/* 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
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
#include "rtsp-stream.h"
#define GST_RTSP_STREAM_GET_PRIVATE(obj) \
{
GMutex lock;
guint idx;
- GstPad *srcpad;
+ /* Only one pad is ever set */
+ GstPad *srcpad, *sinkpad;
GstElement *payloader;
guint buffer_size;
- gboolean is_joined;
+ GstBin *joined_bin;
+
+ /* TRUE if this stream is running on
+ * the client side of an RTSP link (for RECORD) */
+ gboolean client_side;
gchar *control;
GstRTSPProfile profiles;
/* 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;
- /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
- * sockets */
- GstElement *udpsrc_v4[2];
+ /* SRTP encoder/decoder */
+ GstElement *srtpenc;
+ GstElement *srtpdec;
+ GHashTable *keys;
- /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
- * sockets */
+ /* for UDP unicast */
+ GstElement *udpsrc_v4[2];
GstElement *udpsrc_v6[2];
-
+ GstElement *udpqueue[2];
GstElement *udpsink[2];
+ /* for UDP multicast */
+ GstElement *mcast_udpsrc_v4[2];
+ GstElement *mcast_udpsrc_v6[2];
+ GstElement *mcast_udpqueue[2];
+ GstElement *mcast_udpsink[2];
+
/* for TCP transport */
GstElement *appsrc[2];
+ GstClockTime appsrc_base_time[2];
GstElement *appqueue[2];
GstElement *appsink[2];
GstElement *tee[2];
GstElement *funnel[2];
- /* server ports for sending/receiving over ipv4 */
- GstRTSPRange server_port_v4;
- GstRTSPAddress *server_addr_v4;
- gboolean have_ipv4;
+ /* retransmission */
+ GstElement *rtxsend;
+ guint rtx_pt;
+ GstClockTime rtx_time;
+
+ /* pool used to manage unicast and multicast addresses */
+ GstRTSPAddressPool *pool;
- /* server ports for sending/receiving over ipv6 */
- GstRTSPRange server_port_v6;
+ /* unicast server addr/port */
+ GstRTSPAddress *server_addr_v4;
GstRTSPAddress *server_addr_v6;
- gboolean have_ipv6;
/* multicast addresses */
- GstRTSPAddressPool *pool;
- GstRTSPAddress *addr_v4;
- GstRTSPAddress *addr_v6;
+ GstRTSPAddress *mcast_addr_v4;
+ GstRTSPAddress *mcast_addr_v6;
+
+ gchar *multicast_iface;
/* the caps of the stream */
gulong caps_sig;
/* transports we stream to */
guint n_active;
GList *transports;
+ guint transports_cookie;
+ GList *tr_cache_rtp;
+ GList *tr_cache_rtcp;
+ guint tr_cache_cookie_rtp;
+ guint tr_cache_cookie_rtcp;
gint dscp_qos;
/* stream blocking */
gulong blocked_id;
gboolean blocking;
+
+ /* pt->caps map for RECORD streams */
+ GHashTable *ptmap;
+
+ GstRTSPPublishClockMode publish_clock_mode;
};
#define DEFAULT_CONTROL NULL
PROP_LAST
};
+enum
+{
+ SIGNAL_NEW_RTP_ENCODER,
+ SIGNAL_NEW_RTCP_ENCODER,
+ SIGNAL_LAST
+};
+
GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
#define GST_CAT_DEFAULT rtsp_stream_debug
static void gst_rtsp_stream_finalize (GObject * obj);
+static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
+
G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
static void
"Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
+ g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+ gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
+ g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
priv->control = g_strdup (DEFAULT_CONTROL);
priv->profiles = DEFAULT_PROFILES;
priv->protocols = DEFAULT_PROTOCOLS;
+ priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
g_mutex_init (&priv->lock);
+
+ priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
+ NULL, (GDestroyNotify) gst_caps_unref);
+ priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
+ (GDestroyNotify) gst_caps_unref);
}
static void
GST_DEBUG ("finalize stream %p", stream);
/* we really need to be unjoined now */
- g_return_if_fail (!priv->is_joined);
+ g_return_if_fail (priv->joined_bin == NULL);
- if (priv->addr_v4)
- gst_rtsp_address_free (priv->addr_v4);
- if (priv->addr_v6)
- gst_rtsp_address_free (priv->addr_v6);
+ 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);
+
+ g_free (priv->multicast_iface);
+
gst_object_unref (priv->payloader);
- gst_object_unref (priv->srcpad);
+ if (priv->srcpad)
+ gst_object_unref (priv->srcpad);
+ if (priv->sinkpad)
+ gst_object_unref (priv->sinkpad);
g_free (priv->control);
g_mutex_clear (&priv->lock);
+ g_hash_table_unref (priv->keys);
+ g_hash_table_destroy (priv->ptmap);
+
G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
}
/**
* gst_rtsp_stream_new:
* @idx: an index
- * @srcpad: a #GstPad
+ * @pad: a #GstPad
* @payloader: a #GstElement
*
* Create a new media stream with index @idx that handles RTP data on
- * @srcpad and has a payloader element @payloader.
+ * @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: a new #GstRTSPStream
+ * Returns: (transfer full): a new #GstRTSPStream
*/
GstRTSPStream *
-gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
+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 (srcpad), NULL);
- g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), 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);
- priv->srcpad = gst_object_ref (srcpad);
+ if (GST_PAD_IS_SRC (pad))
+ priv->srcpad = gst_object_ref (pad);
+ else
+ priv->sinkpad = gst_object_ref (pad);
return stream;
}
{
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ if (!stream->priv->srcpad)
+ return NULL;
+
return gst_object_ref (stream->priv->srcpad);
}
/**
+ * gst_rtsp_stream_get_sinkpad:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the sinkpad associated with @stream.
+ *
+ * Returns: (transfer full): the sinkpad. Unref after usage.
+ */
+GstPad *
+gst_rtsp_stream_get_sinkpad (GstRTSPStream * stream)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ if (!stream->priv->sinkpad)
+ return NULL;
+
+ return gst_object_ref (stream->priv->sinkpad);
+}
+
+/**
* gst_rtsp_stream_get_control:
* @stream: a #GstRTSPStream
*
* Get the control string to identify this stream.
*
- * Returns: (transfer full): the control string. free after usage.
+ * Returns: (transfer full): the control string. g_free() after usage.
*/
gchar *
gst_rtsp_stream_get_control (GstRTSPStream * stream)
/* Update the dscp qos property on the udp sinks */
static void
-update_dscp_qos (GstRTSPStream * stream)
+update_dscp_qos (GstRTSPStream * stream, GstElement * udpsink[2])
{
GstRTSPStreamPrivate *priv;
- g_return_if_fail (GST_IS_RTSP_STREAM (stream));
-
priv = stream->priv;
- if (priv->udpsink[0]) {
- g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
- NULL);
+ if (udpsink[0]) {
+ g_object_set (G_OBJECT (udpsink[0]), "qos-dscp", priv->dscp_qos, NULL);
}
- if (priv->udpsink[1]) {
- g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
- NULL);
+ if (udpsink[1]) {
+ g_object_set (G_OBJECT (udpsink[1]), "qos-dscp", priv->dscp_qos, NULL);
}
}
priv->dscp_qos = dscp_qos;
- update_dscp_qos (stream);
+ update_dscp_qos (stream, priv->udpsink);
}
/**
/**
* gst_rtsp_stream_is_transport_supported:
* @stream: a #GstRTSPStream
- * @transport: a #GstRTSPTransport
+ * @transport: (transfer none): a #GstRTSPTransport
*
* Check if @transport can be handled by stream
*
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_address_pool:
* @stream: a #GstRTSPStream
- * @pool: a #GstRTSPAddressPool
+ * @pool: (transfer none): a #GstRTSPAddressPool
*
* configure @pool to be used as the address pool of @stream.
*/
}
/**
- * gst_rtsp_stream_get_multicast_address:
+ * gst_rtsp_stream_set_multicast_iface:
+ * @stream: a #GstRTSPStream
+ * @multicast_iface: (transfer none): a multicast interface
+ *
+ * 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
- * @family: the #GSocketFamily
*
- * Get the multicast address of @stream for @family.
+ * Get the multicast interface used for @stream.
*
- * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
- * allocated. gst_rtsp_address_free() after usage.
+ * Returns: (transfer full): the multicast interface for @stream. g_free() after
+ * usage.
*/
-GstRTSPAddress *
-gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
+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;
+}
+
+
+static GstRTSPAddress *
+gst_rtsp_stream_get_multicast_address_locked (GstRTSPStream * stream,
GSocketFamily family)
{
GstRTSPStreamPrivate *priv;
GstRTSPAddress **addrp;
GstRTSPAddressFlags flags;
- g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
-
priv = stream->priv;
if (family == G_SOCKET_FAMILY_IPV6) {
flags = GST_RTSP_ADDRESS_FLAG_IPV6;
- addrp = &priv->addr_v4;
+ addrp = &priv->mcast_addr_v6;
} else {
flags = GST_RTSP_ADDRESS_FLAG_IPV4;
- addrp = &priv->addr_v6;
+ addrp = &priv->mcast_addr_v4;
}
- g_mutex_lock (&priv->lock);
if (*addrp == NULL) {
if (priv->pool == NULL)
goto no_pool;
*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 (&priv->lock);
return result;
no_pool:
{
GST_ERROR_OBJECT (stream, "no address pool specified");
- g_mutex_unlock (&priv->lock);
return NULL;
}
no_address:
{
GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
- g_mutex_unlock (&priv->lock);
return NULL;
}
}
/**
+ * gst_rtsp_stream_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)
+{
+ GstRTSPAddress *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ g_mutex_lock (&stream->priv->lock);
+ result = gst_rtsp_stream_get_multicast_address_locked (stream, family);
+ g_mutex_unlock (&stream->priv->lock);
+
+ return result;
+}
+
+/**
* gst_rtsp_stream_reserve_address:
* @stream: a #GstRTSPStream
* @address: an address
* @n_ports: n_ports
* @ttl: a TTL
*
- * Reserve @address and @port as the address and port of @stream.
+ * 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: the #GstRTSPAddress of @stream or %NULL when the address could be
- * reserved. gst_rtsp_address_free() after usage.
+ * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
+ * the address could be reserved. gst_rtsp_address_free() after usage.
*/
GstRTSPAddress *
gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
}
if (family == G_SOCKET_FAMILY_IPV6)
- addrp = &priv->addr_v4;
+ addrp = &priv->mcast_addr_v6;
else
- addrp = &priv->addr_v6;
+ addrp = &priv->mcast_addr_v4;
g_mutex_lock (&priv->lock);
if (*addrp == NULL) {
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 (strcmp ((*addrp)->address, address) ||
(*addrp)->port != port || (*addrp)->n_ports != n_ports ||
}
}
+/* must be called with lock */
+static void
+set_sockets_for_udpsinks (GstElement * udpsink[2], GSocket * rtp_socket,
+ GSocket * rtcp_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[0]), multisink_socket, rtp_socket, NULL);
+ g_object_set (G_OBJECT (udpsink[1]), multisink_socket, rtcp_socket, NULL);
+}
+
static gboolean
-alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
- GSocketFamily family, GstElement * udpsrc_out[2],
- GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
- GstRTSPAddress ** server_addr_out)
+create_and_configure_udpsinks (GstRTSPStream * stream, GstElement * udpsink[2])
{
- GstStateChangeReturn ret;
- GstElement *udpsrc0, *udpsrc1;
+ GstRTSPStreamPrivate *priv = stream->priv;
GstElement *udpsink0, *udpsink1;
+
+ udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
+ udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
+
+ if (!udpsink0 || !udpsink1)
+ goto no_udp_protocol;
+
+ /* configure sinks */
+
+ g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
+
+ g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
+
+ g_object_set (G_OBJECT (udpsink0), "buffer-size", priv->buffer_size, NULL);
+
+ g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
+ /* Needs to be async for RECORD streams, otherwise we will never go to
+ * PLAYING because the sinks will wait for data while the udpsrc can't
+ * provide data with timestamps in PAUSED. */
+ if (priv->sinkpad)
+ g_object_set (G_OBJECT (udpsink0), "async", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
+
+ /* 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 (udpsink0), "auto-multicast", TRUE, NULL);
+ g_object_set (G_OBJECT (udpsink1), "auto-multicast", TRUE, NULL);
+
+ g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
+
+ udpsink[0] = udpsink0;
+ udpsink[1] = udpsink1;
+
+ /* update the dscp qos field in the sinks */
+ update_dscp_qos (stream, udpsink);
+
+ return TRUE;
+
+ /* ERRORS */
+no_udp_protocol:
+ {
+ return FALSE;
+ }
+}
+
+/* must be called with lock */
+static gboolean
+create_and_configure_udpsources (GstElement * udpsrc_out[2],
+ GSocket * rtp_socket, GSocket * rtcp_socket)
+{
+ GstStateChangeReturn ret;
+
+ udpsrc_out[0] = gst_element_factory_make ("udpsrc", NULL);
+ udpsrc_out[1] = gst_element_factory_make ("udpsrc", NULL);
+
+ if (udpsrc_out[0] == NULL || udpsrc_out[1] == NULL)
+ goto error;
+
+ g_object_set (G_OBJECT (udpsrc_out[0]), "socket", rtp_socket, NULL);
+ g_object_set (G_OBJECT (udpsrc_out[1]), "socket", rtcp_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_out[0]), "auto-multicast", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsrc_out[1]), "auto-multicast", FALSE, NULL);
+
+ g_object_set (G_OBJECT (udpsrc_out[0]), "loop", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsrc_out[1]), "loop", FALSE, NULL);
+
+ ret = gst_element_set_state (udpsrc_out[0], GST_STATE_READY);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto error;
+ ret = gst_element_set_state (udpsrc_out[1], GST_STATE_READY);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto error;
+
+ return TRUE;
+
+ /* ERRORS */
+error:
+ {
+ if (udpsrc_out[0]) {
+ gst_element_set_state (udpsrc_out[0], GST_STATE_NULL);
+ g_clear_object (&udpsrc_out[0]);
+ }
+ if (udpsrc_out[1]) {
+ gst_element_set_state (udpsrc_out[1], GST_STATE_NULL);
+ g_clear_object (&udpsrc_out[1]);
+ }
+ return FALSE;
+ }
+}
+
+static gboolean
+alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
+ GstElement * udpsrc_out[2], GstElement * udpsink_out[2],
+ GstRTSPAddress ** server_addr_out, gboolean multicast)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
GSocket *rtp_socket = NULL;
GSocket *rtcp_socket;
gint tmp_rtp, tmp_rtcp;
GList *rejected_addresses = NULL;
GstRTSPAddress *addr = NULL;
GInetAddress *inetaddr = NULL;
+ gchar *addr_str;
GSocketAddress *rtp_sockaddr = NULL;
GSocketAddress *rtcp_sockaddr = NULL;
- const gchar *multisink_socket;
+ GstRTSPAddressPool *pool;
- if (family == G_SOCKET_FAMILY_IPV6)
- multisink_socket = "socket-v6";
- else
- multisink_socket = "socket";
+ g_assert (!udpsrc_out[0]);
+ g_assert (!udpsrc_out[1]);
+ g_assert ((!udpsink_out[0] && !udpsink_out[1]) ||
+ (udpsink_out[0] && udpsink_out[1]));
+ g_assert (*server_addr_out == NULL);
- udpsrc0 = NULL;
- udpsrc1 = NULL;
- udpsink0 = NULL;
- udpsink1 = NULL;
+ pool = priv->pool;
count = 0;
/* Start with random port */
G_SOCKET_PROTOCOL_UDP, NULL);
if (!rtcp_socket)
goto no_udp_protocol;
-
- if (*server_addr_out)
- gst_rtsp_address_free (*server_addr_out);
+ 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 */
G_SOCKET_PROTOCOL_UDP, NULL);
if (!rtp_socket)
goto no_udp_protocol;
+ g_socket_set_multicast_loopback (rtp_socket, FALSE);
}
- if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
+ if ((pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) || multicast) {
GstRTSPAddressFlags flags;
if (addr)
rejected_addresses = g_list_prepend (rejected_addresses, addr);
- flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
+ if (!pool)
+ goto no_ports;
+
+ 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
}
g_object_unref (rtcp_sockaddr);
- g_clear_object (&inetaddr);
+ if (!addr) {
+ addr = g_slice_new0 (GstRTSPAddress);
+ addr->address = g_inet_address_to_string (inetaddr);
+ addr->port = tmp_rtp;
+ addr->n_ports = 2;
+ }
- udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
- udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
+ addr_str = addr->address;
+ g_clear_object (&inetaddr);
- if (udpsrc0 == NULL || udpsrc1 == NULL)
+ if (!create_and_configure_udpsources (udpsrc_out, rtp_socket, rtcp_socket)) {
goto no_udp_protocol;
+ }
- g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
- g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
-
- ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
- if (ret == GST_STATE_CHANGE_FAILURE)
- goto element_error;
- ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
- if (ret == GST_STATE_CHANGE_FAILURE)
- goto element_error;
-
- /* all fine, do port check */
- g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
- g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
+ g_object_get (G_OBJECT (udpsrc_out[0]), "port", &rtpport, NULL);
+ g_object_get (G_OBJECT (udpsrc_out[1]), "port", &rtcpport, NULL);
/* this should not happen... */
if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
goto port_error;
- if (udpsink_out[0])
- udpsink0 = udpsink_out[0];
- else
- udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
-
- if (!udpsink0)
+ /* This function is called twice (for v4 and v6) but we create only one pair
+ * of udpsinks. */
+ if (!udpsink_out[0]
+ && !create_and_configure_udpsinks (stream, udpsink_out))
goto no_udp_protocol;
- g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
-
- if (udpsink_out[1])
- udpsink1 = udpsink_out[1];
- else
- udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
-
- if (!udpsink1)
- goto no_udp_protocol;
-
- g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
+ if (multicast) {
+ g_object_set (G_OBJECT (udpsink_out[0]), "multicast-iface",
+ priv->multicast_iface, NULL);
+ g_object_set (G_OBJECT (udpsink_out[1]), "multicast-iface",
+ priv->multicast_iface, NULL);
- g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
- g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
+ g_signal_emit_by_name (udpsink_out[0], "add", addr_str, rtpport, NULL);
+ g_signal_emit_by_name (udpsink_out[1], "add", addr_str, rtcpport, NULL);
+ }
- /* we keep these elements, we will further configure them when the
- * client told us to really use the UDP ports. */
- udpsrc_out[0] = udpsrc0;
- udpsrc_out[1] = udpsrc1;
- udpsink_out[0] = udpsink0;
- udpsink_out[1] = udpsink1;
- server_port_out->min = rtpport;
- server_port_out->max = rtcpport;
+ set_sockets_for_udpsinks (udpsink_out, rtp_socket, rtcp_socket, family);
*server_addr_out = addr;
+
g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
g_object_unref (rtp_socket);
{
goto cleanup;
}
-element_error:
- {
- goto cleanup;
- }
cleanup:
{
- if (udpsrc0) {
- gst_element_set_state (udpsrc0, GST_STATE_NULL);
- gst_object_unref (udpsrc0);
- }
- if (udpsrc1) {
- gst_element_set_state (udpsrc1, GST_STATE_NULL);
- gst_object_unref (udpsrc1);
- }
- if (udpsink0) {
- gst_element_set_state (udpsink0, GST_STATE_NULL);
- gst_object_unref (udpsink0);
- }
if (inetaddr)
g_object_unref (inetaddr);
g_list_free_full (rejected_addresses,
}
}
-/* must be called with lock */
-static gboolean
-alloc_ports (GstRTSPStream * stream)
+/**
+ * gst_rtsp_stream_allocate_udp_sockets:
+ * @stream: a #GstRTSPStream
+ * @family: protocol family
+ * @transport_method: transport method
+ *
+ * Allocates RTP and RTCP ports.
+ *
+ * Returns: %TRUE if the RTP and RTCP sockets have been succeccully allocated.
+ * Deprecated: This function shouldn't have been made public
+ */
+gboolean
+gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream,
+ GSocketFamily family, GstRTSPTransport * ct, gboolean use_client_settings)
{
- GstRTSPStreamPrivate *priv = stream->priv;
-
- priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
- G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
- &priv->server_port_v4, &priv->server_addr_v4);
-
- priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
- G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
- &priv->server_port_v6, &priv->server_addr_v6);
-
- return priv->have_ipv4 || priv->have_ipv6;
+ g_warn_if_reached ();
+ return FALSE;
}
/**
- * gst_rtsp_stream_get_server_port:
+ * gst_rtsp_stream_set_client_side:
* @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.
+ * @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_get_server_port (GstRTSPStream * stream,
- GstRTSPRange * server_port, GSocketFamily family)
+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_return_if_fail (priv->is_joined);
-
g_mutex_lock (&priv->lock);
- if (family == G_SOCKET_FAMILY_IPV4) {
- if (server_port)
- *server_port = priv->server_port_v4;
- } else {
- if (server_port)
- *server_port = priv->server_port_v6;
- }
+ priv->client_side = client_side;
g_mutex_unlock (&priv->lock);
}
/**
- * gst_rtsp_stream_get_rtpsession:
+ * gst_rtsp_stream_is_client_side:
* @stream: a #GstRTSPStream
*
- * Get the RTP session of this stream.
+ * 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;
+}
+
+/* must be called with lock */
+static gboolean
+alloc_ports (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ gboolean ret = TRUE;
+
+ if (priv->protocols & GST_RTSP_LOWER_TRANS_UDP) {
+ ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
+ priv->udpsrc_v4, priv->udpsink, &priv->server_addr_v4, FALSE);
+
+ ret |= alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
+ priv->udpsrc_v6, priv->udpsink, &priv->server_addr_v6, FALSE);
+ }
+
+ /* FIXME: Maybe actually consider the return values? */
+ if (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+ ret |= alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
+ priv->mcast_udpsrc_v4, priv->mcast_udpsink, &priv->mcast_addr_v4, TRUE);
+
+ ret |= alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
+ priv->mcast_udpsrc_v6, priv->mcast_udpsink, &priv->mcast_addr_v6, TRUE);
+ }
+
+ 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);
+
+ g_mutex_lock (&priv->lock);
+ if (family == G_SOCKET_FAMILY_IPV4) {
+ if (server_port) {
+ 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) {
+ 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.
*/
}
/**
+ * gst_rtsp_stream_get_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
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
- g_return_if_fail (priv->is_joined);
+ g_return_if_fail (priv->joined_bin != NULL);
g_mutex_lock (&priv->lock);
if (ssrc && priv->session)
g_mutex_unlock (&priv->lock);
}
+/**
+ * gst_rtsp_stream_set_retransmission_time:
+ * @stream: a #GstRTSPStream
+ * @time: a #GstClockTime
+ *
+ * Set the amount of time to store retransmission packets.
+ */
+void
+gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
+ GstClockTime time)
+{
+ GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
+
+ g_mutex_lock (&stream->priv->lock);
+ stream->priv->rtx_time = time;
+ if (stream->priv->rtxsend)
+ g_object_set (stream->priv->rtxsend, "max-size-time",
+ GST_TIME_AS_MSECONDS (time), NULL);
+ g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_retransmission_time:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the amount of time to store retransmission data.
+ *
+ * Returns: the amount of time to store retransmission data.
+ */
+GstClockTime
+gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
+{
+ GstClockTime ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+ g_mutex_lock (&stream->priv->lock);
+ ret = stream->priv->rtx_time;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_set_retransmission_pt:
+ * @stream: a #GstRTSPStream
+ * @rtx_pt: a #guint
+ *
+ * Set the payload type (pt) for retransmission of this stream.
+ */
+void
+gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
+{
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
+
+ g_mutex_lock (&stream->priv->lock);
+ stream->priv->rtx_pt = rtx_pt;
+ if (stream->priv->rtxsend) {
+ guint pt = gst_rtsp_stream_get_pt (stream);
+ gchar *pt_s = g_strdup_printf ("%d", pt);
+ GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
+ pt_s, G_TYPE_UINT, rtx_pt, NULL);
+ g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
+ g_free (pt_s);
+ gst_structure_free (rtx_pt_map);
+ }
+ g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_retransmission_pt:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the payload-type used for retransmission of this stream
+ *
+ * Returns: The retransmission PT.
+ */
+guint
+gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
+{
+ guint rtx_pt;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+ g_mutex_lock (&stream->priv->lock);
+ rtx_pt = stream->priv->rtx_pt;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return rtx_pt;
+}
+
+/**
+ * gst_rtsp_stream_set_buffer_size:
+ * @stream: a #GstRTSPStream
+ * @size: the buffer size
+ *
+ * Set the size of the UDP transmission buffer (in bytes)
+ * Needs to be set before the stream is joined to a bin.
+ *
+ * Since: 1.6
+ */
+void
+gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
+{
+ g_mutex_lock (&stream->priv->lock);
+ stream->priv->buffer_size = size;
+ g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_buffer_size:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the size of the UDP transmission buffer (in bytes)
+ *
+ * Returns: the size of the UDP TX buffer
+ *
+ * Since: 1.6
+ */
+guint
+gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
+{
+ guint buffer_size;
+
+ g_mutex_lock (&stream->priv->lock);
+ buffer_size = stream->priv->buffer_size;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return buffer_size;
+}
+
/* executed from streaming thread */
static void
caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
tr = gst_rtsp_stream_transport_get_transport (trans);
- min = tr->client_port.min;
- max = tr->client_port.max;
+ if (priv->client_side) {
+ /* In client side mode the 'destination' is the RTSP server, so send
+ * to those ports */
+ min = tr->server_port.min;
+ max = tr->server_port.max;
+ } else {
+ min = tr->client_port.min;
+ max = tr->client_port.max;
+ }
if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
result = trans;
}
}
+static void
+on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GST_INFO ("%p: new sender source %p", stream, source);
+#ifndef DUMP_STATS
+ {
+ GstStructure *stats;
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ dump_structure (stats);
+ gst_structure_free (stats);
+ }
+ }
+#endif
+}
+
+static void
+on_sender_ssrc_active (GObject * session, GObject * source,
+ GstRTSPStream * stream)
+{
+#ifndef DUMP_STATS
+ {
+ GstStructure *stats;
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ dump_structure (stats);
+ gst_structure_free (stats);
+ }
+ }
+#endif
+}
+
+static void
+clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
+{
+ if (is_rtp) {
+ g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
+ g_list_free (priv->tr_cache_rtp);
+ priv->tr_cache_rtp = NULL;
+ } else {
+ g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
+ g_list_free (priv->tr_cache_rtcp);
+ priv->tr_cache_rtcp = NULL;
+ }
+}
+
static GstFlowReturn
handle_new_sample (GstAppSink * sink, gpointer user_data)
{
GstSample *sample;
GstBuffer *buffer;
GstRTSPStream *stream;
+ gboolean is_rtp;
sample = gst_app_sink_pull_sample (sink);
if (!sample)
priv = stream->priv;
buffer = gst_sample_get_buffer (sample);
+ is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
+
g_mutex_lock (&priv->lock);
- for (walk = priv->transports; walk; walk = g_list_next (walk)) {
- GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+ if (is_rtp) {
+ if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
+ clear_tr_cache (priv, is_rtp);
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+ priv->tr_cache_rtp =
+ g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
+ }
+ priv->tr_cache_cookie_rtp = priv->transports_cookie;
+ }
+ } else {
+ if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
+ clear_tr_cache (priv, is_rtp);
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+ priv->tr_cache_rtcp =
+ g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
+ }
+ priv->tr_cache_cookie_rtcp = priv->transports_cookie;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
- if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
+ if (is_rtp) {
+ for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
gst_rtsp_stream_transport_send_rtp (tr, buffer);
- } else {
+ }
+ } else {
+ for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
gst_rtsp_stream_transport_send_rtcp (tr, buffer);
}
}
- g_mutex_unlock (&priv->lock);
-
gst_sample_unref (sample);
return GST_FLOW_OK;
handle_new_sample,
};
+static GstElement *
+get_rtp_encoder (GstRTSPStream * stream, guint session)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ if (priv->srtpenc == NULL) {
+ gchar *name;
+
+ name = g_strdup_printf ("srtpenc_%u", session);
+ priv->srtpenc = gst_element_factory_make ("srtpenc", name);
+ g_free (name);
+
+ g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
+ }
+ return gst_object_ref (priv->srtpenc);
+}
+
+static GstElement *
+request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstElement *oldenc, *enc;
+ GstPad *pad;
+ gchar *name;
+
+ if (priv->idx != session)
+ return NULL;
+
+ GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
+
+ oldenc = priv->srtpenc;
+ enc = get_rtp_encoder (stream, session);
+ name = g_strdup_printf ("rtp_sink_%d", session);
+ pad = gst_element_get_request_pad (enc, name);
+ g_free (name);
+ gst_object_unref (pad);
+
+ if (oldenc == NULL)
+ g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
+ enc);
+
+ return enc;
+}
+
+static GstElement *
+request_rtcp_encoder (GstElement * rtpbin, guint session,
+ GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstElement *oldenc, *enc;
+ GstPad *pad;
+ gchar *name;
+
+ if (priv->idx != session)
+ return NULL;
+
+ GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
+
+ oldenc = priv->srtpenc;
+ enc = get_rtp_encoder (stream, session);
+ name = g_strdup_printf ("rtcp_sink_%d", session);
+ pad = gst_element_get_request_pad (enc, name);
+ g_free (name);
+ gst_object_unref (pad);
+
+ if (oldenc == NULL)
+ g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
+ enc);
+
+ return enc;
+}
+
+static GstCaps *
+request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstCaps *caps;
+
+ GST_DEBUG ("request key %08x", ssrc);
+
+ g_mutex_lock (&priv->lock);
+ if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
+ gst_caps_ref (caps);
+ g_mutex_unlock (&priv->lock);
+
+ return caps;
+}
+
+static GstElement *
+request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
+ GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ if (priv->idx != session)
+ return NULL;
+
+ if (priv->srtpdec == NULL) {
+ gchar *name;
+
+ name = g_strdup_printf ("srtpdec_%u", session);
+ priv->srtpdec = gst_element_factory_make ("srtpdec", name);
+ g_free (name);
+
+ g_signal_connect (priv->srtpdec, "request-key",
+ (GCallback) request_key, stream);
+ }
+ return gst_object_ref (priv->srtpdec);
+}
+
/**
- * gst_rtsp_stream_join_bin:
+ * gst_rtsp_stream_request_aux_sender:
* @stream: a #GstRTSPStream
- * @bin: a #GstBin to join
- * @rtpbin: a rtpbin element in @bin
- * @state: the target state of the new elements
+ * @sessid: the session id
*
- * Join the #GstBin @bin that contains the element @rtpbin.
+ * Creating a rtxsend bin
*
- * @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: (transfer full): a #GstElement.
*
- * Returns: %TRUE on success.
+ * Since: 1.6
*/
-gboolean
-gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
- GstElement * rtpbin, GstState state)
+GstElement *
+gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
{
- GstRTSPStreamPrivate *priv;
- gint i;
- guint idx;
+ GstElement *bin;
+ GstPad *pad;
+ GstStructure *pt_map;
gchar *name;
- GstPad *pad, *sinkpad, *selpad;
- GstPadLinkReturn ret;
+ guint pt, rtx_pt;
+ gchar *pt_s;
- 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);
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ pt = gst_rtsp_stream_get_pt (stream);
+ pt_s = g_strdup_printf ("%u", pt);
+ rtx_pt = stream->priv->rtx_pt;
+
+ GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
+
+ bin = gst_bin_new (NULL);
+ stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
+ pt_map = gst_structure_new ("application/x-rtp-pt-map",
+ pt_s, G_TYPE_UINT, rtx_pt, NULL);
+ g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
+ "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
+ g_free (pt_s);
+ gst_structure_free (pt_map);
+ gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
+
+ pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
+ name = g_strdup_printf ("src_%u", sessid);
+ gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
+ g_free (name);
+ gst_object_unref (pad);
+
+ pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
+ name = g_strdup_printf ("sink_%u", sessid);
+ gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
+ g_free (name);
+ gst_object_unref (pad);
+
+ return bin;
+}
+
+/**
+ * gst_rtsp_stream_set_pt_map:
+ * @stream: a #GstRTSPStream
+ * @pt: the pt
+ * @caps: a #GstCaps
+ *
+ * Configure a pt map between @pt and @caps.
+ */
+void
+gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * 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:
+ * @factory: 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);
- if (priv->is_joined)
- goto was_joined;
+ ret = priv->publish_clock_mode;
+ g_mutex_unlock (&priv->lock);
- /* create a session with the same index as the stream */
- idx = priv->idx;
+ return ret;
+}
- GST_INFO ("stream %p joining bin as session %u", stream, idx);
+static GstCaps *
+request_pt_map (GstElement * rtpbin, guint session, guint pt,
+ GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstCaps *caps = NULL;
- if (!alloc_ports (stream))
- goto no_ports;
+ g_mutex_lock (&priv->lock);
- /* update the dscp qos field in the sinks */
- update_dscp_qos (stream);
+ 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);
+ }
+ }
- /* 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_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 (priv->srcpad, priv->send_rtp_sink);
+ ret = gst_pad_link (pad, priv->sinkpad);
if (ret != GST_PAD_LINK_OK)
goto link_failed;
+ priv->recv_rtp_src = gst_object_ref (pad);
- /* get pads from the RTP session element for sending and receiving
- * RTP/RTCP*/
- name = g_strdup_printf ("send_rtp_src_%u", idx);
- priv->send_src[0] = gst_element_get_static_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_rtp_sink_%u", idx);
- priv->recv_sink[0] = 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);
+ return;
- /* get the session */
- g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
+/* 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));
+ }
+}
- 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);
+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 ());
+}
+
+static void
+plug_sink (GstBin * bin, GstElement * tee, GstElement * sink,
+ GstElement ** queue_out)
+{
+ GstPad *pad;
+ GstPad *teepad;
+ GstPad *queuepad;
+
+ gst_bin_add (bin, sink);
+
+ *queue_out = gst_element_factory_make ("queue", NULL);
+ g_object_set (*queue_out, "max-size-buffers", 1, "max-size-bytes", 0,
+ "max-size-time", G_GINT64_CONSTANT (0), NULL);
+ gst_bin_add (bin, *queue_out);
+
+ /* link tee to queue */
+ teepad = gst_element_get_request_pad (tee, "src_%u");
+ pad = gst_element_get_static_pad (*queue_out, "sink");
+ gst_pad_link (teepad, pad);
+ gst_object_unref (pad);
+ gst_object_unref (teepad);
+
+ /* link queue to sink */
+ queuepad = gst_element_get_static_pad (*queue_out, "src");
+ pad = gst_element_get_static_pad (sink, "sink");
+ gst_pad_link (queuepad, pad);
+ gst_object_unref (queuepad);
+ gst_object_unref (pad);
+}
+
+/* must be called with lock */
+static void
+create_sender_part (GstRTSPStream * stream, GstBin * bin, GstState state)
+{
+ GstRTSPStreamPrivate *priv;
+ GstPad *pad;
+ gboolean is_tcp, is_udp;
+ gint i;
+
+ priv = stream->priv;
+
+ is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
+ is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
+ (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
for (i = 0; i < 2; i++) {
- GstPad *teepad, *queuepad;
/* For the sender we create this bit of pipeline for both
* RTP and RTCP. Sync and preroll are enabled on udpsink so
- * we need to add a queue before appsink to make the pipeline
- * not block. For the TCP case, we want to pump data to the
- * client as fast as possible anyway.
+ * we need to add a queue before appsink and udpsink to make
+ * the pipeline not block. For the TCP case, we want to pump
+ * client as fast as possible anyway. This pipeline is used
+ * when both TCP and UDP are present.
*
- * .--------. .-----. .---------.
- * | rtpbin | | tee | | udpsink |
- * | send->sink src->sink |
- * '--------' | | '---------'
+ * .--------. .-----. .---------. .---------.
+ * | rtpbin | | tee | | queue | | udpsink |
+ * | send->sink src->sink src->sink |
+ * '--------' | | '---------' '---------'
* | | .---------. .---------.
* | | | queue | | appsink |
* | src->sink src->sink |
* '-----' '---------' '---------'
*
- * When only UDP is allowed, we skip the tee, queue and appsink and link the
- * udpsink directly to the session.
+ * When only UDP or only TCP is allowed, we skip the tee and queue
+ * and link the udpsink (for UDP) or appsink (for TCP) directly to
+ * the session.
*/
- /* add udpsink */
- gst_bin_add (bin, priv->udpsink[i]);
- sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
- if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
+ /* 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 (is_tcp) {
+ /* make appsink */
+ priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+ g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
+ gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
+ &sink_cb, stream, NULL);
+ }
+
+ /* If we have udp always use a tee because we could have mcast clients
+ * requesting different ports, in which case we'll have to plug more
+ * udpsinks. */
+ if (is_udp) {
/* make tee for RTP/RTCP */
priv->tee[i] = gst_element_factory_make ("tee", NULL);
gst_bin_add (bin, priv->tee[i]);
gst_pad_link (priv->send_src[i], pad);
gst_object_unref (pad);
- /* link tee to udpsink */
- teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
- gst_pad_link (teepad, sinkpad);
- gst_object_unref (teepad);
-
- /* make queue */
- priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
- gst_bin_add (bin, priv->appqueue[i]);
- /* and link to tee */
- teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
- pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
- gst_pad_link (teepad, pad);
- gst_object_unref (pad);
- gst_object_unref (teepad);
+ if (priv->udpsink[i])
+ plug_sink (bin, priv->tee[i], priv->udpsink[i], &priv->udpqueue[i]);
- /* make appsink */
- priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
- g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
- g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
- gst_bin_add (bin, priv->appsink[i]);
- gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
- &sink_cb, stream, NULL);
- /* and link to queue */
- queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
+ if (priv->mcast_udpsink[i])
+ plug_sink (bin, priv->tee[i], priv->mcast_udpsink[i],
+ &priv->mcast_udpqueue[i]);
+
+ if (is_tcp) {
+ g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
+ plug_sink (bin, priv->tee[i], priv->appsink[i], &priv->appqueue[i]);
+ }
+ } else if (is_tcp) {
+ /* only appsink needed, link it to the session */
pad = gst_element_get_static_pad (priv->appsink[i], "sink");
- gst_pad_link (queuepad, pad);
+ gst_pad_link (priv->send_src[i], pad);
gst_object_unref (pad);
- gst_object_unref (queuepad);
- } else {
- /* else only udpsink needed, link it to the session */
- gst_pad_link (priv->send_src[i], sinkpad);
+
+ /* when its only TCP, we need to set sync and preroll to FALSE
+ * for the sink to avoid deadlock. And this is only needed for
+ * sink used for RTCP data, not the RTP data. */
+ if (i == 1)
+ g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
+ }
+
+ /* check if we need to set to a special state */
+ if (state != GST_STATE_NULL) {
+ if (priv->udpsink[i])
+ gst_element_set_state (priv->udpsink[i], state);
+ if (priv->mcast_udpsink[i])
+ gst_element_set_state (priv->mcast_udpsink[i], state);
+ if (priv->appsink[i])
+ gst_element_set_state (priv->appsink[i], state);
+ if (priv->appqueue[i])
+ gst_element_set_state (priv->appqueue[i], state);
+ if (priv->udpqueue[i])
+ gst_element_set_state (priv->udpqueue[i], state);
+ if (priv->mcast_udpqueue[i])
+ gst_element_set_state (priv->mcast_udpqueue[i], state);
+ if (priv->tee[i])
+ gst_element_set_state (priv->tee[i], state);
}
- gst_object_unref (sinkpad);
+ }
+}
+
+/* must be called with lock */
+static void
+plug_src (GstRTSPStream * stream, GstBin * bin, GstElement * src,
+ GstElement * funnel)
+{
+ GstRTSPStreamPrivate *priv;
+ GstPad *pad, *selpad;
+
+ priv = stream->priv;
+
+ if (priv->srcpad) {
+ /* we set and keep these to playing so that they don't cause NO_PREROLL return
+ * values. This is only relevant for PLAY pipelines */
+ gst_element_set_state (src, GST_STATE_PLAYING);
+ gst_element_set_locked_state (src, TRUE);
+ }
+
+ /* add src */
+ gst_bin_add (bin, src);
+
+ /* and link to the funnel */
+ selpad = gst_element_get_request_pad (funnel, "sink_%u");
+ pad = gst_element_get_static_pad (src, "src");
+ gst_pad_link (pad, selpad);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+}
+
+/* must be called with lock */
+static void
+create_receiver_part (GstRTSPStream * stream, GstBin * bin, GstState state)
+{
+ GstRTSPStreamPrivate *priv;
+ GstPad *pad;
+ gboolean is_tcp;
+ gint i;
+ priv = stream->priv;
+
+ is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
+
+ 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.
* | 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 */
priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
gst_bin_add (bin, priv->funnel[i]);
gst_pad_link (pad, priv->recv_sink[i]);
gst_object_unref (pad);
- if (priv->udpsrc_v4[i]) {
- /* we set and keep these to playing so that they don't cause NO_PREROLL return
- * values */
- gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
- gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
- /* add udpsrc */
- gst_bin_add (bin, priv->udpsrc_v4[i]);
-
- /* and link to the funnel v4 */
- selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
- pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
- gst_pad_link (pad, selpad);
- gst_object_unref (pad);
- gst_object_unref (selpad);
- }
+ if (priv->udpsrc_v4[i])
+ plug_src (stream, bin, priv->udpsrc_v4[i], priv->funnel[i]);
+
+ if (priv->udpsrc_v6[i])
+ plug_src (stream, bin, priv->udpsrc_v6[i], priv->funnel[i]);
+
+ if (priv->mcast_udpsrc_v4[i])
+ plug_src (stream, bin, priv->mcast_udpsrc_v4[i], priv->funnel[i]);
+
+ if (priv->mcast_udpsrc_v6[i])
+ plug_src (stream, bin, priv->mcast_udpsrc_v6[i], priv->funnel[i]);
+
+ if (is_tcp) {
+ /* make and add appsrc */
+ priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
+ priv->appsrc_base_time[i] = -1;
+ g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, "is-live",
+ TRUE, NULL);
+ plug_src (stream, bin, priv->appsrc[i], priv->funnel[i]);
+ }
+
+ /* check if we need to set to a special state */
+ if (state != GST_STATE_NULL) {
+ gst_element_set_state (priv->funnel[i], state);
+ }
+ }
+}
+
+static gboolean
+check_mcast_part_for_transport (GstRTSPStream * stream,
+ const GstRTSPTransport * tr)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GInetAddress *inetaddr;
+ GSocketFamily family;
+ GstRTSPAddress *mcast_addr;
+
+ /* Check if it's a ipv4 or ipv6 transport */
+ inetaddr = g_inet_address_new_from_string (tr->destination);
+ family = g_inet_address_get_family (inetaddr);
+ g_object_unref (inetaddr);
+
+ /* Select fields corresponding to the family */
+ if (family == G_SOCKET_FAMILY_IPV4) {
+ mcast_addr = priv->mcast_addr_v4;
+ } else {
+ mcast_addr = priv->mcast_addr_v6;
+ }
+
+ /* We support only one mcast group per family, make sure this transport
+ * matches it. */
+ if (!mcast_addr)
+ goto no_addr;
+
+ if (!g_str_equal (tr->destination, mcast_addr->address) ||
+ tr->port.min != mcast_addr->port ||
+ tr->port.max != mcast_addr->port + mcast_addr->n_ports - 1 ||
+ tr->ttl != mcast_addr->ttl)
+ goto wrong_addr;
+
+ return TRUE;
+
+no_addr:
+ {
+ GST_WARNING_OBJECT (stream, "Adding mcast transport, but no mcast address "
+ "has been reserved");
+ return FALSE;
+ }
+wrong_addr:
+ {
+ 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->udpsrc_v6[i]) {
- gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
- gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
- gst_bin_add (bin, priv->udpsrc_v6[i]);
+ if (!alloc_ports (stream))
+ goto no_ports;
- /* and link to the funnel v6 */
- selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
- pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
- gst_pad_link (pad, selpad);
- gst_object_unref (pad);
- gst_object_unref (selpad);
- }
+ if (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->protocols & GST_RTSP_LOWER_TRANS_TCP) {
- /* make and add appsrc */
- priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
- gst_bin_add (bin, priv->appsrc[i]);
- /* and link to the funnel */
- selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
- pad = gst_element_get_static_pad (priv->appsrc[i], "src");
- gst_pad_link (pad, selpad);
- gst_object_unref (pad);
- gst_object_unref (selpad);
- }
+ if (priv->sinkpad) {
+ g_signal_connect (rtpbin, "request-pt-map",
+ (GCallback) request_pt_map, stream);
+ }
- /* check if we need to set to a special state */
- if (state != GST_STATE_NULL) {
- if (priv->udpsink[i])
- gst_element_set_state (priv->udpsink[i], state);
- if (priv->appsink[i])
- gst_element_set_state (priv->appsink[i], state);
- if (priv->appqueue[i])
- gst_element_set_state (priv->appqueue[i], state);
- if (priv->tee[i])
- gst_element_set_state (priv->tee[i], state);
- if (priv->funnel[i])
- gst_element_set_state (priv->funnel[i], state);
- if (priv->appsrc[i])
- gst_element_set_state (priv->appsrc[i], state);
- }
+ /* get pads from the RTP session element for sending and receiving
+ * RTP/RTCP*/
+ if (priv->srcpad) {
+ /* get a pad for sending RTP */
+ name = g_strdup_printf ("send_rtp_sink_%u", idx);
+ priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
+ g_free (name);
+
+ /* link the RTP pad to the session manager, it should not really fail unless
+ * this is not really an RTP pad */
+ ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
+ if (ret != GST_PAD_LINK_OK)
+ goto link_failed;
+
+ name = g_strdup_printf ("send_rtp_src_%u", idx);
+ priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
+ g_free (name);
+ } else {
+ /* Need to connect our sinkpad from here */
+ g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
+ /* EOS */
+ g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
+
+ name = g_strdup_printf ("recv_rtp_sink_%u", idx);
+ priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
+ g_free (name);
}
- /* be notified of caps changes */
- priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
- (GCallback) caps_notify, stream);
+ 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);
+
+ create_sender_part (stream, bin, state);
+ create_receiver_part (stream, bin, state);
- priv->is_joined = TRUE;
+ 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->joined_bin = gst_object_ref (bin);
g_mutex_unlock (&priv->lock);
return TRUE;
}
}
+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: a #GstBin
- * @rtpbin: a rtpbin #GstElement
+ * @bin: (transfer none): a #GstBin
+ * @rtpbin: (transfer none): a rtpbin #GstElement
*
* Remove the elements of @stream from @bin.
*
priv = stream->priv;
g_mutex_lock (&priv->lock);
- if (!priv->is_joined)
+ 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 */
- g_return_val_if_fail (priv->transports == NULL, FALSE);
+ if (priv->transports != NULL)
+ goto transports_not_removed;
+
+ clear_tr_cache (priv, TRUE);
+ clear_tr_cache (priv, FALSE);
GST_INFO ("stream %p leaving bin", stream);
- gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
- g_signal_handler_disconnect (priv->send_rtp_sink, 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;
+ 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++) {
- if (priv->udpsink[i])
- gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
- if (priv->appsink[i])
- gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
- if (priv->appqueue[i])
- gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
- if (priv->tee[i])
- gst_element_set_state (priv->tee[i], GST_STATE_NULL);
- if (priv->funnel[i])
- gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
- if (priv->appsrc[i])
- gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
- if (priv->udpsrc_v4[i]) {
- /* and set udpsrc to NULL now before removing */
- gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
- gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
- /* removing them should also nicely release the request
- * pads when they finalize */
- gst_bin_remove (bin, priv->udpsrc_v4[i]);
+ 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->udpsrc_v6[i]) {
- gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
- gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
- gst_bin_remove (bin, priv->udpsrc_v6[i]);
- }
- if (priv->udpsink[i])
- gst_bin_remove (bin, priv->udpsink[i]);
- if (priv->appsrc[i])
- gst_bin_remove (bin, priv->appsrc[i]);
- if (priv->appsink[i])
- gst_bin_remove (bin, priv->appsink[i]);
- if (priv->appqueue[i])
- gst_bin_remove (bin, priv->appqueue[i]);
- if (priv->tee[i])
- gst_bin_remove (bin, priv->tee[i]);
- if (priv->funnel[i])
- gst_bin_remove (bin, priv->funnel[i]);
-
- gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
- gst_object_unref (priv->recv_sink[i]);
- priv->recv_sink[i] = NULL;
-
- priv->udpsrc_v4[i] = NULL;
- priv->udpsrc_v6[i] = NULL;
- priv->udpsink[i] = NULL;
- priv->appsrc[i] = NULL;
- priv->appsink[i] = NULL;
- priv->appqueue[i] = NULL;
- priv->tee[i] = NULL;
- priv->funnel[i] = NULL;
- }
- gst_object_unref (priv->send_src[0]);
- priv->send_src[0] = 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]);
gst_caps_unref (priv->caps);
priv->caps = NULL;
- priv->is_joined = FALSE;
+ 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_clear_object (&priv->joined_bin);
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): 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;
}
/**
* @stream: a #GstRTSPStream
* @rtptime: (allow-none): result RTP timestamp
* @seq: (allow-none): result RTP seqnum
- * @clock_rate: the clock rate
+ * @clock_rate: (allow-none): the clock rate
* @running_time: (allow-none): result running-time
*
* Retrieve the current rtptime, seq and running-time. This is used to
g_mutex_lock (&priv->lock);
+ /* First try to extract the information from the last buffer on the sinks.
+ * This will have a more accurate sequence number and timestamp, as between
+ * the payloader and the sink there can be some queues
+ */
+ if (priv->udpsink[0] || priv->appsink[0]) {
+ GstSample *last_sample;
+
+ if (priv->udpsink[0])
+ g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
+ else
+ g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
+
+ if (last_sample) {
+ GstCaps *caps;
+ GstBuffer *buffer;
+ GstSegment *segment;
+ GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
+
+ caps = gst_sample_get_caps (last_sample);
+ buffer = gst_sample_get_buffer (last_sample);
+ segment = gst_sample_get_segment (last_sample);
+
+ if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
+ if (seq) {
+ *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
+ }
+
+ if (rtptime) {
+ *rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
+ }
+
+ gst_rtp_buffer_unmap (&rtp_buffer);
+
+ if (running_time) {
+ *running_time =
+ gst_segment_to_running_time (segment, GST_FORMAT_TIME,
+ GST_BUFFER_TIMESTAMP (buffer));
+ }
+
+ if (clock_rate) {
+ GstStructure *s = gst_caps_get_structure (caps, 0);
+
+ gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
+
+ if (*clock_rate == 0 && running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+ gst_sample_unref (last_sample);
+
+ goto done;
+ } else {
+ gst_sample_unref (last_sample);
+ }
+ }
+ }
+
if (g_object_class_find_property (payobjclass, "stats")) {
g_object_get (priv->payloader, "stats", &stats, NULL);
if (stats == NULL)
if (running_time)
*running_time = GST_CLOCK_TIME_NONE;
}
+
+done:
g_mutex_unlock (&priv->lock);
return TRUE;
* Retrieve the current caps of @stream.
*
* Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
- * after usage.
+ * after usage.
*/
GstCaps *
gst_rtsp_stream_get_caps (GstRTSPStream * stream)
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
priv = stream->priv;
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
- g_return_val_if_fail (priv->is_joined, FALSE);
+ g_return_val_if_fail (priv->joined_bin != NULL, FALSE);
g_mutex_lock (&priv->lock);
if (priv->appsrc[0])
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 {
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
priv = stream->priv;
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
- g_return_val_if_fail (priv->is_joined, FALSE);
+ 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]);
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;
}
tr = gst_rtsp_stream_transport_get_transport (trans);
switch (tr->lower_transport) {
- case GST_RTSP_LOWER_TRANS_UDP:
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
{
+ if (add) {
+ if (!check_mcast_part_for_transport (stream, tr))
+ goto mcast_error;
+ priv->transports = g_list_prepend (priv->transports, trans);
+ } else {
+ priv->transports = g_list_remove (priv->transports, trans);
+ }
+ break;
+ }
+ case GST_RTSP_LOWER_TRANS_UDP:
+ {
gchar *dest;
gint min, max;
guint ttl = 0;
min = tr->port.min;
max = tr->port.max;
ttl = tr->ttl;
+ } else if (priv->client_side) {
+ /* In client side mode the 'destination' is the RTSP server, so send
+ * to those ports */
+ min = tr->server_port.min;
+ max = tr->server_port.max;
} else {
min = tr->client_port.min;
max = tr->client_port.max;
}
if (add) {
- GST_INFO ("adding %s:%d-%d", dest, min, max);
- g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
- g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
if (ttl > 0) {
GST_INFO ("setting ttl-mc %d", ttl);
g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
}
+ GST_INFO ("adding %s:%d-%d", dest, min, max);
+ g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
+ g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
priv->transports = g_list_prepend (priv->transports, trans);
} else {
GST_INFO ("removing %s:%d-%d", dest, min, max);
g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
priv->transports = g_list_remove (priv->transports, trans);
}
+ priv->transports_cookie++;
break;
}
case GST_RTSP_LOWER_TRANS_TCP:
GST_INFO ("removing TCP %s", tr->destination);
priv->transports = g_list_remove (priv->transports, trans);
}
+ priv->transports_cookie++;
break;
default:
goto unknown_transport;
GST_INFO ("Unknown transport %d", tr->lower_transport);
return FALSE;
}
+mcast_error:
+ {
+ return FALSE;
+ }
}
/**
* gst_rtsp_stream_add_transport:
* @stream: a #GstRTSPStream
- * @trans: a #GstRTSPStreamTransport
+ * @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.
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
- g_return_val_if_fail (priv->is_joined, FALSE);
+ g_return_val_if_fail (priv->joined_bin != NULL, FALSE);
g_mutex_lock (&priv->lock);
res = update_transport (stream, trans, TRUE);
/**
* gst_rtsp_stream_remove_transport:
* @stream: a #GstRTSPStream
- * @trans: a #GstRTSPStreamTransport
+ * @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.
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
- g_return_val_if_fail (priv->is_joined, FALSE);
+ g_return_val_if_fail (priv->joined_bin != NULL, FALSE);
g_mutex_lock (&priv->lock);
res = update_transport (stream, trans, FALSE);
}
/**
+ * 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
*
* @stream must be joined to a bin.
*
- * Returns: the RTP socket or %NULL if no socket could be allocated for @family.
- * Unref after usage
+ * 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)
*
* @stream must be joined to a bin.
*
- * Returns: the RTCP socket or %NULL if no socket could be allocated for
- * @family. Unref after usage
+ * 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)
}
/**
+ * 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: user data passed to @func
+ * @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
{
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)
+ 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);
- else
+
+ 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);
default:
break;
}
+ if (changed)
+ goto restart;
}
g_mutex_unlock (&priv->lock);
+ if (func)
+ g_hash_table_unref (visited);
+
return result;
}
return result;
}
+
+/**
+ * gst_rtsp_stream_query_position:
+ * @stream: a #GstRTSPStream
+ *
+ * Query the position of the stream in %GST_FORMAT_TIME. This only considers
+ * the RTP parts of the pipeline and not the RTCP parts.
+ *
+ * Returns: %TRUE if the position could be queried
+ */
+gboolean
+gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
+{
+ GstRTSPStreamPrivate *priv;
+ GstElement *sink;
+ gboolean ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ /* depending on the transport type, it should query corresponding sink */
+ if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
+ (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
+ sink = priv->udpsink[0];
+ else
+ sink = priv->appsink[0];
+
+ if (sink)
+ gst_object_ref (sink);
+ g_mutex_unlock (&priv->lock);
+
+ if (!sink)
+ return FALSE;
+
+ ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
+ gst_object_unref (sink);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_query_stop:
+ * @stream: a #GstRTSPStream
+ *
+ * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
+ * the RTP parts of the pipeline and not the RTCP parts.
+ *
+ * Returns: %TRUE if the stop could be queried
+ */
+gboolean
+gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
+{
+ GstRTSPStreamPrivate *priv;
+ GstElement *sink;
+ GstQuery *query;
+ gboolean ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ /* depending on the transport type, it should query corresponding sink */
+ if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
+ (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
+ sink = priv->udpsink[0];
+ else
+ sink = priv->appsink[0];
+
+ if (sink)
+ gst_object_ref (sink);
+ g_mutex_unlock (&priv->lock);
+
+ if (!sink)
+ return FALSE;
+
+ query = gst_query_new_segment (GST_FORMAT_TIME);
+ if ((ret = gst_element_query (sink, query))) {
+ GstFormat format;
+
+ gst_query_parse_segment (query, NULL, &format, NULL, stop);
+ if (format != GST_FORMAT_TIME)
+ *stop = -1;
+ }
+ gst_query_unref (query);
+ gst_object_unref (sink);
+
+ return ret;
+
+}
projects / platform / upstream / gstreamer.git / blobdiff