
gst_object_unref (pad);
}
return TRUE;
}
/**
* gst_rtsp_stream_query_stop:
* @stream: a #GstRTSPStream
* @stop: (out): current stop of 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;
GstPad *pad = NULL;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
/* query stop position: if no sinks have been added yet,
* we obtain the stop position from the pad otherwise we query the sinks */
priv = stream->priv;
g_mutex_lock (&priv->lock);
/* depending on the transport type, it should query corresponding sink */
if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP)
sink = priv->udpsink[0];
else if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
sink = priv->mcast_udpsink[0];
else
sink = priv->appsink[0];
if (sink) {
gst_object_ref (sink);
} else if (priv->send_src[0]) {
pad = gst_object_ref (priv->send_src[0]);
} else {
g_mutex_unlock (&priv->lock);
GST_WARNING_OBJECT (stream, "Couldn't obtain stop: erroneous pipeline");
return FALSE;
}
g_mutex_unlock (&priv->lock);
if (sink) {
GstQuery *query;
GstFormat format;
gdouble rate;
gint64 start_value;
gint64 stop_value;
query = gst_query_new_segment (GST_FORMAT_TIME);
if (!gst_element_query (sink, query)) {
GST_WARNING_OBJECT (stream, "Couldn't obtain stop: element query failed");
gst_query_unref (query);
gst_object_unref (sink);
return FALSE;
}
gst_query_parse_segment (query, &rate, &format, &start_value, &stop_value);
if (format != GST_FORMAT_TIME)
*stop = -1;
else
*stop = rate > 0.0 ? stop_value : start_value;
gst_query_unref (query);
gst_object_unref (sink);
} else if (pad) {
GstEvent *event;
const GstSegment *segment;
event = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0);
if (!event) {
GST_WARNING_OBJECT (stream, "Couldn't obtain stop: no segment event");
gst_object_unref (pad);
return FALSE;
}
gst_event_parse_segment (event, &segment);
if (segment->format != GST_FORMAT_TIME) {
*stop = -1;
} else {
*stop = segment->stop;
if (*stop == -1)
*stop = segment->duration;
else
*stop = gst_segment_to_stream_time (segment, GST_FORMAT_TIME, *stop);
}
gst_event_unref (event);
gst_object_unref (pad);
}
return TRUE;
}
/**
* gst_rtsp_stream_seekable:
* @stream: a #GstRTSPStream
*
* Checks whether the individual @stream is seekable.
*
* Returns: %TRUE if @stream is seekable, else %FALSE.
*
* Since: 1.14
*/
gboolean
gst_rtsp_stream_seekable (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
GstPad *pad = NULL;
GstQuery *query = NULL;
gboolean seekable = FALSE;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
/* query stop position: if no sinks have been added yet,
* we obtain the stop position from the pad otherwise we query the sinks */
priv = stream->priv;
g_mutex_lock (&priv->lock);
/* depending on the transport type, it should query corresponding sink */
if (priv->srcpad) {
pad = gst_object_ref (priv->srcpad);
} else {
g_mutex_unlock (&priv->lock);
GST_WARNING_OBJECT (stream, "Pad not available, can't query seekability");
goto beach;
}
g_mutex_unlock (&priv->lock);
query = gst_query_new_seeking (GST_FORMAT_TIME);
if (!gst_pad_query (pad, query)) {
GST_WARNING_OBJECT (stream, "seeking query failed");
goto beach;
}
gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL);
beach:
if (pad)
gst_object_unref (pad);
if (query)
gst_query_unref (query);
GST_DEBUG_OBJECT (stream, "Returning %d", seekable);
return seekable;
}
/**
* gst_rtsp_stream_complete_stream:
* @stream: a #GstRTSPStream
* @transport: a #GstRTSPTransport
*
* Add a receiver and sender part to the pipeline based on the transport from
* SETUP.
*
* Returns: %TRUE if the stream has been sucessfully updated.
*
* Since: 1.14
*/
gboolean
gst_rtsp_stream_complete_stream (GstRTSPStream * stream,
const GstRTSPTransport * transport)
{
GstRTSPStreamPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
GST_DEBUG_OBJECT (stream, "complete stream");
g_mutex_lock (&priv->lock);
if (!(priv->allowed_protocols & transport->lower_transport))
goto unallowed_transport;
if (!create_receiver_part (stream, transport))
goto create_receiver_error;
/* in the RECORD case, we only add RTCP sender part */
if (!create_sender_part (stream, transport))
goto create_sender_error;
priv->configured_protocols |= transport->lower_transport;
priv->is_complete = TRUE;
g_mutex_unlock (&priv->lock);
GST_DEBUG_OBJECT (stream, "pipeline sucsessfully updated");
return TRUE;
create_receiver_error:
create_sender_error:
unallowed_transport:
{
g_mutex_unlock (&priv->lock);
return FALSE;
}
}
/**
* gst_rtsp_stream_is_complete:
* @stream: a #GstRTSPStream
*
* Checks whether the stream is complete, contains the receiver and the sender
* parts. As the stream contains sink(s) element(s), it's possible to perform
* seek operations on it.
*
* Returns: %TRUE if the stream contains at least one sink element.
*
* Since: 1.14
*/
gboolean
gst_rtsp_stream_is_complete (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
gboolean ret = FALSE;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
ret = priv->is_complete;
g_mutex_unlock (&priv->lock);
return ret;
}
/**
* gst_rtsp_stream_is_sender:
* @stream: a #GstRTSPStream
*
* Checks whether the stream is a sender.
*
* Returns: %TRUE if the stream is a sender and %FALSE otherwise.
*
* Since: 1.14
*/
gboolean
gst_rtsp_stream_is_sender (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
gboolean ret = FALSE;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
ret = (priv->srcpad != NULL);
g_mutex_unlock (&priv->lock);
return ret;
}
/**
* gst_rtsp_stream_is_receiver:
* @stream: a #GstRTSPStream
*
* Checks whether the stream is a receiver.
*
* Returns: %TRUE if the stream is a receiver and %FALSE otherwise.
*
* Since: 1.14
*/
gboolean
gst_rtsp_stream_is_receiver (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
gboolean ret = FALSE;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
ret = (priv->sinkpad != NULL);
g_mutex_unlock (&priv->lock);
return ret;
}
#define AES_128_KEY_LEN 16
#define AES_256_KEY_LEN 32
#define HMAC_32_KEY_LEN 4
#define HMAC_80_KEY_LEN 10
static gboolean
mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
{
const gchar *srtp_cipher;
const gchar *srtp_auth;
const GstMIKEYPayload *sp;
guint i;
/* loop over Security policy until we find one containing policy */
for (i = 0;; i++) {
if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
break;
if (((GstMIKEYPayloadSP *) sp)->policy == policy)
break;
}
/* the default ciphers */
srtp_cipher = "aes-128-icm";
srtp_auth = "hmac-sha1-80";
/* now override the defaults with what is in the Security Policy */
if (sp != NULL) {
guint len;
guint enc_alg = GST_MIKEY_ENC_AES_CM_128;
/* collect all the params and go over them */
len = gst_mikey_payload_sp_get_n_params (sp);
for (i = 0; i < len; i++) {
const GstMIKEYPayloadSPParam *param =
gst_mikey_payload_sp_get_param (sp, i);
switch (param->type) {
case GST_MIKEY_SP_SRTP_ENC_ALG:
enc_alg = param->val[0];
switch (param->val[0]) {
case GST_MIKEY_ENC_NULL:
srtp_cipher = "null";
break;
case GST_MIKEY_ENC_AES_CM_128:
case GST_MIKEY_ENC_AES_KW_128:
srtp_cipher = "aes-128-icm";
break;
case GST_MIKEY_ENC_AES_GCM_128:
srtp_cipher = "aes-128-gcm";
break;
default:
break;
}
break;
case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
switch (param->val[0]) {
case AES_128_KEY_LEN:
if (enc_alg == GST_MIKEY_ENC_AES_CM_128 ||
enc_alg == GST_MIKEY_ENC_AES_KW_128) {
srtp_cipher = "aes-128-icm";
} else if (enc_alg == GST_MIKEY_ENC_AES_GCM_128) {
srtp_cipher = "aes-128-gcm";
}
break;
case AES_256_KEY_LEN:
if (enc_alg == GST_MIKEY_ENC_AES_CM_128 ||
enc_alg == GST_MIKEY_ENC_AES_KW_128) {
srtp_cipher = "aes-256-icm";
} else if (enc_alg == GST_MIKEY_ENC_AES_GCM_128) {
srtp_cipher = "aes-256-gcm";
}
break;
default:
break;
}
break;
case GST_MIKEY_SP_SRTP_AUTH_ALG:
switch (param->val[0]) {
case GST_MIKEY_MAC_NULL:
srtp_auth = "null";
break;
case GST_MIKEY_MAC_HMAC_SHA_1_160:
srtp_auth = "hmac-sha1-80";
break;
default:
break;
}
break;
case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
switch (param->val[0]) {
case HMAC_32_KEY_LEN:
srtp_auth = "hmac-sha1-32";
break;
case HMAC_80_KEY_LEN:
srtp_auth = "hmac-sha1-80";
break;
default:
break;
}
break;
case GST_MIKEY_SP_SRTP_SRTP_ENC:
break;
case GST_MIKEY_SP_SRTP_SRTCP_ENC:
break;
default:
break;
}
}
}
/* now configure the SRTP parameters */
gst_caps_set_simple (caps,
"srtp-cipher", G_TYPE_STRING, srtp_cipher,
"srtp-auth", G_TYPE_STRING, srtp_auth,
"srtcp-cipher", G_TYPE_STRING, srtp_cipher,
"srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
return TRUE;
}
static gboolean
handle_mikey_data (GstRTSPStream * stream, guint8 * data, gsize size)
{
GstMIKEYMessage *msg;
guint i, n_cs;
GstCaps *caps = NULL;
GstMIKEYPayloadKEMAC *kemac;
const GstMIKEYPayloadKeyData *pkd;
GstBuffer *key;
/* the MIKEY message contains a CSB or crypto session bundle. It is a
* set of Crypto Sessions protected with the same master key.
* In the context of SRTP, an RTP and its RTCP stream is part of a
* crypto session */
if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
goto parse_failed;
/* we can only handle SRTP crypto sessions for now */
if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
goto invalid_map_type;
/* get the number of crypto sessions. This maps SSRC to its
* security parameters */
n_cs = gst_mikey_message_get_n_cs (msg);
if (n_cs == 0)
goto no_crypto_sessions;
/* we also need keys */
if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
(msg, GST_MIKEY_PT_KEMAC, 0)))
goto no_keys;
/* we don't support encrypted keys */
if (kemac->enc_alg != GST_MIKEY_ENC_NULL
|| kemac->mac_alg != GST_MIKEY_MAC_NULL)
goto unsupported_encryption;
/* get Key data sub-payload */
pkd = (const GstMIKEYPayloadKeyData *)
gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
key =
gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
pkd->key_len);
/* go over all crypto sessions and create the security policy for each
* SSRC */
for (i = 0; i < n_cs; i++) {
const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
caps = gst_caps_new_simple ("application/x-srtp",
"ssrc", G_TYPE_UINT, map->ssrc,
"roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
mikey_apply_policy (caps, msg, map->policy);
gst_rtsp_stream_update_crypto (stream, map->ssrc, caps);
gst_caps_unref (caps);
}
gst_mikey_message_unref (msg);
gst_buffer_unref (key);
return TRUE;
/* ERRORS */
parse_failed:
{
GST_DEBUG_OBJECT (stream, "failed to parse MIKEY message");
return FALSE;
}
invalid_map_type:
{
GST_DEBUG_OBJECT (stream, "invalid map type %d", msg->map_type);
goto cleanup_message;
}
no_crypto_sessions:
{
GST_DEBUG_OBJECT (stream, "no crypto sessions");
goto cleanup_message;
}
no_keys:
{
GST_DEBUG_OBJECT (stream, "no keys found");
goto cleanup_message;
}
unsupported_encryption:
{
GST_DEBUG_OBJECT (stream, "unsupported key encryption");
goto cleanup_message;
}
cleanup_message:
{
gst_mikey_message_unref (msg);
return FALSE;
}
}
#define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
static void
strip_chars (gchar * str)
{
gchar *s;
gsize len;
len = strlen (str);
while (len--) {
if (!IS_STRIP_CHAR (str[len]))
break;
str[len] = '\0';
}
for (s = str; *s && IS_STRIP_CHAR (*s); s++);
memmove (str, s, len + 1);
}
/**
* gst_rtsp_stream_handle_keymgmt:
* @stream: a #GstRTSPStream
* @keymgmt: a keymgmt header
*
* Parse and handle a KeyMgmt header.
*
* Since: 1.16
*/
/* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
* key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
*/
gboolean
gst_rtsp_stream_handle_keymgmt (GstRTSPStream * stream, const gchar * keymgmt)
{
gchar **specs;
gint i, j;
specs = g_strsplit (keymgmt, ",", 0);
for (i = 0; specs[i]; i++) {
gchar **split;
split = g_strsplit (specs[i], ";", 0);
for (j = 0; split[j]; j++) {
g_strstrip (split[j]);
if (g_str_has_prefix (split[j], "prot=")) {
g_strstrip (split[j] + 5);
if (!g_str_equal (split[j] + 5, "mikey"))
break;
GST_DEBUG ("found mikey");
} else if (g_str_has_prefix (split[j], "uri=")) {
strip_chars (split[j] + 4);
GST_DEBUG ("found uri '%s'", split[j] + 4);
} else if (g_str_has_prefix (split[j], "data=")) {
guchar *data;
gsize size;
strip_chars (split[j] + 5);
GST_DEBUG ("found data '%s'", split[j] + 5);
data = g_base64_decode_inplace (split[j] + 5, &size);
handle_mikey_data (stream, data, size);
}
}
g_strfreev (split);
}
g_strfreev (specs);
return TRUE;
}
/**
* gst_rtsp_stream_get_ulpfec_pt:
*
* Returns: the payload type used for ULPFEC protection packets
*
* Since: 1.16
*/
guint
gst_rtsp_stream_get_ulpfec_pt (GstRTSPStream * stream)
{
guint res;
g_mutex_lock (&stream->priv->lock);
res = stream->priv->ulpfec_pt;
g_mutex_unlock (&stream->priv->lock);
return res;
}
/**
* gst_rtsp_stream_set_ulpfec_pt:
*
* Set the payload type to be used for ULPFEC protection packets
*
* Since: 1.16
*/
void
gst_rtsp_stream_set_ulpfec_pt (GstRTSPStream * stream, guint pt)
{
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
g_mutex_lock (&stream->priv->lock);
stream->priv->ulpfec_pt = pt;
if (stream->priv->ulpfec_encoder) {
g_object_set (stream->priv->ulpfec_encoder, "pt", pt, NULL);
}
g_mutex_unlock (&stream->priv->lock);
}
/**
* gst_rtsp_stream_request_ulpfec_decoder:
*
* Creating a rtpulpfecdec element
*
* Returns: (transfer full) (nullable): a #GstElement.
*
* Since: 1.16
*/
GstElement *
gst_rtsp_stream_request_ulpfec_decoder (GstRTSPStream * stream,
GstElement * rtpbin, guint sessid)
{
GObject *internal_storage = NULL;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
stream->priv->ulpfec_decoder =
gst_object_ref (gst_element_factory_make ("rtpulpfecdec", NULL));
g_signal_emit_by_name (G_OBJECT (rtpbin), "get-internal-storage", sessid,
&internal_storage);
g_object_set (stream->priv->ulpfec_decoder, "storage", internal_storage,
NULL);
g_object_unref (internal_storage);
update_ulpfec_decoder_pt (stream);
return stream->priv->ulpfec_decoder;
}
/**
* gst_rtsp_stream_request_ulpfec_encoder:
*
* Creating a rtpulpfecenc element
*
* Returns: (transfer full) (nullable): a #GstElement.
*
* Since: 1.16
*/
GstElement *
gst_rtsp_stream_request_ulpfec_encoder (GstRTSPStream * stream, guint sessid)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
if (!stream->priv->ulpfec_percentage)
return NULL;
stream->priv->ulpfec_encoder =
gst_object_ref (gst_element_factory_make ("rtpulpfecenc", NULL));
g_object_set (stream->priv->ulpfec_encoder, "pt", stream->priv->ulpfec_pt,
"percentage", stream->priv->ulpfec_percentage, NULL);
return stream->priv->ulpfec_encoder;
}
/**
* gst_rtsp_stream_set_ulpfec_percentage:
*
* Sets the amount of redundancy to apply when creating ULPFEC
* protection packets.
*
* Since: 1.16
*/
void
gst_rtsp_stream_set_ulpfec_percentage (GstRTSPStream * stream, guint percentage)
{
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
g_mutex_lock (&stream->priv->lock);
stream->priv->ulpfec_percentage = percentage;
if (stream->priv->ulpfec_encoder) {
g_object_set (stream->priv->ulpfec_encoder, "percentage", percentage, NULL);
}
g_mutex_unlock (&stream->priv->lock);
}
/**
* gst_rtsp_stream_get_ulpfec_percentage:
*
* Returns: the amount of redundancy applied when creating ULPFEC
* protection packets.
*
* Since: 1.16
*/
guint
gst_rtsp_stream_get_ulpfec_percentage (GstRTSPStream * stream)
{
guint res;
g_mutex_lock (&stream->priv->lock);
res = stream->priv->ulpfec_percentage;
g_mutex_unlock (&stream->priv->lock);
return res;
}
/**
* gst_rtsp_stream_set_rate_control:
*
* Define whether @stream will follow the Rate-Control=no behaviour as specified
* in the ONVIF replay spec.
*
* Since: 1.18
*/
void
gst_rtsp_stream_set_rate_control (GstRTSPStream * stream, gboolean enabled)
{
GST_DEBUG_OBJECT (stream, "%s rate control",
enabled ? "Enabling" : "Disabling");
g_mutex_lock (&stream->priv->lock);
stream->priv->do_rate_control = enabled;
if (stream->priv->appsink[0])
g_object_set (stream->priv->appsink[0], "sync", enabled, NULL);
if (stream->priv->payloader
&& g_object_class_find_property (G_OBJECT_GET_CLASS (stream->priv->
payloader), "onvif-no-rate-control"))
g_object_set (stream->priv->payloader, "onvif-no-rate-control", !enabled,
NULL);
if (stream->priv->session) {
g_object_set (stream->priv->session, "disable-sr-timestamp", !enabled,
NULL);
}
g_mutex_unlock (&stream->priv->lock);
}
/**
* gst_rtsp_stream_get_rate_control:
*
* Returns: whether @stream will follow the Rate-Control=no behaviour as specified
* in the ONVIF replay spec.
*
* Since: 1.18
*/
gboolean
gst_rtsp_stream_get_rate_control (GstRTSPStream * stream)
{
gboolean ret;
g_mutex_lock (&stream->priv->lock);
ret = stream->priv->do_rate_control;
g_mutex_unlock (&stream->priv->lock);
return ret;
}
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