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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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6417 lines
170 KiB
C
6417 lines
170 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
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* Copyright (C) 2015 Centricular Ltd
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* Author: Sebastian Dröge <sebastian@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:rtsp-stream
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* @short_description: A media stream
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* @see_also: #GstRTSPMedia
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*
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* The #GstRTSPStream object manages the data transport for one stream. It
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* is created from a payloader element and a source pad that produce the RTP
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* packets for the stream.
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*
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* With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
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* and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
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*
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* The #GstRTSPStream will use the configured addresspool, as set with
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* gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
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* stream. With gst_rtsp_stream_get_multicast_address() you can get the
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* configured address.
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*
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* With gst_rtsp_stream_get_server_port () you can get the port that the server
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* will use to receive RTCP. This is the part that the clients will use to send
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* RTCP to.
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*
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* With gst_rtsp_stream_add_transport() destinations can be added where the
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* stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
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* the destination again.
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*
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* Each #GstRTSPStreamTransport spawns one queue that will serve as a backlog of a
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* controllable maximum size when the reflux from the TCP connection's backpressure
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* starts spilling all over.
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*
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* Unlike the backlog in rtspconnection, which we have decided should only contain
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* at most one RTP and one RTCP data message in order to allow control messages to
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* go through unobstructed, this backlog only consists of data messages, allowing
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* us to fill it up without concern.
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*
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* When multiple TCP transports exist, for example in the context of a shared media,
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* we only pop samples from our appsinks when at least one of the transports doesn't
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* experience back pressure: this allows us to pace our sample popping to the speed
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* of the fastest client.
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*
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* When a sample is popped, it is either sent directly on transports that don't
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* experience backpressure, or queued on the transport's backlog otherwise. Samples
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* are then popped from that backlog when the transport reports it has sent the message.
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*
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* Once the backlog reaches an overly large duration, the transport is dropped as
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* the client was deemed too slow.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <stdlib.h>
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#include <stdio.h>
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#include <string.h>
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#include <gio/gio.h>
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#include <gst/app/gstappsrc.h>
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#include <gst/app/gstappsink.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "rtsp-stream.h"
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#include "rtsp-server-internal.h"
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struct _GstRTSPStreamPrivate
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{
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GMutex lock;
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guint idx;
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/* Only one pad is ever set */
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GstPad *srcpad, *sinkpad;
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GstElement *payloader;
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guint buffer_size;
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GstBin *joined_bin;
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/* TRUE if this stream is running on
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* the client side of an RTSP link (for RECORD) */
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gboolean client_side;
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gchar *control;
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/* TRUE if stream is complete. This means that the receiver and the sender
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* parts are present in the stream. */
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gboolean is_complete;
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GstRTSPProfile profiles;
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GstRTSPLowerTrans allowed_protocols;
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GstRTSPLowerTrans configured_protocols;
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/* pads on the rtpbin */
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GstPad *send_rtp_sink;
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GstPad *recv_rtp_src;
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GstPad *recv_sink[2];
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GstPad *send_src[2];
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/* the RTPSession object */
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GObject *session;
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/* SRTP encoder/decoder */
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GstElement *srtpenc;
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GstElement *srtpdec;
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GHashTable *keys;
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/* for UDP unicast */
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GstElement *udpsrc_v4[2];
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GstElement *udpsrc_v6[2];
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GstElement *udpqueue[2];
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GstElement *udpsink[2];
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GSocket *socket_v4[2];
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GSocket *socket_v6[2];
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/* for UDP multicast */
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GstElement *mcast_udpsrc_v4[2];
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GstElement *mcast_udpsrc_v6[2];
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GstElement *mcast_udpqueue[2];
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GstElement *mcast_udpsink[2];
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GSocket *mcast_socket_v4[2];
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GSocket *mcast_socket_v6[2];
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GList *mcast_clients;
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/* for TCP transport */
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GstElement *appsrc[2];
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GstClockTime appsrc_base_time[2];
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GstElement *appqueue[2];
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GstElement *appsink[2];
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GstElement *tee[2];
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GstElement *funnel[2];
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/* retransmission */
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GstElement *rtxsend;
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GstElement *rtxreceive;
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guint rtx_pt;
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GstClockTime rtx_time;
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/* rate control */
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gboolean do_rate_control;
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/* Forward Error Correction with RFC 5109 */
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GstElement *ulpfec_decoder;
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GstElement *ulpfec_encoder;
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guint ulpfec_pt;
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gboolean ulpfec_enabled;
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guint ulpfec_percentage;
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/* pool used to manage unicast and multicast addresses */
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GstRTSPAddressPool *pool;
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/* unicast server addr/port */
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GstRTSPAddress *server_addr_v4;
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GstRTSPAddress *server_addr_v6;
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/* multicast addresses */
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GstRTSPAddress *mcast_addr_v4;
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GstRTSPAddress *mcast_addr_v6;
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gchar *multicast_iface;
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guint max_mcast_ttl;
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gboolean bind_mcast_address;
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/* the caps of the stream */
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gulong caps_sig;
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GstCaps *caps;
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/* transports we stream to */
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guint n_active;
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GList *transports;
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guint transports_cookie;
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GPtrArray *tr_cache;
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guint tr_cache_cookie;
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guint n_tcp_transports;
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gboolean have_buffer[2];
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gint dscp_qos;
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/* Sending logic for TCP */
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GThread *send_thread;
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GCond send_cond;
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GMutex send_lock;
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/* @send_lock is released when pushing data out, we use
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* a cookie to decide whether we should wait on @send_cond
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* before checking the transports' backlogs again
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*/
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guint send_cookie;
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/* Used to control shutdown of @send_thread */
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gboolean continue_sending;
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/* stream blocking */
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gulong blocked_id[2];
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gboolean blocking;
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/* current stream postion */
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GstClockTime position;
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/* pt->caps map for RECORD streams */
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GHashTable *ptmap;
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GstRTSPPublishClockMode publish_clock_mode;
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GThreadPool *send_pool;
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/* Used to provide accurate rtpinfo when the stream is blocking */
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gboolean blocked_buffer;
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guint32 blocked_seqnum;
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guint32 blocked_rtptime;
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GstClockTime blocked_running_time;
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gint blocked_clock_rate;
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/* Whether we should send and receive RTCP */
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gboolean enable_rtcp;
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/* blocking early rtcp packets */
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GstPad *block_early_rtcp_pad;
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gulong block_early_rtcp_probe;
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GstPad *block_early_rtcp_pad_ipv6;
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gulong block_early_rtcp_probe_ipv6;
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};
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#define DEFAULT_CONTROL NULL
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#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
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#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
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GST_RTSP_LOWER_TRANS_TCP
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#define DEFAULT_MAX_MCAST_TTL 255
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#define DEFAULT_BIND_MCAST_ADDRESS FALSE
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#define DEFAULT_DO_RATE_CONTROL TRUE
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#define DEFAULT_ENABLE_RTCP TRUE
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enum
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{
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PROP_0,
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PROP_CONTROL,
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PROP_PROFILES,
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PROP_PROTOCOLS,
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PROP_LAST
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};
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enum
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{
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SIGNAL_NEW_RTP_ENCODER,
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SIGNAL_NEW_RTCP_ENCODER,
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SIGNAL_NEW_RTP_RTCP_DECODER,
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SIGNAL_LAST
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};
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GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
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#define GST_CAT_DEFAULT rtsp_stream_debug
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static GQuark ssrc_stream_map_key;
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static void gst_rtsp_stream_get_property (GObject * object, guint propid,
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GValue * value, GParamSpec * pspec);
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static void gst_rtsp_stream_set_property (GObject * object, guint propid,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtsp_stream_finalize (GObject * obj);
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static gboolean
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update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
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gboolean add);
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static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
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G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
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static void
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gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
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{
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GObjectClass *gobject_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gobject_class->get_property = gst_rtsp_stream_get_property;
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gobject_class->set_property = gst_rtsp_stream_set_property;
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gobject_class->finalize = gst_rtsp_stream_finalize;
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g_object_class_install_property (gobject_class, PROP_CONTROL,
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g_param_spec_string ("control", "Control",
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"The control string for this stream", DEFAULT_CONTROL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_PROFILES,
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g_param_spec_flags ("profiles", "Profiles",
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"Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
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DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
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g_param_spec_flags ("protocols", "Protocols",
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"Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
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DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
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g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
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gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
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g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
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gst_rtsp_stream_signals[SIGNAL_NEW_RTP_RTCP_DECODER] =
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g_signal_new ("new-rtp-rtcp-decoder", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
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GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
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ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
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}
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static void
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gst_rtsp_stream_init (GstRTSPStream * stream)
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{
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GstRTSPStreamPrivate *priv = gst_rtsp_stream_get_instance_private (stream);
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GST_DEBUG ("new stream %p", stream);
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stream->priv = priv;
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priv->dscp_qos = -1;
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priv->control = g_strdup (DEFAULT_CONTROL);
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priv->profiles = DEFAULT_PROFILES;
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priv->allowed_protocols = DEFAULT_PROTOCOLS;
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priv->configured_protocols = 0;
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priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
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priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
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priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
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priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
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priv->enable_rtcp = DEFAULT_ENABLE_RTCP;
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g_mutex_init (&priv->lock);
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priv->continue_sending = TRUE;
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priv->send_cookie = 0;
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g_cond_init (&priv->send_cond);
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g_mutex_init (&priv->send_lock);
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priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
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NULL, (GDestroyNotify) gst_caps_unref);
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priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
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(GDestroyNotify) gst_caps_unref);
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priv->send_pool = NULL;
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priv->block_early_rtcp_pad = NULL;
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priv->block_early_rtcp_probe = 0;
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priv->block_early_rtcp_pad_ipv6 = NULL;
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priv->block_early_rtcp_probe_ipv6 = 0;
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}
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typedef struct _UdpClientAddrInfo UdpClientAddrInfo;
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struct _UdpClientAddrInfo
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{
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gchar *address;
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guint rtp_port;
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guint add_count; /* how often this address has been added */
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};
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static void
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free_mcast_client (gpointer data)
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{
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UdpClientAddrInfo *client = data;
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g_free (client->address);
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g_free (client);
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}
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static void
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gst_rtsp_stream_finalize (GObject * obj)
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{
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GstRTSPStream *stream;
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GstRTSPStreamPrivate *priv;
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guint i;
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stream = GST_RTSP_STREAM (obj);
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priv = stream->priv;
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GST_DEBUG ("finalize stream %p", stream);
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/* we really need to be unjoined now */
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g_return_if_fail (priv->joined_bin == NULL);
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if (priv->send_pool)
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g_thread_pool_free (priv->send_pool, TRUE, TRUE);
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if (priv->mcast_addr_v4)
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gst_rtsp_address_free (priv->mcast_addr_v4);
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if (priv->mcast_addr_v6)
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gst_rtsp_address_free (priv->mcast_addr_v6);
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if (priv->server_addr_v4)
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gst_rtsp_address_free (priv->server_addr_v4);
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if (priv->server_addr_v6)
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gst_rtsp_address_free (priv->server_addr_v6);
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if (priv->pool)
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g_object_unref (priv->pool);
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if (priv->rtxsend)
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g_object_unref (priv->rtxsend);
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if (priv->rtxreceive)
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g_object_unref (priv->rtxreceive);
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if (priv->ulpfec_encoder)
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gst_object_unref (priv->ulpfec_encoder);
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if (priv->ulpfec_decoder)
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gst_object_unref (priv->ulpfec_decoder);
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for (i = 0; i < 2; i++) {
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if (priv->socket_v4[i])
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g_object_unref (priv->socket_v4[i]);
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if (priv->socket_v6[i])
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g_object_unref (priv->socket_v6[i]);
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if (priv->mcast_socket_v4[i])
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g_object_unref (priv->mcast_socket_v4[i]);
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if (priv->mcast_socket_v6[i])
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g_object_unref (priv->mcast_socket_v6[i]);
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}
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g_free (priv->multicast_iface);
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g_list_free_full (priv->mcast_clients, (GDestroyNotify) free_mcast_client);
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gst_object_unref (priv->payloader);
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if (priv->srcpad)
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gst_object_unref (priv->srcpad);
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if (priv->sinkpad)
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gst_object_unref (priv->sinkpad);
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g_free (priv->control);
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g_mutex_clear (&priv->lock);
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g_hash_table_unref (priv->keys);
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g_hash_table_destroy (priv->ptmap);
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g_mutex_clear (&priv->send_lock);
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g_cond_clear (&priv->send_cond);
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if (priv->block_early_rtcp_probe != 0) {
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gst_pad_remove_probe
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(priv->block_early_rtcp_pad, priv->block_early_rtcp_probe);
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gst_object_unref (priv->block_early_rtcp_pad);
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}
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if (priv->block_early_rtcp_probe_ipv6 != 0) {
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gst_pad_remove_probe
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(priv->block_early_rtcp_pad_ipv6, priv->block_early_rtcp_probe_ipv6);
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gst_object_unref (priv->block_early_rtcp_pad_ipv6);
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}
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G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
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}
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static void
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gst_rtsp_stream_get_property (GObject * object, guint propid,
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GValue * value, GParamSpec * pspec)
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{
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GstRTSPStream *stream = GST_RTSP_STREAM (object);
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switch (propid) {
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case PROP_CONTROL:
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g_value_take_string (value, gst_rtsp_stream_get_control (stream));
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break;
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case PROP_PROFILES:
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g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
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break;
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case PROP_PROTOCOLS:
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g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
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|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_stream_set_property (GObject * object, guint propid,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTSPStream *stream = GST_RTSP_STREAM (object);
|
|
|
|
switch (propid) {
|
|
case PROP_CONTROL:
|
|
gst_rtsp_stream_set_control (stream, g_value_get_string (value));
|
|
break;
|
|
case PROP_PROFILES:
|
|
gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
|
|
break;
|
|
case PROP_PROTOCOLS:
|
|
gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_new:
|
|
* @idx: an index
|
|
* @pad: a #GstPad
|
|
* @payloader: a #GstElement
|
|
*
|
|
* Create a new media stream with index @idx that handles RTP data on
|
|
* @pad and has a payloader element @payloader if @pad is a source pad
|
|
* or a depayloader element @payloader if @pad is a sink pad.
|
|
*
|
|
* Returns: (transfer full): a new #GstRTSPStream
|
|
*/
|
|
GstRTSPStream *
|
|
gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * pad)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstRTSPStream *stream;
|
|
|
|
g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
|
|
g_return_val_if_fail (GST_IS_PAD (pad), NULL);
|
|
|
|
stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
|
|
priv = stream->priv;
|
|
priv->idx = idx;
|
|
priv->payloader = gst_object_ref (payloader);
|
|
if (GST_PAD_IS_SRC (pad))
|
|
priv->srcpad = gst_object_ref (pad);
|
|
else
|
|
priv->sinkpad = gst_object_ref (pad);
|
|
|
|
return stream;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_index:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the stream index.
|
|
*
|
|
* Return: the stream index.
|
|
*/
|
|
guint
|
|
gst_rtsp_stream_get_index (GstRTSPStream * stream)
|
|
{
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
|
|
|
|
return stream->priv->idx;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_pt:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the stream payload type.
|
|
*
|
|
* Return: the stream payload type.
|
|
*/
|
|
guint
|
|
gst_rtsp_stream_get_pt (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
guint pt;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
|
|
|
|
return pt;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_srcpad:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the srcpad associated with @stream.
|
|
*
|
|
* Returns: (transfer full) (nullable): the srcpad. Unref after usage.
|
|
*/
|
|
GstPad *
|
|
gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
|
|
{
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
if (!stream->priv->srcpad)
|
|
return NULL;
|
|
|
|
return gst_object_ref (stream->priv->srcpad);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_sinkpad:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the sinkpad associated with @stream.
|
|
*
|
|
* Returns: (transfer full) (nullable): the sinkpad. Unref after usage.
|
|
*/
|
|
GstPad *
|
|
gst_rtsp_stream_get_sinkpad (GstRTSPStream * stream)
|
|
{
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
if (!stream->priv->sinkpad)
|
|
return NULL;
|
|
|
|
return gst_object_ref (stream->priv->sinkpad);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_control:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the control string to identify this stream.
|
|
*
|
|
* Returns: (transfer full) (nullable): the control string. g_free() after usage.
|
|
*/
|
|
gchar *
|
|
gst_rtsp_stream_get_control (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gchar *result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((result = g_strdup (priv->control)) == NULL)
|
|
result = g_strdup_printf ("stream=%u", priv->idx);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_control:
|
|
* @stream: a #GstRTSPStream
|
|
* @control: (nullable): a control string
|
|
*
|
|
* Set the control string in @stream.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
g_free (priv->control);
|
|
priv->control = g_strdup (control);
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_has_control:
|
|
* @stream: a #GstRTSPStream
|
|
* @control: (nullable): a control string
|
|
*
|
|
* Check if @stream has the control string @control.
|
|
*
|
|
* Returns: %TRUE is @stream has @control as the control string
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (priv->control)
|
|
res = (g_strcmp0 (priv->control, control) == 0);
|
|
else {
|
|
guint streamid;
|
|
|
|
if (sscanf (control, "stream=%u", &streamid) > 0)
|
|
res = (streamid == priv->idx);
|
|
else
|
|
res = FALSE;
|
|
}
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_mtu:
|
|
* @stream: a #GstRTSPStream
|
|
* @mtu: a new MTU
|
|
*
|
|
* Configure the mtu in the payloader of @stream to @mtu.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
priv = stream->priv;
|
|
|
|
GST_LOG_OBJECT (stream, "set MTU %u", mtu);
|
|
|
|
g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_mtu:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the configured MTU in the payloader of @stream.
|
|
*
|
|
* Returns: the MTU of the payloader.
|
|
*/
|
|
guint
|
|
gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
guint mtu;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
|
|
|
|
return mtu;
|
|
}
|
|
|
|
/* Update the dscp qos property on the udp sinks */
|
|
static void
|
|
update_dscp_qos (GstRTSPStream * stream, GstElement ** udpsink)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
priv = stream->priv;
|
|
|
|
if (*udpsink) {
|
|
g_object_set (G_OBJECT (*udpsink), "qos-dscp", priv->dscp_qos, NULL);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_dscp_qos:
|
|
* @stream: a #GstRTSPStream
|
|
* @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
|
|
*
|
|
* Configure the dscp qos of the outgoing sockets to @dscp_qos.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
priv = stream->priv;
|
|
|
|
GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
|
|
|
|
if (dscp_qos < -1 || dscp_qos > 63) {
|
|
GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
|
|
return;
|
|
}
|
|
|
|
priv->dscp_qos = dscp_qos;
|
|
|
|
update_dscp_qos (stream, priv->udpsink);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_dscp_qos:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the configured DSCP QoS in of the outgoing sockets.
|
|
*
|
|
* Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
|
|
*/
|
|
gint
|
|
gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
|
|
|
|
priv = stream->priv;
|
|
|
|
return priv->dscp_qos;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_is_transport_supported:
|
|
* @stream: a #GstRTSPStream
|
|
* @transport: (transfer none): a #GstRTSPTransport
|
|
*
|
|
* Check if @transport can be handled by stream
|
|
*
|
|
* Returns: %TRUE if @transport can be handled by @stream.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
|
|
GstRTSPTransport * transport)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
g_return_val_if_fail (transport != NULL, FALSE);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (transport->trans != GST_RTSP_TRANS_RTP)
|
|
goto unsupported_transmode;
|
|
|
|
if (!(transport->profile & priv->profiles))
|
|
goto unsupported_profile;
|
|
|
|
if (!(transport->lower_transport & priv->allowed_protocols))
|
|
goto unsupported_ltrans;
|
|
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
unsupported_transmode:
|
|
{
|
|
GST_DEBUG ("unsupported transport mode %d", transport->trans);
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
unsupported_profile:
|
|
{
|
|
GST_DEBUG ("unsupported profile %d", transport->profile);
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
unsupported_ltrans:
|
|
{
|
|
GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_profiles:
|
|
* @stream: a #GstRTSPStream
|
|
* @profiles: the new profiles
|
|
*
|
|
* Configure the allowed profiles for @stream.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
priv->profiles = profiles;
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_profiles:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the allowed profiles of @stream.
|
|
*
|
|
* Returns: a #GstRTSPProfile
|
|
*/
|
|
GstRTSPProfile
|
|
gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstRTSPProfile res;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
res = priv->profiles;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_protocols:
|
|
* @stream: a #GstRTSPStream
|
|
* @protocols: the new flags
|
|
*
|
|
* Configure the allowed lower transport for @stream.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
|
|
GstRTSPLowerTrans protocols)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
priv->allowed_protocols = protocols;
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_protocols:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the allowed protocols of @stream.
|
|
*
|
|
* Returns: a #GstRTSPLowerTrans
|
|
*/
|
|
GstRTSPLowerTrans
|
|
gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstRTSPLowerTrans res;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
|
|
GST_RTSP_LOWER_TRANS_UNKNOWN);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
res = priv->allowed_protocols;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_address_pool:
|
|
* @stream: a #GstRTSPStream
|
|
* @pool: (transfer none) (nullable): a #GstRTSPAddressPool
|
|
*
|
|
* configure @pool to be used as the address pool of @stream.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
|
|
GstRTSPAddressPool * pool)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstRTSPAddressPool *old;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
priv = stream->priv;
|
|
|
|
GST_LOG_OBJECT (stream, "set address pool %p", pool);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((old = priv->pool) != pool)
|
|
priv->pool = pool ? g_object_ref (pool) : NULL;
|
|
else
|
|
old = NULL;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (old)
|
|
g_object_unref (old);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_address_pool:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the #GstRTSPAddressPool used as the address pool of @stream.
|
|
*
|
|
* Returns: (transfer full) (nullable): the #GstRTSPAddressPool of @stream.
|
|
* g_object_unref() after usage.
|
|
*/
|
|
GstRTSPAddressPool *
|
|
gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstRTSPAddressPool *result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((result = priv->pool))
|
|
g_object_ref (result);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_multicast_iface:
|
|
* @stream: a #GstRTSPStream
|
|
* @multicast_iface: (transfer none) (nullable): a multicast interface name
|
|
*
|
|
* configure @multicast_iface to be used for @stream.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_multicast_iface (GstRTSPStream * stream,
|
|
const gchar * multicast_iface)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gchar *old;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
priv = stream->priv;
|
|
|
|
GST_LOG_OBJECT (stream, "set multicast iface %s",
|
|
GST_STR_NULL (multicast_iface));
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((old = priv->multicast_iface) != multicast_iface)
|
|
priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
|
|
else
|
|
old = NULL;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (old)
|
|
g_free (old);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_multicast_iface:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the multicast interface used for @stream.
|
|
*
|
|
* Returns: (transfer full) (nullable): the multicast interface for @stream.
|
|
* g_free() after usage.
|
|
*/
|
|
gchar *
|
|
gst_rtsp_stream_get_multicast_iface (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gchar *result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((result = priv->multicast_iface))
|
|
result = g_strdup (result);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_multicast_address:
|
|
* @stream: a #GstRTSPStream
|
|
* @family: the #GSocketFamily
|
|
*
|
|
* Get the multicast address of @stream for @family. The original
|
|
* #GstRTSPAddress is cached and copy is returned, so freeing the return value
|
|
* won't release the address from the pool.
|
|
*
|
|
* Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
|
|
* or %NULL when no address could be allocated. gst_rtsp_address_free()
|
|
* after usage.
|
|
*/
|
|
GstRTSPAddress *
|
|
gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
|
|
GSocketFamily family)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstRTSPAddress *result;
|
|
GstRTSPAddress **addrp;
|
|
GstRTSPAddressFlags flags;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
|
|
if (family == G_SOCKET_FAMILY_IPV6) {
|
|
flags = GST_RTSP_ADDRESS_FLAG_IPV6;
|
|
addrp = &priv->mcast_addr_v6;
|
|
} else {
|
|
flags = GST_RTSP_ADDRESS_FLAG_IPV4;
|
|
addrp = &priv->mcast_addr_v4;
|
|
}
|
|
|
|
if (*addrp == NULL) {
|
|
if (priv->pool == NULL)
|
|
goto no_pool;
|
|
|
|
flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
|
|
|
|
*addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
|
|
if (*addrp == NULL)
|
|
goto no_address;
|
|
|
|
/* FIXME: Also reserve the same port with unicast ANY address, since that's
|
|
* where we are going to bind our socket. Probably loop until we find a port
|
|
* available in both mcast and unicast pools. Maybe GstRTSPAddressPool
|
|
* should do it for us when both GST_RTSP_ADDRESS_FLAG_MULTICAST and
|
|
* GST_RTSP_ADDRESS_FLAG_UNICAST are givent. */
|
|
}
|
|
result = gst_rtsp_address_copy (*addrp);
|
|
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
no_pool:
|
|
{
|
|
GST_ERROR_OBJECT (stream, "no address pool specified");
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
return NULL;
|
|
}
|
|
no_address:
|
|
{
|
|
GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_reserve_address:
|
|
* @stream: a #GstRTSPStream
|
|
* @address: an address
|
|
* @port: a port
|
|
* @n_ports: n_ports
|
|
* @ttl: a TTL
|
|
*
|
|
* Reserve @address and @port as the address and port of @stream. The original
|
|
* #GstRTSPAddress is cached and copy is returned, so freeing the return value
|
|
* won't release the address from the pool.
|
|
*
|
|
* Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
|
|
* the address could not be reserved. gst_rtsp_address_free() after
|
|
* usage.
|
|
*/
|
|
GstRTSPAddress *
|
|
gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
|
|
const gchar * address, guint port, guint n_ports, guint ttl)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstRTSPAddress *result;
|
|
GInetAddress *addr;
|
|
GSocketFamily family;
|
|
GstRTSPAddress **addrp;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
g_return_val_if_fail (address != NULL, NULL);
|
|
g_return_val_if_fail (port > 0, NULL);
|
|
g_return_val_if_fail (n_ports > 0, NULL);
|
|
g_return_val_if_fail (ttl > 0, NULL);
|
|
|
|
priv = stream->priv;
|
|
|
|
addr = g_inet_address_new_from_string (address);
|
|
if (!addr) {
|
|
GST_ERROR ("failed to get inet addr from %s", address);
|
|
family = G_SOCKET_FAMILY_IPV4;
|
|
} else {
|
|
family = g_inet_address_get_family (addr);
|
|
g_object_unref (addr);
|
|
}
|
|
|
|
if (family == G_SOCKET_FAMILY_IPV6)
|
|
addrp = &priv->mcast_addr_v6;
|
|
else
|
|
addrp = &priv->mcast_addr_v4;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (*addrp == NULL) {
|
|
GstRTSPAddressPoolResult res;
|
|
|
|
if (priv->pool == NULL)
|
|
goto no_pool;
|
|
|
|
res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
|
|
port, n_ports, ttl, addrp);
|
|
if (res != GST_RTSP_ADDRESS_POOL_OK)
|
|
goto no_address;
|
|
|
|
/* FIXME: Also reserve the same port with unicast ANY address, since that's
|
|
* where we are going to bind our socket. */
|
|
} else {
|
|
if (g_ascii_strcasecmp ((*addrp)->address, address) ||
|
|
(*addrp)->port != port || (*addrp)->n_ports != n_ports ||
|
|
(*addrp)->ttl != ttl)
|
|
goto different_address;
|
|
}
|
|
result = gst_rtsp_address_copy (*addrp);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
no_pool:
|
|
{
|
|
GST_ERROR_OBJECT (stream, "no address pool specified");
|
|
g_mutex_unlock (&priv->lock);
|
|
return NULL;
|
|
}
|
|
no_address:
|
|
{
|
|
GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
|
|
address);
|
|
g_mutex_unlock (&priv->lock);
|
|
return NULL;
|
|
}
|
|
different_address:
|
|
{
|
|
GST_ERROR_OBJECT (stream,
|
|
"address %s is not the same as %s that was already reserved",
|
|
address, (*addrp)->address);
|
|
g_mutex_unlock (&priv->lock);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* must be called with lock */
|
|
static void
|
|
set_socket_for_udpsink (GstElement * udpsink, GSocket * socket,
|
|
GSocketFamily family)
|
|
{
|
|
const gchar *multisink_socket;
|
|
|
|
if (family == G_SOCKET_FAMILY_IPV6)
|
|
multisink_socket = "socket-v6";
|
|
else
|
|
multisink_socket = "socket";
|
|
|
|
g_object_set (G_OBJECT (udpsink), multisink_socket, socket, NULL);
|
|
}
|
|
|
|
/* must be called with lock */
|
|
static void
|
|
set_multicast_socket_for_udpsink (GstElement * udpsink, GSocket * socket,
|
|
GSocketFamily family, const gchar * multicast_iface,
|
|
const gchar * addr_str, gint port, gint mcast_ttl)
|
|
{
|
|
set_socket_for_udpsink (udpsink, socket, family);
|
|
|
|
if (multicast_iface) {
|
|
GST_INFO ("setting multicast-iface %s", multicast_iface);
|
|
g_object_set (G_OBJECT (udpsink), "multicast-iface", multicast_iface, NULL);
|
|
}
|
|
|
|
if (mcast_ttl > 0) {
|
|
GST_INFO ("setting ttl-mc %d", mcast_ttl);
|
|
g_object_set (G_OBJECT (udpsink), "ttl-mc", mcast_ttl, NULL);
|
|
}
|
|
}
|
|
|
|
|
|
/* must be called with lock */
|
|
static void
|
|
set_unicast_socket_for_udpsink (GstElement * udpsink, GSocket * socket,
|
|
GSocketFamily family)
|
|
{
|
|
set_socket_for_udpsink (udpsink, socket, family);
|
|
}
|
|
|
|
static guint16
|
|
get_port_from_socket (GSocket * socket)
|
|
{
|
|
guint16 port;
|
|
GSocketAddress *sockaddr;
|
|
GError *err;
|
|
|
|
GST_DEBUG ("socket: %p", socket);
|
|
sockaddr = g_socket_get_local_address (socket, &err);
|
|
if (sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (sockaddr)) {
|
|
g_clear_object (&sockaddr);
|
|
GST_ERROR ("failed to get sockaddr: %s", err->message);
|
|
g_error_free (err);
|
|
return 0;
|
|
}
|
|
|
|
port = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
|
|
g_object_unref (sockaddr);
|
|
|
|
return port;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
create_and_configure_udpsink (GstRTSPStream * stream, GstElement ** udpsink,
|
|
GSocket * socket_v4, GSocket * socket_v6, gboolean multicast,
|
|
gboolean is_rtp, gint mcast_ttl)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
|
|
*udpsink = gst_element_factory_make ("multiudpsink", NULL);
|
|
|
|
if (!*udpsink)
|
|
goto no_udp_protocol;
|
|
|
|
/* configure sinks */
|
|
|
|
g_object_set (G_OBJECT (*udpsink), "close-socket", FALSE, NULL);
|
|
|
|
g_object_set (G_OBJECT (*udpsink), "send-duplicates", FALSE, NULL);
|
|
|
|
if (is_rtp)
|
|
g_object_set (G_OBJECT (*udpsink), "buffer-size", priv->buffer_size, NULL);
|
|
else
|
|
g_object_set (G_OBJECT (*udpsink), "sync", FALSE, NULL);
|
|
|
|
/* Needs to be async for RECORD streams, otherwise we will never go to
|
|
* PLAYING because the sinks will wait for data while the udpsrc can't
|
|
* provide data with timestamps in PAUSED. */
|
|
if (!is_rtp || priv->sinkpad)
|
|
g_object_set (G_OBJECT (*udpsink), "async", FALSE, NULL);
|
|
|
|
if (multicast) {
|
|
/* join multicast group when adding clients, so we'll start receiving from it.
|
|
* We cannot rely on the udpsrc to join the group since its socket is always a
|
|
* local unicast one. */
|
|
g_object_set (G_OBJECT (*udpsink), "auto-multicast", TRUE, NULL);
|
|
|
|
g_object_set (G_OBJECT (*udpsink), "loop", FALSE, NULL);
|
|
}
|
|
|
|
/* update the dscp qos field in the sinks */
|
|
update_dscp_qos (stream, udpsink);
|
|
|
|
if (priv->server_addr_v4) {
|
|
GST_DEBUG_OBJECT (stream, "udp IPv4, configure udpsinks");
|
|
set_unicast_socket_for_udpsink (*udpsink, socket_v4, G_SOCKET_FAMILY_IPV4);
|
|
}
|
|
|
|
if (priv->server_addr_v6) {
|
|
GST_DEBUG_OBJECT (stream, "udp IPv6, configure udpsinks");
|
|
set_unicast_socket_for_udpsink (*udpsink, socket_v6, G_SOCKET_FAMILY_IPV6);
|
|
}
|
|
|
|
if (multicast) {
|
|
gint port;
|
|
if (priv->mcast_addr_v4) {
|
|
GST_DEBUG_OBJECT (stream, "mcast IPv4, configure udpsinks");
|
|
port = get_port_from_socket (socket_v4);
|
|
if (!port)
|
|
goto get_port_failed;
|
|
set_multicast_socket_for_udpsink (*udpsink, socket_v4,
|
|
G_SOCKET_FAMILY_IPV4, priv->multicast_iface,
|
|
priv->mcast_addr_v4->address, port, mcast_ttl);
|
|
}
|
|
|
|
if (priv->mcast_addr_v6) {
|
|
GST_DEBUG_OBJECT (stream, "mcast IPv6, configure udpsinks");
|
|
port = get_port_from_socket (socket_v6);
|
|
if (!port)
|
|
goto get_port_failed;
|
|
set_multicast_socket_for_udpsink (*udpsink, socket_v6,
|
|
G_SOCKET_FAMILY_IPV6, priv->multicast_iface,
|
|
priv->mcast_addr_v6->address, port, mcast_ttl);
|
|
}
|
|
|
|
}
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_udp_protocol:
|
|
{
|
|
GST_ERROR_OBJECT (stream, "failed to create udpsink element");
|
|
return FALSE;
|
|
}
|
|
get_port_failed:
|
|
{
|
|
GST_ERROR_OBJECT (stream, "failed to get udp port");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* must be called with lock */
|
|
static gboolean
|
|
create_and_configure_udpsource (GstElement ** udpsrc, GSocket * socket)
|
|
{
|
|
GstStateChangeReturn ret;
|
|
|
|
g_assert (socket != NULL);
|
|
|
|
*udpsrc = gst_element_factory_make ("udpsrc", NULL);
|
|
if (*udpsrc == NULL)
|
|
goto error;
|
|
|
|
g_object_set (G_OBJECT (*udpsrc), "socket", socket, NULL);
|
|
|
|
/* The udpsrc cannot do the join because its socket is always a local unicast
|
|
* one. The udpsink sharing the same socket will do it for us. */
|
|
g_object_set (G_OBJECT (*udpsrc), "auto-multicast", FALSE, NULL);
|
|
|
|
g_object_set (G_OBJECT (*udpsrc), "loop", FALSE, NULL);
|
|
|
|
g_object_set (G_OBJECT (*udpsrc), "close-socket", FALSE, NULL);
|
|
|
|
ret = gst_element_set_state (*udpsrc, GST_STATE_READY);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
goto error;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
if (*udpsrc) {
|
|
gst_element_set_state (*udpsrc, GST_STATE_NULL);
|
|
g_clear_object (udpsrc);
|
|
}
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
|
|
GSocket * socket_out[2], GstRTSPAddress ** server_addr_out,
|
|
gboolean multicast, GstRTSPTransport * ct, gboolean use_transport_settings)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GSocket *rtp_socket = NULL;
|
|
GSocket *rtcp_socket = NULL;
|
|
gint tmp_rtp, tmp_rtcp;
|
|
guint count;
|
|
GList *rejected_addresses = NULL;
|
|
GstRTSPAddress *addr = NULL;
|
|
GInetAddress *inetaddr = NULL;
|
|
GSocketAddress *rtp_sockaddr = NULL;
|
|
GSocketAddress *rtcp_sockaddr = NULL;
|
|
GstRTSPAddressPool *pool;
|
|
gboolean transport_settings_defined = FALSE;
|
|
|
|
pool = priv->pool;
|
|
count = 0;
|
|
|
|
/* Start with random port */
|
|
tmp_rtp = 0;
|
|
tmp_rtcp = 0;
|
|
|
|
if (use_transport_settings) {
|
|
if (!multicast)
|
|
goto no_mcast;
|
|
|
|
if (ct == NULL)
|
|
goto no_transport;
|
|
|
|
/* multicast and transport specific case */
|
|
if (ct->destination != NULL) {
|
|
tmp_rtp = ct->port.min;
|
|
tmp_rtcp = ct->port.max;
|
|
|
|
/* check if the provided address is a multicast address */
|
|
inetaddr = g_inet_address_new_from_string (ct->destination);
|
|
if (inetaddr == NULL)
|
|
goto destination_error;
|
|
if (!g_inet_address_get_is_multicast (inetaddr))
|
|
goto destination_no_mcast;
|
|
|
|
|
|
if (!priv->bind_mcast_address) {
|
|
g_clear_object (&inetaddr);
|
|
inetaddr = g_inet_address_new_any (family);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (stream, "use transport settings");
|
|
transport_settings_defined = TRUE;
|
|
}
|
|
}
|
|
|
|
if (priv->enable_rtcp) {
|
|
rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
|
|
G_SOCKET_PROTOCOL_UDP, NULL);
|
|
if (!rtcp_socket)
|
|
goto no_udp_protocol;
|
|
g_socket_set_multicast_loopback (rtcp_socket, FALSE);
|
|
}
|
|
|
|
/* try to allocate UDP ports, the RTP port should be an even
|
|
* number and the RTCP port (if enabled) should be the next (uneven) port */
|
|
again:
|
|
|
|
if (rtp_socket == NULL) {
|
|
rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
|
|
G_SOCKET_PROTOCOL_UDP, NULL);
|
|
if (!rtp_socket)
|
|
goto no_udp_protocol;
|
|
g_socket_set_multicast_loopback (rtp_socket, FALSE);
|
|
}
|
|
|
|
if (!transport_settings_defined) {
|
|
if ((pool && gst_rtsp_address_pool_has_unicast_addresses (pool))
|
|
|| multicast) {
|
|
GstRTSPAddressFlags flags;
|
|
|
|
if (addr) {
|
|
g_assert (*server_addr_out == NULL);
|
|
rejected_addresses = g_list_prepend (rejected_addresses, addr);
|
|
}
|
|
|
|
if (!pool)
|
|
goto no_pool;
|
|
|
|
flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT;
|
|
if (multicast)
|
|
flags |= GST_RTSP_ADDRESS_FLAG_MULTICAST;
|
|
else
|
|
flags |= GST_RTSP_ADDRESS_FLAG_UNICAST;
|
|
|
|
if (family == G_SOCKET_FAMILY_IPV6)
|
|
flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
|
|
else
|
|
flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
|
|
|
|
if (*server_addr_out)
|
|
addr = *server_addr_out;
|
|
else
|
|
addr = gst_rtsp_address_pool_acquire_address (pool, flags,
|
|
priv->enable_rtcp ? 2 : 1);
|
|
|
|
if (addr == NULL)
|
|
goto no_address;
|
|
|
|
tmp_rtp = addr->port;
|
|
|
|
g_clear_object (&inetaddr);
|
|
/* FIXME: Does it really work with the IP_MULTICAST_ALL socket option and
|
|
* socket control message set in udpsrc? */
|
|
if (priv->bind_mcast_address || !multicast)
|
|
inetaddr = g_inet_address_new_from_string (addr->address);
|
|
else
|
|
inetaddr = g_inet_address_new_any (family);
|
|
} else {
|
|
if (tmp_rtp != 0) {
|
|
tmp_rtp += 2;
|
|
if (++count > 20)
|
|
goto no_ports;
|
|
}
|
|
|
|
if (inetaddr == NULL)
|
|
inetaddr = g_inet_address_new_any (family);
|
|
}
|
|
}
|
|
|
|
rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
|
|
if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
|
|
GST_DEBUG_OBJECT (stream, "rtp bind() failed, will try again");
|
|
g_object_unref (rtp_sockaddr);
|
|
if (transport_settings_defined) {
|
|
goto transport_settings_error;
|
|
} else if (*server_addr_out && ((pool
|
|
&& gst_rtsp_address_pool_has_unicast_addresses (pool))
|
|
|| multicast)) {
|
|
goto no_address;
|
|
} else {
|
|
goto again;
|
|
}
|
|
}
|
|
g_object_unref (rtp_sockaddr);
|
|
|
|
rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
|
|
if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
|
|
g_clear_object (&rtp_sockaddr);
|
|
goto socket_error;
|
|
}
|
|
|
|
if (!transport_settings_defined) {
|
|
tmp_rtp =
|
|
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
|
|
|
|
/* check if port is even. RFC 3550 encorages the use of an even/odd port
|
|
* pair, however it's not a strict requirement so this check is not done
|
|
* for the client selected ports. */
|
|
if ((tmp_rtp & 1) != 0) {
|
|
/* port not even, close and allocate another */
|
|
tmp_rtp++;
|
|
g_object_unref (rtp_sockaddr);
|
|
g_clear_object (&rtp_socket);
|
|
goto again;
|
|
}
|
|
}
|
|
g_object_unref (rtp_sockaddr);
|
|
|
|
/* set port */
|
|
if (priv->enable_rtcp) {
|
|
tmp_rtcp = tmp_rtp + 1;
|
|
|
|
rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
|
|
if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
|
|
GST_DEBUG_OBJECT (stream, "rctp bind() failed, will try again");
|
|
g_object_unref (rtcp_sockaddr);
|
|
g_clear_object (&rtp_socket);
|
|
if (transport_settings_defined)
|
|
goto transport_settings_error;
|
|
goto again;
|
|
}
|
|
g_object_unref (rtcp_sockaddr);
|
|
}
|
|
|
|
if (!addr) {
|
|
addr = g_slice_new0 (GstRTSPAddress);
|
|
addr->port = tmp_rtp;
|
|
addr->n_ports = 2;
|
|
if (transport_settings_defined)
|
|
addr->address = g_strdup (ct->destination);
|
|
else
|
|
addr->address = g_inet_address_to_string (inetaddr);
|
|
addr->ttl = ct->ttl;
|
|
}
|
|
|
|
g_clear_object (&inetaddr);
|
|
|
|
if (multicast && (ct->ttl > 0) && (ct->ttl <= priv->max_mcast_ttl)) {
|
|
GST_DEBUG ("setting mcast ttl to %d", ct->ttl);
|
|
g_socket_set_multicast_ttl (rtp_socket, ct->ttl);
|
|
if (rtcp_socket)
|
|
g_socket_set_multicast_ttl (rtcp_socket, ct->ttl);
|
|
}
|
|
|
|
socket_out[0] = rtp_socket;
|
|
socket_out[1] = rtcp_socket;
|
|
*server_addr_out = addr;
|
|
|
|
if (priv->enable_rtcp) {
|
|
GST_DEBUG_OBJECT (stream, "allocated address: %s and ports: %d, %d",
|
|
addr->address, tmp_rtp, tmp_rtcp);
|
|
} else {
|
|
GST_DEBUG_OBJECT (stream, "allocated address: %s and port: %d",
|
|
addr->address, tmp_rtp);
|
|
}
|
|
|
|
g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_mcast:
|
|
{
|
|
GST_ERROR_OBJECT (stream, "failed to allocate UDP ports: wrong transport");
|
|
goto cleanup;
|
|
}
|
|
no_transport:
|
|
{
|
|
GST_ERROR_OBJECT (stream, "failed to allocate UDP ports: no transport");
|
|
goto cleanup;
|
|
}
|
|
destination_error:
|
|
{
|
|
GST_ERROR_OBJECT (stream,
|
|
"failed to allocate UDP ports: destination error");
|
|
goto cleanup;
|
|
}
|
|
destination_no_mcast:
|
|
{
|
|
GST_ERROR_OBJECT (stream,
|
|
"failed to allocate UDP ports: destination not multicast address");
|
|
goto cleanup;
|
|
}
|
|
no_udp_protocol:
|
|
{
|
|
GST_WARNING_OBJECT (stream, "failed to allocate UDP ports: protocol error");
|
|
goto cleanup;
|
|
}
|
|
no_pool:
|
|
{
|
|
GST_WARNING_OBJECT (stream,
|
|
"failed to allocate UDP ports: no address pool specified");
|
|
goto cleanup;
|
|
}
|
|
no_address:
|
|
{
|
|
GST_WARNING_OBJECT (stream, "failed to acquire address from pool");
|
|
goto cleanup;
|
|
}
|
|
no_ports:
|
|
{
|
|
GST_WARNING_OBJECT (stream, "failed to allocate UDP ports: no ports");
|
|
goto cleanup;
|
|
}
|
|
transport_settings_error:
|
|
{
|
|
GST_ERROR_OBJECT (stream,
|
|
"failed to allocate UDP ports with requested transport settings");
|
|
goto cleanup;
|
|
}
|
|
socket_error:
|
|
{
|
|
GST_WARNING_OBJECT (stream, "failed to allocate UDP ports: socket error");
|
|
goto cleanup;
|
|
}
|
|
cleanup:
|
|
{
|
|
if (inetaddr)
|
|
g_object_unref (inetaddr);
|
|
g_list_free_full (rejected_addresses,
|
|
(GDestroyNotify) gst_rtsp_address_free);
|
|
if (addr)
|
|
gst_rtsp_address_free (addr);
|
|
if (rtp_socket)
|
|
g_object_unref (rtp_socket);
|
|
if (rtcp_socket)
|
|
g_object_unref (rtcp_socket);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* must be called with lock */
|
|
static gboolean
|
|
add_mcast_client_addr (GstRTSPStream * stream, const gchar * destination,
|
|
guint rtp_port, guint rtcp_port)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GList *walk;
|
|
UdpClientAddrInfo *client;
|
|
GInetAddress *inet;
|
|
|
|
priv = stream->priv;
|
|
|
|
if (destination == NULL)
|
|
return FALSE;
|
|
|
|
inet = g_inet_address_new_from_string (destination);
|
|
if (inet == NULL)
|
|
goto invalid_address;
|
|
|
|
if (!g_inet_address_get_is_multicast (inet)) {
|
|
g_object_unref (inet);
|
|
goto invalid_address;
|
|
}
|
|
g_object_unref (inet);
|
|
|
|
for (walk = priv->mcast_clients; walk; walk = g_list_next (walk)) {
|
|
UdpClientAddrInfo *cli = walk->data;
|
|
|
|
if ((g_strcmp0 (cli->address, destination) == 0) &&
|
|
(cli->rtp_port == rtp_port)) {
|
|
GST_DEBUG ("requested destination already exists: %s:%u-%u",
|
|
destination, rtp_port, rtcp_port);
|
|
cli->add_count++;
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
client = g_new0 (UdpClientAddrInfo, 1);
|
|
client->address = g_strdup (destination);
|
|
client->rtp_port = rtp_port;
|
|
client->add_count = 1;
|
|
priv->mcast_clients = g_list_prepend (priv->mcast_clients, client);
|
|
|
|
GST_DEBUG ("added mcast client %s:%u-%u", destination, rtp_port, rtcp_port);
|
|
|
|
return TRUE;
|
|
|
|
invalid_address:
|
|
{
|
|
GST_WARNING_OBJECT (stream, "Multicast address is invalid: %s",
|
|
destination);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* must be called with lock */
|
|
static gboolean
|
|
remove_mcast_client_addr (GstRTSPStream * stream, const gchar * destination,
|
|
guint rtp_port, guint rtcp_port)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GList *walk;
|
|
|
|
priv = stream->priv;
|
|
|
|
if (destination == NULL)
|
|
goto no_destination;
|
|
|
|
for (walk = priv->mcast_clients; walk; walk = g_list_next (walk)) {
|
|
UdpClientAddrInfo *cli = walk->data;
|
|
|
|
if ((g_strcmp0 (cli->address, destination) == 0) &&
|
|
(cli->rtp_port == rtp_port)) {
|
|
cli->add_count--;
|
|
|
|
if (!cli->add_count) {
|
|
priv->mcast_clients = g_list_remove (priv->mcast_clients, cli);
|
|
free_mcast_client (cli);
|
|
}
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
GST_WARNING_OBJECT (stream, "Address not found");
|
|
return FALSE;
|
|
|
|
no_destination:
|
|
{
|
|
GST_WARNING_OBJECT (stream, "No destination has been provided");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_rtsp_stream_allocate_udp_sockets:
|
|
* @stream: a #GstRTSPStream
|
|
* @family: protocol family
|
|
* @transport: transport method
|
|
* @use_client_settings: Whether to use client settings or not
|
|
*
|
|
* Allocates RTP and RTCP ports.
|
|
*
|
|
* Returns: %TRUE if the RTP and RTCP sockets have been succeccully allocated.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream,
|
|
GSocketFamily family, GstRTSPTransport * ct,
|
|
gboolean use_transport_settings)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gboolean ret = FALSE;
|
|
GstRTSPLowerTrans transport;
|
|
gboolean allocated = FALSE;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
g_return_val_if_fail (ct != NULL, FALSE);
|
|
priv = stream->priv;
|
|
|
|
transport = ct->lower_transport;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
|
|
if (family == G_SOCKET_FAMILY_IPV4 && priv->mcast_socket_v4[0])
|
|
allocated = TRUE;
|
|
else if (family == G_SOCKET_FAMILY_IPV6 && priv->mcast_socket_v6[0])
|
|
allocated = TRUE;
|
|
} else if (transport == GST_RTSP_LOWER_TRANS_UDP) {
|
|
if (family == G_SOCKET_FAMILY_IPV4 && priv->socket_v4[0])
|
|
allocated = TRUE;
|
|
else if (family == G_SOCKET_FAMILY_IPV6 && priv->socket_v6[0])
|
|
allocated = TRUE;
|
|
}
|
|
|
|
if (allocated) {
|
|
GST_DEBUG_OBJECT (stream, "Allocated already");
|
|
g_mutex_unlock (&priv->lock);
|
|
return TRUE;
|
|
}
|
|
|
|
if (family == G_SOCKET_FAMILY_IPV4) {
|
|
/* IPv4 */
|
|
if (transport == GST_RTSP_LOWER_TRANS_UDP) {
|
|
/* UDP unicast */
|
|
GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_UDP, ipv4");
|
|
ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
|
|
priv->socket_v4, &priv->server_addr_v4, FALSE, ct, FALSE);
|
|
} else {
|
|
/* multicast */
|
|
GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_MCAST_UDP, ipv4");
|
|
ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
|
|
priv->mcast_socket_v4, &priv->mcast_addr_v4, TRUE, ct,
|
|
use_transport_settings);
|
|
}
|
|
} else {
|
|
/* IPv6 */
|
|
if (transport == GST_RTSP_LOWER_TRANS_UDP) {
|
|
/* unicast */
|
|
GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_UDP, ipv6");
|
|
ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
|
|
priv->socket_v6, &priv->server_addr_v6, FALSE, ct, FALSE);
|
|
|
|
} else {
|
|
/* multicast */
|
|
GST_DEBUG_OBJECT (stream, "GST_RTSP_LOWER_TRANS_MCAST_UDP, ipv6");
|
|
ret = alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
|
|
priv->mcast_socket_v6, &priv->mcast_addr_v6, TRUE, ct,
|
|
use_transport_settings);
|
|
}
|
|
}
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_client_side:
|
|
* @stream: a #GstRTSPStream
|
|
* @client_side: TRUE if this #GstRTSPStream is running on the 'client' side of
|
|
* an RTSP connection.
|
|
*
|
|
* Sets the #GstRTSPStream as a 'client side' stream - used for sending
|
|
* streams to an RTSP server via RECORD. This has the practical effect
|
|
* of changing which UDP port numbers are used when setting up the local
|
|
* side of the stream sending to be either the 'server' or 'client' pair
|
|
* of a configured UDP transport.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_client_side (GstRTSPStream * stream, gboolean client_side)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
priv->client_side = client_side;
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_is_client_side:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* See gst_rtsp_stream_set_client_side()
|
|
*
|
|
* Returns: TRUE if this #GstRTSPStream is client-side.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_is_client_side (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gboolean ret;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
ret = priv->client_side;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_server_port:
|
|
* @stream: a #GstRTSPStream
|
|
* @server_port: (out): result server port
|
|
* @family: the port family to get
|
|
*
|
|
* Fill @server_port with the port pair used by the server. This function can
|
|
* only be called when @stream has been joined.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
|
|
GstRTSPRange * server_port, GSocketFamily family)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
priv = stream->priv;
|
|
g_return_if_fail (priv->joined_bin != NULL);
|
|
|
|
if (server_port) {
|
|
server_port->min = 0;
|
|
server_port->max = 0;
|
|
}
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (family == G_SOCKET_FAMILY_IPV4) {
|
|
if (server_port && priv->server_addr_v4) {
|
|
server_port->min = priv->server_addr_v4->port;
|
|
if (priv->enable_rtcp) {
|
|
server_port->max =
|
|
priv->server_addr_v4->port + priv->server_addr_v4->n_ports - 1;
|
|
}
|
|
}
|
|
} else {
|
|
if (server_port && priv->server_addr_v6) {
|
|
server_port->min = priv->server_addr_v6->port;
|
|
if (priv->enable_rtcp) {
|
|
server_port->max =
|
|
priv->server_addr_v6->port + priv->server_addr_v6->n_ports - 1;
|
|
}
|
|
}
|
|
}
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_rtpsession:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the RTP session of this stream.
|
|
*
|
|
* Returns: (transfer full) (nullable): The RTP session of this stream. Unref after usage.
|
|
*/
|
|
GObject *
|
|
gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GObject *session;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((session = priv->session))
|
|
g_object_ref (session);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return session;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_srtp_encoder:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the SRTP encoder for this stream.
|
|
*
|
|
* Returns: (transfer full) (nullable): The SRTP encoder for this stream. Unref after usage.
|
|
*/
|
|
GstElement *
|
|
gst_rtsp_stream_get_srtp_encoder (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstElement *encoder;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((encoder = priv->srtpenc))
|
|
g_object_ref (encoder);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return encoder;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_ssrc:
|
|
* @stream: a #GstRTSPStream
|
|
* @ssrc: (out): result ssrc
|
|
*
|
|
* Get the SSRC used by the RTP session of this stream. This function can only
|
|
* be called when @stream has been joined.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
priv = stream->priv;
|
|
g_return_if_fail (priv->joined_bin != NULL);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (ssrc && priv->session)
|
|
g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_retransmission_time:
|
|
* @stream: a #GstRTSPStream
|
|
* @time: a #GstClockTime
|
|
*
|
|
* Set the amount of time to store retransmission packets.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
|
|
GstClockTime time)
|
|
{
|
|
GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
stream->priv->rtx_time = time;
|
|
if (stream->priv->rtxsend)
|
|
g_object_set (stream->priv->rtxsend, "max-size-time",
|
|
GST_TIME_AS_MSECONDS (time), NULL);
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_retransmission_time:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the amount of time to store retransmission data.
|
|
*
|
|
* Returns: the amount of time to store retransmission data.
|
|
*/
|
|
GstClockTime
|
|
gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
|
|
{
|
|
GstClockTime ret;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
ret = stream->priv->rtx_time;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_retransmission_pt:
|
|
* @stream: a #GstRTSPStream
|
|
* @rtx_pt: a #guint
|
|
*
|
|
* Set the payload type (pt) for retransmission of this stream.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
|
|
{
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
stream->priv->rtx_pt = rtx_pt;
|
|
if (stream->priv->rtxsend) {
|
|
guint pt = gst_rtsp_stream_get_pt (stream);
|
|
gchar *pt_s = g_strdup_printf ("%d", pt);
|
|
GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
|
|
pt_s, G_TYPE_UINT, rtx_pt, NULL);
|
|
g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
|
|
g_free (pt_s);
|
|
gst_structure_free (rtx_pt_map);
|
|
}
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_retransmission_pt:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the payload-type used for retransmission of this stream
|
|
*
|
|
* Returns: The retransmission PT.
|
|
*/
|
|
guint
|
|
gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
|
|
{
|
|
guint rtx_pt;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
rtx_pt = stream->priv->rtx_pt;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
|
|
return rtx_pt;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_buffer_size:
|
|
* @stream: a #GstRTSPStream
|
|
* @size: the buffer size
|
|
*
|
|
* Set the size of the UDP transmission buffer (in bytes)
|
|
* Needs to be set before the stream is joined to a bin.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
|
|
{
|
|
g_mutex_lock (&stream->priv->lock);
|
|
stream->priv->buffer_size = size;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_buffer_size:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the size of the UDP transmission buffer (in bytes)
|
|
*
|
|
* Returns: the size of the UDP TX buffer
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
guint
|
|
gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
|
|
{
|
|
guint buffer_size;
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
buffer_size = stream->priv->buffer_size;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
|
|
return buffer_size;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_max_mcast_ttl:
|
|
* @stream: a #GstRTSPStream
|
|
* @ttl: the new multicast ttl value
|
|
*
|
|
* Set the maximum time-to-live value of outgoing multicast packets.
|
|
*
|
|
* Returns: %TRUE if the requested ttl has been set successfully.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_set_max_mcast_ttl (GstRTSPStream * stream, guint ttl)
|
|
{
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
if (ttl == 0 || ttl > DEFAULT_MAX_MCAST_TTL) {
|
|
GST_WARNING_OBJECT (stream, "The reqested mcast TTL value is not valid.");
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
return FALSE;
|
|
}
|
|
stream->priv->max_mcast_ttl = ttl;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_max_mcast_ttl:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the the maximum time-to-live value of outgoing multicast packets.
|
|
*
|
|
* Returns: the maximum time-to-live value of outgoing multicast packets.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
guint
|
|
gst_rtsp_stream_get_max_mcast_ttl (GstRTSPStream * stream)
|
|
{
|
|
guint ttl;
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
ttl = stream->priv->max_mcast_ttl;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
|
|
return ttl;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_verify_mcast_ttl:
|
|
* @stream: a #GstRTSPStream
|
|
* @ttl: a requested multicast ttl
|
|
*
|
|
* Check if the requested multicast ttl value is allowed.
|
|
*
|
|
* Returns: TRUE if the requested ttl value is allowed.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_verify_mcast_ttl (GstRTSPStream * stream, guint ttl)
|
|
{
|
|
gboolean res = FALSE;
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
if ((ttl > 0) && (ttl <= stream->priv->max_mcast_ttl))
|
|
res = TRUE;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_bind_mcast_address:
|
|
* @stream: a #GstRTSPStream,
|
|
* @bind_mcast_addr: the new value
|
|
*
|
|
* Decide whether the multicast socket should be bound to a multicast address or
|
|
* INADDR_ANY.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_bind_mcast_address (GstRTSPStream * stream,
|
|
gboolean bind_mcast_addr)
|
|
{
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
stream->priv->bind_mcast_address = bind_mcast_addr;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_is_bind_mcast_address:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Check if multicast sockets are configured to be bound to multicast addresses.
|
|
*
|
|
* Returns: %TRUE if multicast sockets are configured to be bound to multicast addresses.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_is_bind_mcast_address (GstRTSPStream * stream)
|
|
{
|
|
gboolean result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
result = stream->priv->bind_mcast_address;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
|
|
return result;
|
|
}
|
|
|
|
void
|
|
gst_rtsp_stream_set_enable_rtcp (GstRTSPStream * stream, gboolean enable)
|
|
{
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
stream->priv->enable_rtcp = enable;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
}
|
|
|
|
/* executed from streaming thread */
|
|
static void
|
|
caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GstCaps *newcaps, *oldcaps;
|
|
|
|
newcaps = gst_pad_get_current_caps (pad);
|
|
|
|
GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
|
|
newcaps);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
oldcaps = priv->caps;
|
|
priv->caps = newcaps;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (oldcaps)
|
|
gst_caps_unref (oldcaps);
|
|
}
|
|
|
|
static void
|
|
dump_structure (const GstStructure * s)
|
|
{
|
|
gchar *sstr;
|
|
|
|
sstr = gst_structure_to_string (s);
|
|
GST_INFO ("structure: %s", sstr);
|
|
g_free (sstr);
|
|
}
|
|
|
|
static GstRTSPStreamTransport *
|
|
find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GList *walk;
|
|
GstRTSPStreamTransport *result = NULL;
|
|
const gchar *tmp;
|
|
gchar *dest;
|
|
guint port;
|
|
|
|
if (rtcp_from == NULL)
|
|
return NULL;
|
|
|
|
tmp = g_strrstr (rtcp_from, ":");
|
|
if (tmp == NULL)
|
|
return NULL;
|
|
|
|
port = atoi (tmp + 1);
|
|
dest = g_strndup (rtcp_from, tmp - rtcp_from);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
GST_INFO ("finding %s:%d in %d transports", dest, port,
|
|
g_list_length (priv->transports));
|
|
|
|
for (walk = priv->transports; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamTransport *trans = walk->data;
|
|
const GstRTSPTransport *tr;
|
|
gint min, max;
|
|
|
|
tr = gst_rtsp_stream_transport_get_transport (trans);
|
|
|
|
if (priv->client_side) {
|
|
/* In client side mode the 'destination' is the RTSP server, so send
|
|
* to those ports */
|
|
min = tr->server_port.min;
|
|
max = tr->server_port.max;
|
|
} else {
|
|
min = tr->client_port.min;
|
|
max = tr->client_port.max;
|
|
}
|
|
|
|
if ((g_ascii_strcasecmp (tr->destination, dest) == 0) &&
|
|
(min == port || max == port)) {
|
|
result = trans;
|
|
break;
|
|
}
|
|
}
|
|
if (result)
|
|
g_object_ref (result);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
g_free (dest);
|
|
|
|
return result;
|
|
}
|
|
|
|
static GstRTSPStreamTransport *
|
|
check_transport (GObject * source, GstRTSPStream * stream)
|
|
{
|
|
GstStructure *stats;
|
|
GstRTSPStreamTransport *trans;
|
|
|
|
/* see if we have a stream to match with the origin of the RTCP packet */
|
|
trans = g_object_get_qdata (source, ssrc_stream_map_key);
|
|
if (trans == NULL) {
|
|
g_object_get (source, "stats", &stats, NULL);
|
|
if (stats) {
|
|
const gchar *rtcp_from;
|
|
|
|
dump_structure (stats);
|
|
|
|
rtcp_from = gst_structure_get_string (stats, "rtcp-from");
|
|
if ((trans = find_transport (stream, rtcp_from))) {
|
|
GST_INFO ("%p: found transport %p for source %p", stream, trans,
|
|
source);
|
|
g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
|
|
g_object_unref);
|
|
}
|
|
gst_structure_free (stats);
|
|
}
|
|
}
|
|
return trans;
|
|
}
|
|
|
|
|
|
static void
|
|
on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamTransport *trans;
|
|
|
|
GST_INFO ("%p: new source %p", stream, source);
|
|
|
|
trans = check_transport (source, stream);
|
|
|
|
if (trans)
|
|
GST_INFO ("%p: source %p for transport %p", stream, source, trans);
|
|
}
|
|
|
|
static void
|
|
on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
|
|
{
|
|
GST_INFO ("%p: new SDES %p", stream, source);
|
|
}
|
|
|
|
static void
|
|
on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamTransport *trans;
|
|
|
|
trans = check_transport (source, stream);
|
|
|
|
if (trans) {
|
|
GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
|
|
gst_rtsp_stream_transport_keep_alive (trans);
|
|
}
|
|
#ifdef DUMP_STATS
|
|
{
|
|
GstStructure *stats;
|
|
g_object_get (source, "stats", &stats, NULL);
|
|
if (stats) {
|
|
dump_structure (stats);
|
|
gst_structure_free (stats);
|
|
}
|
|
}
|
|
#endif
|
|
}
|
|
|
|
static void
|
|
on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
|
|
{
|
|
GST_INFO ("%p: source %p bye", stream, source);
|
|
}
|
|
|
|
static void
|
|
on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamTransport *trans;
|
|
|
|
GST_INFO ("%p: source %p bye timeout", stream, source);
|
|
|
|
if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
|
|
gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
|
|
g_object_set_qdata (source, ssrc_stream_map_key, NULL);
|
|
}
|
|
}
|
|
|
|
static void
|
|
on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamTransport *trans;
|
|
|
|
GST_INFO ("%p: source %p timeout", stream, source);
|
|
|
|
if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
|
|
gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
|
|
g_object_set_qdata (source, ssrc_stream_map_key, NULL);
|
|
}
|
|
}
|
|
|
|
static void
|
|
on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
|
|
{
|
|
GST_INFO ("%p: new sender source %p", stream, source);
|
|
#ifndef DUMP_STATS
|
|
{
|
|
GstStructure *stats;
|
|
g_object_get (source, "stats", &stats, NULL);
|
|
if (stats) {
|
|
dump_structure (stats);
|
|
gst_structure_free (stats);
|
|
}
|
|
}
|
|
#endif
|
|
}
|
|
|
|
static void
|
|
on_sender_ssrc_active (GObject * session, GObject * source,
|
|
GstRTSPStream * stream)
|
|
{
|
|
#ifndef DUMP_STATS
|
|
{
|
|
GstStructure *stats;
|
|
g_object_get (source, "stats", &stats, NULL);
|
|
if (stats) {
|
|
dump_structure (stats);
|
|
gst_structure_free (stats);
|
|
}
|
|
}
|
|
#endif
|
|
}
|
|
|
|
static void
|
|
clear_tr_cache (GstRTSPStreamPrivate * priv)
|
|
{
|
|
if (priv->tr_cache)
|
|
g_ptr_array_unref (priv->tr_cache);
|
|
priv->tr_cache = NULL;
|
|
}
|
|
|
|
/* With lock taken */
|
|
static gboolean
|
|
any_transport_ready (GstRTSPStream * stream, gboolean is_rtp)
|
|
{
|
|
gboolean ret = TRUE;
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GPtrArray *transports;
|
|
gint index;
|
|
|
|
transports = priv->tr_cache;
|
|
|
|
if (!transports)
|
|
goto done;
|
|
|
|
for (index = 0; index < transports->len; index++) {
|
|
GstRTSPStreamTransport *tr = g_ptr_array_index (transports, index);
|
|
if (!gst_rtsp_stream_transport_check_back_pressure (tr, is_rtp)) {
|
|
ret = TRUE;
|
|
break;
|
|
} else {
|
|
ret = FALSE;
|
|
}
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
/* Must be called *without* priv->lock */
|
|
static gboolean
|
|
push_data (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
|
|
GstBuffer * buffer, GstBufferList * buffer_list, gboolean is_rtp)
|
|
{
|
|
gboolean send_ret = TRUE;
|
|
|
|
if (is_rtp) {
|
|
if (buffer)
|
|
send_ret = gst_rtsp_stream_transport_send_rtp (trans, buffer);
|
|
if (buffer_list)
|
|
send_ret = gst_rtsp_stream_transport_send_rtp_list (trans, buffer_list);
|
|
} else {
|
|
if (buffer)
|
|
send_ret = gst_rtsp_stream_transport_send_rtcp (trans, buffer);
|
|
if (buffer_list)
|
|
send_ret = gst_rtsp_stream_transport_send_rtcp_list (trans, buffer_list);
|
|
}
|
|
|
|
return send_ret;
|
|
}
|
|
|
|
/* With priv->lock */
|
|
static void
|
|
ensure_cached_transports (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GList *walk;
|
|
|
|
if (priv->tr_cache_cookie != priv->transports_cookie) {
|
|
clear_tr_cache (priv);
|
|
priv->tr_cache =
|
|
g_ptr_array_new_full (priv->n_tcp_transports, g_object_unref);
|
|
|
|
for (walk = priv->transports; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
|
|
const GstRTSPTransport *t = gst_rtsp_stream_transport_get_transport (tr);
|
|
|
|
if (t->lower_transport != GST_RTSP_LOWER_TRANS_TCP)
|
|
continue;
|
|
|
|
g_ptr_array_add (priv->tr_cache, g_object_ref (tr));
|
|
}
|
|
priv->tr_cache_cookie = priv->transports_cookie;
|
|
}
|
|
}
|
|
|
|
/* Must be called *without* priv->lock */
|
|
static void
|
|
check_transport_backlog (GstRTSPStream * stream, GstRTSPStreamTransport * trans)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
gboolean send_ret = TRUE;
|
|
|
|
gst_rtsp_stream_transport_lock_backlog (trans);
|
|
|
|
if (!gst_rtsp_stream_transport_backlog_is_empty (trans)) {
|
|
GstBuffer *buffer;
|
|
GstBufferList *buffer_list;
|
|
gboolean is_rtp;
|
|
gboolean popped;
|
|
|
|
is_rtp = gst_rtsp_stream_transport_backlog_peek_is_rtp (trans);
|
|
|
|
if (!gst_rtsp_stream_transport_check_back_pressure (trans, is_rtp)) {
|
|
popped =
|
|
gst_rtsp_stream_transport_backlog_pop (trans, &buffer, &buffer_list,
|
|
&is_rtp);
|
|
|
|
g_assert (popped == TRUE);
|
|
|
|
send_ret = push_data (stream, trans, buffer, buffer_list, is_rtp);
|
|
|
|
gst_clear_buffer (&buffer);
|
|
gst_clear_buffer_list (&buffer_list);
|
|
}
|
|
}
|
|
|
|
gst_rtsp_stream_transport_unlock_backlog (trans);
|
|
|
|
if (!send_ret) {
|
|
/* remove transport on send error */
|
|
g_mutex_lock (&priv->lock);
|
|
update_transport (stream, trans, FALSE);
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|
|
}
|
|
|
|
/* Must be called with priv->lock */
|
|
static void
|
|
send_tcp_message (GstRTSPStream * stream, gint idx)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GstAppSink *sink;
|
|
GstSample *sample;
|
|
GstBuffer *buffer;
|
|
GstBufferList *buffer_list;
|
|
gboolean is_rtp;
|
|
GPtrArray *transports;
|
|
|
|
if (!priv->have_buffer[idx])
|
|
return;
|
|
|
|
ensure_cached_transports (stream);
|
|
|
|
is_rtp = (idx == 0);
|
|
|
|
if (!any_transport_ready (stream, is_rtp))
|
|
return;
|
|
|
|
priv->have_buffer[idx] = FALSE;
|
|
|
|
if (priv->appsink[idx] == NULL) {
|
|
/* session expired */
|
|
return;
|
|
}
|
|
|
|
sink = GST_APP_SINK (priv->appsink[idx]);
|
|
sample = gst_app_sink_pull_sample (sink);
|
|
if (!sample) {
|
|
return;
|
|
}
|
|
|
|
buffer = gst_sample_get_buffer (sample);
|
|
buffer_list = gst_sample_get_buffer_list (sample);
|
|
|
|
/* We will get one message-sent notification per buffer or
|
|
* complete buffer-list. We handle each buffer-list as a unit */
|
|
|
|
transports = priv->tr_cache;
|
|
if (transports)
|
|
g_ptr_array_ref (transports);
|
|
|
|
if (transports) {
|
|
gint index;
|
|
|
|
for (index = 0; index < transports->len; index++) {
|
|
GstRTSPStreamTransport *tr = g_ptr_array_index (transports, index);
|
|
GstBuffer *buf_ref = NULL;
|
|
GstBufferList *buflist_ref = NULL;
|
|
|
|
gst_rtsp_stream_transport_lock_backlog (tr);
|
|
|
|
if (buffer)
|
|
buf_ref = gst_buffer_ref (buffer);
|
|
if (buffer_list)
|
|
buflist_ref = gst_buffer_list_ref (buffer_list);
|
|
|
|
if (!gst_rtsp_stream_transport_backlog_push (tr,
|
|
buf_ref, buflist_ref, is_rtp)) {
|
|
GST_ERROR_OBJECT (stream,
|
|
"Dropping slow transport %" GST_PTR_FORMAT, tr);
|
|
update_transport (stream, tr, FALSE);
|
|
}
|
|
|
|
gst_rtsp_stream_transport_unlock_backlog (tr);
|
|
}
|
|
}
|
|
gst_sample_unref (sample);
|
|
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (transports) {
|
|
gint index;
|
|
|
|
for (index = 0; index < transports->len; index++) {
|
|
GstRTSPStreamTransport *tr = g_ptr_array_index (transports, index);
|
|
|
|
check_transport_backlog (stream, tr);
|
|
}
|
|
g_ptr_array_unref (transports);
|
|
}
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
}
|
|
|
|
static gpointer
|
|
send_func (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->send_lock);
|
|
|
|
while (priv->continue_sending) {
|
|
int i;
|
|
int idx = -1;
|
|
guint cookie;
|
|
|
|
cookie = priv->send_cookie;
|
|
g_mutex_unlock (&priv->send_lock);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
/* iterate from 1 and down, so we prioritize RTCP over RTP */
|
|
for (i = 1; i >= 0; i--) {
|
|
if (priv->have_buffer[i]) {
|
|
/* send message */
|
|
idx = i;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (idx != -1) {
|
|
send_tcp_message (stream, idx);
|
|
}
|
|
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
g_mutex_lock (&priv->send_lock);
|
|
while (cookie == priv->send_cookie && priv->continue_sending) {
|
|
g_cond_wait (&priv->send_cond, &priv->send_lock);
|
|
}
|
|
}
|
|
|
|
g_mutex_unlock (&priv->send_lock);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
handle_new_sample (GstAppSink * sink, gpointer user_data)
|
|
{
|
|
GstRTSPStream *stream = user_data;
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
int i;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
for (i = 0; i < 2; i++) {
|
|
if (GST_ELEMENT_CAST (sink) == priv->appsink[i]) {
|
|
priv->have_buffer[i] = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (priv->send_thread == NULL) {
|
|
priv->send_thread = g_thread_new (NULL, (GThreadFunc) send_func, user_data);
|
|
}
|
|
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
g_mutex_lock (&priv->send_lock);
|
|
priv->send_cookie++;
|
|
g_cond_signal (&priv->send_cond);
|
|
g_mutex_unlock (&priv->send_lock);
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstAppSinkCallbacks sink_cb = {
|
|
NULL, /* not interested in EOS */
|
|
NULL, /* not interested in preroll samples */
|
|
handle_new_sample,
|
|
};
|
|
|
|
static GstElement *
|
|
get_rtp_encoder (GstRTSPStream * stream, guint session)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
|
|
if (priv->srtpenc == NULL) {
|
|
gchar *name;
|
|
|
|
name = g_strdup_printf ("srtpenc_%u", session);
|
|
priv->srtpenc = gst_element_factory_make ("srtpenc", name);
|
|
g_free (name);
|
|
|
|
g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
|
|
}
|
|
return gst_object_ref (priv->srtpenc);
|
|
}
|
|
|
|
static GstElement *
|
|
request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GstElement *oldenc, *enc;
|
|
GstPad *pad;
|
|
gchar *name;
|
|
|
|
if (priv->idx != session)
|
|
return NULL;
|
|
|
|
GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
|
|
|
|
oldenc = priv->srtpenc;
|
|
enc = get_rtp_encoder (stream, session);
|
|
name = g_strdup_printf ("rtp_sink_%d", session);
|
|
pad = gst_element_request_pad_simple (enc, name);
|
|
g_free (name);
|
|
gst_object_unref (pad);
|
|
|
|
if (oldenc == NULL)
|
|
g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
|
|
enc);
|
|
|
|
return enc;
|
|
}
|
|
|
|
static GstElement *
|
|
request_rtcp_encoder (GstElement * rtpbin, guint session,
|
|
GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GstElement *oldenc, *enc;
|
|
GstPad *pad;
|
|
gchar *name;
|
|
|
|
if (priv->idx != session)
|
|
return NULL;
|
|
|
|
GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
|
|
|
|
oldenc = priv->srtpenc;
|
|
enc = get_rtp_encoder (stream, session);
|
|
name = g_strdup_printf ("rtcp_sink_%d", session);
|
|
pad = gst_element_request_pad_simple (enc, name);
|
|
g_free (name);
|
|
gst_object_unref (pad);
|
|
|
|
if (oldenc == NULL)
|
|
g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
|
|
enc);
|
|
|
|
return enc;
|
|
}
|
|
|
|
static GstCaps *
|
|
request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GstCaps *caps;
|
|
|
|
GST_DEBUG ("request key %08x", ssrc);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
|
|
gst_caps_ref (caps);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static GstElement *
|
|
request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
|
|
GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
|
|
if (priv->idx != session)
|
|
return NULL;
|
|
|
|
if (priv->srtpdec == NULL) {
|
|
gchar *name;
|
|
|
|
name = g_strdup_printf ("srtpdec_%u", session);
|
|
priv->srtpdec = gst_element_factory_make ("srtpdec", name);
|
|
g_free (name);
|
|
|
|
g_signal_connect (priv->srtpdec, "request-key",
|
|
(GCallback) request_key, stream);
|
|
|
|
g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_RTCP_DECODER],
|
|
0, priv->srtpdec);
|
|
|
|
}
|
|
return gst_object_ref (priv->srtpdec);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_request_aux_sender:
|
|
* @stream: a #GstRTSPStream
|
|
* @sessid: the session id
|
|
*
|
|
* Creating a rtxsend bin
|
|
*
|
|
* Returns: (transfer full) (nullable): a #GstElement.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
GstElement *
|
|
gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
|
|
{
|
|
GstElement *bin;
|
|
GstPad *pad;
|
|
GstStructure *pt_map;
|
|
gchar *name;
|
|
guint pt, rtx_pt;
|
|
gchar *pt_s;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
pt = gst_rtsp_stream_get_pt (stream);
|
|
pt_s = g_strdup_printf ("%u", pt);
|
|
rtx_pt = stream->priv->rtx_pt;
|
|
|
|
GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
|
|
|
|
bin = gst_bin_new (NULL);
|
|
stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
|
|
pt_map = gst_structure_new ("application/x-rtp-pt-map",
|
|
pt_s, G_TYPE_UINT, rtx_pt, NULL);
|
|
g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
|
|
"max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
|
|
g_free (pt_s);
|
|
gst_structure_free (pt_map);
|
|
gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
|
|
|
|
pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
|
|
name = g_strdup_printf ("src_%u", sessid);
|
|
gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
|
|
g_free (name);
|
|
gst_object_unref (pad);
|
|
|
|
pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
|
|
name = g_strdup_printf ("sink_%u", sessid);
|
|
gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
|
|
g_free (name);
|
|
gst_object_unref (pad);
|
|
|
|
return bin;
|
|
}
|
|
|
|
static void
|
|
add_rtx_pt (gpointer key, GstCaps * caps, GstStructure * pt_map)
|
|
{
|
|
guint pt = GPOINTER_TO_INT (key);
|
|
const GstStructure *s = gst_caps_get_structure (caps, 0);
|
|
const gchar *apt;
|
|
|
|
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "RTX") &&
|
|
(apt = gst_structure_get_string (s, "apt"))) {
|
|
gst_structure_set (pt_map, apt, G_TYPE_UINT, pt, NULL);
|
|
}
|
|
}
|
|
|
|
/* Call with priv->lock taken */
|
|
static void
|
|
update_rtx_receive_pt_map (GstRTSPStream * stream)
|
|
{
|
|
GstStructure *pt_map;
|
|
|
|
if (!stream->priv->rtxreceive)
|
|
goto done;
|
|
|
|
pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
|
|
g_hash_table_foreach (stream->priv->ptmap, (GHFunc) add_rtx_pt, pt_map);
|
|
g_object_set (stream->priv->rtxreceive, "payload-type-map", pt_map, NULL);
|
|
gst_structure_free (pt_map);
|
|
|
|
done:
|
|
return;
|
|
}
|
|
|
|
static void
|
|
retrieve_ulpfec_pt (gpointer key, GstCaps * caps, GstElement * ulpfec_decoder)
|
|
{
|
|
guint pt = GPOINTER_TO_INT (key);
|
|
const GstStructure *s = gst_caps_get_structure (caps, 0);
|
|
|
|
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
|
|
g_object_set (ulpfec_decoder, "pt", pt, NULL);
|
|
}
|
|
|
|
static void
|
|
update_ulpfec_decoder_pt (GstRTSPStream * stream)
|
|
{
|
|
if (!stream->priv->ulpfec_decoder)
|
|
goto done;
|
|
|
|
g_hash_table_foreach (stream->priv->ptmap, (GHFunc) retrieve_ulpfec_pt,
|
|
stream->priv->ulpfec_decoder);
|
|
|
|
done:
|
|
return;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_request_aux_receiver:
|
|
* @stream: a #GstRTSPStream
|
|
* @sessid: the session id
|
|
*
|
|
* Creating a rtxreceive bin
|
|
*
|
|
* Returns: (transfer full) (nullable): a #GstElement.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
GstElement *
|
|
gst_rtsp_stream_request_aux_receiver (GstRTSPStream * stream, guint sessid)
|
|
{
|
|
GstElement *bin;
|
|
GstPad *pad;
|
|
gchar *name;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
bin = gst_bin_new (NULL);
|
|
stream->priv->rtxreceive = gst_element_factory_make ("rtprtxreceive", NULL);
|
|
update_rtx_receive_pt_map (stream);
|
|
update_ulpfec_decoder_pt (stream);
|
|
gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxreceive));
|
|
|
|
pad = gst_element_get_static_pad (stream->priv->rtxreceive, "src");
|
|
name = g_strdup_printf ("src_%u", sessid);
|
|
gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
|
|
g_free (name);
|
|
gst_object_unref (pad);
|
|
|
|
pad = gst_element_get_static_pad (stream->priv->rtxreceive, "sink");
|
|
name = g_strdup_printf ("sink_%u", sessid);
|
|
gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
|
|
g_free (name);
|
|
gst_object_unref (pad);
|
|
|
|
return bin;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_pt_map:
|
|
* @stream: a #GstRTSPStream
|
|
* @pt: the pt
|
|
* @caps: a #GstCaps
|
|
*
|
|
* Configure a pt map between @pt and @caps.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
|
|
if (!GST_IS_CAPS (caps))
|
|
return;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
|
|
update_rtx_receive_pt_map (stream);
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_publish_clock_mode:
|
|
* @stream: a #GstRTSPStream
|
|
* @mode: the clock publish mode
|
|
*
|
|
* Sets if and how the stream clock should be published according to RFC7273.
|
|
*
|
|
* Since: 1.8
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_publish_clock_mode (GstRTSPStream * stream,
|
|
GstRTSPPublishClockMode mode)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
priv->publish_clock_mode = mode;
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_publish_clock_mode:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Gets if and how the stream clock should be published according to RFC7273.
|
|
*
|
|
* Returns: The GstRTSPPublishClockMode
|
|
*
|
|
* Since: 1.8
|
|
*/
|
|
GstRTSPPublishClockMode
|
|
gst_rtsp_stream_get_publish_clock_mode (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstRTSPPublishClockMode ret;
|
|
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
ret = priv->publish_clock_mode;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstCaps *
|
|
request_pt_map (GstElement * rtpbin, guint session, guint pt,
|
|
GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GstCaps *caps = NULL;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
if (priv->idx == session) {
|
|
caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
|
|
if (caps) {
|
|
GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
|
|
gst_caps_ref (caps);
|
|
} else {
|
|
GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
|
|
}
|
|
}
|
|
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static void
|
|
pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
gchar *name;
|
|
GstPadLinkReturn ret;
|
|
guint sessid;
|
|
|
|
GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
|
|
GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
|
|
|
|
name = gst_pad_get_name (pad);
|
|
if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
|
|
g_free (name);
|
|
return;
|
|
}
|
|
g_free (name);
|
|
|
|
if (priv->idx != sessid)
|
|
return;
|
|
|
|
if (gst_pad_is_linked (priv->sinkpad)) {
|
|
GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
|
|
GST_DEBUG_PAD_NAME (priv->sinkpad));
|
|
return;
|
|
}
|
|
|
|
/* link the RTP pad to the session manager, it should not really fail unless
|
|
* this is not really an RTP pad */
|
|
ret = gst_pad_link (pad, priv->sinkpad);
|
|
if (ret != GST_PAD_LINK_OK)
|
|
goto link_failed;
|
|
priv->recv_rtp_src = gst_object_ref (pad);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
link_failed:
|
|
{
|
|
GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
|
|
GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
|
|
}
|
|
}
|
|
|
|
static void
|
|
on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
|
|
GstRTSPStream * stream)
|
|
{
|
|
/* TODO: What to do here other than this? */
|
|
GST_DEBUG ("Stream %p: Got EOS", stream);
|
|
gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
|
|
}
|
|
|
|
typedef struct _ProbeData ProbeData;
|
|
|
|
struct _ProbeData
|
|
{
|
|
GstRTSPStream *stream;
|
|
/* existing sink, already linked to tee */
|
|
GstElement *sink1;
|
|
/* new sink, about to be linked */
|
|
GstElement *sink2;
|
|
/* new queue element, that will be linked to tee and sink1 */
|
|
GstElement **queue1;
|
|
/* new queue element, that will be linked to tee and sink2 */
|
|
GstElement **queue2;
|
|
GstPad *sink_pad;
|
|
GstPad *tee_pad;
|
|
guint index;
|
|
};
|
|
|
|
static void
|
|
free_cb_data (gpointer user_data)
|
|
{
|
|
ProbeData *data = user_data;
|
|
|
|
gst_object_unref (data->stream);
|
|
gst_object_unref (data->sink1);
|
|
gst_object_unref (data->sink2);
|
|
gst_object_unref (data->sink_pad);
|
|
gst_object_unref (data->tee_pad);
|
|
g_free (data);
|
|
}
|
|
|
|
|
|
static void
|
|
create_and_plug_queue_to_unlinked_stream (GstRTSPStream * stream,
|
|
GstElement * tee, GstElement * sink, GstElement ** queue)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GstPad *tee_pad;
|
|
GstPad *queue_pad;
|
|
GstPad *sink_pad;
|
|
|
|
/* create queue for the new stream */
|
|
*queue = gst_element_factory_make ("queue", NULL);
|
|
g_object_set (*queue, "max-size-buffers", 1, "max-size-bytes", 0,
|
|
"max-size-time", G_GINT64_CONSTANT (0), NULL);
|
|
gst_bin_add (priv->joined_bin, *queue);
|
|
|
|
/* link tee to queue */
|
|
tee_pad = gst_element_request_pad_simple (tee, "src_%u");
|
|
queue_pad = gst_element_get_static_pad (*queue, "sink");
|
|
gst_pad_link (tee_pad, queue_pad);
|
|
gst_object_unref (queue_pad);
|
|
gst_object_unref (tee_pad);
|
|
|
|
/* link queue to sink */
|
|
queue_pad = gst_element_get_static_pad (*queue, "src");
|
|
sink_pad = gst_element_get_static_pad (sink, "sink");
|
|
gst_pad_link (queue_pad, sink_pad);
|
|
gst_object_unref (queue_pad);
|
|
gst_object_unref (sink_pad);
|
|
|
|
gst_element_sync_state_with_parent (sink);
|
|
gst_element_sync_state_with_parent (*queue);
|
|
}
|
|
|
|
static GstPadProbeReturn
|
|
create_and_plug_queue_to_linked_stream_probe_cb (GstPad * inpad,
|
|
GstPadProbeInfo * info, gpointer user_data)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
ProbeData *data = user_data;
|
|
GstRTSPStream *stream;
|
|
GstElement **queue1;
|
|
GstElement **queue2;
|
|
GstPad *sink_pad;
|
|
GstPad *tee_pad;
|
|
GstPad *queue_pad;
|
|
guint index;
|
|
|
|
stream = data->stream;
|
|
priv = stream->priv;
|
|
queue1 = data->queue1;
|
|
queue2 = data->queue2;
|
|
sink_pad = data->sink_pad;
|
|
tee_pad = data->tee_pad;
|
|
index = data->index;
|
|
|
|
/* unlink tee and the existing sink:
|
|
* .-----. .---------.
|
|
* | tee | | sink1 |
|
|
* sink src->sink |
|
|
* '-----' '---------'
|
|
*/
|
|
g_assert (gst_pad_unlink (tee_pad, sink_pad));
|
|
|
|
/* add queue to the already existing stream */
|
|
*queue1 = gst_element_factory_make ("queue", NULL);
|
|
g_object_set (*queue1, "max-size-buffers", 1, "max-size-bytes", 0,
|
|
"max-size-time", G_GINT64_CONSTANT (0), NULL);
|
|
gst_bin_add (priv->joined_bin, *queue1);
|
|
|
|
/* link tee, queue and sink:
|
|
* .-----. .---------. .---------.
|
|
* | tee | | queue1 | | sink1 |
|
|
* sink src->sink src->sink |
|
|
* '-----' '---------' '---------'
|
|
*/
|
|
queue_pad = gst_element_get_static_pad (*queue1, "sink");
|
|
gst_pad_link (tee_pad, queue_pad);
|
|
gst_object_unref (queue_pad);
|
|
queue_pad = gst_element_get_static_pad (*queue1, "src");
|
|
gst_pad_link (queue_pad, sink_pad);
|
|
gst_object_unref (queue_pad);
|
|
|
|
gst_element_sync_state_with_parent (*queue1);
|
|
|
|
/* create queue and link it to tee and the new sink */
|
|
create_and_plug_queue_to_unlinked_stream (stream,
|
|
priv->tee[index], data->sink2, queue2);
|
|
|
|
/* the final stream:
|
|
*
|
|
* .-----. .---------. .---------.
|
|
* | tee | | queue1 | | sink1 |
|
|
* sink src->sink src->sink |
|
|
* | | '---------' '---------'
|
|
* | | .---------. .---------.
|
|
* | | | queue2 | | sink2 |
|
|
* | src->sink src->sink |
|
|
* '-----' '---------' '---------'
|
|
*/
|
|
|
|
return GST_PAD_PROBE_REMOVE;
|
|
}
|
|
|
|
static void
|
|
create_and_plug_queue_to_linked_stream (GstRTSPStream * stream,
|
|
GstElement * sink1, GstElement * sink2, guint index, GstElement ** queue1,
|
|
GstElement ** queue2)
|
|
{
|
|
ProbeData *data;
|
|
|
|
data = g_new0 (ProbeData, 1);
|
|
data->stream = gst_object_ref (stream);
|
|
data->sink1 = gst_object_ref (sink1);
|
|
data->sink2 = gst_object_ref (sink2);
|
|
data->queue1 = queue1;
|
|
data->queue2 = queue2;
|
|
data->index = index;
|
|
|
|
data->sink_pad = gst_element_get_static_pad (sink1, "sink");
|
|
g_assert (data->sink_pad);
|
|
data->tee_pad = gst_pad_get_peer (data->sink_pad);
|
|
g_assert (data->tee_pad);
|
|
|
|
gst_pad_add_probe (data->tee_pad, GST_PAD_PROBE_TYPE_IDLE,
|
|
create_and_plug_queue_to_linked_stream_probe_cb, data, free_cb_data);
|
|
}
|
|
|
|
static void
|
|
plug_udp_sink (GstRTSPStream * stream, GstElement * sink_to_plug,
|
|
GstElement ** queue_to_plug, guint index, gboolean is_mcast)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GstElement *existing_sink;
|
|
|
|
if (is_mcast)
|
|
existing_sink = priv->udpsink[index];
|
|
else
|
|
existing_sink = priv->mcast_udpsink[index];
|
|
|
|
GST_DEBUG_OBJECT (stream, "plug %s sink", is_mcast ? "mcast" : "udp");
|
|
|
|
/* add sink to the bin */
|
|
gst_bin_add (priv->joined_bin, sink_to_plug);
|
|
|
|
if (priv->appsink[index] && existing_sink) {
|
|
|
|
/* queues are already added for the existing stream, add one for
|
|
the newly added udp stream */
|
|
create_and_plug_queue_to_unlinked_stream (stream, priv->tee[index],
|
|
sink_to_plug, queue_to_plug);
|
|
|
|
} else if (priv->appsink[index] || existing_sink) {
|
|
GstElement **queue;
|
|
GstElement *element;
|
|
|
|
/* add queue to the already existing stream plus the newly created udp
|
|
stream */
|
|
if (priv->appsink[index]) {
|
|
element = priv->appsink[index];
|
|
queue = &priv->appqueue[index];
|
|
} else {
|
|
element = existing_sink;
|
|
if (is_mcast)
|
|
queue = &priv->udpqueue[index];
|
|
else
|
|
queue = &priv->mcast_udpqueue[index];
|
|
}
|
|
|
|
create_and_plug_queue_to_linked_stream (stream, element, sink_to_plug,
|
|
index, queue, queue_to_plug);
|
|
|
|
} else {
|
|
GstPad *tee_pad;
|
|
GstPad *sink_pad;
|
|
|
|
GST_DEBUG_OBJECT (stream, "creating first stream");
|
|
|
|
/* no need to add queues */
|
|
tee_pad = gst_element_request_pad_simple (priv->tee[index], "src_%u");
|
|
sink_pad = gst_element_get_static_pad (sink_to_plug, "sink");
|
|
gst_pad_link (tee_pad, sink_pad);
|
|
gst_object_unref (tee_pad);
|
|
gst_object_unref (sink_pad);
|
|
}
|
|
|
|
gst_element_sync_state_with_parent (sink_to_plug);
|
|
}
|
|
|
|
static void
|
|
plug_tcp_sink (GstRTSPStream * stream, guint index)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
|
|
GST_DEBUG_OBJECT (stream, "plug tcp sink");
|
|
|
|
/* add sink to the bin */
|
|
gst_bin_add (priv->joined_bin, priv->appsink[index]);
|
|
|
|
if (priv->mcast_udpsink[index] && priv->udpsink[index]) {
|
|
|
|
/* queues are already added for the existing stream, add one for
|
|
the newly added tcp stream */
|
|
create_and_plug_queue_to_unlinked_stream (stream,
|
|
priv->tee[index], priv->appsink[index], &priv->appqueue[index]);
|
|
|
|
} else if (priv->mcast_udpsink[index] || priv->udpsink[index]) {
|
|
GstElement **queue;
|
|
GstElement *element;
|
|
|
|
/* add queue to the already existing stream plus the newly created tcp
|
|
stream */
|
|
if (priv->mcast_udpsink[index]) {
|
|
element = priv->mcast_udpsink[index];
|
|
queue = &priv->mcast_udpqueue[index];
|
|
} else {
|
|
element = priv->udpsink[index];
|
|
queue = &priv->udpqueue[index];
|
|
}
|
|
|
|
create_and_plug_queue_to_linked_stream (stream, element,
|
|
priv->appsink[index], index, queue, &priv->appqueue[index]);
|
|
|
|
} else {
|
|
GstPad *tee_pad;
|
|
GstPad *sink_pad;
|
|
|
|
/* no need to add queues */
|
|
tee_pad = gst_element_request_pad_simple (priv->tee[index], "src_%u");
|
|
sink_pad = gst_element_get_static_pad (priv->appsink[index], "sink");
|
|
gst_pad_link (tee_pad, sink_pad);
|
|
gst_object_unref (tee_pad);
|
|
gst_object_unref (sink_pad);
|
|
}
|
|
|
|
gst_element_sync_state_with_parent (priv->appsink[index]);
|
|
}
|
|
|
|
static void
|
|
plug_sink (GstRTSPStream * stream, const GstRTSPTransport * transport,
|
|
guint index)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gboolean is_tcp, is_udp, is_mcast;
|
|
priv = stream->priv;
|
|
|
|
is_tcp = transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP;
|
|
is_udp = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP;
|
|
is_mcast = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST;
|
|
|
|
if (is_udp)
|
|
plug_udp_sink (stream, priv->udpsink[index],
|
|
&priv->udpqueue[index], index, FALSE);
|
|
|
|
else if (is_mcast)
|
|
plug_udp_sink (stream, priv->mcast_udpsink[index],
|
|
&priv->mcast_udpqueue[index], index, TRUE);
|
|
|
|
else if (is_tcp)
|
|
plug_tcp_sink (stream, index);
|
|
}
|
|
|
|
/* must be called with lock */
|
|
static gboolean
|
|
create_sender_part (GstRTSPStream * stream, const GstRTSPTransport * transport)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstPad *pad;
|
|
GstBin *bin;
|
|
gboolean is_tcp, is_udp, is_mcast;
|
|
gint mcast_ttl = 0;
|
|
gint i;
|
|
|
|
GST_DEBUG_OBJECT (stream, "create sender part");
|
|
priv = stream->priv;
|
|
bin = priv->joined_bin;
|
|
|
|
is_tcp = transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP;
|
|
is_udp = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP;
|
|
is_mcast = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST;
|
|
|
|
if (is_mcast)
|
|
mcast_ttl = transport->ttl;
|
|
|
|
GST_DEBUG_OBJECT (stream, "tcp: %d, udp: %d, mcast: %d (ttl: %d)", is_tcp,
|
|
is_udp, is_mcast, mcast_ttl);
|
|
|
|
if (is_udp && !priv->server_addr_v4 && !priv->server_addr_v6) {
|
|
GST_WARNING_OBJECT (stream, "no sockets assigned for UDP");
|
|
return FALSE;
|
|
}
|
|
|
|
if (is_mcast && !priv->mcast_addr_v4 && !priv->mcast_addr_v6) {
|
|
GST_WARNING_OBJECT (stream, "no sockets assigned for UDP multicast");
|
|
return FALSE;
|
|
}
|
|
|
|
if (g_object_class_find_property (G_OBJECT_GET_CLASS (priv->payloader),
|
|
"onvif-no-rate-control"))
|
|
g_object_set (priv->payloader, "onvif-no-rate-control",
|
|
!priv->do_rate_control, NULL);
|
|
|
|
for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
|
|
gboolean link_tee = FALSE;
|
|
/* For the sender we create this bit of pipeline for both
|
|
* RTP and RTCP (when enabled).
|
|
* Initially there will be only one active transport for
|
|
* the stream, so the pipeline will look like this:
|
|
*
|
|
* .--------. .-----. .---------.
|
|
* | rtpbin | | tee | | sink |
|
|
* | send->sink src->sink |
|
|
* '--------' '-----' '---------'
|
|
*
|
|
* For each new transport, the already existing branch will
|
|
* be reconfigured by adding a queue element:
|
|
*
|
|
* .--------. .-----. .---------. .---------.
|
|
* | rtpbin | | tee | | queue | | udpsink |
|
|
* | send->sink src->sink src->sink |
|
|
* '--------' | | '---------' '---------'
|
|
* | | .---------. .---------.
|
|
* | | | queue | | udpsink |
|
|
* | src->sink src->sink |
|
|
* | | '---------' '---------'
|
|
* | | .---------. .---------.
|
|
* | | | queue | | appsink |
|
|
* | src->sink src->sink |
|
|
* '-----' '---------' '---------'
|
|
*/
|
|
|
|
/* Only link the RTP send src if we're going to send RTP, link
|
|
* the RTCP send src always */
|
|
if (!priv->srcpad && i == 0)
|
|
continue;
|
|
|
|
if (!priv->tee[i]) {
|
|
/* make tee for RTP/RTCP */
|
|
priv->tee[i] = gst_element_factory_make ("tee", NULL);
|
|
gst_bin_add (bin, priv->tee[i]);
|
|
link_tee = TRUE;
|
|
}
|
|
|
|
if (is_udp && !priv->udpsink[i]) {
|
|
/* we create only one pair of udpsinks for IPv4 and IPv6 */
|
|
create_and_configure_udpsink (stream, &priv->udpsink[i],
|
|
priv->socket_v4[i], priv->socket_v6[i], FALSE, (i == 0), mcast_ttl);
|
|
plug_sink (stream, transport, i);
|
|
} else if (is_mcast && !priv->mcast_udpsink[i]) {
|
|
/* we create only one pair of mcast-udpsinks for IPv4 and IPv6 */
|
|
create_and_configure_udpsink (stream, &priv->mcast_udpsink[i],
|
|
priv->mcast_socket_v4[i], priv->mcast_socket_v6[i], TRUE, (i == 0),
|
|
mcast_ttl);
|
|
plug_sink (stream, transport, i);
|
|
} else if (is_tcp && !priv->appsink[i]) {
|
|
/* make appsink */
|
|
priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
|
|
g_object_set (priv->appsink[i], "emit-signals", FALSE, "buffer-list",
|
|
TRUE, "max-buffers", 1, NULL);
|
|
|
|
if (i == 0)
|
|
g_object_set (priv->appsink[i], "sync", priv->do_rate_control, NULL);
|
|
|
|
/* we need to set sync and preroll to FALSE for the sink to avoid
|
|
* deadlock. This is only needed for sink sending RTCP data. */
|
|
if (i == 1)
|
|
g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
|
|
|
|
gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
|
|
&sink_cb, stream, NULL);
|
|
plug_sink (stream, transport, i);
|
|
}
|
|
|
|
if (link_tee) {
|
|
/* and link to rtpbin send pad */
|
|
gst_element_sync_state_with_parent (priv->tee[i]);
|
|
pad = gst_element_get_static_pad (priv->tee[i], "sink");
|
|
gst_pad_link (priv->send_src[i], pad);
|
|
gst_object_unref (pad);
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* must be called with lock */
|
|
static void
|
|
plug_src (GstRTSPStream * stream, GstBin * bin, GstElement * src,
|
|
GstElement * funnel)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstPad *pad, *selpad;
|
|
gulong id = 0;
|
|
|
|
priv = stream->priv;
|
|
|
|
/* add src */
|
|
gst_bin_add (bin, src);
|
|
|
|
pad = gst_element_get_static_pad (src, "src");
|
|
if (priv->srcpad) {
|
|
/* block pad so src can't push data while it's not yet linked */
|
|
id = gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BLOCK |
|
|
GST_PAD_PROBE_TYPE_BUFFER, NULL, NULL, NULL);
|
|
/* we set and keep these to playing so that they don't cause NO_PREROLL return
|
|
* values. This is only relevant for PLAY pipelines */
|
|
gst_element_set_state (src, GST_STATE_PLAYING);
|
|
gst_element_set_locked_state (src, TRUE);
|
|
}
|
|
|
|
/* and link to the funnel */
|
|
selpad = gst_element_request_pad_simple (funnel, "sink_%u");
|
|
gst_pad_link (pad, selpad);
|
|
if (id != 0)
|
|
gst_pad_remove_probe (pad, id);
|
|
gst_object_unref (pad);
|
|
gst_object_unref (selpad);
|
|
}
|
|
|
|
/* must be called with lock */
|
|
static gboolean
|
|
create_receiver_part (GstRTSPStream * stream, const GstRTSPTransport *
|
|
transport)
|
|
{
|
|
gboolean ret = FALSE;
|
|
GstRTSPStreamPrivate *priv;
|
|
GstPad *pad;
|
|
GstBin *bin;
|
|
gboolean tcp;
|
|
gboolean udp;
|
|
gboolean mcast;
|
|
gboolean secure;
|
|
gint i;
|
|
GstCaps *rtp_caps;
|
|
GstCaps *rtcp_caps;
|
|
|
|
GST_DEBUG_OBJECT (stream, "create receiver part");
|
|
priv = stream->priv;
|
|
bin = priv->joined_bin;
|
|
|
|
tcp = transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP;
|
|
udp = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP;
|
|
mcast = transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST;
|
|
secure = (priv->profiles & GST_RTSP_PROFILE_SAVP)
|
|
|| (priv->profiles & GST_RTSP_PROFILE_SAVPF);
|
|
|
|
if (secure) {
|
|
rtp_caps = gst_caps_new_empty_simple ("application/x-srtp");
|
|
rtcp_caps = gst_caps_new_empty_simple ("application/x-srtcp");
|
|
} else {
|
|
rtp_caps = gst_caps_new_empty_simple ("application/x-rtp");
|
|
rtcp_caps = gst_caps_new_empty_simple ("application/x-rtcp");
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (stream,
|
|
"RTP caps: %" GST_PTR_FORMAT " RTCP caps: %" GST_PTR_FORMAT, rtp_caps,
|
|
rtcp_caps);
|
|
|
|
for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
|
|
/* For the receiver we create this bit of pipeline for both
|
|
* RTP and RTCP (when enabled). We receive RTP/RTCP on appsrc and udpsrc
|
|
* and it is all funneled into the rtpbin receive pad.
|
|
*
|
|
*
|
|
* .--------. .--------. .--------.
|
|
* | udpsrc | | funnel | | rtpbin |
|
|
* | RTP src->sink src->sink |
|
|
* '--------' | | | |
|
|
* .--------. | | | |
|
|
* | appsrc | | | | |
|
|
* | RTP src->sink | | |
|
|
* '--------' '--------' | |
|
|
* | |
|
|
* .--------. .--------. | |
|
|
* | udpsrc | | funnel | | |
|
|
* | RTCP src->sink src->sink |
|
|
* '--------' | | '--------'
|
|
* .--------. | |
|
|
* | appsrc | | |
|
|
* | RTCP src->sink |
|
|
* '--------' '--------'
|
|
*/
|
|
|
|
if (!priv->sinkpad && i == 0) {
|
|
/* Only connect recv RTP sink if we expect to receive RTP. Connect recv
|
|
* RTCP sink always */
|
|
continue;
|
|
}
|
|
|
|
/* make funnel for the RTP/RTCP receivers */
|
|
if (!priv->funnel[i]) {
|
|
priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
|
|
gst_bin_add (bin, priv->funnel[i]);
|
|
|
|
pad = gst_element_get_static_pad (priv->funnel[i], "src");
|
|
gst_pad_link (pad, priv->recv_sink[i]);
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
if (udp && !priv->udpsrc_v4[i] && priv->server_addr_v4) {
|
|
GST_DEBUG_OBJECT (stream, "udp IPv4, create and configure udpsources");
|
|
if (!create_and_configure_udpsource (&priv->udpsrc_v4[i],
|
|
priv->socket_v4[i]))
|
|
goto done;
|
|
|
|
if (i == 0) {
|
|
g_object_set (priv->udpsrc_v4[i], "caps", rtp_caps, NULL);
|
|
} else {
|
|
g_object_set (priv->udpsrc_v4[i], "caps", rtcp_caps, NULL);
|
|
|
|
/* block early rtcp packets, pipeline not ready */
|
|
g_assert (priv->block_early_rtcp_pad == NULL);
|
|
priv->block_early_rtcp_pad = gst_element_get_static_pad
|
|
(priv->udpsrc_v4[i], "src");
|
|
priv->block_early_rtcp_probe = gst_pad_add_probe
|
|
(priv->block_early_rtcp_pad,
|
|
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER, NULL, NULL,
|
|
NULL);
|
|
}
|
|
|
|
plug_src (stream, bin, priv->udpsrc_v4[i], priv->funnel[i]);
|
|
}
|
|
|
|
if (udp && !priv->udpsrc_v6[i] && priv->server_addr_v6) {
|
|
GST_DEBUG_OBJECT (stream, "udp IPv6, create and configure udpsources");
|
|
if (!create_and_configure_udpsource (&priv->udpsrc_v6[i],
|
|
priv->socket_v6[i]))
|
|
goto done;
|
|
|
|
if (i == 0) {
|
|
g_object_set (priv->udpsrc_v6[i], "caps", rtp_caps, NULL);
|
|
} else {
|
|
g_object_set (priv->udpsrc_v6[i], "caps", rtcp_caps, NULL);
|
|
|
|
/* block early rtcp packets, pipeline not ready */
|
|
g_assert (priv->block_early_rtcp_pad_ipv6 == NULL);
|
|
priv->block_early_rtcp_pad_ipv6 = gst_element_get_static_pad
|
|
(priv->udpsrc_v6[i], "src");
|
|
priv->block_early_rtcp_probe_ipv6 = gst_pad_add_probe
|
|
(priv->block_early_rtcp_pad_ipv6,
|
|
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER, NULL, NULL,
|
|
NULL);
|
|
}
|
|
|
|
plug_src (stream, bin, priv->udpsrc_v6[i], priv->funnel[i]);
|
|
}
|
|
|
|
if (mcast && !priv->mcast_udpsrc_v4[i] && priv->mcast_addr_v4) {
|
|
GST_DEBUG_OBJECT (stream, "mcast IPv4, create and configure udpsources");
|
|
if (!create_and_configure_udpsource (&priv->mcast_udpsrc_v4[i],
|
|
priv->mcast_socket_v4[i]))
|
|
goto done;
|
|
|
|
if (i == 0) {
|
|
g_object_set (priv->mcast_udpsrc_v4[i], "caps", rtp_caps, NULL);
|
|
} else {
|
|
g_object_set (priv->mcast_udpsrc_v4[i], "caps", rtcp_caps, NULL);
|
|
}
|
|
|
|
plug_src (stream, bin, priv->mcast_udpsrc_v4[i], priv->funnel[i]);
|
|
}
|
|
|
|
if (mcast && !priv->mcast_udpsrc_v6[i] && priv->mcast_addr_v6) {
|
|
GST_DEBUG_OBJECT (stream, "mcast IPv6, create and configure udpsources");
|
|
if (!create_and_configure_udpsource (&priv->mcast_udpsrc_v6[i],
|
|
priv->mcast_socket_v6[i]))
|
|
goto done;
|
|
|
|
if (i == 0) {
|
|
g_object_set (priv->mcast_udpsrc_v6[i], "caps", rtp_caps, NULL);
|
|
} else {
|
|
g_object_set (priv->mcast_udpsrc_v6[i], "caps", rtcp_caps, NULL);
|
|
}
|
|
|
|
plug_src (stream, bin, priv->mcast_udpsrc_v6[i], priv->funnel[i]);
|
|
}
|
|
|
|
if (tcp && !priv->appsrc[i]) {
|
|
/* make and add appsrc */
|
|
priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
|
|
priv->appsrc_base_time[i] = -1;
|
|
g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, "is-live",
|
|
TRUE, NULL);
|
|
plug_src (stream, bin, priv->appsrc[i], priv->funnel[i]);
|
|
}
|
|
|
|
gst_element_sync_state_with_parent (priv->funnel[i]);
|
|
}
|
|
|
|
ret = TRUE;
|
|
|
|
done:
|
|
gst_caps_unref (rtp_caps);
|
|
gst_caps_unref (rtcp_caps);
|
|
return ret;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtsp_stream_is_tcp_receiver (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gboolean ret = FALSE;
|
|
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
ret = (priv->sinkpad != NULL && priv->appsrc[0] != NULL);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
check_mcast_client_addr (GstRTSPStream * stream, const GstRTSPTransport * tr)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GList *walk;
|
|
|
|
if (priv->mcast_clients == NULL)
|
|
goto no_addr;
|
|
|
|
if (tr == NULL)
|
|
goto no_transport;
|
|
|
|
if (tr->destination == NULL)
|
|
goto no_destination;
|
|
|
|
for (walk = priv->mcast_clients; walk; walk = g_list_next (walk)) {
|
|
UdpClientAddrInfo *cli = walk->data;
|
|
|
|
if ((g_strcmp0 (cli->address, tr->destination) == 0) &&
|
|
(cli->rtp_port == tr->port.min))
|
|
return TRUE;
|
|
}
|
|
|
|
return FALSE;
|
|
|
|
no_addr:
|
|
{
|
|
GST_WARNING_OBJECT (stream, "Adding mcast transport, but no mcast address "
|
|
"has been reserved");
|
|
return FALSE;
|
|
}
|
|
no_transport:
|
|
{
|
|
GST_WARNING_OBJECT (stream, "Adding mcast transport, but no transport "
|
|
"has been provided");
|
|
return FALSE;
|
|
}
|
|
no_destination:
|
|
{
|
|
GST_WARNING_OBJECT (stream, "Adding mcast transport, but it doesn't match "
|
|
"the reserved address");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_join_bin:
|
|
* @stream: a #GstRTSPStream
|
|
* @bin: (transfer none): a #GstBin to join
|
|
* @rtpbin: (transfer none): a rtpbin element in @bin
|
|
* @state: the target state of the new elements
|
|
*
|
|
* Join the #GstBin @bin that contains the element @rtpbin.
|
|
*
|
|
* @stream will link to @rtpbin, which must be inside @bin. The elements
|
|
* added to @bin will be set to the state given in @state.
|
|
*
|
|
* Returns: %TRUE on success.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
|
|
GstElement * rtpbin, GstState state)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
guint idx;
|
|
gchar *name;
|
|
GstPadLinkReturn ret;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
|
|
g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (priv->joined_bin != NULL)
|
|
goto was_joined;
|
|
|
|
/* create a session with the same index as the stream */
|
|
idx = priv->idx;
|
|
|
|
GST_INFO ("stream %p joining bin as session %u", stream, idx);
|
|
|
|
if (priv->profiles & GST_RTSP_PROFILE_SAVP
|
|
|| priv->profiles & GST_RTSP_PROFILE_SAVPF) {
|
|
/* For SRTP */
|
|
g_signal_connect (rtpbin, "request-rtp-encoder",
|
|
(GCallback) request_rtp_encoder, stream);
|
|
g_signal_connect (rtpbin, "request-rtcp-encoder",
|
|
(GCallback) request_rtcp_encoder, stream);
|
|
g_signal_connect (rtpbin, "request-rtp-decoder",
|
|
(GCallback) request_rtp_rtcp_decoder, stream);
|
|
g_signal_connect (rtpbin, "request-rtcp-decoder",
|
|
(GCallback) request_rtp_rtcp_decoder, stream);
|
|
}
|
|
|
|
if (priv->sinkpad) {
|
|
g_signal_connect (rtpbin, "request-pt-map",
|
|
(GCallback) request_pt_map, stream);
|
|
}
|
|
|
|
/* get pads from the RTP session element for sending and receiving
|
|
* RTP/RTCP*/
|
|
if (priv->srcpad) {
|
|
/* get a pad for sending RTP */
|
|
name = g_strdup_printf ("send_rtp_sink_%u", idx);
|
|
priv->send_rtp_sink = gst_element_request_pad_simple (rtpbin, name);
|
|
g_free (name);
|
|
|
|
/* link the RTP pad to the session manager, it should not really fail unless
|
|
* this is not really an RTP pad */
|
|
ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
|
|
if (ret != GST_PAD_LINK_OK)
|
|
goto link_failed;
|
|
|
|
name = g_strdup_printf ("send_rtp_src_%u", idx);
|
|
priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
|
|
g_free (name);
|
|
} else {
|
|
/* RECORD case: need to connect our sinkpad from here */
|
|
g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
|
|
/* EOS */
|
|
g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
|
|
|
|
name = g_strdup_printf ("recv_rtp_sink_%u", idx);
|
|
priv->recv_sink[0] = gst_element_request_pad_simple (rtpbin, name);
|
|
g_free (name);
|
|
}
|
|
|
|
if (priv->enable_rtcp) {
|
|
name = g_strdup_printf ("send_rtcp_src_%u", idx);
|
|
priv->send_src[1] = gst_element_request_pad_simple (rtpbin, name);
|
|
g_free (name);
|
|
|
|
name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
|
|
priv->recv_sink[1] = gst_element_request_pad_simple (rtpbin, name);
|
|
g_free (name);
|
|
}
|
|
|
|
/* get the session */
|
|
g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
|
|
|
|
g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
|
|
stream);
|
|
g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
|
|
stream);
|
|
g_signal_connect (priv->session, "on-ssrc-active",
|
|
(GCallback) on_ssrc_active, stream);
|
|
g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
|
|
stream);
|
|
g_signal_connect (priv->session, "on-bye-timeout",
|
|
(GCallback) on_bye_timeout, stream);
|
|
g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
|
|
stream);
|
|
|
|
/* signal for sender ssrc */
|
|
g_signal_connect (priv->session, "on-new-sender-ssrc",
|
|
(GCallback) on_new_sender_ssrc, stream);
|
|
g_signal_connect (priv->session, "on-sender-ssrc-active",
|
|
(GCallback) on_sender_ssrc_active, stream);
|
|
|
|
g_object_set (priv->session, "disable-sr-timestamp", !priv->do_rate_control,
|
|
NULL);
|
|
|
|
if (priv->srcpad) {
|
|
/* be notified of caps changes */
|
|
priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
|
|
(GCallback) caps_notify, stream);
|
|
priv->caps = gst_pad_get_current_caps (priv->send_src[0]);
|
|
}
|
|
|
|
priv->joined_bin = bin;
|
|
GST_DEBUG_OBJECT (stream, "successfully joined bin");
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
was_joined:
|
|
{
|
|
g_mutex_unlock (&priv->lock);
|
|
return TRUE;
|
|
}
|
|
link_failed:
|
|
{
|
|
GST_WARNING ("failed to link stream %u", idx);
|
|
gst_object_unref (priv->send_rtp_sink);
|
|
priv->send_rtp_sink = NULL;
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
clear_element (GstBin * bin, GstElement ** elementptr)
|
|
{
|
|
if (*elementptr) {
|
|
gst_element_set_locked_state (*elementptr, FALSE);
|
|
gst_element_set_state (*elementptr, GST_STATE_NULL);
|
|
if (GST_ELEMENT_PARENT (*elementptr))
|
|
gst_bin_remove (bin, *elementptr);
|
|
else
|
|
gst_object_unref (*elementptr);
|
|
*elementptr = NULL;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_leave_bin:
|
|
* @stream: a #GstRTSPStream
|
|
* @bin: (transfer none): a #GstBin
|
|
* @rtpbin: (transfer none): a rtpbin #GstElement
|
|
*
|
|
* Remove the elements of @stream from @bin.
|
|
*
|
|
* Return: %TRUE on success.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
|
|
GstElement * rtpbin)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gint i;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
|
|
g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->send_lock);
|
|
priv->continue_sending = FALSE;
|
|
priv->send_cookie++;
|
|
g_cond_signal (&priv->send_cond);
|
|
g_mutex_unlock (&priv->send_lock);
|
|
|
|
if (priv->send_thread) {
|
|
g_thread_join (priv->send_thread);
|
|
}
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (priv->joined_bin == NULL)
|
|
goto was_not_joined;
|
|
if (priv->joined_bin != bin)
|
|
goto wrong_bin;
|
|
|
|
priv->joined_bin = NULL;
|
|
|
|
/* all transports must be removed by now */
|
|
if (priv->transports != NULL)
|
|
goto transports_not_removed;
|
|
|
|
if (priv->send_pool) {
|
|
GThreadPool *slask;
|
|
|
|
slask = priv->send_pool;
|
|
priv->send_pool = NULL;
|
|
g_mutex_unlock (&priv->lock);
|
|
g_thread_pool_free (slask, TRUE, TRUE);
|
|
g_mutex_lock (&priv->lock);
|
|
}
|
|
|
|
clear_tr_cache (priv);
|
|
|
|
GST_INFO ("stream %p leaving bin", stream);
|
|
|
|
if (priv->srcpad) {
|
|
gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
|
|
|
|
g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
|
|
gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
|
|
gst_object_unref (priv->send_rtp_sink);
|
|
priv->send_rtp_sink = NULL;
|
|
} else if (priv->recv_rtp_src) {
|
|
gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
|
|
gst_object_unref (priv->recv_rtp_src);
|
|
priv->recv_rtp_src = NULL;
|
|
}
|
|
|
|
for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
|
|
clear_element (bin, &priv->udpsrc_v4[i]);
|
|
clear_element (bin, &priv->udpsrc_v6[i]);
|
|
clear_element (bin, &priv->udpqueue[i]);
|
|
clear_element (bin, &priv->udpsink[i]);
|
|
|
|
clear_element (bin, &priv->mcast_udpsrc_v4[i]);
|
|
clear_element (bin, &priv->mcast_udpsrc_v6[i]);
|
|
clear_element (bin, &priv->mcast_udpqueue[i]);
|
|
clear_element (bin, &priv->mcast_udpsink[i]);
|
|
|
|
clear_element (bin, &priv->appsrc[i]);
|
|
clear_element (bin, &priv->appqueue[i]);
|
|
clear_element (bin, &priv->appsink[i]);
|
|
|
|
clear_element (bin, &priv->tee[i]);
|
|
clear_element (bin, &priv->funnel[i]);
|
|
|
|
if (priv->sinkpad || i == 1) {
|
|
gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
|
|
gst_object_unref (priv->recv_sink[i]);
|
|
priv->recv_sink[i] = NULL;
|
|
}
|
|
}
|
|
|
|
if (priv->srcpad) {
|
|
gst_object_unref (priv->send_src[0]);
|
|
priv->send_src[0] = NULL;
|
|
}
|
|
|
|
if (priv->enable_rtcp) {
|
|
gst_element_release_request_pad (rtpbin, priv->send_src[1]);
|
|
gst_object_unref (priv->send_src[1]);
|
|
priv->send_src[1] = NULL;
|
|
}
|
|
|
|
g_object_unref (priv->session);
|
|
priv->session = NULL;
|
|
if (priv->caps)
|
|
gst_caps_unref (priv->caps);
|
|
priv->caps = NULL;
|
|
|
|
if (priv->srtpenc)
|
|
gst_object_unref (priv->srtpenc);
|
|
if (priv->srtpdec)
|
|
gst_object_unref (priv->srtpdec);
|
|
|
|
if (priv->mcast_addr_v4)
|
|
gst_rtsp_address_free (priv->mcast_addr_v4);
|
|
priv->mcast_addr_v4 = NULL;
|
|
if (priv->mcast_addr_v6)
|
|
gst_rtsp_address_free (priv->mcast_addr_v6);
|
|
priv->mcast_addr_v6 = NULL;
|
|
if (priv->server_addr_v4)
|
|
gst_rtsp_address_free (priv->server_addr_v4);
|
|
priv->server_addr_v4 = NULL;
|
|
if (priv->server_addr_v6)
|
|
gst_rtsp_address_free (priv->server_addr_v6);
|
|
priv->server_addr_v6 = NULL;
|
|
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return TRUE;
|
|
|
|
was_not_joined:
|
|
{
|
|
g_mutex_unlock (&priv->lock);
|
|
return TRUE;
|
|
}
|
|
transports_not_removed:
|
|
{
|
|
GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
wrong_bin:
|
|
{
|
|
GST_ERROR_OBJECT (stream, "leaving the wrong bin");
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_joined_bin:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the previous joined bin with gst_rtsp_stream_join_bin() or NULL.
|
|
*
|
|
* Return: (transfer full) (nullable): the joined bin or NULL.
|
|
*/
|
|
GstBin *
|
|
gst_rtsp_stream_get_joined_bin (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstBin *bin = NULL;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
bin = priv->joined_bin ? gst_object_ref (priv->joined_bin) : NULL;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return bin;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_rtpinfo:
|
|
* @stream: a #GstRTSPStream
|
|
* @rtptime: (allow-none) (out caller-allocates): result RTP timestamp
|
|
* @seq: (allow-none) (out caller-allocates): result RTP seqnum
|
|
* @clock_rate: (allow-none) (out caller-allocates): the clock rate
|
|
* @running_time: (out caller-allocates): result running-time
|
|
*
|
|
* Retrieve the current rtptime, seq and running-time. This is used to
|
|
* construct a RTPInfo reply header.
|
|
*
|
|
* Returns: %TRUE when rtptime, seq and running-time could be determined.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
|
|
guint * rtptime, guint * seq, guint * clock_rate,
|
|
GstClockTime * running_time)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstStructure *stats;
|
|
GObjectClass *payobjclass;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
priv = stream->priv;
|
|
|
|
payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
/* First try to extract the information from the last buffer on the sinks.
|
|
* This will have a more accurate sequence number and timestamp, as between
|
|
* the payloader and the sink there can be some queues
|
|
*/
|
|
if (priv->udpsink[0] || priv->mcast_udpsink[0] || priv->appsink[0]) {
|
|
GstSample *last_sample;
|
|
|
|
if (priv->udpsink[0])
|
|
g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
|
|
else if (priv->mcast_udpsink[0])
|
|
g_object_get (priv->mcast_udpsink[0], "last-sample", &last_sample, NULL);
|
|
else
|
|
g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
|
|
|
|
if (last_sample) {
|
|
GstCaps *caps;
|
|
GstBuffer *buffer;
|
|
GstSegment *segment;
|
|
GstStructure *s;
|
|
GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
|
|
|
|
caps = gst_sample_get_caps (last_sample);
|
|
buffer = gst_sample_get_buffer (last_sample);
|
|
segment = gst_sample_get_segment (last_sample);
|
|
s = gst_caps_get_structure (caps, 0);
|
|
|
|
if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
|
|
guint ssrc_buf = gst_rtp_buffer_get_ssrc (&rtp_buffer);
|
|
guint ssrc_stream = 0;
|
|
if (gst_structure_has_field_typed (s, "ssrc", G_TYPE_UINT) &&
|
|
gst_structure_get_uint (s, "ssrc", &ssrc_stream) &&
|
|
ssrc_buf != ssrc_stream) {
|
|
/* Skip buffers from auxiliary streams. */
|
|
GST_DEBUG_OBJECT (stream,
|
|
"not a buffer from the payloader, SSRC: %08x", ssrc_buf);
|
|
|
|
gst_rtp_buffer_unmap (&rtp_buffer);
|
|
gst_sample_unref (last_sample);
|
|
goto stats;
|
|
}
|
|
|
|
if (seq) {
|
|
*seq = gst_rtp_buffer_get_seq (&rtp_buffer);
|
|
}
|
|
|
|
if (rtptime) {
|
|
*rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
|
|
}
|
|
|
|
gst_rtp_buffer_unmap (&rtp_buffer);
|
|
|
|
if (running_time) {
|
|
*running_time =
|
|
gst_segment_to_running_time (segment, GST_FORMAT_TIME,
|
|
GST_BUFFER_TIMESTAMP (buffer));
|
|
}
|
|
|
|
if (clock_rate) {
|
|
gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
|
|
|
|
if (*clock_rate == 0 && running_time)
|
|
*running_time = GST_CLOCK_TIME_NONE;
|
|
}
|
|
gst_sample_unref (last_sample);
|
|
|
|
goto done;
|
|
} else {
|
|
gst_sample_unref (last_sample);
|
|
}
|
|
} else if (priv->blocking) {
|
|
if (seq) {
|
|
if (!priv->blocked_buffer)
|
|
goto stats;
|
|
*seq = priv->blocked_seqnum;
|
|
}
|
|
|
|
if (rtptime) {
|
|
if (!priv->blocked_buffer)
|
|
goto stats;
|
|
*rtptime = priv->blocked_rtptime;
|
|
}
|
|
|
|
if (running_time) {
|
|
if (!GST_CLOCK_TIME_IS_VALID (priv->blocked_running_time))
|
|
goto stats;
|
|
*running_time = priv->blocked_running_time;
|
|
}
|
|
|
|
if (clock_rate) {
|
|
*clock_rate = priv->blocked_clock_rate;
|
|
|
|
if (*clock_rate == 0 && running_time)
|
|
*running_time = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
stats:
|
|
if (g_object_class_find_property (payobjclass, "stats")) {
|
|
g_object_get (priv->payloader, "stats", &stats, NULL);
|
|
if (stats == NULL)
|
|
goto no_stats;
|
|
|
|
if (seq)
|
|
gst_structure_get_uint (stats, "seqnum-offset", seq);
|
|
|
|
if (rtptime)
|
|
gst_structure_get_uint (stats, "timestamp", rtptime);
|
|
|
|
if (running_time)
|
|
gst_structure_get_clock_time (stats, "running-time", running_time);
|
|
|
|
if (clock_rate) {
|
|
gst_structure_get_uint (stats, "clock-rate", clock_rate);
|
|
if (*clock_rate == 0 && running_time)
|
|
*running_time = GST_CLOCK_TIME_NONE;
|
|
}
|
|
gst_structure_free (stats);
|
|
} else {
|
|
if (!g_object_class_find_property (payobjclass, "seqnum") ||
|
|
!g_object_class_find_property (payobjclass, "timestamp"))
|
|
goto no_stats;
|
|
|
|
if (seq)
|
|
g_object_get (priv->payloader, "seqnum", seq, NULL);
|
|
|
|
if (rtptime)
|
|
g_object_get (priv->payloader, "timestamp", rtptime, NULL);
|
|
|
|
if (running_time)
|
|
*running_time = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
done:
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_stats:
|
|
{
|
|
GST_WARNING ("Could not get payloader stats");
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_rates:
|
|
* @stream: a #GstRTSPStream
|
|
* @rate: (optional) (out caller-allocates): the configured rate
|
|
* @applied_rate: (optional) (out caller-allocates): the configured applied_rate
|
|
*
|
|
* Retrieve the current rate and/or applied_rate.
|
|
*
|
|
* Returns: %TRUE if rate and/or applied_rate could be determined.
|
|
* Since: 1.18
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_get_rates (GstRTSPStream * stream, gdouble * rate,
|
|
gdouble * applied_rate)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstEvent *event;
|
|
const GstSegment *segment;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
if (!rate && !applied_rate) {
|
|
GST_WARNING_OBJECT (stream, "rate and applied_rate are both NULL");
|
|
return FALSE;
|
|
}
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
if (!priv->send_rtp_sink)
|
|
goto no_rtp_sink_pad;
|
|
|
|
event = gst_pad_get_sticky_event (priv->send_rtp_sink, GST_EVENT_SEGMENT, 0);
|
|
if (!event)
|
|
goto no_sticky_event;
|
|
|
|
gst_event_parse_segment (event, &segment);
|
|
if (rate)
|
|
*rate = segment->rate;
|
|
if (applied_rate)
|
|
*applied_rate = segment->applied_rate;
|
|
|
|
gst_event_unref (event);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_rtp_sink_pad:
|
|
{
|
|
GST_WARNING_OBJECT (stream, "no send_rtp_sink pad yet");
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
no_sticky_event:
|
|
{
|
|
GST_WARNING_OBJECT (stream, "no segment event on send_rtp_sink pad");
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_caps:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Retrieve the current caps of @stream.
|
|
*
|
|
* Returns: (transfer full) (nullable): the #GstCaps of @stream.
|
|
* use gst_caps_unref() after usage.
|
|
*/
|
|
GstCaps *
|
|
gst_rtsp_stream_get_caps (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstCaps *result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((result = priv->caps))
|
|
gst_caps_ref (result);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_recv_rtp:
|
|
* @stream: a #GstRTSPStream
|
|
* @buffer: (transfer full): a #GstBuffer
|
|
*
|
|
* Handle an RTP buffer for the stream. This method is usually called when a
|
|
* message has been received from a client using the TCP transport.
|
|
*
|
|
* This function takes ownership of @buffer.
|
|
*
|
|
* Returns: a GstFlowReturn.
|
|
*/
|
|
GstFlowReturn
|
|
gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstFlowReturn ret;
|
|
GstElement *element;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
|
|
priv = stream->priv;
|
|
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
|
|
g_return_val_if_fail (priv->joined_bin != NULL, FALSE);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (priv->appsrc[0])
|
|
element = gst_object_ref (priv->appsrc[0]);
|
|
else
|
|
element = NULL;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (element) {
|
|
if (priv->appsrc_base_time[0] == -1) {
|
|
/* Take current running_time. This timestamp will be put on
|
|
* the first buffer of each stream because we are a live source and so we
|
|
* timestamp with the running_time. When we are dealing with TCP, we also
|
|
* only timestamp the first buffer (using the DISCONT flag) because a server
|
|
* typically bursts data, for which we don't want to compensate by speeding
|
|
* up the media. The other timestamps will be interpollated from this one
|
|
* using the RTP timestamps. */
|
|
GST_OBJECT_LOCK (element);
|
|
if (GST_ELEMENT_CLOCK (element)) {
|
|
GstClockTime now;
|
|
GstClockTime base_time;
|
|
|
|
now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
|
|
base_time = GST_ELEMENT_CAST (element)->base_time;
|
|
|
|
priv->appsrc_base_time[0] = now - base_time;
|
|
GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
|
|
GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
|
|
", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
|
|
GST_TIME_ARGS (base_time));
|
|
}
|
|
GST_OBJECT_UNLOCK (element);
|
|
}
|
|
|
|
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
|
|
gst_object_unref (element);
|
|
} else {
|
|
ret = GST_FLOW_OK;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_recv_rtcp:
|
|
* @stream: a #GstRTSPStream
|
|
* @buffer: (transfer full): a #GstBuffer
|
|
*
|
|
* Handle an RTCP buffer for the stream. This method is usually called when a
|
|
* message has been received from a client using the TCP transport.
|
|
*
|
|
* This function takes ownership of @buffer.
|
|
*
|
|
* Returns: a GstFlowReturn.
|
|
*/
|
|
GstFlowReturn
|
|
gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstFlowReturn ret;
|
|
GstElement *element;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
|
|
priv = stream->priv;
|
|
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
|
|
|
|
if (priv->joined_bin == NULL) {
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_NOT_LINKED;
|
|
}
|
|
g_mutex_lock (&priv->lock);
|
|
if (priv->appsrc[1])
|
|
element = gst_object_ref (priv->appsrc[1]);
|
|
else
|
|
element = NULL;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (element) {
|
|
if (priv->appsrc_base_time[1] == -1) {
|
|
/* Take current running_time. This timestamp will be put on
|
|
* the first buffer of each stream because we are a live source and so we
|
|
* timestamp with the running_time. When we are dealing with TCP, we also
|
|
* only timestamp the first buffer (using the DISCONT flag) because a server
|
|
* typically bursts data, for which we don't want to compensate by speeding
|
|
* up the media. The other timestamps will be interpollated from this one
|
|
* using the RTP timestamps. */
|
|
GST_OBJECT_LOCK (element);
|
|
if (GST_ELEMENT_CLOCK (element)) {
|
|
GstClockTime now;
|
|
GstClockTime base_time;
|
|
|
|
now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
|
|
base_time = GST_ELEMENT_CAST (element)->base_time;
|
|
|
|
priv->appsrc_base_time[1] = now - base_time;
|
|
GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
|
|
GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
|
|
", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
|
|
GST_TIME_ARGS (base_time));
|
|
}
|
|
GST_OBJECT_UNLOCK (element);
|
|
}
|
|
|
|
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
|
|
gst_object_unref (element);
|
|
} else {
|
|
ret = GST_FLOW_OK;
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
/* must be called with lock */
|
|
static inline void
|
|
add_client (GstElement * rtp_sink, GstElement * rtcp_sink, const gchar * host,
|
|
gint rtp_port, gint rtcp_port)
|
|
{
|
|
if (rtp_sink != NULL)
|
|
g_signal_emit_by_name (rtp_sink, "add", host, rtp_port, NULL);
|
|
if (rtcp_sink != NULL)
|
|
g_signal_emit_by_name (rtcp_sink, "add", host, rtcp_port, NULL);
|
|
}
|
|
|
|
/* must be called with lock */
|
|
static void
|
|
remove_client (GstElement * rtp_sink, GstElement * rtcp_sink,
|
|
const gchar * host, gint rtp_port, gint rtcp_port)
|
|
{
|
|
if (rtp_sink != NULL)
|
|
g_signal_emit_by_name (rtp_sink, "remove", host, rtp_port, NULL);
|
|
if (rtcp_sink != NULL)
|
|
g_signal_emit_by_name (rtcp_sink, "remove", host, rtcp_port, NULL);
|
|
}
|
|
|
|
/* must be called with lock */
|
|
static gboolean
|
|
update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
|
|
gboolean add)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
const GstRTSPTransport *tr;
|
|
gchar *dest;
|
|
gint min, max;
|
|
GList *tr_element;
|
|
|
|
tr = gst_rtsp_stream_transport_get_transport (trans);
|
|
dest = tr->destination;
|
|
|
|
tr_element = g_list_find (priv->transports, trans);
|
|
|
|
if (add && tr_element)
|
|
return TRUE;
|
|
else if (!add && !tr_element)
|
|
return FALSE;
|
|
|
|
switch (tr->lower_transport) {
|
|
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
|
|
{
|
|
min = tr->port.min;
|
|
max = tr->port.max;
|
|
|
|
if (add) {
|
|
GST_INFO ("adding %s:%d-%d", dest, min, max);
|
|
if (!check_mcast_client_addr (stream, tr))
|
|
goto mcast_error;
|
|
add_client (priv->mcast_udpsink[0], priv->mcast_udpsink[1], dest, min,
|
|
max);
|
|
|
|
if (tr->ttl > 0) {
|
|
GST_INFO ("setting ttl-mc %d", tr->ttl);
|
|
if (priv->mcast_udpsink[0])
|
|
g_object_set (G_OBJECT (priv->mcast_udpsink[0]), "ttl-mc", tr->ttl,
|
|
NULL);
|
|
if (priv->mcast_udpsink[1])
|
|
g_object_set (G_OBJECT (priv->mcast_udpsink[1]), "ttl-mc", tr->ttl,
|
|
NULL);
|
|
}
|
|
priv->transports = g_list_prepend (priv->transports, trans);
|
|
} else {
|
|
GST_INFO ("removing %s:%d-%d", dest, min, max);
|
|
if (!remove_mcast_client_addr (stream, dest, min, max))
|
|
GST_WARNING_OBJECT (stream,
|
|
"Failed to remove multicast address: %s:%d-%d", dest, min, max);
|
|
priv->transports = g_list_delete_link (priv->transports, tr_element);
|
|
remove_client (priv->mcast_udpsink[0], priv->mcast_udpsink[1], dest,
|
|
min, max);
|
|
}
|
|
break;
|
|
}
|
|
case GST_RTSP_LOWER_TRANS_UDP:
|
|
{
|
|
if (priv->client_side) {
|
|
/* In client side mode the 'destination' is the RTSP server, so send
|
|
* to those ports */
|
|
min = tr->server_port.min;
|
|
max = tr->server_port.max;
|
|
} else {
|
|
min = tr->client_port.min;
|
|
max = tr->client_port.max;
|
|
}
|
|
|
|
if (add) {
|
|
GST_INFO ("adding %s:%d-%d", dest, min, max);
|
|
add_client (priv->udpsink[0], priv->udpsink[1], dest, min, max);
|
|
priv->transports = g_list_prepend (priv->transports, trans);
|
|
} else {
|
|
GST_INFO ("removing %s:%d-%d", dest, min, max);
|
|
priv->transports = g_list_delete_link (priv->transports, tr_element);
|
|
remove_client (priv->udpsink[0], priv->udpsink[1], dest, min, max);
|
|
}
|
|
priv->transports_cookie++;
|
|
break;
|
|
}
|
|
case GST_RTSP_LOWER_TRANS_TCP:
|
|
if (add) {
|
|
GST_INFO ("adding TCP %s", tr->destination);
|
|
priv->transports = g_list_prepend (priv->transports, trans);
|
|
priv->n_tcp_transports++;
|
|
} else {
|
|
GST_INFO ("removing TCP %s", tr->destination);
|
|
priv->transports = g_list_delete_link (priv->transports, tr_element);
|
|
|
|
gst_rtsp_stream_transport_lock_backlog (trans);
|
|
gst_rtsp_stream_transport_clear_backlog (trans);
|
|
gst_rtsp_stream_transport_unlock_backlog (trans);
|
|
|
|
priv->n_tcp_transports--;
|
|
}
|
|
priv->transports_cookie++;
|
|
break;
|
|
default:
|
|
goto unknown_transport;
|
|
}
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
unknown_transport:
|
|
{
|
|
GST_INFO ("Unknown transport %d", tr->lower_transport);
|
|
return FALSE;
|
|
}
|
|
mcast_error:
|
|
{
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
on_message_sent (GstRTSPStreamTransport * trans, gpointer user_data)
|
|
{
|
|
GstRTSPStream *stream = GST_RTSP_STREAM (user_data);
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
|
|
GST_DEBUG_OBJECT (stream, "message send complete");
|
|
|
|
check_transport_backlog (stream, trans);
|
|
|
|
g_mutex_lock (&priv->send_lock);
|
|
priv->send_cookie++;
|
|
g_cond_signal (&priv->send_cond);
|
|
g_mutex_unlock (&priv->send_lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_add_transport:
|
|
* @stream: a #GstRTSPStream
|
|
* @trans: (transfer none): a #GstRTSPStreamTransport
|
|
*
|
|
* Add the transport in @trans to @stream. The media of @stream will
|
|
* then also be send to the values configured in @trans. Adding the
|
|
* same transport twice will not add it a second time.
|
|
*
|
|
* @stream must be joined to a bin.
|
|
*
|
|
* @trans must contain a valid #GstRTSPTransport.
|
|
*
|
|
* Returns: %TRUE if @trans was added
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_add_transport (GstRTSPStream * stream,
|
|
GstRTSPStreamTransport * trans)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
priv = stream->priv;
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
|
|
g_return_val_if_fail (priv->joined_bin != NULL, FALSE);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
res = update_transport (stream, trans, TRUE);
|
|
if (res)
|
|
gst_rtsp_stream_transport_set_message_sent_full (trans, on_message_sent,
|
|
stream, NULL);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_remove_transport:
|
|
* @stream: a #GstRTSPStream
|
|
* @trans: (transfer none): a #GstRTSPStreamTransport
|
|
*
|
|
* Remove the transport in @trans from @stream. The media of @stream will
|
|
* not be sent to the values configured in @trans.
|
|
*
|
|
* @stream must be joined to a bin.
|
|
*
|
|
* @trans must contain a valid #GstRTSPTransport.
|
|
*
|
|
* Returns: %TRUE if @trans was removed
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
|
|
GstRTSPStreamTransport * trans)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
priv = stream->priv;
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
|
|
g_return_val_if_fail (priv->joined_bin != NULL, FALSE);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
res = update_transport (stream, trans, FALSE);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_update_crypto:
|
|
* @stream: a #GstRTSPStream
|
|
* @ssrc: the SSRC
|
|
* @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
|
|
*
|
|
* Update the new crypto information for @ssrc in @stream. If information
|
|
* for @ssrc did not exist, it will be added. If information
|
|
* for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
|
|
* be removed from @stream.
|
|
*
|
|
* Returns: %TRUE if @crypto could be updated
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
|
|
guint ssrc, GstCaps * crypto)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
|
|
|
|
priv = stream->priv;
|
|
|
|
GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (crypto)
|
|
g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
|
|
gst_caps_ref (crypto));
|
|
else
|
|
g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_rtp_socket:
|
|
* @stream: a #GstRTSPStream
|
|
* @family: the socket family
|
|
*
|
|
* Get the RTP socket from @stream for a @family.
|
|
*
|
|
* @stream must be joined to a bin.
|
|
*
|
|
* Returns: (transfer full) (nullable): the RTP socket or %NULL if no
|
|
* socket could be allocated for @family. Unref after usage
|
|
*/
|
|
GSocket *
|
|
gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GSocket *socket;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
|
|
family == G_SOCKET_FAMILY_IPV6, NULL);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (family == G_SOCKET_FAMILY_IPV6)
|
|
socket = priv->socket_v6[0];
|
|
else
|
|
socket = priv->socket_v4[0];
|
|
|
|
if (socket != NULL)
|
|
socket = g_object_ref (socket);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return socket;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_rtcp_socket:
|
|
* @stream: a #GstRTSPStream
|
|
* @family: the socket family
|
|
*
|
|
* Get the RTCP socket from @stream for a @family.
|
|
*
|
|
* @stream must be joined to a bin.
|
|
*
|
|
* Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
|
|
* socket could be allocated for @family. Unref after usage
|
|
*/
|
|
GSocket *
|
|
gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GSocket *socket;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
|
|
family == G_SOCKET_FAMILY_IPV6, NULL);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (family == G_SOCKET_FAMILY_IPV6)
|
|
socket = priv->socket_v6[1];
|
|
else
|
|
socket = priv->socket_v4[1];
|
|
|
|
if (socket != NULL)
|
|
socket = g_object_ref (socket);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return socket;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_rtp_multicast_socket:
|
|
* @stream: a #GstRTSPStream
|
|
* @family: the socket family
|
|
*
|
|
* Get the multicast RTP socket from @stream for a @family.
|
|
*
|
|
* Returns: (transfer full) (nullable): the multicast RTP socket or %NULL if no
|
|
*
|
|
* socket could be allocated for @family. Unref after usage
|
|
*/
|
|
GSocket *
|
|
gst_rtsp_stream_get_rtp_multicast_socket (GstRTSPStream * stream,
|
|
GSocketFamily family)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GSocket *socket;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
|
|
family == G_SOCKET_FAMILY_IPV6, NULL);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (family == G_SOCKET_FAMILY_IPV6)
|
|
socket = priv->mcast_socket_v6[0];
|
|
else
|
|
socket = priv->mcast_socket_v4[0];
|
|
|
|
if (socket != NULL)
|
|
socket = g_object_ref (socket);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return socket;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_rtcp_multicast_socket:
|
|
* @stream: a #GstRTSPStream
|
|
* @family: the socket family
|
|
*
|
|
* Get the multicast RTCP socket from @stream for a @family.
|
|
*
|
|
* Returns: (transfer full) (nullable): the multicast RTCP socket or %NULL if no
|
|
* socket could be allocated for @family. Unref after usage
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
GSocket *
|
|
gst_rtsp_stream_get_rtcp_multicast_socket (GstRTSPStream * stream,
|
|
GSocketFamily family)
|
|
{
|
|
GstRTSPStreamPrivate *priv = stream->priv;
|
|
GSocket *socket;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
|
|
family == G_SOCKET_FAMILY_IPV6, NULL);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if (family == G_SOCKET_FAMILY_IPV6)
|
|
socket = priv->mcast_socket_v6[1];
|
|
else
|
|
socket = priv->mcast_socket_v4[1];
|
|
|
|
if (socket != NULL)
|
|
socket = g_object_ref (socket);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return socket;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_add_multicast_client_address:
|
|
* @stream: a #GstRTSPStream
|
|
* @destination: (transfer none): a multicast address to add
|
|
* @rtp_port: RTP port
|
|
* @rtcp_port: RTCP port
|
|
* @family: socket family
|
|
*
|
|
* Add multicast client address to stream. At this point, the sockets that
|
|
* will stream RTP and RTCP data to @destination are supposed to be
|
|
* allocated.
|
|
*
|
|
* Returns: %TRUE if @destination can be addedd and handled by @stream.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_add_multicast_client_address (GstRTSPStream * stream,
|
|
const gchar * destination, guint rtp_port, guint rtcp_port,
|
|
GSocketFamily family)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
g_return_val_if_fail (destination != NULL, FALSE);
|
|
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
if ((family == G_SOCKET_FAMILY_IPV4) && (priv->mcast_socket_v4[0] == NULL))
|
|
goto socket_error;
|
|
else if ((family == G_SOCKET_FAMILY_IPV6) &&
|
|
(priv->mcast_socket_v6[0] == NULL))
|
|
goto socket_error;
|
|
|
|
if (!add_mcast_client_addr (stream, destination, rtp_port, rtcp_port))
|
|
goto add_addr_error;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return TRUE;
|
|
|
|
socket_error:
|
|
{
|
|
GST_WARNING_OBJECT (stream,
|
|
"Failed to add multicast address: no udp socket");
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
add_addr_error:
|
|
{
|
|
GST_WARNING_OBJECT (stream,
|
|
"Failed to add multicast address: invalid address");
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_multicast_client_addresses
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get all multicast client addresses that RTP data will be sent to
|
|
*
|
|
* Returns: A comma separated list of host:port pairs with destinations
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
gchar *
|
|
gst_rtsp_stream_get_multicast_client_addresses (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GString *str;
|
|
GList *clients;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
priv = stream->priv;
|
|
str = g_string_new ("");
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
clients = priv->mcast_clients;
|
|
while (clients != NULL) {
|
|
UdpClientAddrInfo *client;
|
|
|
|
client = (UdpClientAddrInfo *) clients->data;
|
|
clients = g_list_next (clients);
|
|
g_string_append_printf (str, "%s:%d%s", client->address, client->rtp_port,
|
|
(clients != NULL ? "," : ""));
|
|
}
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return g_string_free (str, FALSE);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_seqnum:
|
|
* @stream: a #GstRTSPStream
|
|
* @seqnum: a new sequence number
|
|
*
|
|
* Configure the sequence number in the payloader of @stream to @seqnum.
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
priv = stream->priv;
|
|
|
|
g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_seqnum:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Get the configured sequence number in the payloader of @stream.
|
|
*
|
|
* Returns: the sequence number of the payloader.
|
|
*/
|
|
guint16
|
|
gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
guint seqnum;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
|
|
|
|
return seqnum;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_transport_filter:
|
|
* @stream: a #GstRTSPStream
|
|
* @func: (scope call) (allow-none): a callback
|
|
* @user_data: (closure): user data passed to @func
|
|
*
|
|
* Call @func for each transport managed by @stream. The result value of @func
|
|
* determines what happens to the transport. @func will be called with @stream
|
|
* locked so no further actions on @stream can be performed from @func.
|
|
*
|
|
* If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
|
|
* @stream.
|
|
*
|
|
* If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
|
|
*
|
|
* If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
|
|
* will also be added with an additional ref to the result #GList of this
|
|
* function..
|
|
*
|
|
* When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
|
|
*
|
|
* Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
|
|
* transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
|
|
* element in the #GList should be unreffed before the list is freed.
|
|
*/
|
|
GList *
|
|
gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
|
|
GstRTSPStreamTransportFilterFunc func, gpointer user_data)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GList *result, *walk, *next;
|
|
GHashTable *visited = NULL;
|
|
guint cookie;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
priv = stream->priv;
|
|
|
|
result = NULL;
|
|
if (func)
|
|
visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
restart:
|
|
cookie = priv->transports_cookie;
|
|
for (walk = priv->transports; walk; walk = next) {
|
|
GstRTSPStreamTransport *trans = walk->data;
|
|
GstRTSPFilterResult res;
|
|
gboolean changed;
|
|
|
|
next = g_list_next (walk);
|
|
|
|
if (func) {
|
|
/* only visit each transport once */
|
|
if (g_hash_table_contains (visited, trans))
|
|
continue;
|
|
|
|
g_hash_table_add (visited, g_object_ref (trans));
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
res = func (stream, trans, user_data);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
} else
|
|
res = GST_RTSP_FILTER_REF;
|
|
|
|
changed = (cookie != priv->transports_cookie);
|
|
|
|
switch (res) {
|
|
case GST_RTSP_FILTER_REMOVE:
|
|
update_transport (stream, trans, FALSE);
|
|
break;
|
|
case GST_RTSP_FILTER_REF:
|
|
result = g_list_prepend (result, g_object_ref (trans));
|
|
break;
|
|
case GST_RTSP_FILTER_KEEP:
|
|
default:
|
|
break;
|
|
}
|
|
if (changed)
|
|
goto restart;
|
|
}
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (func)
|
|
g_hash_table_unref (visited);
|
|
|
|
return result;
|
|
}
|
|
|
|
static GstPadProbeReturn
|
|
rtp_pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstRTSPStream *stream;
|
|
GstBuffer *buffer = NULL;
|
|
GstPadProbeReturn ret = GST_PAD_PROBE_OK;
|
|
GstEvent *event;
|
|
|
|
stream = user_data;
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
if ((info->type & GST_PAD_PROBE_TYPE_BUFFER)) {
|
|
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
|
|
|
|
buffer = gst_pad_probe_info_get_buffer (info);
|
|
if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)) {
|
|
priv->blocked_buffer = TRUE;
|
|
priv->blocked_seqnum = gst_rtp_buffer_get_seq (&rtp);
|
|
priv->blocked_rtptime = gst_rtp_buffer_get_timestamp (&rtp);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
}
|
|
priv->position = GST_BUFFER_TIMESTAMP (buffer);
|
|
} else if ((info->type & GST_PAD_PROBE_TYPE_BUFFER_LIST)) {
|
|
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
|
|
|
|
GstBufferList *list = gst_pad_probe_info_get_buffer_list (info);
|
|
buffer = gst_buffer_list_get (list, 0);
|
|
if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)) {
|
|
priv->blocked_buffer = TRUE;
|
|
priv->blocked_seqnum = gst_rtp_buffer_get_seq (&rtp);
|
|
priv->blocked_rtptime = gst_rtp_buffer_get_timestamp (&rtp);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
}
|
|
priv->position = GST_BUFFER_TIMESTAMP (buffer);
|
|
} else if ((info->type & GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM)) {
|
|
if (GST_EVENT_TYPE (info->data) == GST_EVENT_GAP) {
|
|
gst_event_parse_gap (info->data, &priv->position, NULL);
|
|
} else {
|
|
ret = GST_PAD_PROBE_PASS;
|
|
g_mutex_unlock (&priv->lock);
|
|
goto done;
|
|
}
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
event = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0);
|
|
if (event) {
|
|
const GstSegment *segment;
|
|
|
|
gst_event_parse_segment (event, &segment);
|
|
priv->blocked_running_time =
|
|
gst_segment_to_stream_time (segment, GST_FORMAT_TIME, priv->position);
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
event = gst_pad_get_sticky_event (pad, GST_EVENT_CAPS, 0);
|
|
if (event) {
|
|
GstCaps *caps;
|
|
GstStructure *s;
|
|
|
|
gst_event_parse_caps (event, &caps);
|
|
s = gst_caps_get_structure (caps, 0);
|
|
gst_structure_get_int (s, "clock-rate", &priv->blocked_clock_rate);
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
priv->blocking = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (pad, "Now blocking");
|
|
|
|
GST_DEBUG_OBJECT (stream, "position: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (priv->position));
|
|
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
gst_element_post_message (priv->payloader,
|
|
gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
|
|
gst_structure_new ("GstRTSPStreamBlocking", "is_complete",
|
|
G_TYPE_BOOLEAN, priv->is_complete, NULL)));
|
|
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
static GstPadProbeReturn
|
|
rtcp_pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstRTSPStream *stream;
|
|
GstPadProbeReturn ret = GST_PAD_PROBE_OK;
|
|
|
|
stream = user_data;
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
if ((info->type & GST_PAD_PROBE_TYPE_BUFFER) ||
|
|
(info->type & GST_PAD_PROBE_TYPE_BUFFER_LIST)) {
|
|
GST_DEBUG_OBJECT (pad, "Now blocking on buffer");
|
|
} else if ((info->type & GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM)) {
|
|
if (GST_EVENT_TYPE (info->data) == GST_EVENT_GAP) {
|
|
GST_DEBUG_OBJECT (pad, "Now blocking on gap event");
|
|
ret = GST_PAD_PROBE_OK;
|
|
} else {
|
|
ret = GST_PAD_PROBE_PASS;
|
|
g_mutex_unlock (&priv->lock);
|
|
goto done;
|
|
}
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
|
|
static void
|
|
set_blocked (GstRTSPStream * stream, gboolean blocked)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
int i;
|
|
|
|
GST_DEBUG_OBJECT (stream, "blocked: %d", blocked);
|
|
|
|
priv = stream->priv;
|
|
|
|
if (blocked) {
|
|
/* if receiver */
|
|
if (priv->sinkpad) {
|
|
priv->blocking = TRUE;
|
|
return;
|
|
}
|
|
for (i = 0; i < 2; i++) {
|
|
if (priv->blocked_id[i] != 0)
|
|
continue;
|
|
if (priv->send_src[i]) {
|
|
priv->blocking = FALSE;
|
|
priv->blocked_buffer = FALSE;
|
|
priv->blocked_running_time = GST_CLOCK_TIME_NONE;
|
|
priv->blocked_clock_rate = 0;
|
|
|
|
if (i == 0) {
|
|
priv->blocked_id[i] = gst_pad_add_probe (priv->send_src[i],
|
|
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
|
|
GST_PAD_PROBE_TYPE_BUFFER_LIST |
|
|
GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, rtp_pad_blocking,
|
|
g_object_ref (stream), g_object_unref);
|
|
} else {
|
|
priv->blocked_id[i] = gst_pad_add_probe (priv->send_src[i],
|
|
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
|
|
GST_PAD_PROBE_TYPE_BUFFER_LIST |
|
|
GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, rtcp_pad_blocking,
|
|
g_object_ref (stream), g_object_unref);
|
|
}
|
|
}
|
|
}
|
|
} else {
|
|
for (i = 0; i < 2; i++) {
|
|
if (priv->blocked_id[i] != 0) {
|
|
gst_pad_remove_probe (priv->send_src[i], priv->blocked_id[i]);
|
|
priv->blocked_id[i] = 0;
|
|
}
|
|
}
|
|
priv->blocking = FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_blocked:
|
|
* @stream: a #GstRTSPStream
|
|
* @blocked: boolean indicating we should block or unblock
|
|
*
|
|
* Blocks or unblocks the dataflow on @stream.
|
|
*
|
|
* Returns: %TRUE on success
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
set_blocked (stream, blocked);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_ublock_linked:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Unblocks the dataflow on @stream if it is linked.
|
|
*
|
|
* Returns: %TRUE on success
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_unblock_linked (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
if (priv->send_src[0] && gst_pad_is_linked (priv->send_src[0]))
|
|
set_blocked (stream, FALSE);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_is_blocking:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Check if @stream is blocking on a #GstBuffer.
|
|
*
|
|
* Returns: %TRUE if @stream is blocking
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gboolean result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
result = priv->blocking;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_query_position:
|
|
* @stream: a #GstRTSPStream
|
|
* @position: (out): current position of a #GstRTSPStream
|
|
*
|
|
* Query the position of the stream in %GST_FORMAT_TIME. This only considers
|
|
* the RTP parts of the pipeline and not the RTCP parts.
|
|
*
|
|
* Returns: %TRUE if the position could be queried
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstElement *sink;
|
|
GstPad *pad = NULL;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
/* query position: if no sinks have been added yet,
|
|
* we obtain the position from the pad otherwise we query the sinks */
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
if (priv->blocking && GST_CLOCK_TIME_IS_VALID (priv->blocked_running_time)) {
|
|
*position = priv->blocked_running_time;
|
|
g_mutex_unlock (&priv->lock);
|
|
return TRUE;
|
|
}
|
|
|
|
/* depending on the transport type, it should query corresponding sink */
|
|
if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP)
|
|
sink = priv->udpsink[0];
|
|
else if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
|
|
sink = priv->mcast_udpsink[0];
|
|
else
|
|
sink = priv->appsink[0];
|
|
|
|
if (sink) {
|
|
gst_object_ref (sink);
|
|
} else if (priv->send_src[0]) {
|
|
pad = gst_object_ref (priv->send_src[0]);
|
|
} else {
|
|
g_mutex_unlock (&priv->lock);
|
|
GST_WARNING_OBJECT (stream, "Couldn't obtain position: erroneous pipeline");
|
|
return FALSE;
|
|
}
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (sink) {
|
|
if (!gst_element_query_position (sink, GST_FORMAT_TIME, position)) {
|
|
GST_WARNING_OBJECT (stream,
|
|
"Couldn't obtain position: position query failed");
|
|
gst_object_unref (sink);
|
|
return FALSE;
|
|
}
|
|
gst_object_unref (sink);
|
|
} else if (pad) {
|
|
GstEvent *event;
|
|
const GstSegment *segment;
|
|
|
|
event = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0);
|
|
if (!event) {
|
|
GST_WARNING_OBJECT (stream, "Couldn't obtain position: no segment event");
|
|
gst_object_unref (pad);
|
|
return FALSE;
|
|
}
|
|
|
|
gst_event_parse_segment (event, &segment);
|
|
if (segment->format != GST_FORMAT_TIME) {
|
|
*position = -1;
|
|
} else {
|
|
g_mutex_lock (&priv->lock);
|
|
*position = priv->position;
|
|
g_mutex_unlock (&priv->lock);
|
|
*position =
|
|
gst_segment_to_stream_time (segment, GST_FORMAT_TIME, *position);
|
|
}
|
|
gst_event_unref (event);
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_query_stop:
|
|
* @stream: a #GstRTSPStream
|
|
* @stop: (out): current stop of a #GstRTSPStream
|
|
*
|
|
* Query the stop of the stream in %GST_FORMAT_TIME. This only considers
|
|
* the RTP parts of the pipeline and not the RTCP parts.
|
|
*
|
|
* Returns: %TRUE if the stop could be queried
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstElement *sink;
|
|
GstPad *pad = NULL;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
/* query stop position: if no sinks have been added yet,
|
|
* we obtain the stop position from the pad otherwise we query the sinks */
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
/* depending on the transport type, it should query corresponding sink */
|
|
if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP)
|
|
sink = priv->udpsink[0];
|
|
else if (priv->configured_protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
|
|
sink = priv->mcast_udpsink[0];
|
|
else
|
|
sink = priv->appsink[0];
|
|
|
|
if (sink) {
|
|
gst_object_ref (sink);
|
|
} else if (priv->send_src[0]) {
|
|
pad = gst_object_ref (priv->send_src[0]);
|
|
} else {
|
|
g_mutex_unlock (&priv->lock);
|
|
GST_WARNING_OBJECT (stream, "Couldn't obtain stop: erroneous pipeline");
|
|
return FALSE;
|
|
}
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (sink) {
|
|
GstQuery *query;
|
|
GstFormat format;
|
|
gdouble rate;
|
|
gint64 start_value;
|
|
gint64 stop_value;
|
|
|
|
query = gst_query_new_segment (GST_FORMAT_TIME);
|
|
if (!gst_element_query (sink, query)) {
|
|
GST_WARNING_OBJECT (stream, "Couldn't obtain stop: element query failed");
|
|
gst_query_unref (query);
|
|
gst_object_unref (sink);
|
|
return FALSE;
|
|
}
|
|
gst_query_parse_segment (query, &rate, &format, &start_value, &stop_value);
|
|
if (format != GST_FORMAT_TIME)
|
|
*stop = -1;
|
|
else
|
|
*stop = rate > 0.0 ? stop_value : start_value;
|
|
gst_query_unref (query);
|
|
gst_object_unref (sink);
|
|
} else if (pad) {
|
|
GstEvent *event;
|
|
const GstSegment *segment;
|
|
|
|
event = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0);
|
|
if (!event) {
|
|
GST_WARNING_OBJECT (stream, "Couldn't obtain stop: no segment event");
|
|
gst_object_unref (pad);
|
|
return FALSE;
|
|
}
|
|
gst_event_parse_segment (event, &segment);
|
|
if (segment->format != GST_FORMAT_TIME) {
|
|
*stop = -1;
|
|
} else {
|
|
*stop = segment->stop;
|
|
if (*stop == -1)
|
|
*stop = segment->duration;
|
|
else
|
|
*stop = gst_segment_to_stream_time (segment, GST_FORMAT_TIME, *stop);
|
|
}
|
|
gst_event_unref (event);
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_seekable:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Checks whether the individual @stream is seekable.
|
|
*
|
|
* Returns: %TRUE if @stream is seekable, else %FALSE.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_seekable (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
GstPad *pad = NULL;
|
|
GstQuery *query = NULL;
|
|
gboolean seekable = FALSE;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
/* query stop position: if no sinks have been added yet,
|
|
* we obtain the stop position from the pad otherwise we query the sinks */
|
|
|
|
priv = stream->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
/* depending on the transport type, it should query corresponding sink */
|
|
if (priv->srcpad) {
|
|
pad = gst_object_ref (priv->srcpad);
|
|
} else {
|
|
g_mutex_unlock (&priv->lock);
|
|
GST_WARNING_OBJECT (stream, "Pad not available, can't query seekability");
|
|
goto beach;
|
|
}
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
query = gst_query_new_seeking (GST_FORMAT_TIME);
|
|
if (!gst_pad_query (pad, query)) {
|
|
GST_WARNING_OBJECT (stream, "seeking query failed");
|
|
goto beach;
|
|
}
|
|
gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL);
|
|
|
|
beach:
|
|
if (pad)
|
|
gst_object_unref (pad);
|
|
if (query)
|
|
gst_query_unref (query);
|
|
|
|
GST_DEBUG_OBJECT (stream, "Returning %d", seekable);
|
|
|
|
return seekable;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_complete_stream:
|
|
* @stream: a #GstRTSPStream
|
|
* @transport: a #GstRTSPTransport
|
|
*
|
|
* Add a receiver and sender part to the pipeline based on the transport from
|
|
* SETUP.
|
|
*
|
|
* Returns: %TRUE if the stream has been successfully updated.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_complete_stream (GstRTSPStream * stream,
|
|
const GstRTSPTransport * transport)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
priv = stream->priv;
|
|
GST_DEBUG_OBJECT (stream, "complete stream");
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
if (!(priv->allowed_protocols & transport->lower_transport))
|
|
goto unallowed_transport;
|
|
|
|
if (!create_receiver_part (stream, transport))
|
|
goto create_receiver_error;
|
|
|
|
/* in the RECORD case, we only add RTCP sender part */
|
|
if (!create_sender_part (stream, transport))
|
|
goto create_sender_error;
|
|
|
|
priv->configured_protocols |= transport->lower_transport;
|
|
|
|
priv->is_complete = TRUE;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
GST_DEBUG_OBJECT (stream, "pipeline successfully updated");
|
|
return TRUE;
|
|
|
|
create_receiver_error:
|
|
create_sender_error:
|
|
unallowed_transport:
|
|
{
|
|
g_mutex_unlock (&priv->lock);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_is_complete:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Checks whether the stream is complete, contains the receiver and the sender
|
|
* parts. As the stream contains sink(s) element(s), it's possible to perform
|
|
* seek operations on it.
|
|
*
|
|
* Returns: %TRUE if the stream contains at least one sink element.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_is_complete (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gboolean ret = FALSE;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
ret = priv->is_complete;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_is_sender:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Checks whether the stream is a sender.
|
|
*
|
|
* Returns: %TRUE if the stream is a sender and %FALSE otherwise.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_is_sender (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gboolean ret = FALSE;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
ret = (priv->srcpad != NULL);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_is_receiver:
|
|
* @stream: a #GstRTSPStream
|
|
*
|
|
* Checks whether the stream is a receiver.
|
|
*
|
|
* Returns: %TRUE if the stream is a receiver and %FALSE otherwise.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_is_receiver (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
gboolean ret = FALSE;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
|
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
ret = (priv->sinkpad != NULL);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return ret;
|
|
}
|
|
|
|
#define AES_128_KEY_LEN 16
|
|
#define AES_256_KEY_LEN 32
|
|
|
|
#define HMAC_32_KEY_LEN 4
|
|
#define HMAC_80_KEY_LEN 10
|
|
|
|
static gboolean
|
|
mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
|
|
{
|
|
const gchar *srtp_cipher;
|
|
const gchar *srtp_auth;
|
|
const GstMIKEYPayload *sp;
|
|
guint i;
|
|
|
|
/* loop over Security policy until we find one containing policy */
|
|
for (i = 0;; i++) {
|
|
if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
|
|
break;
|
|
|
|
if (((GstMIKEYPayloadSP *) sp)->policy == policy)
|
|
break;
|
|
}
|
|
|
|
/* the default ciphers */
|
|
srtp_cipher = "aes-128-icm";
|
|
srtp_auth = "hmac-sha1-80";
|
|
|
|
/* now override the defaults with what is in the Security Policy */
|
|
if (sp != NULL) {
|
|
guint len;
|
|
guint enc_alg = GST_MIKEY_ENC_AES_CM_128;
|
|
|
|
/* collect all the params and go over them */
|
|
len = gst_mikey_payload_sp_get_n_params (sp);
|
|
for (i = 0; i < len; i++) {
|
|
const GstMIKEYPayloadSPParam *param =
|
|
gst_mikey_payload_sp_get_param (sp, i);
|
|
|
|
switch (param->type) {
|
|
case GST_MIKEY_SP_SRTP_ENC_ALG:
|
|
enc_alg = param->val[0];
|
|
switch (param->val[0]) {
|
|
case GST_MIKEY_ENC_NULL:
|
|
srtp_cipher = "null";
|
|
break;
|
|
case GST_MIKEY_ENC_AES_CM_128:
|
|
case GST_MIKEY_ENC_AES_KW_128:
|
|
srtp_cipher = "aes-128-icm";
|
|
break;
|
|
case GST_MIKEY_ENC_AES_GCM_128:
|
|
srtp_cipher = "aes-128-gcm";
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
|
|
switch (param->val[0]) {
|
|
case AES_128_KEY_LEN:
|
|
if (enc_alg == GST_MIKEY_ENC_AES_CM_128 ||
|
|
enc_alg == GST_MIKEY_ENC_AES_KW_128) {
|
|
srtp_cipher = "aes-128-icm";
|
|
} else if (enc_alg == GST_MIKEY_ENC_AES_GCM_128) {
|
|
srtp_cipher = "aes-128-gcm";
|
|
}
|
|
break;
|
|
case AES_256_KEY_LEN:
|
|
if (enc_alg == GST_MIKEY_ENC_AES_CM_128 ||
|
|
enc_alg == GST_MIKEY_ENC_AES_KW_128) {
|
|
srtp_cipher = "aes-256-icm";
|
|
} else if (enc_alg == GST_MIKEY_ENC_AES_GCM_128) {
|
|
srtp_cipher = "aes-256-gcm";
|
|
}
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
case GST_MIKEY_SP_SRTP_AUTH_ALG:
|
|
switch (param->val[0]) {
|
|
case GST_MIKEY_MAC_NULL:
|
|
srtp_auth = "null";
|
|
break;
|
|
case GST_MIKEY_MAC_HMAC_SHA_1_160:
|
|
srtp_auth = "hmac-sha1-80";
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
|
|
switch (param->val[0]) {
|
|
case HMAC_32_KEY_LEN:
|
|
srtp_auth = "hmac-sha1-32";
|
|
break;
|
|
case HMAC_80_KEY_LEN:
|
|
srtp_auth = "hmac-sha1-80";
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
case GST_MIKEY_SP_SRTP_SRTP_ENC:
|
|
break;
|
|
case GST_MIKEY_SP_SRTP_SRTCP_ENC:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
/* now configure the SRTP parameters */
|
|
gst_caps_set_simple (caps,
|
|
"srtp-cipher", G_TYPE_STRING, srtp_cipher,
|
|
"srtp-auth", G_TYPE_STRING, srtp_auth,
|
|
"srtcp-cipher", G_TYPE_STRING, srtp_cipher,
|
|
"srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
handle_mikey_data (GstRTSPStream * stream, guint8 * data, gsize size)
|
|
{
|
|
GstMIKEYMessage *msg;
|
|
guint i, n_cs;
|
|
GstCaps *caps = NULL;
|
|
GstMIKEYPayloadKEMAC *kemac;
|
|
const GstMIKEYPayloadKeyData *pkd;
|
|
GstBuffer *key;
|
|
|
|
/* the MIKEY message contains a CSB or crypto session bundle. It is a
|
|
* set of Crypto Sessions protected with the same master key.
|
|
* In the context of SRTP, an RTP and its RTCP stream is part of a
|
|
* crypto session */
|
|
if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
|
|
goto parse_failed;
|
|
|
|
/* we can only handle SRTP crypto sessions for now */
|
|
if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
|
|
goto invalid_map_type;
|
|
|
|
/* get the number of crypto sessions. This maps SSRC to its
|
|
* security parameters */
|
|
n_cs = gst_mikey_message_get_n_cs (msg);
|
|
if (n_cs == 0)
|
|
goto no_crypto_sessions;
|
|
|
|
/* we also need keys */
|
|
if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
|
|
(msg, GST_MIKEY_PT_KEMAC, 0)))
|
|
goto no_keys;
|
|
|
|
/* we don't support encrypted keys */
|
|
if (kemac->enc_alg != GST_MIKEY_ENC_NULL
|
|
|| kemac->mac_alg != GST_MIKEY_MAC_NULL)
|
|
goto unsupported_encryption;
|
|
|
|
/* get Key data sub-payload */
|
|
pkd = (const GstMIKEYPayloadKeyData *)
|
|
gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
|
|
|
|
key = gst_buffer_new_memdup (pkd->key_data, pkd->key_len);
|
|
|
|
/* go over all crypto sessions and create the security policy for each
|
|
* SSRC */
|
|
for (i = 0; i < n_cs; i++) {
|
|
const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
|
|
|
|
caps = gst_caps_new_simple ("application/x-srtp",
|
|
"ssrc", G_TYPE_UINT, map->ssrc,
|
|
"roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
|
|
mikey_apply_policy (caps, msg, map->policy);
|
|
|
|
gst_rtsp_stream_update_crypto (stream, map->ssrc, caps);
|
|
gst_caps_unref (caps);
|
|
}
|
|
gst_mikey_message_unref (msg);
|
|
gst_buffer_unref (key);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
parse_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (stream, "failed to parse MIKEY message");
|
|
return FALSE;
|
|
}
|
|
invalid_map_type:
|
|
{
|
|
GST_DEBUG_OBJECT (stream, "invalid map type %d", msg->map_type);
|
|
goto cleanup_message;
|
|
}
|
|
no_crypto_sessions:
|
|
{
|
|
GST_DEBUG_OBJECT (stream, "no crypto sessions");
|
|
goto cleanup_message;
|
|
}
|
|
no_keys:
|
|
{
|
|
GST_DEBUG_OBJECT (stream, "no keys found");
|
|
goto cleanup_message;
|
|
}
|
|
unsupported_encryption:
|
|
{
|
|
GST_DEBUG_OBJECT (stream, "unsupported key encryption");
|
|
goto cleanup_message;
|
|
}
|
|
cleanup_message:
|
|
{
|
|
gst_mikey_message_unref (msg);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
#define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
|
|
|
|
static void
|
|
strip_chars (gchar * str)
|
|
{
|
|
gchar *s;
|
|
gsize len;
|
|
|
|
len = strlen (str);
|
|
while (len--) {
|
|
if (!IS_STRIP_CHAR (str[len]))
|
|
break;
|
|
str[len] = '\0';
|
|
}
|
|
for (s = str; *s && IS_STRIP_CHAR (*s); s++);
|
|
memmove (str, s, len + 1);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_handle_keymgmt:
|
|
* @stream: a #GstRTSPStream
|
|
* @keymgmt: a keymgmt header
|
|
*
|
|
* Parse and handle a KeyMgmt header.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
/* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
|
|
* key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_handle_keymgmt (GstRTSPStream * stream, const gchar * keymgmt)
|
|
{
|
|
gchar **specs;
|
|
gint i, j;
|
|
|
|
specs = g_strsplit (keymgmt, ",", 0);
|
|
for (i = 0; specs[i]; i++) {
|
|
gchar **split;
|
|
|
|
split = g_strsplit (specs[i], ";", 0);
|
|
for (j = 0; split[j]; j++) {
|
|
g_strstrip (split[j]);
|
|
if (g_str_has_prefix (split[j], "prot=")) {
|
|
g_strstrip (split[j] + 5);
|
|
if (!g_str_equal (split[j] + 5, "mikey"))
|
|
break;
|
|
GST_DEBUG ("found mikey");
|
|
} else if (g_str_has_prefix (split[j], "uri=")) {
|
|
strip_chars (split[j] + 4);
|
|
GST_DEBUG ("found uri '%s'", split[j] + 4);
|
|
} else if (g_str_has_prefix (split[j], "data=")) {
|
|
guchar *data;
|
|
gsize size;
|
|
strip_chars (split[j] + 5);
|
|
GST_DEBUG ("found data '%s'", split[j] + 5);
|
|
data = g_base64_decode_inplace (split[j] + 5, &size);
|
|
handle_mikey_data (stream, data, size);
|
|
}
|
|
}
|
|
g_strfreev (split);
|
|
}
|
|
g_strfreev (specs);
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_ulpfec_pt:
|
|
*
|
|
* Returns: the payload type used for ULPFEC protection packets
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
guint
|
|
gst_rtsp_stream_get_ulpfec_pt (GstRTSPStream * stream)
|
|
{
|
|
guint res;
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
res = stream->priv->ulpfec_pt;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_ulpfec_pt:
|
|
*
|
|
* Set the payload type to be used for ULPFEC protection packets
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_ulpfec_pt (GstRTSPStream * stream, guint pt)
|
|
{
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
stream->priv->ulpfec_pt = pt;
|
|
if (stream->priv->ulpfec_encoder) {
|
|
g_object_set (stream->priv->ulpfec_encoder, "pt", pt, NULL);
|
|
}
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_request_ulpfec_decoder:
|
|
*
|
|
* Creating a rtpulpfecdec element
|
|
*
|
|
* Returns: (transfer full) (nullable): a #GstElement.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
GstElement *
|
|
gst_rtsp_stream_request_ulpfec_decoder (GstRTSPStream * stream,
|
|
GstElement * rtpbin, guint sessid)
|
|
{
|
|
GObject *internal_storage = NULL;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
stream->priv->ulpfec_decoder =
|
|
gst_object_ref (gst_element_factory_make ("rtpulpfecdec", NULL));
|
|
|
|
g_signal_emit_by_name (G_OBJECT (rtpbin), "get-internal-storage", sessid,
|
|
&internal_storage);
|
|
g_object_set (stream->priv->ulpfec_decoder, "storage", internal_storage,
|
|
NULL);
|
|
g_object_unref (internal_storage);
|
|
update_ulpfec_decoder_pt (stream);
|
|
|
|
return stream->priv->ulpfec_decoder;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_request_ulpfec_encoder:
|
|
*
|
|
* Creating a rtpulpfecenc element
|
|
*
|
|
* Returns: (transfer full) (nullable): a #GstElement.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
GstElement *
|
|
gst_rtsp_stream_request_ulpfec_encoder (GstRTSPStream * stream, guint sessid)
|
|
{
|
|
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
|
|
|
|
if (!stream->priv->ulpfec_percentage)
|
|
return NULL;
|
|
|
|
stream->priv->ulpfec_encoder =
|
|
gst_object_ref (gst_element_factory_make ("rtpulpfecenc", NULL));
|
|
|
|
g_object_set (stream->priv->ulpfec_encoder, "pt", stream->priv->ulpfec_pt,
|
|
"percentage", stream->priv->ulpfec_percentage, NULL);
|
|
|
|
return stream->priv->ulpfec_encoder;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_ulpfec_percentage:
|
|
*
|
|
* Sets the amount of redundancy to apply when creating ULPFEC
|
|
* protection packets.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_ulpfec_percentage (GstRTSPStream * stream, guint percentage)
|
|
{
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
stream->priv->ulpfec_percentage = percentage;
|
|
if (stream->priv->ulpfec_encoder) {
|
|
g_object_set (stream->priv->ulpfec_encoder, "percentage", percentage, NULL);
|
|
}
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_ulpfec_percentage:
|
|
*
|
|
* Returns: the amount of redundancy applied when creating ULPFEC
|
|
* protection packets.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
guint
|
|
gst_rtsp_stream_get_ulpfec_percentage (GstRTSPStream * stream)
|
|
{
|
|
guint res;
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
res = stream->priv->ulpfec_percentage;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_set_rate_control:
|
|
*
|
|
* Define whether @stream will follow the Rate-Control=no behaviour as specified
|
|
* in the ONVIF replay spec.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
void
|
|
gst_rtsp_stream_set_rate_control (GstRTSPStream * stream, gboolean enabled)
|
|
{
|
|
GST_DEBUG_OBJECT (stream, "%s rate control",
|
|
enabled ? "Enabling" : "Disabling");
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
stream->priv->do_rate_control = enabled;
|
|
if (stream->priv->appsink[0])
|
|
g_object_set (stream->priv->appsink[0], "sync", enabled, NULL);
|
|
if (stream->priv->payloader
|
|
&& g_object_class_find_property (G_OBJECT_GET_CLASS (stream->
|
|
priv->payloader), "onvif-no-rate-control"))
|
|
g_object_set (stream->priv->payloader, "onvif-no-rate-control", !enabled,
|
|
NULL);
|
|
if (stream->priv->session) {
|
|
g_object_set (stream->priv->session, "disable-sr-timestamp", !enabled,
|
|
NULL);
|
|
}
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_get_rate_control:
|
|
*
|
|
* Returns: whether @stream will follow the Rate-Control=no behaviour as specified
|
|
* in the ONVIF replay spec.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
gboolean
|
|
gst_rtsp_stream_get_rate_control (GstRTSPStream * stream)
|
|
{
|
|
gboolean ret;
|
|
|
|
g_mutex_lock (&stream->priv->lock);
|
|
ret = stream->priv->do_rate_control;
|
|
g_mutex_unlock (&stream->priv->lock);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_stream_unblock_rtcp:
|
|
*
|
|
* Remove blocking probe from the RTCP source. When creating an UDP source for
|
|
* RTCP it is initially blocked until this function is called.
|
|
* This functions should be called once the pipeline is ready for handling RTCP
|
|
* packets.
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
void
|
|
gst_rtsp_stream_unblock_rtcp (GstRTSPStream * stream)
|
|
{
|
|
GstRTSPStreamPrivate *priv;
|
|
|
|
priv = stream->priv;
|
|
g_mutex_lock (&priv->lock);
|
|
if (priv->block_early_rtcp_probe != 0) {
|
|
gst_pad_remove_probe
|
|
(priv->block_early_rtcp_pad, priv->block_early_rtcp_probe);
|
|
priv->block_early_rtcp_probe = 0;
|
|
gst_object_unref (priv->block_early_rtcp_pad);
|
|
priv->block_early_rtcp_pad = NULL;
|
|
}
|
|
if (priv->block_early_rtcp_probe_ipv6 != 0) {
|
|
gst_pad_remove_probe
|
|
(priv->block_early_rtcp_pad_ipv6, priv->block_early_rtcp_probe_ipv6);
|
|
priv->block_early_rtcp_probe_ipv6 = 0;
|
|
gst_object_unref (priv->block_early_rtcp_pad_ipv6);
|
|
priv->block_early_rtcp_pad_ipv6 = NULL;
|
|
}
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|