/* 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
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
+/**
+ * SECTION:rtsp-stream
+ * @short_description: A media stream
+ * @see_also: #GstRTSPMedia
+ *
+ * The #GstRTSPStream object manages the data transport for one stream. It
+ * is created from a payloader element and a source pad that produce the RTP
+ * packets for the stream.
+ *
+ * With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
+ * and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
+ *
+ * The #GstRTSPStream will use the configured addresspool, as set with
+ * gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
+ * stream. With gst_rtsp_stream_get_multicast_address() you can get the
+ * configured address.
+ *
+ * With gst_rtsp_stream_get_server_port () you can get the port that the server
+ * will use to receive RTCP. This is the part that the clients will use to send
+ * RTCP to.
+ *
+ * With gst_rtsp_stream_add_transport() destinations can be added where the
+ * stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
+ * the destination again.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
-#include <string.h>
#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
#include <gio/gio.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
#include "rtsp-stream.h"
#define GST_RTSP_STREAM_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
+typedef struct
+{
+ GstRTSPStreamTransport *transport;
+
+ /* RTP and RTCP source */
+ GstElement *udpsrc[2];
+ GstPad *selpad[2];
+} GstRTSPMulticastTransportSource;
+
struct _GstRTSPStreamPrivate
{
GMutex lock;
guint idx;
- GstPad *srcpad;
+ /* Only one pad is ever set */
+ GstPad *srcpad, *sinkpad;
GstElement *payloader;
- gboolean is_ipv6;
guint buffer_size;
gboolean is_joined;
+ gchar *control;
+
+ GstRTSPProfile profiles;
+ GstRTSPLowerTrans protocols;
/* pads on the rtpbin */
GstPad *send_rtp_sink;
+ GstPad *recv_rtp_src;
GstPad *recv_sink[2];
GstPad *send_src[2];
/* the RTPSession object */
GObject *session;
- /* sinks used for sending and receiving RTP and RTCP, they share
+ /* SRTP encoder/decoder */
+ GstElement *srtpenc;
+ GstElement *srtpdec;
+ GHashTable *keys;
+
+ /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
* sockets */
- GstElement *udpsrc[2];
+ GstElement *udpsrc_v4[2];
+
+ /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
+ * sockets */
+ GstElement *udpsrc_v6[2];
+
+ GstElement *udpqueue[2];
GstElement *udpsink[2];
+
/* for TCP transport */
GstElement *appsrc[2];
+ GstClockTime appsrc_base_time[2];
GstElement *appqueue[2];
GstElement *appsink[2];
GstElement *tee[2];
GstElement *funnel[2];
- /* server ports for sending/receiving */
- GstRTSPRange server_port;
+ /* retransmission */
+ GstElement *rtxsend;
+ guint rtx_pt;
+ GstClockTime rtx_time;
+
+ /* server ports for sending/receiving over ipv4 */
+ GstRTSPRange server_port_v4;
+ GstRTSPAddress *server_addr_v4;
+ gboolean have_ipv4;
+
+ /* server ports for sending/receiving over ipv6 */
+ GstRTSPRange server_port_v6;
+ GstRTSPAddress *server_addr_v6;
+ gboolean have_ipv6;
/* multicast addresses */
GstRTSPAddressPool *pool;
- GstRTSPAddress *addr;
+ GstRTSPAddress *addr_v4;
+ GstRTSPAddress *addr_v6;
/* 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;
+
+
+ /* UDP sources for UDP multicast transports */
+ GList *transport_sources;
+
+ gint dscp_qos;
+
+ /* stream blocking */
+ gulong blocked_id;
+ gboolean blocking;
+
+ /* pt->caps map for RECORD streams */
+ GHashTable *ptmap;
};
+#define DEFAULT_CONTROL NULL
+#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+ GST_RTSP_LOWER_TRANS_TCP
enum
{
PROP_0,
+ PROP_CONTROL,
+ PROP_PROFILES,
+ PROP_PROTOCOLS,
PROP_LAST
};
+enum
+{
+ SIGNAL_NEW_RTP_ENCODER,
+ SIGNAL_NEW_RTCP_ENCODER,
+ SIGNAL_LAST
+};
+
GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
#define GST_CAT_DEFAULT rtsp_stream_debug
static GQuark ssrc_stream_map_key;
+static void gst_rtsp_stream_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_stream_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+
static void gst_rtsp_stream_finalize (GObject * obj);
+static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
+
G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
static void
gobject_class = G_OBJECT_CLASS (klass);
+ gobject_class->get_property = gst_rtsp_stream_get_property;
+ gobject_class->set_property = gst_rtsp_stream_set_property;
gobject_class->finalize = gst_rtsp_stream_finalize;
+ g_object_class_install_property (gobject_class, PROP_CONTROL,
+ g_param_spec_string ("control", "Control",
+ "The control string for this stream", DEFAULT_CONTROL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROFILES,
+ g_param_spec_flags ("profiles", "Profiles",
+ "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+ DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+ g_param_spec_flags ("protocols", "Protocols",
+ "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+ DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
+ g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+ gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
+ g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
stream->priv = priv;
+ priv->dscp_qos = -1;
+ priv->control = g_strdup (DEFAULT_CONTROL);
+ priv->profiles = DEFAULT_PROFILES;
+ priv->protocols = DEFAULT_PROTOCOLS;
+
g_mutex_init (&priv->lock);
+
+ priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
+ NULL, (GDestroyNotify) gst_caps_unref);
+ priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
+ (GDestroyNotify) gst_caps_unref);
}
static void
/* we really need to be unjoined now */
g_return_if_fail (!priv->is_joined);
- if (priv->addr)
- gst_rtsp_address_free (priv->addr);
+ if (priv->addr_v4)
+ gst_rtsp_address_free (priv->addr_v4);
+ if (priv->addr_v6)
+ gst_rtsp_address_free (priv->addr_v6);
+ if (priv->server_addr_v4)
+ gst_rtsp_address_free (priv->server_addr_v4);
+ if (priv->server_addr_v6)
+ gst_rtsp_address_free (priv->server_addr_v6);
if (priv->pool)
g_object_unref (priv->pool);
+ if (priv->rtxsend)
+ g_object_unref (priv->rtxsend);
+
gst_object_unref (priv->payloader);
- 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);
}
+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
- * @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;
}
}
/**
+ * gst_rtsp_stream_get_pt:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the stream payload type.
+ *
+ * Return: the stream payload type.
+ */
+guint
+gst_rtsp_stream_get_pt (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ guint pt;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
+
+ return pt;
+}
+
+/**
+ * gst_rtsp_stream_get_srcpad:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the srcpad associated with @stream.
+ *
+ * Returns: (transfer full): the srcpad. Unref after usage.
+ */
+GstPad *
+gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ if (!stream->priv->srcpad)
+ return NULL;
+
+ return gst_object_ref (stream->priv->srcpad);
+}
+
+/**
+ * gst_rtsp_stream_get_sinkpad:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the sinkpad associated with @stream.
+ *
+ * Returns: (transfer full): the sinkpad. Unref after usage.
+ */
+GstPad *
+gst_rtsp_stream_get_sinkpad (GstRTSPStream * stream)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ if (!stream->priv->sinkpad)
+ return NULL;
+
+ return gst_object_ref (stream->priv->sinkpad);
+}
+
+/**
+ * gst_rtsp_stream_get_control:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the control string to identify this stream.
+ *
+ * Returns: (transfer full): the control string. g_free() after usage.
+ */
+gchar *
+gst_rtsp_stream_get_control (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = g_strdup (priv->control)) == NULL)
+ result = g_strdup_printf ("stream=%u", priv->idx);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_set_control:
+ * @stream: a #GstRTSPStream
+ * @control: a control string
+ *
+ * Set the control string in @stream.
+ */
+void
+gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_free (priv->control);
+ priv->control = g_strdup (control);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_has_control:
+ * @stream: a #GstRTSPStream
+ * @control: a control string
+ *
+ * Check if @stream has the control string @control.
+ *
+ * Returns: %TRUE is @stream has @control as the control string
+ */
+gboolean
+gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->control)
+ res = (g_strcmp0 (priv->control, control) == 0);
+ else {
+ guint streamid;
+
+ if (sscanf (control, "stream=%u", &streamid) > 0)
+ res = (streamid == priv->idx);
+ else
+ res = FALSE;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
* gst_rtsp_stream_set_mtu:
* @stream: a #GstRTSPStream
* @mtu: a new MTU
return mtu;
}
+/* Update the dscp qos property on the udp sinks */
+static void
+update_dscp_qos (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ if (priv->udpsink[0]) {
+ g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
+ NULL);
+ }
+
+ if (priv->udpsink[1]) {
+ g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
+ NULL);
+ }
+}
+
/**
- * gst_rtsp_stream_set_address_pool:
+ * gst_rtsp_stream_set_dscp_qos:
* @stream: a #GstRTSPStream
- * @pool: a #GstRTSPAddressPool
+ * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
*
- * configure @pool to be used as the address pool of @stream.
+ * Configure the dscp qos of the outgoing sockets to @dscp_qos.
*/
void
-gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
- GstRTSPAddressPool * pool)
+gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
{
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);
+ GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
- 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 (dscp_qos < -1 || dscp_qos > 63) {
+ GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
+ return;
+ }
- if (old)
- g_object_unref (old);
+ priv->dscp_qos = dscp_qos;
+
+ update_dscp_qos (stream);
}
/**
- * gst_rtsp_stream_get_address_pool:
+ * gst_rtsp_stream_get_dscp_qos:
* @stream: a #GstRTSPStream
*
- * Get the #GstRTSPAddressPool used as the address pool of @stream.
+ * Get the configured DSCP QoS in of the outgoing sockets.
*
- * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
- * usage.
+ * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
*/
-GstRTSPAddressPool *
-gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
+gint
+gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv;
- GstRTSPAddressPool *result;
- g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
priv = stream->priv;
- g_mutex_lock (&priv->lock);
- if ((result = priv->pool))
- g_object_ref (result);
- g_mutex_unlock (&priv->lock);
-
- return result;
+ return priv->dscp_qos;
}
/**
- * gst_rtsp_stream_get_address:
+ * gst_rtsp_stream_is_transport_supported:
* @stream: a #GstRTSPStream
+ * @transport: (transfer none): a #GstRTSPTransport
*
- * Get the multicast address of @stream.
+ * Check if @transport can be handled by stream
*
- * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
- * allocated. gst_rtsp_address_free() after usage.
+ * Returns: %TRUE if @transport can be handled by @stream.
*/
-GstRTSPAddress *
-gst_rtsp_stream_get_address (GstRTSPStream * stream)
+gboolean
+gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
+ GstRTSPTransport * transport)
{
GstRTSPStreamPrivate *priv;
- GstRTSPAddress *result;
- g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
priv = stream->priv;
g_mutex_lock (&priv->lock);
- if (priv->addr == NULL) {
- if (priv->pool == NULL)
- goto no_pool;
+ if (transport->trans != GST_RTSP_TRANS_RTP)
+ goto unsupported_transmode;
+
+ if (!(transport->profile & priv->profiles))
+ goto unsupported_profile;
+
+ if (!(transport->lower_transport & priv->protocols))
+ goto unsupported_ltrans;
- priv->addr = gst_rtsp_address_pool_acquire_address (priv->pool,
- GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
- if (priv->addr == NULL)
- goto no_address;
- }
- result = gst_rtsp_address_copy (priv->addr);
g_mutex_unlock (&priv->lock);
- return result;
+ return TRUE;
/* ERRORS */
-no_pool:
+unsupported_transmode:
{
- GST_ERROR_OBJECT (stream, "no address pool specified");
+ GST_DEBUG ("unsupported transport mode %d", transport->trans);
g_mutex_unlock (&priv->lock);
- return NULL;
+ return FALSE;
}
-no_address:
+unsupported_profile:
{
- GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
+ GST_DEBUG ("unsupported profile %d", transport->profile);
g_mutex_unlock (&priv->lock);
- return NULL;
+ return FALSE;
+ }
+unsupported_ltrans:
+ {
+ GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
}
}
/**
- * gst_rtsp_stream_reserve_address:
+ * gst_rtsp_stream_set_profiles:
* @stream: a #GstRTSPStream
+ * @profiles: the new profiles
*
- * Get a specific multicast address of @stream.
- *
- * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
- * allocated. gst_rtsp_address_free() after usage.
+ * Configure the allowed profiles for @stream.
*/
-GstRTSPAddress *
-gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
- const gchar * address, guint port, guint n_ports, guint ttl)
+void
+gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
{
GstRTSPStreamPrivate *priv;
- GstRTSPAddress *result;
- 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);
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
priv = stream->priv;
g_mutex_lock (&priv->lock);
- if (priv->addr == NULL) {
- if (priv->pool == NULL)
- goto no_pool;
-
- priv->addr = gst_rtsp_address_pool_reserve_address (priv->pool, address,
- port, n_ports, ttl);
- if (priv->addr == NULL)
- goto no_address;
- } else {
- if (strcmp (priv->addr->address, address) ||
- priv->addr->port != port || priv->addr->n_ports != n_ports ||
- priv->addr->ttl != ttl)
- goto different_address;
- }
- result = gst_rtsp_address_copy (priv->addr);
+ priv->profiles = profiles;
g_mutex_unlock (&priv->lock);
+}
- return result;
-
- /* ERRORS */
-no_pool:
- {
- GST_ERROR_OBJECT (stream, "no address pool specified");
+/**
+ * gst_rtsp_stream_get_profiles:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the allowed profiles of @stream.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPProfile res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->profiles;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_set_protocols:
+ * @stream: a #GstRTSPStream
+ * @protocols: the new flags
+ *
+ * Configure the allowed lower transport for @stream.
+ */
+void
+gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
+ GstRTSPLowerTrans protocols)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->protocols = protocols;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_protocols:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the allowed protocols of @stream.
+ *
+ * Returns: a #GstRTSPLowerTrans
+ */
+GstRTSPLowerTrans
+gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPLowerTrans res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
+ GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->protocols;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_stream_set_address_pool:
+ * @stream: a #GstRTSPStream
+ * @pool: (transfer none): a #GstRTSPAddressPool
+ *
+ * configure @pool to be used as the address pool of @stream.
+ */
+void
+gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
+ GstRTSPAddressPool * pool)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddressPool *old;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ GST_LOG_OBJECT (stream, "set address pool %p", pool);
+
+ g_mutex_lock (&priv->lock);
+ if ((old = priv->pool) != pool)
+ priv->pool = pool ? g_object_ref (pool) : NULL;
+ else
+ old = NULL;
+ g_mutex_unlock (&priv->lock);
+
+ if (old)
+ g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_stream_get_address_pool:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the #GstRTSPAddressPool used as the address pool of @stream.
+ *
+ * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
+ * usage.
+ */
+GstRTSPAddressPool *
+gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddressPool *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->pool))
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_get_multicast_address:
+ * @stream: a #GstRTSPStream
+ * @family: the #GSocketFamily
+ *
+ * Get the multicast address of @stream for @family.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
+ * or %NULL when no address could be allocated. gst_rtsp_address_free()
+ * after usage.
+ */
+GstRTSPAddress *
+gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
+ GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddress *result;
+ GstRTSPAddress **addrp;
+ GstRTSPAddressFlags flags;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ if (family == G_SOCKET_FAMILY_IPV6) {
+ flags = GST_RTSP_ADDRESS_FLAG_IPV6;
+ addrp = &priv->addr_v6;
+ } else {
+ flags = GST_RTSP_ADDRESS_FLAG_IPV4;
+ addrp = &priv->addr_v4;
+ }
+
+ g_mutex_lock (&priv->lock);
+ if (*addrp == NULL) {
+ if (priv->pool == NULL)
+ goto no_pool;
+
+ flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
+
+ *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
+ if (*addrp == NULL)
+ goto no_address;
+ }
+ result = gst_rtsp_address_copy (*addrp);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+
+ /* ERRORS */
+no_pool:
+ {
+ GST_ERROR_OBJECT (stream, "no address pool specified");
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+no_address:
+ {
+ GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_stream_reserve_address:
+ * @stream: a #GstRTSPStream
+ * @address: an address
+ * @port: a port
+ * @n_ports: n_ports
+ * @ttl: a TTL
+ *
+ * Reserve @address and @port as the address and port of @stream.
+ *
+ * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
+ * the address could be reserved. gst_rtsp_address_free() after usage.
+ */
+GstRTSPAddress *
+gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
+ const gchar * address, guint port, guint n_ports, guint ttl)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPAddress *result;
+ GInetAddress *addr;
+ GSocketFamily family;
+ GstRTSPAddress **addrp;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (address != NULL, NULL);
+ g_return_val_if_fail (port > 0, NULL);
+ g_return_val_if_fail (n_ports > 0, NULL);
+ g_return_val_if_fail (ttl > 0, NULL);
+
+ priv = stream->priv;
+
+ addr = g_inet_address_new_from_string (address);
+ if (!addr) {
+ GST_ERROR ("failed to get inet addr from %s", address);
+ family = G_SOCKET_FAMILY_IPV4;
+ } else {
+ family = g_inet_address_get_family (addr);
+ g_object_unref (addr);
+ }
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ addrp = &priv->addr_v6;
+ else
+ addrp = &priv->addr_v4;
+
+ g_mutex_lock (&priv->lock);
+ if (*addrp == NULL) {
+ GstRTSPAddressPoolResult res;
+
+ if (priv->pool == NULL)
+ goto no_pool;
+
+ res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
+ port, n_ports, ttl, addrp);
+ if (res != GST_RTSP_ADDRESS_POOL_OK)
+ goto no_address;
+ } else {
+ if (strcmp ((*addrp)->address, address) ||
+ (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
+ (*addrp)->ttl != ttl)
+ goto different_address;
+ }
+ result = gst_rtsp_address_copy (*addrp);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+
+ /* ERRORS */
+no_pool:
+ {
+ GST_ERROR_OBJECT (stream, "no address pool specified");
g_mutex_unlock (&priv->lock);
return NULL;
}
}
}
-/* must be called with lock */
static gboolean
-alloc_ports (GstRTSPStream * stream)
+alloc_ports_one_family (GstRTSPStream * stream, GstRTSPAddressPool * pool,
+ gint buffer_size, GSocketFamily family, GstElement * udpsrc_out[2],
+ GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
+ GstRTSPAddress ** server_addr_out)
{
GstRTSPStreamPrivate *priv = stream->priv;
GstStateChangeReturn ret;
GstElement *udpsrc0, *udpsrc1;
GstElement *udpsink0, *udpsink1;
+ GSocket *rtp_socket = NULL;
+ GSocket *rtcp_socket;
gint tmp_rtp, tmp_rtcp;
guint count;
gint rtpport, rtcpport;
- GSocket *socket;
- const gchar *host;
+ GList *rejected_addresses = NULL;
+ GstRTSPAddress *addr = NULL;
+ GInetAddress *inetaddr = NULL;
+ GSocketAddress *rtp_sockaddr = NULL;
+ GSocketAddress *rtcp_sockaddr = NULL;
+ const gchar *multisink_socket;
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ multisink_socket = "socket-v6";
+ else
+ multisink_socket = "socket";
udpsrc0 = NULL;
udpsrc1 = NULL;
/* Start with random port */
tmp_rtp = 0;
- if (priv->is_ipv6)
- host = "udp://[::0]";
- else
- host = "udp://0.0.0.0";
+ rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ if (!rtcp_socket)
+ goto no_udp_protocol;
+
+ if (*server_addr_out)
+ gst_rtsp_address_free (*server_addr_out);
/* try to allocate 2 UDP ports, the RTP port should be an even
* number and the RTCP port should be the next (uneven) port */
again:
- udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
- if (udpsrc0 == NULL)
- goto no_udp_protocol;
- g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
- ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
- if (ret == GST_STATE_CHANGE_FAILURE) {
+ if (rtp_socket == NULL) {
+ rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ if (!rtp_socket)
+ goto no_udp_protocol;
+ }
+
+ if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
+ GstRTSPAddressFlags flags;
+
+ if (addr)
+ rejected_addresses = g_list_prepend (rejected_addresses, addr);
+
+ flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
+ if (family == G_SOCKET_FAMILY_IPV6)
+ flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
+ else
+ flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
+
+ addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
+
+ if (addr == NULL)
+ goto no_ports;
+
+ tmp_rtp = addr->port;
+
+ g_clear_object (&inetaddr);
+ inetaddr = g_inet_address_new_from_string (addr->address);
+ } else {
if (tmp_rtp != 0) {
tmp_rtp += 2;
if (++count > 20)
goto no_ports;
+ }
- gst_element_set_state (udpsrc0, GST_STATE_NULL);
- gst_object_unref (udpsrc0);
+ if (inetaddr == NULL)
+ inetaddr = g_inet_address_new_any (family);
+ }
- goto again;
- }
- goto no_udp_protocol;
+ rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
+ if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
+ g_object_unref (rtp_sockaddr);
+ goto again;
+ }
+ g_object_unref (rtp_sockaddr);
+
+ rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
+ if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
+ g_clear_object (&rtp_sockaddr);
+ goto socket_error;
}
- g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
+ tmp_rtp =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
+ g_object_unref (rtp_sockaddr);
/* check if port is even */
if ((tmp_rtp & 1) != 0) {
/* port not even, close and allocate another */
- if (++count > 20)
- goto no_ports;
-
- gst_element_set_state (udpsrc0, GST_STATE_NULL);
- gst_object_unref (udpsrc0);
-
tmp_rtp++;
+ g_clear_object (&rtp_socket);
goto again;
}
- /* allocate port+1 for RTCP now */
- udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
- if (udpsrc1 == NULL)
- goto no_udp_rtcp_protocol;
-
/* set port */
tmp_rtcp = tmp_rtp + 1;
- g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
- ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
- /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
- if (ret == GST_STATE_CHANGE_FAILURE) {
+ rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
+ if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
+ g_object_unref (rtcp_sockaddr);
+ g_clear_object (&rtp_socket);
+ goto again;
+ }
+ g_object_unref (rtcp_sockaddr);
- if (++count > 20)
- goto no_ports;
+ g_clear_object (&inetaddr);
- gst_element_set_state (udpsrc0, GST_STATE_NULL);
- gst_object_unref (udpsrc0);
+ udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
+ udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
- gst_element_set_state (udpsrc1, GST_STATE_NULL);
- gst_object_unref (udpsrc1);
+ if (udpsrc0 == NULL || udpsrc1 == NULL)
+ goto no_udp_protocol;
+
+ g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
+ g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
+
+ ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto element_error;
+ ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto element_error;
- tmp_rtp += 2;
- goto again;
- }
/* all fine, do port check */
g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
goto port_error;
- udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
+ if (udpsink_out[0])
+ udpsink0 = udpsink_out[0];
+ else
+ udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
+
if (!udpsink0)
goto no_udp_protocol;
- g_object_get (G_OBJECT (udpsrc0), "used-socket", &socket, NULL);
- g_object_set (G_OBJECT (udpsink0), "socket", socket, NULL);
- g_object_unref (socket);
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);
- udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
if (!udpsink1)
goto no_udp_protocol;
- if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
- "send-duplicates")) {
- g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
- } else {
- g_warning
- ("old multiudpsink version found without send-duplicates property");
- }
+ 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 (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
- "buffer-size")) {
- g_object_set (G_OBJECT (udpsink0), "buffer-size", priv->buffer_size, NULL);
- } else {
- GST_WARNING ("multiudpsink version found without buffer-size property");
- }
-
- g_object_get (G_OBJECT (udpsrc1), "used-socket", &socket, NULL);
- g_object_set (G_OBJECT (udpsink1), "socket", socket, NULL);
- g_object_unref (socket);
g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
+ /* Needs to be async for RECORD streams, otherwise we will never go to
+ * PLAYING because the sinks will wait for data while the udpsrc can't
+ * provide data with timestamps in PAUSED. */
+ if (priv->sinkpad)
+ g_object_set (G_OBJECT (udpsink0), "async", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
/* we keep these elements, we will further configure them when the
* client told us to really use the UDP ports. */
- priv->udpsrc[0] = udpsrc0;
- priv->udpsrc[1] = udpsrc1;
- priv->udpsink[0] = udpsink0;
- priv->udpsink[1] = udpsink1;
- priv->server_port.min = rtpport;
- priv->server_port.max = rtcpport;
+ udpsrc_out[0] = udpsrc0;
+ udpsrc_out[1] = udpsrc1;
+ udpsink_out[0] = udpsink0;
+ udpsink_out[1] = udpsink1;
+
+ server_port_out->min = rtpport;
+ server_port_out->max = rtcpport;
+
+ *server_addr_out = addr;
+ g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
+
+ g_object_unref (rtp_socket);
+ g_object_unref (rtcp_socket);
return TRUE;
{
goto cleanup;
}
-no_udp_rtcp_protocol:
+port_error:
{
goto cleanup;
}
-port_error:
+socket_error:
+ {
+ goto cleanup;
+ }
+element_error:
{
goto cleanup;
}
gst_element_set_state (udpsink0, GST_STATE_NULL);
gst_object_unref (udpsink0);
}
- if (udpsink1) {
- gst_element_set_state (udpsink1, GST_STATE_NULL);
- gst_object_unref (udpsink1);
- }
+ if (inetaddr)
+ g_object_unref (inetaddr);
+ g_list_free_full (rejected_addresses,
+ (GDestroyNotify) gst_rtsp_address_free);
+ if (addr)
+ gst_rtsp_address_free (addr);
+ if (rtp_socket)
+ g_object_unref (rtp_socket);
+ if (rtcp_socket)
+ g_object_unref (rtcp_socket);
return FALSE;
}
}
+/* must be called with lock */
+static gboolean
+alloc_ports (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ priv->have_ipv4 =
+ alloc_ports_one_family (stream, priv->pool, priv->buffer_size,
+ G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
+ &priv->server_port_v4, &priv->server_addr_v4);
+
+ priv->have_ipv6 =
+ alloc_ports_one_family (stream, priv->pool, priv->buffer_size,
+ G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
+ &priv->server_port_v6, &priv->server_addr_v6);
+
+ return priv->have_ipv4 || priv->have_ipv6;
+}
+
/**
* gst_rtsp_stream_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)
+ GstRTSPRange * server_port, GSocketFamily family)
{
GstRTSPStreamPrivate *priv;
g_return_if_fail (priv->is_joined);
g_mutex_lock (&priv->lock);
- if (server_port)
- *server_port = priv->server_port;
+ if (family == G_SOCKET_FAMILY_IPV4) {
+ if (server_port)
+ *server_port = priv->server_port_v4;
+ } else {
+ if (server_port)
+ *server_port = priv->server_port_v6;
+ }
g_mutex_unlock (&priv->lock);
}
/**
+ * gst_rtsp_stream_get_rtpsession:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the RTP session of this stream.
+ *
+ * Returns: (transfer full): The RTP session of this stream. Unref after usage.
+ */
+GObject *
+gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GObject *session;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((session = priv->session))
+ g_object_ref (session);
+ g_mutex_unlock (&priv->lock);
+
+ return session;
+}
+
+/**
* gst_rtsp_stream_get_ssrc:
* @stream: a #GstRTSPStream
* @ssrc: (out): result ssrc
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)
break;
}
}
+ if (result)
+ g_object_ref (result);
g_mutex_unlock (&priv->lock);
g_free (dest);
if ((trans = find_transport (stream, rtcp_from))) {
GST_INFO ("%p: found transport %p for source %p", stream, trans,
source);
- g_object_set_qdata (source, ssrc_stream_map_key, trans);
+ g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
+ g_object_unref);
}
gst_structure_free (stats);
}
}
-static void
-on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+static void
+on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ GST_INFO ("%p: new source %p", stream, source);
+
+ trans = check_transport (source, stream);
+
+ if (trans)
+ GST_INFO ("%p: source %p for transport %p", stream, source, trans);
+}
+
+static void
+on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GST_INFO ("%p: new SDES %p", stream, source);
+}
+
+static void
+on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ trans = check_transport (source, stream);
+
+ if (trans) {
+ GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
+ gst_rtsp_stream_transport_keep_alive (trans);
+ }
+#ifdef DUMP_STATS
+ {
+ GstStructure *stats;
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ dump_structure (stats);
+ gst_structure_free (stats);
+ }
+ }
+#endif
+}
+
+static void
+on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GST_INFO ("%p: source %p bye", stream, source);
+}
+
+static void
+on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ GST_INFO ("%p: source %p bye timeout", stream, source);
+
+ if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
+ gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
+ g_object_set_qdata (source, ssrc_stream_map_key, NULL);
+ }
+}
+
+static void
+on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GstRTSPStreamTransport *trans;
+
+ GST_INFO ("%p: source %p timeout", stream, source);
+
+ if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
+ gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
+ g_object_set_qdata (source, ssrc_stream_map_key, NULL);
+ }
+}
+
+static void
+on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+ GST_INFO ("%p: new sender source %p", stream, source);
+#ifndef DUMP_STATS
+ {
+ GstStructure *stats;
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ dump_structure (stats);
+ gst_structure_free (stats);
+ }
+ }
+#endif
+}
+
+static void
+on_sender_ssrc_active (GObject * session, GObject * source,
+ GstRTSPStream * stream)
+{
+#ifndef DUMP_STATS
+ {
+ GstStructure *stats;
+ g_object_get (source, "stats", &stats, NULL);
+ if (stats) {
+ dump_structure (stats);
+ gst_structure_free (stats);
+ }
+ }
+#endif
+}
+
+static void
+clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
+{
+ if (is_rtp) {
+ g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
+ g_list_free (priv->tr_cache_rtp);
+ priv->tr_cache_rtp = NULL;
+ } else {
+ g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
+ g_list_free (priv->tr_cache_rtcp);
+ priv->tr_cache_rtcp = NULL;
+ }
+}
+
+static GstFlowReturn
+handle_new_sample (GstAppSink * sink, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GList *walk;
+ GstSample *sample;
+ GstBuffer *buffer;
+ GstRTSPStream *stream;
+ gboolean is_rtp;
+
+ sample = gst_app_sink_pull_sample (sink);
+ if (!sample)
+ return GST_FLOW_OK;
+
+ stream = (GstRTSPStream *) user_data;
+ priv = stream->priv;
+ buffer = gst_sample_get_buffer (sample);
+
+ is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
+
+ g_mutex_lock (&priv->lock);
+ if (is_rtp) {
+ if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
+ clear_tr_cache (priv, is_rtp);
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+ priv->tr_cache_rtp =
+ g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
+ }
+ priv->tr_cache_cookie_rtp = priv->transports_cookie;
+ }
+ } else {
+ if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
+ clear_tr_cache (priv, is_rtp);
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+ priv->tr_cache_rtcp =
+ g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
+ }
+ priv->tr_cache_cookie_rtcp = priv->transports_cookie;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (is_rtp) {
+ for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+ gst_rtsp_stream_transport_send_rtp (tr, buffer);
+ }
+ } else {
+ for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+ gst_rtsp_stream_transport_send_rtcp (tr, buffer);
+ }
+ }
+ gst_sample_unref (sample);
+
+ return GST_FLOW_OK;
+}
+
+static GstAppSinkCallbacks sink_cb = {
+ NULL, /* not interested in EOS */
+ NULL, /* not interested in preroll samples */
+ handle_new_sample,
+};
+
+static GstElement *
+get_rtp_encoder (GstRTSPStream * stream, guint session)
{
- GstRTSPStreamTransport *trans;
+ GstRTSPStreamPrivate *priv = stream->priv;
- GST_INFO ("%p: new source %p", stream, source);
+ if (priv->srtpenc == NULL) {
+ gchar *name;
- trans = check_transport (source, stream);
+ name = g_strdup_printf ("srtpenc_%u", session);
+ priv->srtpenc = gst_element_factory_make ("srtpenc", name);
+ g_free (name);
- if (trans)
- GST_INFO ("%p: source %p for transport %p", stream, source, trans);
+ g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
+ }
+ return gst_object_ref (priv->srtpenc);
}
-static void
-on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
+static GstElement *
+request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
{
- GST_INFO ("%p: new SDES %p", stream, source);
+ 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 void
-on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
+static GstElement *
+request_rtcp_encoder (GstElement * rtpbin, guint session,
+ GstRTSPStream * stream)
{
- GstRTSPStreamTransport *trans;
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstElement *oldenc, *enc;
+ GstPad *pad;
+ gchar *name;
- trans = check_transport (source, stream);
+ if (priv->idx != session)
+ return NULL;
- 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
+ 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 void
-on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+static GstCaps *
+request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
{
- GST_INFO ("%p: source %p bye", stream, source);
+ 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 void
-on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+static GstElement *
+request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
+ GstRTSPStream * stream)
{
- GstRTSPStreamTransport *trans;
+ GstRTSPStreamPrivate *priv = stream->priv;
- GST_INFO ("%p: source %p bye timeout", stream, source);
+ if (priv->idx != session)
+ return NULL;
- if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
- gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
+ 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);
}
-static void
-on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+/**
+ * gst_rtsp_stream_request_aux_sender:
+ * @stream: a #GstRTSPStream
+ * @sessid: the session id
+ *
+ * Creating a rtxsend bin
+ *
+ * Returns: (transfer full): a #GstElement.
+ *
+ * Since: 1.6
+ */
+GstElement *
+gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
{
- GstRTSPStreamTransport *trans;
+ GstElement *bin;
+ GstPad *pad;
+ GstStructure *pt_map;
+ gchar *name;
+ guint pt, rtx_pt;
+ gchar *pt_s;
- GST_INFO ("%p: source %p timeout", stream, source);
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
- if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
- gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
- }
+ 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 GstFlowReturn
-handle_new_sample (GstAppSink * sink, gpointer user_data)
+/**
+ * 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;
- GList *walk;
- GstSample *sample;
- GstBuffer *buffer;
- GstRTSPStream *stream;
+ GstRTSPStreamPrivate *priv = stream->priv;
- sample = gst_app_sink_pull_sample (sink);
- if (!sample)
- return GST_FLOW_OK;
+ g_mutex_lock (&priv->lock);
+ g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
+ g_mutex_unlock (&priv->lock);
+}
- stream = (GstRTSPStream *) user_data;
- priv = stream->priv;
- buffer = gst_sample_get_buffer (sample);
+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);
- for (walk = priv->transports; walk; walk = g_list_next (walk)) {
- GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
- if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
- gst_rtsp_stream_transport_send_rtp (tr, buffer);
+ 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_rtsp_stream_transport_send_rtcp (tr, buffer);
+ GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
}
}
+
g_mutex_unlock (&priv->lock);
- gst_sample_unref (sample);
+ return caps;
+}
- return GST_FLOW_OK;
+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 GstAppSinkCallbacks sink_cb = {
- NULL, /* not interested in EOS */
- NULL, /* not interested in preroll samples */
- handle_new_sample,
-};
+static void
+on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
+ GstRTSPStream * stream)
+{
+ /* TODO: What to do here other than this? */
+ GST_DEBUG ("Stream %p: Got EOS", stream);
+ gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
+}
/**
* gst_rtsp_stream_join_bin:
* @stream: a #GstRTSPStream
- * @bin: a #GstBin to join
- * @rtpbin: a rtpbin element in @bin
+ * @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.
+ * 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.
GstElement * rtpbin, GstState state)
{
GstRTSPStreamPrivate *priv;
- gint i, idx;
+ gint i;
+ guint idx;
gchar *name;
- GstPad *pad, *teepad, *queuepad, *selpad;
+ GstPad *pad, *sinkpad, *selpad;
GstPadLinkReturn ret;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
/* create a session with the same index as the stream */
idx = priv->idx;
- GST_INFO ("stream %p joining bin as session %d", stream, idx);
+ GST_INFO ("stream %p joining bin as session %u", stream, idx);
if (!alloc_ports (stream))
goto no_ports;
+ /* update the dscp qos field in the sinks */
+ update_dscp_qos (stream);
+
+ if (priv->profiles & GST_RTSP_PROFILE_SAVP
+ || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
+ /* For SRTP */
+ g_signal_connect (rtpbin, "request-rtp-encoder",
+ (GCallback) request_rtp_encoder, stream);
+ g_signal_connect (rtpbin, "request-rtcp-encoder",
+ (GCallback) request_rtcp_encoder, stream);
+ g_signal_connect (rtpbin, "request-rtp-decoder",
+ (GCallback) request_rtp_rtcp_decoder, stream);
+ g_signal_connect (rtpbin, "request-rtcp-decoder",
+ (GCallback) request_rtp_rtcp_decoder, stream);
+ }
+
+ if (priv->sinkpad) {
+ g_signal_connect (rtpbin, "request-pt-map",
+ (GCallback) request_pt_map, stream);
+ }
+
/* get 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;
+
+ if (priv->srcpad) {
+ /* 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;
+ } 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);
+ }
/* 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 ("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);
g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
stream);
+ /* signal for sender ssrc */
+ g_signal_connect (priv->session, "on-new-sender-ssrc",
+ (GCallback) on_new_sender_ssrc, stream);
+ g_signal_connect (priv->session, "on-sender-ssrc-active",
+ (GCallback) on_sender_ssrc_active, stream);
+
for (i = 0; i < 2; i++) {
+ GstPad *teepad, *queuepad;
/* For the sender we create this bit of pipeline for both
* RTP and RTCP. Sync and preroll are enabled on udpsink so
- * we need to add a queue before appsink 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
+ * data to the client as fast as possible.
*
- * .--------. .-----. .---------.
- * | 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.
*/
- /* make tee for RTP/RTCP */
- priv->tee[i] = gst_element_factory_make ("tee", NULL);
- gst_bin_add (bin, priv->tee[i]);
-
- /* and link to rtpbin send pad */
- pad = gst_element_get_static_pad (priv->tee[i], "sink");
- gst_pad_link (priv->send_src[i], pad);
- gst_object_unref (pad);
-
/* add udpsink */
gst_bin_add (bin, priv->udpsink[i]);
-
- /* link tee to udpsink */
- teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
- pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
- gst_pad_link (teepad, pad);
- gst_object_unref (pad);
- 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);
-
- /* 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");
- pad = gst_element_get_static_pad (priv->appsink[i], "sink");
- gst_pad_link (queuepad, pad);
- gst_object_unref (pad);
- gst_object_unref (queuepad);
+ sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
+
+ if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
+ /* make tee for RTP/RTCP */
+ priv->tee[i] = gst_element_factory_make ("tee", NULL);
+ gst_bin_add (bin, priv->tee[i]);
+
+ /* and link to rtpbin send pad */
+ pad = gst_element_get_static_pad (priv->tee[i], "sink");
+ gst_pad_link (priv->send_src[i], pad);
+ gst_object_unref (pad);
+
+ priv->udpqueue[i] = gst_element_factory_make ("queue", NULL);
+ g_object_set (priv->udpqueue[i], "max-size-buffers",
+ 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0), NULL);
+ gst_bin_add (bin, priv->udpqueue[i]);
+ /* link tee to udpqueue */
+ teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
+ pad = gst_element_get_static_pad (priv->udpqueue[i], "sink");
+ gst_pad_link (teepad, pad);
+ gst_object_unref (pad);
+ gst_object_unref (teepad);
+
+ /* link udpqueue to udpsink */
+ queuepad = gst_element_get_static_pad (priv->udpqueue[i], "src");
+ gst_pad_link (queuepad, sinkpad);
+ gst_object_unref (queuepad);
+
+ /* make queue */
+ priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
+ g_object_set (priv->appqueue[i], "max-size-buffers",
+ 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0), NULL);
+ gst_bin_add (bin, priv->appqueue[i]);
+ /* and link 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);
+
+ /* 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");
+ pad = gst_element_get_static_pad (priv->appsink[i], "sink");
+ gst_pad_link (queuepad, 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);
+ }
+ gst_object_unref (sinkpad);
/* For the receiver we create this bit of pipeline for both
* RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
gst_pad_link (pad, priv->recv_sink[i]);
gst_object_unref (pad);
- /* we set and keep these to playing so that they don't cause NO_PREROLL return
- * values */
- gst_element_set_state (priv->udpsrc[i], GST_STATE_PLAYING);
- gst_element_set_locked_state (priv->udpsrc[i], TRUE);
- /* add udpsrc */
- gst_bin_add (bin, priv->udpsrc[i]);
- /* and link to the funnel */
- selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
- pad = gst_element_get_static_pad (priv->udpsrc[i], "src");
- gst_pad_link (pad, selpad);
- gst_object_unref (pad);
- gst_object_unref (selpad);
-
- /* 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->udpsrc_v4[i]) {
+ if (priv->srcpad) {
+ /* we set and keep these to playing so that they don't cause NO_PREROLL return
+ * values. This is only relevant for PLAY pipelines */
+ gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
+ gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
+ }
+ /* add udpsrc */
+ gst_bin_add (bin, priv->udpsrc_v4[i]);
+
+ /* and link to the funnel v4 */
+ selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+ pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
+ gst_pad_link (pad, selpad);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+ }
+
+ if (priv->udpsrc_v6[i]) {
+ if (priv->srcpad) {
+ gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
+ gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
+ }
+ gst_bin_add (bin, priv->udpsrc_v6[i]);
+
+ /* and link to the funnel v6 */
+ selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+ pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
+ gst_pad_link (pad, selpad);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+ }
+
+ if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
+ /* make and add appsrc */
+ priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
+ priv->appsrc_base_time[i] = -1;
+ g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, NULL);
+ gst_bin_add (bin, priv->appsrc[i]);
+ /* and link to the funnel */
+ selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+ pad = gst_element_get_static_pad (priv->appsrc[i], "src");
+ gst_pad_link (pad, selpad);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+ }
/* check if we need to set to a special state */
if (state != GST_STATE_NULL) {
- gst_element_set_state (priv->udpsink[i], state);
- gst_element_set_state (priv->appsink[i], state);
- gst_element_set_state (priv->appqueue[i], state);
- gst_element_set_state (priv->tee[i], state);
- gst_element_set_state (priv->funnel[i], state);
- gst_element_set_state (priv->appsrc[i], state);
+ 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->udpqueue[i])
+ gst_element_set_state (priv->udpqueue[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);
}
}
/* be notified of caps changes */
- priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
+ priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
(GCallback) caps_notify, stream);
priv->is_joined = TRUE;
no_ports:
{
g_mutex_unlock (&priv->lock);
- GST_WARNING ("failed to allocate ports %d", idx);
+ GST_WARNING ("failed to allocate ports %u", idx);
return FALSE;
}
link_failed:
{
- GST_WARNING ("failed to link stream %d", idx);
+ 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);
/**
* 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. @bin must be set
- * to the NULL state before calling this.
+ * Remove the elements of @stream from @bin.
*
* Return: %TRUE on success.
*/
{
GstRTSPStreamPrivate *priv;
gint i;
+ GList *l;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
goto was_not_joined;
/* 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);
+ if (priv->srcpad) {
+ gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
+ } 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;
+ }
+ 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;
for (i = 0; i < 2; i++) {
- /* and set udpsrc to NULL now before removing */
- gst_element_set_locked_state (priv->udpsrc[i], FALSE);
- gst_element_set_state (priv->udpsrc[i], GST_STATE_NULL);
-
- /* removing them should also nicely release the request
- * pads when they finalize */
- gst_bin_remove (bin, priv->udpsrc[i]);
- gst_bin_remove (bin, priv->udpsink[i]);
- gst_bin_remove (bin, priv->appsrc[i]);
- gst_bin_remove (bin, priv->appsink[i]);
- gst_bin_remove (bin, priv->appqueue[i]);
- gst_bin_remove (bin, priv->tee[i]);
- gst_bin_remove (bin, priv->funnel[i]);
+ if (priv->udpsink[i])
+ gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
+ if (priv->appsink[i])
+ gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
+ if (priv->appqueue[i])
+ gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
+ if (priv->udpqueue[i])
+ gst_element_set_state (priv->udpqueue[i], GST_STATE_NULL);
+ if (priv->tee[i])
+ gst_element_set_state (priv->tee[i], GST_STATE_NULL);
+ if (priv->funnel[i])
+ gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
+ if (priv->appsrc[i])
+ gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
+ if (priv->udpsrc_v4[i]) {
+ /* 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]);
+ }
+ 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]);
+ }
+
+ for (l = priv->transport_sources; l; l = l->next) {
+ GstRTSPMulticastTransportSource *s = l->data;
+
+ if (!s->udpsrc[i])
+ continue;
+
+ gst_element_set_locked_state (s->udpsrc[i], FALSE);
+ gst_element_set_state (s->udpsrc[i], GST_STATE_NULL);
+ gst_bin_remove (bin, s->udpsrc[i]);
+ }
+
+ if (priv->udpsink[i])
+ 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->udpqueue[i])
+ gst_bin_remove (bin, priv->udpqueue[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[i] = NULL;
+ priv->udpsrc_v4[i] = NULL;
+ priv->udpsrc_v6[i] = NULL;
priv->udpsink[i] = NULL;
priv->appsrc[i] = NULL;
priv->appsink[i] = NULL;
priv->appqueue[i] = NULL;
+ priv->udpqueue[i] = NULL;
priv->tee[i] = NULL;
priv->funnel[i] = NULL;
}
+
+ for (l = priv->transport_sources; l; l = l->next) {
+ GstRTSPMulticastTransportSource *s = l->data;
+ g_slice_free (GstRTSPMulticastTransportSource, s);
+ }
+ g_list_free (priv->transport_sources);
+ priv->transport_sources = NULL;
+
gst_object_unref (priv->send_src[0]);
priv->send_src[0] = NULL;
priv->send_src[1] = NULL;
g_object_unref (priv->session);
+ priv->session = NULL;
if (priv->caps)
gst_caps_unref (priv->caps);
+ priv->caps = NULL;
+
+ if (priv->srtpenc)
+ gst_object_unref (priv->srtpenc);
+ if (priv->srtpdec)
+ gst_object_unref (priv->srtpdec);
priv->is_joined = FALSE;
g_mutex_unlock (&priv->lock);
was_not_joined:
{
+ g_mutex_unlock (&priv->lock);
return TRUE;
}
+transports_not_removed:
+ {
+ GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
}
/**
* gst_rtsp_stream_get_rtpinfo:
* @stream: a #GstRTSPStream
- * @rtptime: result RTP timestamp
- * @seq: result RTP seqnum
+ * @rtptime: (allow-none): result RTP timestamp
+ * @seq: (allow-none): result RTP seqnum
+ * @clock_rate: (allow-none): the clock rate
+ * @running_time: (allow-none): result running-time
*
- * Retrieve the current rtptime and seq. This is used to
+ * Retrieve the current rtptime, seq and running-time. This is used to
* construct a RTPInfo reply header.
*
- * Returns: %TRUE when rtptime and seq could be determined.
+ * Returns: %TRUE when rtptime, seq and running-time could be determined.
*/
gboolean
gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
- guint * rtptime, guint * seq)
+ 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);
- g_return_val_if_fail (rtptime != NULL, FALSE);
- g_return_val_if_fail (seq != NULL, FALSE);
priv = stream->priv;
payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
- if (!g_object_class_find_property (payobjclass, "seqnum") ||
- !g_object_class_find_property (payobjclass, "timestamp"))
- return FALSE;
+ g_mutex_lock (&priv->lock);
+
+ /* First try to extract the information from the last buffer on the sinks.
+ * This will have a more accurate sequence number and timestamp, as between
+ * the payloader and the sink there can be some queues
+ */
+ if (priv->udpsink[0] || priv->appsink[0]) {
+ GstSample *last_sample;
+
+ if (priv->udpsink[0])
+ g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
+ else
+ g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
+
+ if (last_sample) {
+ GstCaps *caps;
+ GstBuffer *buffer;
+ GstSegment *segment;
+ GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
+
+ caps = gst_sample_get_caps (last_sample);
+ buffer = gst_sample_get_buffer (last_sample);
+ segment = gst_sample_get_segment (last_sample);
+
+ if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
+ if (seq) {
+ *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
+ }
+
+ if (rtptime) {
+ *rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
+ }
+
+ gst_rtp_buffer_unmap (&rtp_buffer);
+
+ if (running_time) {
+ *running_time =
+ gst_segment_to_running_time (segment, GST_FORMAT_TIME,
+ GST_BUFFER_TIMESTAMP (buffer));
+ }
+
+ if (clock_rate) {
+ GstStructure *s = gst_caps_get_structure (caps, 0);
+
+ gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
+
+ if (*clock_rate == 0 && running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+ gst_sample_unref (last_sample);
+
+ goto done;
+ } else {
+ gst_sample_unref (last_sample);
+ }
+ }
+ }
+
+ if (g_object_class_find_property (payobjclass, "stats")) {
+ g_object_get (priv->payloader, "stats", &stats, NULL);
+ if (stats == NULL)
+ goto no_stats;
+
+ if (seq)
+ gst_structure_get_uint (stats, "seqnum", seq);
+
+ if (rtptime)
+ gst_structure_get_uint (stats, "timestamp", rtptime);
+
+ if (running_time)
+ gst_structure_get_clock_time (stats, "running-time", running_time);
+
+ if (clock_rate) {
+ gst_structure_get_uint (stats, "clock-rate", clock_rate);
+ if (*clock_rate == 0 && running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+ gst_structure_free (stats);
+ } else {
+ if (!g_object_class_find_property (payobjclass, "seqnum") ||
+ !g_object_class_find_property (payobjclass, "timestamp"))
+ goto no_stats;
+
+ if (seq)
+ g_object_get (priv->payloader, "seqnum", seq, NULL);
+
+ if (rtptime)
+ g_object_get (priv->payloader, "timestamp", rtptime, NULL);
+
+ if (running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
- g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
+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;
+ }
}
/**
* 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 (priv->is_joined, FALSE);
g_mutex_lock (&priv->lock);
- element = gst_object_ref (priv->appsrc[0]);
+ if (priv->appsrc[0])
+ element = gst_object_ref (priv->appsrc[0]);
+ else
+ element = NULL;
g_mutex_unlock (&priv->lock);
- ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
-
- gst_object_unref (element);
+ 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;
}
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->is_joined) {
+ gst_buffer_unref (buffer);
+ return GST_FLOW_NOT_LINKED;
+ }
g_mutex_lock (&priv->lock);
- element = gst_object_ref (priv->appsrc[1]);
+ if (priv->appsrc[1])
+ element = gst_object_ref (priv->appsrc[1]);
+ else
+ element = NULL;
g_mutex_unlock (&priv->lock);
- ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
-
- gst_object_unref (element);
+ 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:
{
+ GstRTSPMulticastTransportSource *source;
+ GstBin *bin;
+
+ bin = GST_BIN (gst_object_get_parent (GST_OBJECT (priv->funnel[0])));
+
+ if (add) {
+ gchar *host;
+ gint i;
+ GstPad *selpad, *pad;
+
+ source = g_slice_new0 (GstRTSPMulticastTransportSource);
+ source->transport = trans;
+
+ for (i = 0; i < 2; i++) {
+ host =
+ g_strdup_printf ("udp://%s:%d", tr->destination,
+ (i == 0) ? tr->port.min : tr->port.max);
+ source->udpsrc[i] =
+ gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
+ g_free (host);
+
+ if (priv->srcpad) {
+ /* we set and keep these to playing so that they don't cause NO_PREROLL return
+ * values. This is only relevant for PLAY pipelines */
+ gst_element_set_state (source->udpsrc[i], GST_STATE_PLAYING);
+ gst_element_set_locked_state (source->udpsrc[i], TRUE);
+ }
+ /* add udpsrc */
+ gst_bin_add (bin, source->udpsrc[i]);
+
+ /* and link to the funnel v4 */
+ source->selpad[i] = selpad =
+ gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+ pad = gst_element_get_static_pad (source->udpsrc[i], "src");
+ gst_pad_link (pad, selpad);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+ }
+
+ priv->transport_sources =
+ g_list_prepend (priv->transport_sources, source);
+ } else {
+ GList *l;
+
+ for (l = priv->transport_sources; l; l = l->next) {
+ source = l->data;
+
+ if (source->transport == trans) {
+ priv->transport_sources =
+ g_list_delete_link (priv->transport_sources, l);
+ break;
+ }
+ }
+
+ if (l != NULL) {
+ gint i;
+
+ for (i = 0; i < 2; i++) {
+ /* Will automatically unlink everything */
+ gst_bin_remove (bin,
+ GST_ELEMENT (gst_object_ref (source->udpsrc[i])));
+
+ gst_element_set_state (source->udpsrc[i], GST_STATE_NULL);
+ gst_object_unref (source->udpsrc[i]);
+
+ gst_element_release_request_pad (priv->funnel[i],
+ source->selpad[i]);
+ }
+
+ g_slice_free (GstRTSPMulticastTransportSource, source);
+ }
+ }
+
+ gst_object_unref (bin);
+
+ /* fall through for the generic case */
+ }
+ case GST_RTSP_LOWER_TRANS_UDP:
+ {
gchar *dest;
gint min, max;
guint ttl = 0;
}
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_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.
/**
* 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.
return res;
}
+
+/**
+ * gst_rtsp_stream_update_crypto:
+ * @stream: a #GstRTSPStream
+ * @ssrc: the SSRC
+ * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
+ *
+ * Update the new crypto information for @ssrc in @stream. If information
+ * for @ssrc did not exist, it will be added. If information
+ * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
+ * be removed from @stream.
+ *
+ * Returns: %TRUE if @crypto could be updated
+ */
+gboolean
+gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
+ guint ssrc, GstCaps * crypto)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
+
+ priv = stream->priv;
+
+ GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
+
+ g_mutex_lock (&priv->lock);
+ if (crypto)
+ g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
+ gst_caps_ref (crypto));
+ else
+ g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_get_rtp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
+ * socket could be allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+ GSocket *socket;
+ const gchar *name;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+ g_return_val_if_fail (priv->udpsink[0], NULL);
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ name = "socket-v6";
+ else
+ name = "socket";
+
+ g_object_get (priv->udpsink[0], name, &socket, NULL);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_get_rtcp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTCP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
+ * socket could be allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+ GSocket *socket;
+ const gchar *name;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+ g_return_val_if_fail (priv->udpsink[1], NULL);
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ name = "socket-v6";
+ else
+ name = "socket";
+
+ g_object_get (priv->udpsink[1], name, &socket, NULL);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_set_seqnum:
+ * @stream: a #GstRTSPStream
+ * @seqnum: a new sequence number
+ *
+ * Configure the sequence number in the payloader of @stream to @seqnum.
+ */
+void
+gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
+}
+
+/**
+ * gst_rtsp_stream_get_seqnum:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured sequence number in the payloader of @stream.
+ *
+ * Returns: the sequence number of the payloader.
+ */
+guint16
+gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ guint seqnum;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
+
+ return seqnum;
+}
+
+/**
+ * gst_rtsp_stream_transport_filter:
+ * @stream: a #GstRTSPStream
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: (closure): user data passed to @func
+ *
+ * Call @func for each transport managed by @stream. The result value of @func
+ * determines what happens to the transport. @func will be called with @stream
+ * locked so no further actions on @stream can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
+ * @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
+ *
+ * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
+ * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
+ GstRTSPStreamTransportFilterFunc func, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GList *result, *walk, *next;
+ GHashTable *visited = NULL;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ g_mutex_lock (&priv->lock);
+restart:
+ cookie = priv->transports_cookie;
+ for (walk = priv->transports; walk; walk = next) {
+ GstRTSPStreamTransport *trans = walk->data;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ next = g_list_next (walk);
+
+ if (func) {
+ /* only visit each transport once */
+ if (g_hash_table_contains (visited, trans))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (trans));
+ g_mutex_unlock (&priv->lock);
+
+ res = func (stream, trans, user_data);
+
+ g_mutex_lock (&priv->lock);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->transports_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ update_transport (stream, trans, FALSE);
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (trans));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
+
+static GstPadProbeReturn
+pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPStream *stream;
+
+ stream = user_data;
+ priv = stream->priv;
+
+ GST_DEBUG_OBJECT (pad, "now blocking");
+
+ g_mutex_lock (&priv->lock);
+ priv->blocking = TRUE;
+ g_mutex_unlock (&priv->lock);
+
+ gst_element_post_message (priv->payloader,
+ gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
+ gst_structure_new_empty ("GstRTSPStreamBlocking")));
+
+ return GST_PAD_PROBE_OK;
+}
+
+/**
+ * gst_rtsp_stream_set_blocked:
+ * @stream: a #GstRTSPStream
+ * @blocked: boolean indicating we should block or unblock
+ *
+ * Blocks or unblocks the dataflow on @stream.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (blocked) {
+ priv->blocking = FALSE;
+ if (priv->blocked_id == 0) {
+ priv->blocked_id = gst_pad_add_probe (priv->srcpad,
+ GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
+ GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
+ g_object_ref (stream), g_object_unref);
+ }
+ } else {
+ if (priv->blocked_id != 0) {
+ gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
+ priv->blocked_id = 0;
+ priv->blocking = FALSE;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_is_blocking:
+ * @stream: a #GstRTSPStream
+ *
+ * Check if @stream is blocking on a #GstBuffer.
+ *
+ * Returns: %TRUE if @stream is blocking
+ */
+gboolean
+gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = priv->blocking;
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_query_position:
+ * @stream: a #GstRTSPStream
+ *
+ * Query the position of the stream in %GST_FORMAT_TIME. This only considers
+ * the RTP parts of the pipeline and not the RTCP parts.
+ *
+ * Returns: %TRUE if the position could be queried
+ */
+gboolean
+gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
+{
+ GstRTSPStreamPrivate *priv;
+ GstElement *sink;
+ gboolean ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((sink = priv->udpsink[0]))
+ 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);
+ if ((sink = priv->udpsink[0]))
+ 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