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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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66df66daa2
Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize), (gst_rtspsrc_create_stream), (rtsp_auth_method_to_string), (gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth), (gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_uri_set_uri): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (append_auth_header), (rtsp_connection_send), (rtsp_connection_free), (rtsp_connection_set_auth): * gst/rtsp/rtspconnection.h: * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri): * gst/rtsp/rtspurl.h: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work on hosts that require authentication.
2761 lines
75 KiB
C
2761 lines
75 KiB
C
/* GStreamer
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* Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
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* <2006> Lutz Mueller <lutz at topfrose dot de>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/*
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* Unless otherwise indicated, Source Code is licensed under MIT license.
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* See further explanation attached in License Statement (distributed in the file
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* LICENSE).
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
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* of the Software, and to permit persons to whom the Software is furnished to do
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* so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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/**
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* SECTION:element-rtspsrc
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*
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* <refsect2>
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* <para>
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* Makes a connection to an RTSP server and read the data.
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* rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
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* RealMedia/Quicktime/Microsoft extensions.
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* </para>
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* <para>
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* RTSP supports transport over TCP or UDP in unicast or multicast mode. By
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* default rtspsrc will negotiate a connection in the following order:
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* UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
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* protocols can be controlled with the "protocols" property.
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* </para>
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* <para>
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* rtspsrc currently understands SDP as the format of the session description.
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* For each stream listed in the SDP a new rtp_stream%d pad will be created
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* with caps derived from the SDP media description. This is a caps of mime type
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* "application/x-rtp" that can be connected to any available RTP depayloader
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* element.
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* </para>
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* <para>
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* rtspsrc will internally instantiate an RTP session manager element
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* that will handle the RTCP messages to and from the server, jitter removal,
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* packet reordering along with providing a clock for the pipeline.
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* This feature is however currently not yet implemented.
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* </para>
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* <para>
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* rtspsrc acts like a live source and will therefore only generate data in the
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* PLAYING state.
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* </para>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
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* </programlisting>
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* Establish a connection to an RTSP server and send the raw RTP packets to a fakesink.
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2006-08-18 (0.10.5)
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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 <unistd.h>
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#include <stdlib.h>
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#include <string.h>
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#include "gstrtspsrc.h"
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#include "sdp.h"
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/* define for experimental real support */
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#undef WITH_EXT_REAL
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#include "rtspextwms.h"
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#ifdef WITH_EXT_REAL
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#include "rtspextreal.h"
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#endif
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GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
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#define GST_CAT_DEFAULT (rtspsrc_debug)
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/* elementfactory information */
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static const GstElementDetails gst_rtspsrc_details =
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GST_ELEMENT_DETAILS ("RTSP packet receiver",
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"Source/Network",
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"Receive data over the network via RTSP (RFC 2326)",
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"Wim Taymans <wim@fluendo.com>\n"
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"Thijs Vermeir <thijs.vermeir@barco.com>\n"
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"Lutz Mueller <lutz@topfrose.de>");
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static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream%d",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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#define DEFAULT_LOCATION NULL
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#define DEFAULT_PROTOCOLS RTSP_LOWER_TRANS_UDP | RTSP_LOWER_TRANS_UDP_MCAST | RTSP_LOWER_TRANS_TCP
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#define DEFAULT_DEBUG FALSE
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#define DEFAULT_RETRY 20
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#define DEFAULT_TIMEOUT 5000000
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enum
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{
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PROP_0,
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PROP_LOCATION,
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PROP_PROTOCOLS,
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PROP_DEBUG,
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PROP_RETRY,
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PROP_TIMEOUT,
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};
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#define GST_TYPE_RTSP_LOWER_TRANS (gst_rtsp_lower_trans_get_type())
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static GType
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gst_rtsp_lower_trans_get_type (void)
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{
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static GType rtsp_lower_trans_type = 0;
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static const GFlagsValue rtsp_lower_trans[] = {
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{RTSP_LOWER_TRANS_UDP, "UDP Unicast Mode", "udp-unicast"},
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{RTSP_LOWER_TRANS_UDP_MCAST, "UDP Multicast Mode", "udp-multicast"},
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{RTSP_LOWER_TRANS_TCP, "TCP interleaved mode", "tcp"},
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{0, NULL, NULL},
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};
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if (!rtsp_lower_trans_type) {
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rtsp_lower_trans_type =
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g_flags_register_static ("GstRTSPLowerTrans", rtsp_lower_trans);
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}
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return rtsp_lower_trans_type;
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}
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static void gst_rtspsrc_base_init (gpointer g_class);
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static void gst_rtspsrc_finalize (GObject * object);
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static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
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gpointer iface_data);
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static GstCaps *gst_rtspsrc_media_to_caps (gint pt, SDPMedia * media);
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static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
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static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
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RTSPMessage * response);
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static gboolean gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
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RTSPMessage * response, RTSPStatusCode * code);
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static gboolean gst_rtspsrc_open (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_play (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_pause (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_close (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
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const gchar * uri);
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static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
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static void gst_rtspsrc_loop (GstRTSPSrc * src);
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/* commands we send to out loop to notify it of events */
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#define CMD_WAIT 0
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#define CMD_RECONNECT 1
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#define CMD_STOP 2
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/*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
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static void
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_do_init (GType rtspsrc_type)
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{
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static const GInterfaceInfo urihandler_info = {
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gst_rtspsrc_uri_handler_init,
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NULL,
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NULL
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};
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GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
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g_type_add_interface_static (rtspsrc_type, GST_TYPE_URI_HANDLER,
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&urihandler_info);
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}
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GST_BOILERPLATE_FULL (GstRTSPSrc, gst_rtspsrc, GstBin, GST_TYPE_BIN, _do_init);
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static void
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gst_rtspsrc_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtptemplate));
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gst_element_class_set_details (element_class, &gst_rtspsrc_details);
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}
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static void
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gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBinClass *gstbin_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbin_class = (GstBinClass *) klass;
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gobject_class->set_property = gst_rtspsrc_set_property;
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gobject_class->get_property = gst_rtspsrc_get_property;
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gobject_class->finalize = gst_rtspsrc_finalize;
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g_object_class_install_property (gobject_class, PROP_LOCATION,
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g_param_spec_string ("location", "RTSP Location",
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"Location of the RTSP url to read",
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DEFAULT_LOCATION, G_PARAM_READWRITE));
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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_CONSTRUCT));
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g_object_class_install_property (gobject_class, PROP_DEBUG,
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g_param_spec_boolean ("debug", "Debug",
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"Dump request and response messages to stdout",
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DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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g_object_class_install_property (gobject_class, PROP_RETRY,
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g_param_spec_uint ("retry", "Retry",
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"Max number of retries when allocating RTP ports.",
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0, G_MAXUINT16, DEFAULT_RETRY,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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g_object_class_install_property (gobject_class, PROP_TIMEOUT,
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g_param_spec_uint64 ("timeout", "Timeout",
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"Retry TCP transport after timeout microseconds (0 = disabled)",
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0, G_MAXUINT64, DEFAULT_TIMEOUT,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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gstelement_class->change_state = gst_rtspsrc_change_state;
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gstbin_class->handle_message = gst_rtspsrc_handle_message;
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}
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static void
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gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
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{
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src->stream_rec_lock = g_new (GStaticRecMutex, 1);
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g_static_rec_mutex_init (src->stream_rec_lock);
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src->loop_cond = g_cond_new ();
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src->location = g_strdup (DEFAULT_LOCATION);
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src->url = NULL;
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/* install WMS extension by default */
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src->extension = rtsp_ext_wms_get_context ();
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#ifdef WITH_EXT_REAL
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src->extension = rtsp_ext_real_get_context ();
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#endif
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src->extension->src = (gpointer) src;
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}
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static void
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gst_rtspsrc_finalize (GObject * object)
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{
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GstRTSPSrc *rtspsrc;
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rtspsrc = GST_RTSPSRC (object);
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g_static_rec_mutex_free (rtspsrc->stream_rec_lock);
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g_free (rtspsrc->stream_rec_lock);
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g_cond_free (rtspsrc->loop_cond);
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g_free (rtspsrc->location);
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g_free (rtspsrc->req_location);
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g_free (rtspsrc->content_base);
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rtsp_url_free (rtspsrc->url);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
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GParamSpec * pspec)
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{
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GstRTSPSrc *rtspsrc;
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rtspsrc = GST_RTSPSRC (object);
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switch (prop_id) {
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case PROP_LOCATION:
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gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
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g_value_get_string (value));
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break;
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case PROP_PROTOCOLS:
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rtspsrc->protocols = g_value_get_flags (value);
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break;
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case PROP_DEBUG:
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rtspsrc->debug = g_value_get_boolean (value);
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break;
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case PROP_RETRY:
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rtspsrc->retry = g_value_get_uint (value);
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break;
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case PROP_TIMEOUT:
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rtspsrc->timeout = g_value_get_uint64 (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstRTSPSrc *rtspsrc;
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rtspsrc = GST_RTSPSRC (object);
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switch (prop_id) {
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case PROP_LOCATION:
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g_value_set_string (value, rtspsrc->location);
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break;
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case PROP_PROTOCOLS:
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g_value_set_flags (value, rtspsrc->protocols);
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break;
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case PROP_DEBUG:
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g_value_set_boolean (value, rtspsrc->debug);
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break;
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case PROP_RETRY:
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g_value_set_uint (value, rtspsrc->retry);
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break;
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case PROP_TIMEOUT:
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g_value_set_uint64 (value, rtspsrc->timeout);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static gint
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find_stream_by_pt (GstRTSPStream * stream, gconstpointer a)
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{
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gint pt = GPOINTER_TO_INT (a);
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if (stream->pt == pt)
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return 0;
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return -1;
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}
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static GstRTSPStream *
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gst_rtspsrc_create_stream (GstRTSPSrc * src, SDPMessage * sdp, gint idx)
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{
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GstRTSPStream *stream;
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gchar *control_url;
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gchar *payload;
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SDPMedia *media;
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/* get media, should not return NULL */
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media = sdp_message_get_media (sdp, idx);
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if (media == NULL)
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return NULL;
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stream = g_new0 (GstRTSPStream, 1);
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stream->parent = src;
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/* we mark the pad as not linked, we will mark it as OK when we add the pad to
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* the element. */
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stream->last_ret = GST_FLOW_NOT_LINKED;
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stream->added = FALSE;
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stream->id = src->numstreams++;
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/* we must have a payload. No payload means we cannot create caps */
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/* FIXME, handle multiple formats. */
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if ((payload = sdp_media_get_format (media, 0))) {
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stream->pt = atoi (payload);
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/* convert caps */
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stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
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if (stream->pt >= 96) {
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/* If we have a dynamic payload type, see if we have a stream with the
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* same payload number. If there is one, they are part of the same
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* container and we only need to add one pad. */
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if (g_list_find_custom (src->streams, GINT_TO_POINTER (stream->pt),
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(GCompareFunc) find_stream_by_pt)) {
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stream->container = TRUE;
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}
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}
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}
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/* get control url to construct the setup url. The setup url is used to
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* configure the transport of the stream and is used to identity the stream in
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* the RTP-Info header field returned from PLAY. */
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control_url = sdp_media_get_attribute_val (media, "control");
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GST_DEBUG_OBJECT (src, "stream %d", stream->id);
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GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
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GST_DEBUG_OBJECT (src, " container: %d", stream->container);
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GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
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GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
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if (control_url != NULL) {
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/* If the control_url starts with a '/' or a non rtsp: protocol we will most
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* likely build a URL that the server will fail to understand, this is ok,
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* we will fail then. */
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if (g_str_has_prefix (control_url, "rtsp://"))
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stream->setup_url = g_strdup (control_url);
|
|
else if (src->content_base)
|
|
stream->setup_url =
|
|
g_strdup_printf ("%s%s", src->content_base, control_url);
|
|
else
|
|
stream->setup_url =
|
|
g_strdup_printf ("%s/%s", src->req_location, control_url);
|
|
}
|
|
GST_DEBUG_OBJECT (src, " setup: %s", GST_STR_NULL (stream->setup_url));
|
|
|
|
/* we keep track of all streams */
|
|
src->streams = g_list_append (src->streams, stream);
|
|
|
|
return stream;
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
|
|
{
|
|
gint i;
|
|
|
|
GST_DEBUG_OBJECT (src, "free stream %p", stream);
|
|
|
|
if (stream->caps)
|
|
gst_caps_unref (stream->caps);
|
|
|
|
g_free (stream->setup_url);
|
|
|
|
for (i = 0; i < 2; i++) {
|
|
GstElement *udpsrc = stream->udpsrc[i];
|
|
|
|
if (udpsrc) {
|
|
GstPad *pad;
|
|
|
|
/* unlink the pad */
|
|
pad = gst_element_get_pad (udpsrc, "src");
|
|
if (stream->channelpad[i]) {
|
|
gst_pad_unlink (pad, stream->channelpad[i]);
|
|
gst_object_unref (stream->channelpad[i]);
|
|
stream->channelpad[i] = NULL;
|
|
}
|
|
|
|
gst_element_set_state (udpsrc, GST_STATE_NULL);
|
|
gst_bin_remove (GST_BIN_CAST (src), udpsrc);
|
|
gst_object_unref (stream->udpsrc[i]);
|
|
stream->udpsrc[i] = NULL;
|
|
}
|
|
}
|
|
if (stream->sess) {
|
|
gst_element_set_state (stream->sess, GST_STATE_NULL);
|
|
gst_bin_remove (GST_BIN_CAST (src), stream->sess);
|
|
stream->sess = NULL;
|
|
}
|
|
if (stream->srcpad) {
|
|
gst_pad_set_active (stream->srcpad, FALSE);
|
|
if (stream->added) {
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
|
|
stream->added = FALSE;
|
|
}
|
|
stream->srcpad = NULL;
|
|
}
|
|
g_free (stream);
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_cleanup (GstRTSPSrc * src)
|
|
{
|
|
GList *walk;
|
|
|
|
GST_DEBUG_OBJECT (src, "cleanup");
|
|
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
|
|
gst_rtspsrc_stream_free (src, stream);
|
|
}
|
|
g_list_free (src->streams);
|
|
src->streams = NULL;
|
|
src->numstreams = 0;
|
|
if (src->props)
|
|
gst_structure_free (src->props);
|
|
src->props = NULL;
|
|
}
|
|
|
|
|
|
#define PARSE_INT(p, del, res) \
|
|
G_STMT_START { \
|
|
gchar *t = p; \
|
|
p = strstr (p, del); \
|
|
if (p == NULL) \
|
|
res = -1; \
|
|
else { \
|
|
*p = '\0'; \
|
|
p++; \
|
|
res = atoi (t); \
|
|
} \
|
|
} G_STMT_END
|
|
|
|
#define PARSE_STRING(p, del, res) \
|
|
G_STMT_START { \
|
|
gchar *t = p; \
|
|
p = strstr (p, del); \
|
|
if (p == NULL) \
|
|
res = NULL; \
|
|
else { \
|
|
*p = '\0'; \
|
|
p++; \
|
|
res = t; \
|
|
} \
|
|
} G_STMT_END
|
|
|
|
#define SKIP_SPACES(p) \
|
|
while (*p && g_ascii_isspace (*p)) \
|
|
p++;
|
|
|
|
/* rtpmap contains:
|
|
*
|
|
* <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
|
|
*/
|
|
static gboolean
|
|
gst_rtspsrc_parse_rtpmap (gchar * rtpmap, gint * payload, gchar ** name,
|
|
gint * rate, gchar ** params)
|
|
{
|
|
gchar *p, *t;
|
|
|
|
t = p = rtpmap;
|
|
|
|
PARSE_INT (p, " ", *payload);
|
|
if (*payload == -1)
|
|
return FALSE;
|
|
|
|
SKIP_SPACES (p);
|
|
if (*p == '\0')
|
|
return FALSE;
|
|
|
|
PARSE_STRING (p, "/", *name);
|
|
if (*name == NULL) {
|
|
/* no rate, assume 0 then */
|
|
*name = p;
|
|
*rate = -1;
|
|
return TRUE;
|
|
}
|
|
|
|
t = p;
|
|
p = strstr (p, "/");
|
|
if (p == NULL) {
|
|
*rate = atoi (t);
|
|
return TRUE;
|
|
}
|
|
*p = '\0';
|
|
p++;
|
|
*rate = atoi (t);
|
|
|
|
t = p;
|
|
if (*p == '\0')
|
|
return TRUE;
|
|
*params = t;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/*
|
|
* Mapping of caps to and from SDP fields:
|
|
*
|
|
* m=<media> <UDP port> RTP/AVP <payload>
|
|
* a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
|
|
* a=fmtp:<payload> <param>[=<value>];...
|
|
*/
|
|
static GstCaps *
|
|
gst_rtspsrc_media_to_caps (gint pt, SDPMedia * media)
|
|
{
|
|
GstCaps *caps;
|
|
gchar *rtpmap;
|
|
gchar *fmtp;
|
|
gchar *name = NULL;
|
|
gint rate = -1;
|
|
gchar *params = NULL;
|
|
gchar *tmp;
|
|
GstStructure *s;
|
|
|
|
/* dynamic payloads need rtpmap */
|
|
if (pt >= 96) {
|
|
gint payload = 0;
|
|
gboolean ret;
|
|
|
|
if ((rtpmap = sdp_media_get_attribute_val (media, "rtpmap"))) {
|
|
ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
|
|
if (ret) {
|
|
if (payload != pt) {
|
|
/* FIXME, not fatal? */
|
|
g_warning ("rtpmap of wrong payload type");
|
|
name = NULL;
|
|
rate = -1;
|
|
params = NULL;
|
|
}
|
|
} else {
|
|
/* FIXME, not fatal? */
|
|
g_warning ("error parsing rtpmap");
|
|
}
|
|
} else
|
|
goto no_rtpmap;
|
|
}
|
|
|
|
tmp = g_ascii_strdown (media->media, -1);
|
|
caps = gst_caps_new_simple ("application/x-unknown",
|
|
"media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
|
|
g_free (tmp);
|
|
s = gst_caps_get_structure (caps, 0);
|
|
|
|
if (rate != -1)
|
|
gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
|
|
|
|
/* encoding name must be upper case */
|
|
if (name != NULL) {
|
|
tmp = g_ascii_strup (name, -1);
|
|
gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
|
|
g_free (tmp);
|
|
}
|
|
|
|
/* params must be lower case */
|
|
if (params != NULL) {
|
|
tmp = g_ascii_strdown (params, -1);
|
|
gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
|
|
g_free (tmp);
|
|
}
|
|
|
|
/* parse optional fmtp: field */
|
|
if ((fmtp = sdp_media_get_attribute_val (media, "fmtp"))) {
|
|
gchar *p;
|
|
gint payload = 0;
|
|
|
|
p = fmtp;
|
|
|
|
/* p is now of the format <payload> <param>[=<value>];... */
|
|
PARSE_INT (p, " ", payload);
|
|
if (payload != -1 && payload == pt) {
|
|
gchar **pairs;
|
|
gint i;
|
|
|
|
/* <param>[=<value>] are separated with ';' */
|
|
pairs = g_strsplit (p, ";", 0);
|
|
for (i = 0; pairs[i]; i++) {
|
|
gchar *valpos;
|
|
gchar *val, *key;
|
|
|
|
/* the key may not have a '=', the value can have other '='s */
|
|
valpos = strstr (pairs[i], "=");
|
|
if (valpos) {
|
|
/* we have a '=' and thus a value, remove the '=' with \0 */
|
|
*valpos = '\0';
|
|
/* value is everything between '=' and ';'. FIXME, strip? */
|
|
val = g_strstrip (valpos + 1);
|
|
} else {
|
|
/* simple <param>;.. is translated into <param>=1;... */
|
|
val = "1";
|
|
}
|
|
/* strip the key of spaces, convert key to lowercase but not the value. */
|
|
key = g_strstrip (pairs[i]);
|
|
tmp = g_ascii_strdown (key, -1);
|
|
gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
|
|
g_free (tmp);
|
|
}
|
|
g_strfreev (pairs);
|
|
}
|
|
}
|
|
return caps;
|
|
|
|
/* ERRORS */
|
|
no_rtpmap:
|
|
{
|
|
g_warning ("rtpmap type not given for dynamic payload %d", pt);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
|
|
gint * rtpport, gint * rtcpport)
|
|
{
|
|
GstRTSPSrc *src;
|
|
GstStateChangeReturn ret;
|
|
GstElement *tmp, *udpsrc0, *udpsrc1;
|
|
gint tmp_rtp, tmp_rtcp;
|
|
guint count;
|
|
|
|
src = stream->parent;
|
|
|
|
tmp = NULL;
|
|
udpsrc0 = NULL;
|
|
udpsrc1 = NULL;
|
|
count = 0;
|
|
|
|
/* try to allocate 2 UDP ports, the RTP port should be an even
|
|
* number and the RTCP port should be the next (uneven) port */
|
|
again:
|
|
udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL);
|
|
if (udpsrc0 == NULL)
|
|
goto no_udp_protocol;
|
|
|
|
ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
goto start_udp_failure;
|
|
|
|
g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
|
|
GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
|
|
|
|
/* check if port is even */
|
|
if ((tmp_rtp & 0x01) != 0) {
|
|
/* port not even, close and allocate another */
|
|
count++;
|
|
if (count > src->retry)
|
|
goto no_ports;
|
|
|
|
GST_DEBUG_OBJECT (src, "RTP port not even, retry %d", count);
|
|
/* have to keep port allocated so we can get a new one */
|
|
if (tmp != NULL) {
|
|
GST_DEBUG_OBJECT (src, "free temp");
|
|
gst_element_set_state (tmp, GST_STATE_NULL);
|
|
gst_object_unref (tmp);
|
|
}
|
|
tmp = udpsrc0;
|
|
GST_DEBUG_OBJECT (src, "retry %d", count);
|
|
goto again;
|
|
}
|
|
/* free leftover temp element/port */
|
|
if (tmp) {
|
|
gst_element_set_state (tmp, GST_STATE_NULL);
|
|
gst_object_unref (tmp);
|
|
tmp = NULL;
|
|
}
|
|
|
|
/* allocate port+1 for RTCP now */
|
|
udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
|
|
if (udpsrc1 == NULL)
|
|
goto no_udp_rtcp_protocol;
|
|
|
|
/* set port */
|
|
tmp_rtcp = tmp_rtp + 1;
|
|
g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
|
|
|
|
GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
|
|
ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
|
|
/* FIXME, this could fail if the next port is not free, we
|
|
* should retry with another port then */
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
goto start_rtcp_failure;
|
|
|
|
/* all fine, do port check */
|
|
g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
|
|
g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
|
|
|
|
/* this should not happen */
|
|
if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
|
|
goto port_error;
|
|
|
|
/* we keep these elements, we configure all in configure_transport when the
|
|
* server told us to really use the UDP ports. */
|
|
stream->udpsrc[0] = gst_object_ref (udpsrc0);
|
|
stream->udpsrc[1] = gst_object_ref (udpsrc1);
|
|
|
|
/* they are ours now */
|
|
gst_object_sink (udpsrc0);
|
|
gst_object_sink (udpsrc1);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_udp_protocol:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "could not get UDP source");
|
|
goto cleanup;
|
|
}
|
|
start_udp_failure:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "could not start UDP source");
|
|
goto cleanup;
|
|
}
|
|
no_ports:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
|
|
count);
|
|
goto cleanup;
|
|
}
|
|
no_udp_rtcp_protocol:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
|
|
goto cleanup;
|
|
}
|
|
start_rtcp_failure:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "could not start UDP source for RTCP");
|
|
goto cleanup;
|
|
}
|
|
port_error:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
|
|
tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
|
|
goto cleanup;
|
|
}
|
|
cleanup:
|
|
{
|
|
if (tmp) {
|
|
gst_element_set_state (tmp, GST_STATE_NULL);
|
|
gst_object_unref (tmp);
|
|
}
|
|
if (udpsrc0) {
|
|
gst_element_set_state (udpsrc0, GST_STATE_NULL);
|
|
gst_object_unref (udpsrc0);
|
|
}
|
|
if (udpsrc1) {
|
|
gst_element_set_state (udpsrc1, GST_STATE_NULL);
|
|
gst_object_unref (udpsrc1);
|
|
}
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
pad_unblocked (GstPad * pad, gboolean blocked, GstRTSPSrc * src)
|
|
{
|
|
GST_DEBUG_OBJECT (src, "pad %s:%s unblocked", GST_DEBUG_PAD_NAME (pad));
|
|
}
|
|
|
|
static void
|
|
pad_blocked (GstPad * pad, gboolean blocked, GstRTSPSrc * src)
|
|
{
|
|
GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
|
|
GST_DEBUG_PAD_NAME (pad));
|
|
|
|
/* activate the streams */
|
|
GST_OBJECT_LOCK (src);
|
|
if (!src->need_activate)
|
|
goto was_ok;
|
|
|
|
src->need_activate = FALSE;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
gst_rtspsrc_activate_streams (src);
|
|
|
|
return;
|
|
|
|
was_ok:
|
|
{
|
|
GST_OBJECT_UNLOCK (src);
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* sets up all elements needed for streaming over the specified transport.
|
|
* Does not yet expose the element pads, this will be done when there is actuall
|
|
* dataflow detected, which might never happen when UDP is blocked in a
|
|
* firewall, for example.
|
|
*/
|
|
static gboolean
|
|
gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
|
|
RTSPTransport * transport)
|
|
{
|
|
GstRTSPSrc *src;
|
|
GstPad *outpad = NULL;
|
|
GstPadTemplate *template;
|
|
GstStateChangeReturn ret;
|
|
gchar *name;
|
|
GstStructure *s;
|
|
const gchar *mime, *manager;
|
|
RTSPResult res;
|
|
|
|
src = stream->parent;
|
|
|
|
GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
|
|
|
|
s = gst_caps_get_structure (stream->caps, 0);
|
|
|
|
/* get the proper mime type for this stream now */
|
|
if ((res = rtsp_transport_get_mime (transport->trans, &mime)) < 0)
|
|
goto no_mime;
|
|
if (!mime)
|
|
goto no_mime;
|
|
|
|
/* configure the final mime type */
|
|
GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
|
|
gst_structure_set_name (s, mime);
|
|
|
|
/* find a manager */
|
|
if ((res = rtsp_transport_get_manager (transport->trans, &manager)) < 0)
|
|
goto no_manager;
|
|
|
|
if (manager) {
|
|
GST_DEBUG_OBJECT (src, "using manager %s", manager);
|
|
/* FIXME, the session manager needs to be shared with all the streams */
|
|
if (!(stream->sess = gst_element_factory_make (manager, NULL)))
|
|
goto no_element;
|
|
|
|
/* we manage this element */
|
|
gst_bin_add (GST_BIN_CAST (src), stream->sess);
|
|
|
|
ret = gst_element_set_state (stream->sess, GST_STATE_PAUSED);
|
|
if (ret != GST_STATE_CHANGE_SUCCESS)
|
|
goto start_session_failure;
|
|
|
|
/* we stream directly to the manager, FIXME, pad names should not be
|
|
* hardcoded. */
|
|
stream->channelpad[0] = gst_element_get_pad (stream->sess, "sinkrtp");
|
|
stream->channelpad[1] = gst_element_get_pad (stream->sess, "sinkrtcp");
|
|
}
|
|
|
|
if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
|
|
gint i;
|
|
|
|
/* configure for interleaved delivery, nothing needs to be done
|
|
* here, the loop function will call the chain functions of the
|
|
* session manager. */
|
|
stream->channel[0] = transport->interleaved.min;
|
|
stream->channel[1] = transport->interleaved.max;
|
|
GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
|
|
stream->channel[0], stream->channel[1]);
|
|
|
|
/* we can remove the allocated UDP ports now */
|
|
for (i = 0; i < 2; i++) {
|
|
if (stream->udpsrc[i]) {
|
|
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
|
|
gst_object_unref (stream->udpsrc[i]);
|
|
stream->udpsrc[i] = NULL;
|
|
}
|
|
}
|
|
|
|
/* no session manager, send data to srcpad directly */
|
|
if (!stream->channelpad[0]) {
|
|
GST_DEBUG_OBJECT (src, "no manager, creating pad");
|
|
|
|
/* create a new pad we will use to stream to */
|
|
name = g_strdup_printf ("stream%d", stream->id);
|
|
template = gst_static_pad_template_get (&rtptemplate);
|
|
stream->channelpad[0] = gst_pad_new_from_template (template, name);
|
|
gst_object_unref (template);
|
|
g_free (name);
|
|
|
|
/* set caps and activate */
|
|
gst_pad_use_fixed_caps (stream->channelpad[0]);
|
|
gst_pad_set_caps (stream->channelpad[0], stream->caps);
|
|
gst_pad_set_active (stream->channelpad[0], TRUE);
|
|
|
|
outpad = gst_object_ref (stream->channelpad[0]);
|
|
} else {
|
|
GST_DEBUG_OBJECT (src, "using manager source pad");
|
|
outpad = gst_element_get_pad (stream->sess, "srcrtp");
|
|
}
|
|
} else {
|
|
/* multicast was selected, create UDP sources and join the multicast
|
|
* group. */
|
|
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST) {
|
|
gchar *uri;
|
|
|
|
GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
|
|
|
|
/* creating UDP source */
|
|
if (transport->port.min != -1) {
|
|
uri = g_strdup_printf ("udp://%s:%d", transport->destination,
|
|
transport->port.min);
|
|
stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
|
|
g_free (uri);
|
|
if (stream->udpsrc[0] == NULL)
|
|
goto no_element;
|
|
|
|
/* take ownership */
|
|
gst_object_ref (stream->udpsrc[0]);
|
|
gst_object_sink (stream->udpsrc[0]);
|
|
|
|
/* change state */
|
|
gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
|
|
}
|
|
|
|
/* creating another UDP source */
|
|
if (transport->port.max != -1) {
|
|
uri = g_strdup_printf ("udp://%s:%d", transport->destination,
|
|
transport->port.max);
|
|
stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
|
|
g_free (uri);
|
|
if (stream->udpsrc[1] == NULL)
|
|
goto no_element;
|
|
|
|
/* take ownership */
|
|
gst_object_ref (stream->udpsrc[0]);
|
|
gst_object_sink (stream->udpsrc[0]);
|
|
|
|
gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
|
|
}
|
|
}
|
|
|
|
/* we manage the UDP elements now. For unicast, the UDP sources where
|
|
* allocated in the stream when we suggested a transport. */
|
|
if (stream->udpsrc[0]) {
|
|
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
|
|
|
|
GST_DEBUG_OBJECT (src, "setting up UDP source");
|
|
|
|
/* set caps */
|
|
g_object_set (G_OBJECT (stream->udpsrc[0]), "caps", stream->caps, NULL);
|
|
|
|
/* configure a timeout on the UDP port. When the timeout message is
|
|
* posted, we assume UDP transport is not possible. We reconnect using TCP
|
|
* if we can. */
|
|
g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->timeout,
|
|
NULL);
|
|
|
|
/* get output pad of the UDP source. */
|
|
outpad = gst_element_get_pad (stream->udpsrc[0], "src");
|
|
|
|
/* save it so we can unblock */
|
|
stream->blockedpad = outpad;
|
|
|
|
/* configure pad block on the pad. As soon as there is dataflow on the
|
|
* UDP source, we know that UDP is not blocked by a firewall and we can
|
|
* configure all the streams to let the application autoplug decoders. */
|
|
gst_pad_set_blocked_async (outpad, TRUE,
|
|
(GstPadBlockCallback) pad_blocked, src);
|
|
|
|
if (stream->channelpad[0]) {
|
|
GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
|
|
/* configure for UDP delivery, we need to connect the UDP pads to
|
|
* the session plugin. */
|
|
gst_pad_link (outpad, stream->channelpad[0]);
|
|
gst_object_unref (outpad);
|
|
/* the real output pad is that of the session manager */
|
|
outpad = gst_element_get_pad (stream->sess, "srcrtp");
|
|
} else {
|
|
GST_DEBUG_OBJECT (src, "using UDP src pad as output");
|
|
}
|
|
}
|
|
|
|
if (stream->udpsrc[1]) {
|
|
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
|
|
|
|
if (stream->channelpad[1]) {
|
|
GstPad *pad;
|
|
|
|
GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
|
|
|
|
pad = gst_element_get_pad (stream->udpsrc[1], "src");
|
|
gst_pad_link (pad, stream->channelpad[1]);
|
|
gst_object_unref (pad);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (outpad) {
|
|
GST_DEBUG_OBJECT (src, "creating ghostpad");
|
|
|
|
gst_pad_use_fixed_caps (outpad);
|
|
gst_pad_set_caps (outpad, stream->caps);
|
|
|
|
/* create ghostpad, don't add just yet, this will be done when we activate
|
|
* the stream. */
|
|
name = g_strdup_printf ("stream%d", stream->id);
|
|
template = gst_static_pad_template_get (&rtptemplate);
|
|
stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
|
|
gst_object_unref (template);
|
|
g_free (name);
|
|
|
|
gst_object_unref (outpad);
|
|
}
|
|
/* mark pad as ok */
|
|
stream->last_ret = GST_FLOW_OK;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_mime:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "unknown transport");
|
|
return FALSE;
|
|
}
|
|
no_manager:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "cannot get a session manager");
|
|
return FALSE;
|
|
}
|
|
no_element:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no rtpdec element found");
|
|
return FALSE;
|
|
}
|
|
start_session_failure:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "could not start session");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* Adds the source pads of all configured streams to the element.
|
|
* This code is performed when we detected dataflow.
|
|
*
|
|
* We detect dataflow from either the _loop function or with pad probes on the
|
|
* udp sources.
|
|
*/
|
|
static gboolean
|
|
gst_rtspsrc_activate_streams (GstRTSPSrc * src)
|
|
{
|
|
GList *walk;
|
|
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
|
|
if (stream->srcpad) {
|
|
gst_pad_set_active (stream->srcpad, TRUE);
|
|
/* add the pad */
|
|
if (!stream->added) {
|
|
gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
|
|
stream->added = TRUE;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* if we got here all was configured. We have dynamic pads so we notify that
|
|
* we are done */
|
|
gst_element_no_more_pads (GST_ELEMENT_CAST (src));
|
|
|
|
/* unblock all pads */
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
|
|
if (stream->blockedpad) {
|
|
gst_pad_set_blocked_async (stream->blockedpad, FALSE,
|
|
(GstPadBlockCallback) pad_unblocked, src);
|
|
stream->blockedpad = NULL;
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gint
|
|
find_stream_by_channel (GstRTSPStream * stream, gconstpointer a)
|
|
{
|
|
gint channel = GPOINTER_TO_INT (a);
|
|
|
|
if (stream->channel[0] == channel || stream->channel[1] == channel)
|
|
return 0;
|
|
|
|
return -1;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
|
|
GstFlowReturn ret)
|
|
{
|
|
GList *streams;
|
|
|
|
/* store the value */
|
|
stream->last_ret = ret;
|
|
|
|
/* if it's success we can return the value right away */
|
|
if (GST_FLOW_IS_SUCCESS (ret))
|
|
goto done;
|
|
|
|
/* any other error that is not-linked can be returned right
|
|
* away */
|
|
if (ret != GST_FLOW_NOT_LINKED)
|
|
goto done;
|
|
|
|
/* only return NOT_LINKED if all other pads returned NOT_LINKED */
|
|
for (streams = src->streams; streams; streams = g_list_next (streams)) {
|
|
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
|
|
|
|
ret = ostream->last_ret;
|
|
/* some other return value (must be SUCCESS but we can return
|
|
* other values as well) */
|
|
if (ret != GST_FLOW_NOT_LINKED)
|
|
goto done;
|
|
}
|
|
/* if we get here, all other pads were unlinked and we return
|
|
* NOT_LINKED then */
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
|
|
{
|
|
GList *streams;
|
|
|
|
for (streams = src->streams; streams; streams = g_list_next (streams)) {
|
|
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
|
|
|
|
/* only pads that have a connection to the outside world */
|
|
if (ostream->srcpad == NULL)
|
|
continue;
|
|
|
|
if (ostream->channelpad[0]) {
|
|
gst_event_ref (event);
|
|
if (GST_PAD_IS_SRC (ostream->channelpad[0]))
|
|
gst_pad_push_event (ostream->channelpad[0], event);
|
|
else
|
|
gst_pad_send_event (ostream->channelpad[0], event);
|
|
}
|
|
|
|
if (ostream->channelpad[1]) {
|
|
gst_event_ref (event);
|
|
if (GST_PAD_IS_SRC (ostream->channelpad[1]))
|
|
gst_pad_push_event (ostream->channelpad[1], event);
|
|
else
|
|
gst_pad_send_event (ostream->channelpad[1], event);
|
|
}
|
|
}
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
|
|
{
|
|
RTSPMessage response = { 0 };
|
|
RTSPResult res;
|
|
gint channel;
|
|
GList *lstream;
|
|
GstRTSPStream *stream;
|
|
GstPad *outpad = NULL;
|
|
guint8 *data;
|
|
guint size;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstCaps *caps = NULL;
|
|
GstBuffer *buf;
|
|
|
|
do {
|
|
GST_DEBUG_OBJECT (src, "doing receive");
|
|
if ((res = rtsp_connection_receive (src->connection, &response)) < 0)
|
|
goto receive_error;
|
|
|
|
GST_DEBUG_OBJECT (src, "got packet type %d", response.type);
|
|
}
|
|
while (response.type != RTSP_MESSAGE_DATA);
|
|
|
|
channel = response.type_data.data.channel;
|
|
|
|
lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (channel),
|
|
(GCompareFunc) find_stream_by_channel);
|
|
if (!lstream)
|
|
goto unknown_stream;
|
|
|
|
stream = (GstRTSPStream *) lstream->data;
|
|
if (channel == stream->channel[0]) {
|
|
outpad = stream->channelpad[0];
|
|
caps = stream->caps;
|
|
} else if (channel == stream->channel[1]) {
|
|
outpad = stream->channelpad[1];
|
|
}
|
|
|
|
/* take a look at the body to figure out what we have */
|
|
rtsp_message_get_body (&response, &data, &size);
|
|
if (size < 2)
|
|
goto invalid_length;
|
|
|
|
/* channels are not correct on some servers, do extra check */
|
|
if (data[1] >= 200 && data[1] <= 204) {
|
|
/* hmm RTCP message switch to the RTCP pad of the same stream. */
|
|
outpad = stream->channelpad[1];
|
|
}
|
|
|
|
/* we have no clue what this is, just ignore then. */
|
|
if (outpad == NULL)
|
|
goto unknown_stream;
|
|
|
|
/* and chain buffer to internal element */
|
|
rtsp_message_steal_body (&response, &data, &size);
|
|
|
|
/* strip the trailing \0 */
|
|
size -= 1;
|
|
|
|
buf = gst_buffer_new ();
|
|
GST_BUFFER_DATA (buf) = data;
|
|
GST_BUFFER_MALLOCDATA (buf) = data;
|
|
GST_BUFFER_SIZE (buf) = size;
|
|
|
|
/* don't need message anymore */
|
|
rtsp_message_unset (&response);
|
|
|
|
if (caps)
|
|
gst_buffer_set_caps (buf, caps);
|
|
|
|
GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
|
|
channel);
|
|
|
|
if (src->need_activate) {
|
|
gst_rtspsrc_activate_streams (src);
|
|
src->need_activate = FALSE;
|
|
}
|
|
|
|
/* chain to the peer pad */
|
|
if (GST_PAD_IS_SINK (outpad))
|
|
ret = gst_pad_chain (outpad, buf);
|
|
else
|
|
ret = gst_pad_push (outpad, buf);
|
|
|
|
/* combine all stream flows */
|
|
ret = gst_rtspsrc_combine_flows (src, stream, ret);
|
|
if (ret != GST_FLOW_OK)
|
|
goto need_pause;
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
unknown_stream:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
|
|
rtsp_message_unset (&response);
|
|
return;
|
|
}
|
|
receive_error:
|
|
{
|
|
gchar *str = rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
|
|
("Could not receive message. (%s)", str));
|
|
g_free (str);
|
|
|
|
if (src->debug)
|
|
rtsp_message_dump (&response);
|
|
|
|
rtsp_message_unset (&response);
|
|
ret = GST_FLOW_UNEXPECTED;
|
|
goto need_pause;
|
|
}
|
|
invalid_length:
|
|
{
|
|
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
|
|
("Short message received."));
|
|
rtsp_message_unset (&response);
|
|
return;
|
|
}
|
|
need_pause:
|
|
{
|
|
const gchar *reason = gst_flow_get_name (ret);
|
|
|
|
GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
|
|
src->running = FALSE;
|
|
gst_task_pause (src->task);
|
|
if (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED) {
|
|
if (ret == GST_FLOW_UNEXPECTED) {
|
|
/* perform EOS logic */
|
|
if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
|
|
gst_element_post_message (GST_ELEMENT_CAST (src),
|
|
gst_message_new_segment_done (GST_OBJECT_CAST (src),
|
|
src->segment.format, src->segment.last_stop));
|
|
} else {
|
|
gst_rtspsrc_push_event (src, gst_event_new_eos ());
|
|
}
|
|
} else {
|
|
/* for fatal errors we post an error message, post the error
|
|
* first so the app knows about the error first. */
|
|
GST_ELEMENT_ERROR (src, STREAM, FAILED,
|
|
("Internal data flow error."),
|
|
("streaming task paused, reason %s (%d)", reason, ret));
|
|
gst_rtspsrc_push_event (src, gst_event_new_eos ());
|
|
}
|
|
}
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_loop_udp (GstRTSPSrc * src)
|
|
{
|
|
gboolean restart = FALSE;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
if (src->loop_cmd == CMD_STOP)
|
|
goto stopping;
|
|
|
|
while (src->loop_cmd == CMD_WAIT) {
|
|
GST_DEBUG_OBJECT (src, "waiting");
|
|
GST_RTSP_LOOP_WAIT (src);
|
|
GST_DEBUG_OBJECT (src, "waiting done");
|
|
if (src->loop_cmd == CMD_STOP)
|
|
goto stopping;
|
|
}
|
|
if (src->loop_cmd == CMD_RECONNECT) {
|
|
/* FIXME, when we get here we have to reconnect using tcp */
|
|
src->loop_cmd = CMD_WAIT;
|
|
|
|
/* only restart when the pads were not yet activated, else we were
|
|
* streaming over UDP */
|
|
restart = src->need_activate;
|
|
}
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
/* no need to restart, we're done */
|
|
if (!restart)
|
|
goto done;
|
|
|
|
/* We post a warning message now to inform the user
|
|
* that nothing happened. It's most likely a firewall thing. */
|
|
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
|
|
("Could not receive any UDP packets for %.4f seconds, maybe your "
|
|
"firewall is blocking it. Retrying using a TCP connection.",
|
|
gst_guint64_to_gdouble (src->timeout / 1000000)));
|
|
/* we can try only TCP now */
|
|
src->cur_protocols = RTSP_LOWER_TRANS_TCP;
|
|
|
|
/* pause to prepare for a restart */
|
|
gst_rtspsrc_pause (src);
|
|
|
|
if (src->task) {
|
|
/* stop task, we cannot join as this would deadlock */
|
|
gst_task_stop (src->task);
|
|
/* and free the task so that _close will not stop/join it again. */
|
|
gst_object_unref (GST_OBJECT (src->task));
|
|
src->task = NULL;
|
|
}
|
|
/* close and cleanup our state */
|
|
gst_rtspsrc_close (src);
|
|
|
|
/* see if we have TCP left to try */
|
|
if (!(src->cur_protocols & RTSP_LOWER_TRANS_TCP))
|
|
goto no_protocols;
|
|
|
|
/* open new connection using tcp */
|
|
if (!gst_rtspsrc_open (src))
|
|
goto open_failed;
|
|
|
|
/* start playback */
|
|
if (!gst_rtspsrc_play (src))
|
|
goto play_failed;
|
|
|
|
done:
|
|
return;
|
|
|
|
/* ERRORS */
|
|
stopping:
|
|
{
|
|
GST_OBJECT_UNLOCK (src);
|
|
src->running = FALSE;
|
|
gst_task_pause (src->task);
|
|
return;
|
|
}
|
|
no_protocols:
|
|
{
|
|
src->cur_protocols = 0;
|
|
/* no transport possible, post an error and stop */
|
|
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
|
|
("Could not connect to server, no protocols left"));
|
|
return;
|
|
}
|
|
open_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "open failed");
|
|
return;
|
|
}
|
|
play_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "play failed");
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
|
|
{
|
|
GST_OBJECT_LOCK (src);
|
|
src->loop_cmd = cmd;
|
|
GST_RTSP_LOOP_SIGNAL (src);
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_loop (GstRTSPSrc * src)
|
|
{
|
|
if (src->interleaved)
|
|
gst_rtspsrc_loop_interleaved (src);
|
|
else
|
|
gst_rtspsrc_loop_udp (src);
|
|
}
|
|
|
|
static RTSPResult
|
|
gst_rtspsrc_handle_request (GstRTSPSrc * src, RTSPMessage * request)
|
|
{
|
|
RTSPMessage response = { 0 };
|
|
RTSPResult res;
|
|
|
|
res = rtsp_message_init_response (&response, RTSP_STS_OK, "OK", request);
|
|
if (res < 0)
|
|
goto send_error;
|
|
|
|
if (src->debug)
|
|
rtsp_message_dump (&response);
|
|
|
|
if ((res = rtsp_connection_send (src->connection, &response)) < 0)
|
|
goto send_error;
|
|
|
|
return RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
send_error:
|
|
{
|
|
gchar *str = rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
g_free (str);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
#ifndef GST_DISABLE_GST_DEBUG
|
|
const gchar *
|
|
rtsp_auth_method_to_string (RTSPAuthMethod method)
|
|
{
|
|
gint index = 0;
|
|
|
|
while (method != 0) {
|
|
index++;
|
|
method >>= 1;
|
|
}
|
|
switch (index) {
|
|
case 0:
|
|
return "None";
|
|
case 1:
|
|
return "Basic";
|
|
case 2:
|
|
return "Digest";
|
|
}
|
|
|
|
return "Unknown";
|
|
}
|
|
#endif
|
|
|
|
/* Parse a WWW-Authenticate Response header and determine the
|
|
* available authentication methods
|
|
* FIXME: To implement digest or other auth types, we should extract
|
|
* the authentication tokens that the server provided for each method
|
|
* into an array of structures and give those to the connection object.
|
|
*
|
|
* This code should also cope with the fact that each WWW-Authenticate
|
|
* header can contain multiple challenge methods + tokens
|
|
*
|
|
* At the moment, we just do a minimal check for Basic auth and don't
|
|
* even parse out the realm */
|
|
static void
|
|
gst_rtspsrc_parse_auth_hdr (gchar * hdr, RTSPAuthMethod * methods)
|
|
{
|
|
gchar *start;
|
|
|
|
g_return_if_fail (hdr != NULL);
|
|
g_return_if_fail (methods != NULL);
|
|
|
|
/* Skip whitespace at the start of the string */
|
|
for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
|
|
|
|
if (g_ascii_strncasecmp (start, "basic", 5) == 0)
|
|
*methods |= RTSP_AUTH_BASIC;
|
|
}
|
|
|
|
/**
|
|
* gst_rtspsrc_setup_auth:
|
|
* @src: the rtsp source
|
|
*
|
|
* Configure a username and password and auth method on the
|
|
* connection object based on a response we received from the
|
|
* peer.
|
|
*
|
|
* Currently, this requires that a username and password were supplied
|
|
* in the uri. In the future, they may be requested on demand by sending
|
|
* a message up the bus.
|
|
*
|
|
* Returns: TRUE if authentication information could be set up correctly.
|
|
*/
|
|
static gboolean
|
|
gst_rtspsrc_setup_auth (GstRTSPSrc * src, RTSPMessage * response)
|
|
{
|
|
gchar *user = NULL;
|
|
gchar *pass = NULL;
|
|
RTSPAuthMethod avail_methods = RTSP_AUTH_NONE;
|
|
RTSPAuthMethod method;
|
|
RTSPResult auth_result;
|
|
gchar *hdr;
|
|
|
|
/* Identify the available auth methods and see if any are supported */
|
|
if (rtsp_message_get_header (response, RTSP_HDR_WWW_AUTHENTICATE, &hdr) ==
|
|
RTSP_OK) {
|
|
gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods);
|
|
}
|
|
|
|
if (avail_methods == RTSP_AUTH_NONE)
|
|
goto no_auth_available;
|
|
|
|
/* FIXME: For digest auth, if the response indicates that the session
|
|
* data are stale, we just update them in the connection object and
|
|
* return TRUE to retry the request */
|
|
|
|
/* Do we have username and password available? */
|
|
if (src->url != NULL && !src->tried_url_auth) {
|
|
user = src->url->user;
|
|
pass = src->url->passwd;
|
|
src->tried_url_auth = TRUE;
|
|
GST_DEBUG_OBJECT (src,
|
|
"Attempting authentication using credentials from the URL");
|
|
}
|
|
|
|
/* FIXME: If the url didn't contain username and password or we tried them
|
|
* already, request a username and passwd from the application via some kind
|
|
* of credentials request message */
|
|
|
|
/* If we don't have a username and passwd at this point, bail out. */
|
|
if (user == NULL || pass == NULL)
|
|
goto no_user_pass;
|
|
|
|
/* Try to configure for each available authentication method, strongest to
|
|
* weakest */
|
|
for (method = RTSP_AUTH_MAX; method != RTSP_AUTH_NONE; method >>= 1) {
|
|
/* Check if this method is available on the server */
|
|
if ((method & avail_methods) == 0)
|
|
continue;
|
|
|
|
/* Pass the credentials to the connection to try on the next request */
|
|
auth_result =
|
|
rtsp_connection_set_auth (src->connection, method, user, pass);
|
|
/* INVAL indicates an invalid username/passwd were supplied, so we'll just
|
|
* ignore it and end up retrying later */
|
|
if (auth_result == RTSP_OK || auth_result == RTSP_EINVAL) {
|
|
GST_DEBUG_OBJECT (src, "Attempting %s authentication",
|
|
rtsp_auth_method_to_string (method));
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (method == RTSP_AUTH_NONE)
|
|
goto no_auth_available;
|
|
|
|
return TRUE;
|
|
|
|
no_auth_available:
|
|
/* Output an error indicating that we couldn't connect because there were
|
|
* no supported authentication protocols */
|
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
|
|
("No supported authentication protocol was found"));
|
|
return FALSE;
|
|
no_user_pass:
|
|
/* We don't fire an error message, we just return FALSE and let the
|
|
* normal NOT_AUTHORIZED error be propagated */
|
|
return FALSE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtspsrc_send:
|
|
* @src: the rtsp source
|
|
* @request: must point to a valid request
|
|
* @response: must point to an empty #RTSPMessage
|
|
*
|
|
* send @request and retrieve the response in @response. optionally @code can be
|
|
* non-NULL in which case it will contain the status code of the response.
|
|
*
|
|
* If This function returns TRUE, @response will contain a valid response
|
|
* message that should be cleaned with rtsp_message_unset() after usage.
|
|
*
|
|
* If @code is NULL, this function will return FALSE (with an invalid @response
|
|
* message) if the response code was not 200 (OK).
|
|
*
|
|
* If the attempt results in an authentication failure, then this will attempt
|
|
* to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
|
|
* the request.
|
|
*
|
|
* Returns: TRUE if the processing was successful.
|
|
*/
|
|
gboolean
|
|
gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
|
|
RTSPMessage * response, RTSPStatusCode * code)
|
|
{
|
|
RTSPStatusCode int_code = RTSP_STS_OK;
|
|
gboolean res;
|
|
gboolean retry;
|
|
|
|
do {
|
|
retry = FALSE;
|
|
res = gst_rtspsrc_try_send (src, request, response, &int_code);
|
|
|
|
if (int_code == RTSP_STS_UNAUTHORIZED) {
|
|
if (gst_rtspsrc_setup_auth (src, response)) {
|
|
/* Try the request/response again after configuring the auth info
|
|
* and loop again */
|
|
retry = TRUE;
|
|
}
|
|
}
|
|
} while (retry == TRUE);
|
|
|
|
/* If the user requested the code, let them handle errors, otherwise
|
|
* post an error below */
|
|
if (code != NULL)
|
|
*code = int_code;
|
|
else if (int_code != RTSP_STS_OK)
|
|
goto error_response;
|
|
|
|
return res;
|
|
|
|
error_response:
|
|
{
|
|
switch (response->type_data.response.code) {
|
|
case RTSP_STS_NOT_FOUND:
|
|
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
|
|
response->type_data.response.reason));
|
|
break;
|
|
default:
|
|
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
|
|
("Got error response: %d (%s).", response->type_data.response.code,
|
|
response->type_data.response.reason));
|
|
break;
|
|
}
|
|
/* we return FALSE so we should unset the response ourselves */
|
|
rtsp_message_unset (response);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
|
|
RTSPMessage * response, RTSPStatusCode * code)
|
|
{
|
|
RTSPResult res;
|
|
RTSPStatusCode thecode;
|
|
gchar *content_base = NULL;
|
|
|
|
if (src->extension && src->extension->before_send)
|
|
src->extension->before_send (src->extension, request);
|
|
|
|
if (src->debug)
|
|
rtsp_message_dump (request);
|
|
|
|
if ((res = rtsp_connection_send (src->connection, request)) < 0)
|
|
goto send_error;
|
|
|
|
next:
|
|
if ((res = rtsp_connection_receive (src->connection, response)) < 0)
|
|
goto receive_error;
|
|
|
|
if (src->debug)
|
|
rtsp_message_dump (response);
|
|
|
|
switch (response->type) {
|
|
case RTSP_MESSAGE_REQUEST:
|
|
/* FIXME, handle server request, reply with OK, for now */
|
|
if ((res = gst_rtspsrc_handle_request (src, response)) < 0)
|
|
goto handle_request_failed;
|
|
goto next;
|
|
case RTSP_MESSAGE_RESPONSE:
|
|
/* ok, a response is good */
|
|
break;
|
|
default:
|
|
case RTSP_MESSAGE_DATA:
|
|
/* get next response */
|
|
goto next;
|
|
}
|
|
|
|
thecode = response->type_data.response.code;
|
|
|
|
/* if the caller wanted the result code, we store it. */
|
|
if (code)
|
|
*code = thecode;
|
|
|
|
/* If the request didn't succeed, bail out before doing any more */
|
|
if (thecode != RTSP_STS_OK)
|
|
return FALSE;
|
|
|
|
/* store new content base if any */
|
|
rtsp_message_get_header (response, RTSP_HDR_CONTENT_BASE, &content_base);
|
|
g_free (src->content_base);
|
|
src->content_base = g_strdup (content_base);
|
|
|
|
if (src->extension && src->extension->after_send)
|
|
src->extension->after_send (src->extension, request, response);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
send_error:
|
|
{
|
|
gchar *str = rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
g_free (str);
|
|
return FALSE;
|
|
}
|
|
receive_error:
|
|
{
|
|
gchar *str = rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
|
|
("Could not receive message. (%s)", str));
|
|
g_free (str);
|
|
return FALSE;
|
|
}
|
|
handle_request_failed:
|
|
{
|
|
/* ERROR was posted */
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* parse the response and collect all the supported methods. We need this
|
|
* information so that we don't try to send an unsupported request to the
|
|
* server.
|
|
*/
|
|
static gboolean
|
|
gst_rtspsrc_parse_methods (GstRTSPSrc * src, RTSPMessage * response)
|
|
{
|
|
gchar *respoptions = NULL;
|
|
gchar **options;
|
|
gint i;
|
|
|
|
/* clear supported methods */
|
|
src->methods = 0;
|
|
|
|
/* Try Allow Header first */
|
|
rtsp_message_get_header (response, RTSP_HDR_ALLOW, &respoptions);
|
|
if (!respoptions)
|
|
/* Then maybe Public Header... */
|
|
rtsp_message_get_header (response, RTSP_HDR_PUBLIC, &respoptions);
|
|
if (!respoptions) {
|
|
/* this field is not required, assume the server supports
|
|
* DESCRIBE, SETUP and PLAY */
|
|
GST_DEBUG_OBJECT (src, "could not get OPTIONS");
|
|
src->methods = RTSP_DESCRIBE | RTSP_SETUP | RTSP_PLAY | RTSP_PAUSE;
|
|
goto done;
|
|
}
|
|
|
|
/* If we get here, the server gave a list of supported methods, parse
|
|
* them here. The string is like:
|
|
*
|
|
* OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ...
|
|
*/
|
|
options = g_strsplit (respoptions, ",", 0);
|
|
|
|
for (i = 0; options[i]; i++) {
|
|
gchar *stripped;
|
|
gint method;
|
|
|
|
stripped = g_strstrip (options[i]);
|
|
method = rtsp_find_method (stripped);
|
|
|
|
/* keep bitfield of supported methods */
|
|
if (method != RTSP_INVALID)
|
|
src->methods |= method;
|
|
}
|
|
g_strfreev (options);
|
|
|
|
/* we need describe and setup */
|
|
if (!(src->methods & RTSP_DESCRIBE))
|
|
goto no_describe;
|
|
if (!(src->methods & RTSP_SETUP))
|
|
goto no_setup;
|
|
|
|
done:
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_describe:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
|
|
("Server does not support DESCRIBE."));
|
|
return FALSE;
|
|
}
|
|
no_setup:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
|
|
("Server does not support SETUP."));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static RTSPResult
|
|
gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
|
|
RTSPLowerTrans protocols, gchar ** transports)
|
|
{
|
|
gchar *result;
|
|
RTSPResult res;
|
|
|
|
*transports = NULL;
|
|
if (src->extension && src->extension->get_transports)
|
|
if ((res =
|
|
src->extension->get_transports (src->extension, protocols,
|
|
transports)) < 0)
|
|
goto failed;
|
|
|
|
/* extension listed transports, use those */
|
|
if (*transports != NULL)
|
|
return RTSP_OK;
|
|
|
|
/* the default RTSP transports */
|
|
result = g_strdup ("");
|
|
if (protocols & RTSP_LOWER_TRANS_UDP) {
|
|
gchar *new;
|
|
|
|
GST_DEBUG_OBJECT (src, "adding UDP unicast");
|
|
|
|
new =
|
|
g_strconcat (result, "RTP/AVP/UDP;unicast;client_port=%%u1-%%u2", NULL);
|
|
g_free (result);
|
|
result = new;
|
|
}
|
|
if (protocols & RTSP_LOWER_TRANS_UDP_MCAST) {
|
|
gchar *new;
|
|
|
|
GST_DEBUG_OBJECT (src, "adding UDP multicast");
|
|
|
|
/* we don't have to allocate any UDP ports yet, if the selected transport
|
|
* turns out to be multicast we can create them and join the multicast
|
|
* group indicated in the transport reply */
|
|
new = g_strconcat (result, result[0] ? "," : "",
|
|
"RTP/AVP/UDP;multicast", NULL);
|
|
g_free (result);
|
|
result = new;
|
|
}
|
|
if (protocols & RTSP_LOWER_TRANS_TCP) {
|
|
gchar *new;
|
|
|
|
GST_DEBUG_OBJECT (src, "adding TCP");
|
|
|
|
new = g_strconcat (result, result[0] ? "," : "",
|
|
"RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2", NULL);
|
|
g_free (result);
|
|
result = new;
|
|
}
|
|
*transports = result;
|
|
|
|
return RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
failed:
|
|
{
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static RTSPResult
|
|
gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports)
|
|
{
|
|
GstRTSPSrc *src;
|
|
gint nr_udp, nr_int;
|
|
gchar *next, *p;
|
|
gint rtpport = 0, rtcpport = 0;
|
|
GString *str;
|
|
|
|
src = stream->parent;
|
|
|
|
/* find number of placeholders first */
|
|
if (strstr (*transports, "%%i2"))
|
|
nr_int = 2;
|
|
else if (strstr (*transports, "%%i1"))
|
|
nr_int = 1;
|
|
else
|
|
nr_int = 0;
|
|
|
|
if (strstr (*transports, "%%u2"))
|
|
nr_udp = 2;
|
|
else if (strstr (*transports, "%%u1"))
|
|
nr_udp = 1;
|
|
else
|
|
nr_udp = 0;
|
|
|
|
if (nr_udp == 0 && nr_int == 0)
|
|
goto done;
|
|
|
|
if (nr_udp > 0) {
|
|
if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
|
|
goto failed;
|
|
}
|
|
|
|
str = g_string_new ("");
|
|
p = *transports;
|
|
while ((next = strstr (p, "%%"))) {
|
|
g_string_append_len (str, p, next - p);
|
|
if (next[2] == 'u') {
|
|
if (next[3] == '1')
|
|
g_string_append_printf (str, "%d", rtpport);
|
|
else if (next[3] == '2')
|
|
g_string_append_printf (str, "%d", rtcpport);
|
|
}
|
|
if (next[2] == 'i') {
|
|
if (next[3] == '1')
|
|
g_string_append_printf (str, "%d", src->free_channel);
|
|
else if (next[3] == '2')
|
|
g_string_append_printf (str, "%d", src->free_channel + 1);
|
|
}
|
|
|
|
p = next + 4;
|
|
}
|
|
|
|
g_free (*transports);
|
|
*transports = g_string_free (str, FALSE);
|
|
|
|
done:
|
|
return RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
failed:
|
|
{
|
|
return RTSP_ERROR;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_setup_streams (GstRTSPSrc * src)
|
|
{
|
|
GList *walk;
|
|
RTSPResult res;
|
|
RTSPMessage request = { 0 };
|
|
RTSPMessage response = { 0 };
|
|
GstRTSPStream *stream = NULL;
|
|
RTSPLowerTrans protocols;
|
|
|
|
/* we initially allow all configured lower transports. based on the URL
|
|
* transports and the replies from the server we narrow them down. */
|
|
protocols = src->url->transports & src->cur_protocols;
|
|
|
|
/* reset some state */
|
|
src->free_channel = 0;
|
|
src->interleaved = FALSE;
|
|
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
gchar *transports;
|
|
|
|
stream = (GstRTSPStream *) walk->data;
|
|
|
|
/* see if we need to configure this stream */
|
|
if (src->extension && src->extension->configure_stream) {
|
|
if (!src->extension->configure_stream (src->extension, stream)) {
|
|
GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
|
|
stream);
|
|
continue;
|
|
}
|
|
}
|
|
|
|
/* merge/overwrite global caps */
|
|
if (stream->caps) {
|
|
guint j, num;
|
|
GstStructure *s;
|
|
|
|
s = gst_caps_get_structure (stream->caps, 0);
|
|
|
|
num = gst_structure_n_fields (src->props);
|
|
for (j = 0; j < num; j++) {
|
|
const gchar *name;
|
|
const GValue *val;
|
|
|
|
name = gst_structure_nth_field_name (src->props, j);
|
|
val = gst_structure_get_value (src->props, name);
|
|
gst_structure_set_value (s, name, val);
|
|
|
|
GST_DEBUG_OBJECT (src, "copied %s", name);
|
|
}
|
|
}
|
|
|
|
/* skip setup if we have no URL for it */
|
|
if (stream->setup_url == NULL) {
|
|
GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
|
|
continue;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
|
|
stream->setup_url);
|
|
|
|
/* create a string with all the transports */
|
|
res = gst_rtspsrc_create_transports_string (src, protocols, &transports);
|
|
if (res < 0)
|
|
goto setup_transport_failed;
|
|
|
|
/* replace placeholders with real values, this function will optionally
|
|
* allocate UDP ports and other info needed to execute the setup request */
|
|
res = gst_rtspsrc_prepare_transports (stream, &transports);
|
|
if (res < 0)
|
|
goto setup_transport_failed;
|
|
|
|
/* create SETUP request */
|
|
res = rtsp_message_init_request (&request, RTSP_SETUP, stream->setup_url);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
/* select transport, copy is made when adding to header so we can free it. */
|
|
rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
|
|
g_free (transports);
|
|
|
|
if (!gst_rtspsrc_send (src, &request, &response, NULL))
|
|
goto send_error;
|
|
|
|
/* parse response transport */
|
|
{
|
|
gchar *resptrans = NULL;
|
|
RTSPTransport transport = { 0 };
|
|
|
|
rtsp_message_get_header (&response, RTSP_HDR_TRANSPORT, &resptrans);
|
|
if (!resptrans)
|
|
goto no_transport;
|
|
|
|
/* parse transport */
|
|
if (rtsp_transport_parse (resptrans, &transport) != RTSP_OK)
|
|
continue;
|
|
|
|
/* update allowed transports for other streams. once the transport of
|
|
* one stream has been determined, we make sure that all other streams
|
|
* are configured in the same way */
|
|
switch (transport.lower_transport) {
|
|
case RTSP_LOWER_TRANS_TCP:
|
|
GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
|
|
protocols = RTSP_LOWER_TRANS_TCP;
|
|
src->interleaved = TRUE;
|
|
/* update free channels */
|
|
src->free_channel =
|
|
MAX (transport.interleaved.min, src->free_channel);
|
|
src->free_channel =
|
|
MAX (transport.interleaved.max, src->free_channel);
|
|
src->free_channel++;
|
|
break;
|
|
case RTSP_LOWER_TRANS_UDP_MCAST:
|
|
/* only allow multicast for other streams */
|
|
GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
|
|
protocols = RTSP_LOWER_TRANS_UDP_MCAST;
|
|
break;
|
|
case RTSP_LOWER_TRANS_UDP:
|
|
/* only allow unicast for other streams */
|
|
GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
|
|
protocols = RTSP_LOWER_TRANS_UDP;
|
|
break;
|
|
default:
|
|
GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
|
|
transport.lower_transport);
|
|
break;
|
|
}
|
|
|
|
if (!stream->container || !src->interleaved) {
|
|
/* now configure the stream with the selected transport */
|
|
if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
|
|
GST_DEBUG_OBJECT (src,
|
|
"could not configure stream %p transport, skipping stream",
|
|
stream);
|
|
}
|
|
}
|
|
/* clean up our transport struct */
|
|
rtsp_transport_init (&transport);
|
|
}
|
|
}
|
|
if (src->extension && src->extension->stream_select)
|
|
src->extension->stream_select (src->extension);
|
|
|
|
/* we need to activate the streams when we detect activity */
|
|
src->need_activate = TRUE;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
create_request_failed:
|
|
{
|
|
gchar *str = rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
|
|
("Could not create request. (%s)", str));
|
|
g_free (str);
|
|
goto cleanup_error;
|
|
}
|
|
setup_transport_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
("Could not setup transport."));
|
|
goto cleanup_error;
|
|
}
|
|
send_error:
|
|
{
|
|
gchar *str = rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
g_free (str);
|
|
goto cleanup_error;
|
|
}
|
|
no_transport:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
("Server did not select transport."));
|
|
goto cleanup_error;
|
|
}
|
|
cleanup_error:
|
|
{
|
|
rtsp_message_unset (&request);
|
|
rtsp_message_unset (&response);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_open (GstRTSPSrc * src)
|
|
{
|
|
RTSPResult res;
|
|
RTSPMessage request = { 0 };
|
|
RTSPMessage response = { 0 };
|
|
guint8 *data;
|
|
guint size;
|
|
gint i, n_streams;
|
|
SDPMessage sdp = { 0 };
|
|
GstRTSPStream *stream = NULL;
|
|
gchar *respcont = NULL;
|
|
|
|
/* reset our state */
|
|
gst_segment_init (&src->segment, GST_FORMAT_TIME);
|
|
|
|
/* can't continue without a valid url */
|
|
if (G_UNLIKELY (src->url == NULL))
|
|
goto no_url;
|
|
src->tried_url_auth = FALSE;
|
|
|
|
/* create connection */
|
|
GST_DEBUG_OBJECT (src, "creating connection (%s)...", src->req_location);
|
|
if ((res = rtsp_connection_create (src->url, &src->connection)) < 0)
|
|
goto could_not_create;
|
|
|
|
/* connect */
|
|
GST_DEBUG_OBJECT (src, "connecting (%s)...", src->req_location);
|
|
if ((res = rtsp_connection_connect (src->connection)) < 0)
|
|
goto could_not_connect;
|
|
|
|
/* create OPTIONS */
|
|
GST_DEBUG_OBJECT (src, "create options...");
|
|
res = rtsp_message_init_request (&request, RTSP_OPTIONS, src->req_location);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
/* send OPTIONS */
|
|
GST_DEBUG_OBJECT (src, "send options...");
|
|
if (!gst_rtspsrc_send (src, &request, &response, NULL))
|
|
goto send_error;
|
|
|
|
/* parse OPTIONS */
|
|
if (!gst_rtspsrc_parse_methods (src, &response))
|
|
goto methods_error;
|
|
|
|
/* create DESCRIBE */
|
|
GST_DEBUG_OBJECT (src, "create describe...");
|
|
res = rtsp_message_init_request (&request, RTSP_DESCRIBE, src->req_location);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
/* we only accept SDP for now */
|
|
rtsp_message_add_header (&request, RTSP_HDR_ACCEPT, "application/sdp");
|
|
|
|
/* prepare global stream caps properties */
|
|
if (src->props)
|
|
gst_structure_remove_all_fields (src->props);
|
|
else
|
|
src->props = gst_structure_empty_new ("RTSP Properties");
|
|
|
|
/* send DESCRIBE */
|
|
GST_DEBUG_OBJECT (src, "send describe...");
|
|
if (!gst_rtspsrc_send (src, &request, &response, NULL))
|
|
goto send_error;
|
|
|
|
/* check if reply is SDP */
|
|
rtsp_message_get_header (&response, RTSP_HDR_CONTENT_TYPE, &respcont);
|
|
/* could not be set but since the request returned OK, we assume it
|
|
* was SDP, else check it. */
|
|
if (respcont) {
|
|
if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
|
|
goto wrong_content_type;
|
|
}
|
|
|
|
/* get message body and parse as SDP */
|
|
rtsp_message_get_body (&response, &data, &size);
|
|
|
|
GST_DEBUG_OBJECT (src, "parse SDP...");
|
|
sdp_message_init (&sdp);
|
|
sdp_message_parse_buffer (data, size, &sdp);
|
|
|
|
if (src->debug)
|
|
sdp_message_dump (&sdp);
|
|
|
|
if (src->extension && src->extension->parse_sdp)
|
|
src->extension->parse_sdp (src->extension, &sdp);
|
|
|
|
/* create streams */
|
|
n_streams = sdp_message_medias_len (&sdp);
|
|
for (i = 0; i < n_streams; i++) {
|
|
stream = gst_rtspsrc_create_stream (src, &sdp, i);
|
|
}
|
|
|
|
/* setup streams */
|
|
gst_rtspsrc_setup_streams (src);
|
|
|
|
/* clean up any messages */
|
|
rtsp_message_unset (&request);
|
|
rtsp_message_unset (&response);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_url:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
|
|
("No valid RTSP URL was provided"));
|
|
goto cleanup_error;
|
|
}
|
|
could_not_create:
|
|
{
|
|
gchar *str = rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
|
|
("Could not create connection. (%s)", str));
|
|
g_free (str);
|
|
goto cleanup_error;
|
|
}
|
|
could_not_connect:
|
|
{
|
|
gchar *str = rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
|
|
("Could not connect to server. (%s)", str));
|
|
g_free (str);
|
|
goto cleanup_error;
|
|
}
|
|
create_request_failed:
|
|
{
|
|
gchar *str = rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
|
|
("Could not create request. (%s)", str));
|
|
g_free (str);
|
|
goto cleanup_error;
|
|
}
|
|
send_error:
|
|
{
|
|
/* Don't post a message - the rtsp_send method will have
|
|
* taken care of it because we passed NULL for the response code */
|
|
goto cleanup_error;
|
|
}
|
|
methods_error:
|
|
{
|
|
/* error was posted */
|
|
goto cleanup_error;
|
|
}
|
|
wrong_content_type:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
("Server does not support SDP, got %s.", respcont));
|
|
goto cleanup_error;
|
|
}
|
|
cleanup_error:
|
|
{
|
|
rtsp_message_unset (&request);
|
|
rtsp_message_unset (&response);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_close (GstRTSPSrc * src)
|
|
{
|
|
RTSPMessage request = { 0 };
|
|
RTSPMessage response = { 0 };
|
|
RTSPResult res;
|
|
|
|
GST_DEBUG_OBJECT (src, "TEARDOWN...");
|
|
|
|
gst_rtspsrc_loop_send_cmd (src, CMD_STOP);
|
|
|
|
/* stop task if any */
|
|
if (src->task) {
|
|
gst_task_stop (src->task);
|
|
|
|
/* make sure it is not running */
|
|
g_static_rec_mutex_lock (src->stream_rec_lock);
|
|
g_static_rec_mutex_unlock (src->stream_rec_lock);
|
|
|
|
/* no wait for the task to finish */
|
|
gst_task_join (src->task);
|
|
|
|
/* and free the task */
|
|
gst_object_unref (GST_OBJECT (src->task));
|
|
src->task = NULL;
|
|
}
|
|
|
|
if (src->methods & RTSP_PLAY) {
|
|
/* do TEARDOWN */
|
|
res =
|
|
rtsp_message_init_request (&request, RTSP_TEARDOWN, src->req_location);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
if (!gst_rtspsrc_send (src, &request, &response, NULL))
|
|
goto send_error;
|
|
|
|
/* FIXME, parse result? */
|
|
rtsp_message_unset (&request);
|
|
rtsp_message_unset (&response);
|
|
}
|
|
|
|
/* close connection */
|
|
GST_DEBUG_OBJECT (src, "closing connection...");
|
|
if ((res = rtsp_connection_close (src->connection)) < 0)
|
|
goto close_failed;
|
|
|
|
/* free connection */
|
|
rtsp_connection_free (src->connection);
|
|
src->connection = NULL;
|
|
|
|
/* cleanup */
|
|
gst_rtspsrc_cleanup (src);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
create_request_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
|
|
("Could not create request."));
|
|
return FALSE;
|
|
}
|
|
send_error:
|
|
{
|
|
rtsp_message_unset (&request);
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send message."));
|
|
return FALSE;
|
|
}
|
|
close_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, CLOSE, (NULL), ("Close failed."));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* RTP-Info is of the format:
|
|
*
|
|
* url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
|
|
*/
|
|
static gboolean
|
|
gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
|
|
{
|
|
gchar **infos;
|
|
gint i;
|
|
|
|
infos = g_strsplit (rtpinfo, ",", 0);
|
|
for (i = 0; infos[i]; i++) {
|
|
/* FIXME, do something here:
|
|
* parse url, find stream for url.
|
|
* parse seq and rtptime. The seq number should be configured in the rtp
|
|
* depayloader or session manager to detect gaps. Same for the rtptime, it
|
|
* should be used to create an initial time newsegment.
|
|
*/
|
|
}
|
|
g_strfreev (infos);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_play (GstRTSPSrc * src)
|
|
{
|
|
RTSPMessage request = { 0 };
|
|
RTSPMessage response = { 0 };
|
|
RTSPResult res;
|
|
gchar *rtpinfo;
|
|
|
|
if (!(src->methods & RTSP_PLAY))
|
|
return TRUE;
|
|
|
|
GST_DEBUG_OBJECT (src, "PLAY...");
|
|
|
|
/* do play */
|
|
res = rtsp_message_init_request (&request, RTSP_PLAY, src->req_location);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
rtsp_message_add_header (&request, RTSP_HDR_RANGE, "npt=0-");
|
|
|
|
if (!gst_rtspsrc_send (src, &request, &response, NULL))
|
|
goto send_error;
|
|
|
|
rtsp_message_unset (&request);
|
|
|
|
/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
|
|
* for the RTP packets. If this is not present, we assume all starts from 0...
|
|
* FIXME, this is info for the RTP session manager ideally. */
|
|
rtsp_message_get_header (&response, RTSP_HDR_RTP_INFO, &rtpinfo);
|
|
if (rtpinfo)
|
|
gst_rtspsrc_parse_rtpinfo (src, rtpinfo);
|
|
|
|
rtsp_message_unset (&response);
|
|
|
|
/* for interleaved transport, we receive the data on the RTSP connection
|
|
* instead of UDP. We start a task to select and read from that connection.
|
|
* For UDP we start the task as well to look for server info and UDP timeouts. */
|
|
if (src->task == NULL) {
|
|
src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_loop, src);
|
|
gst_task_set_lock (src->task, src->stream_rec_lock);
|
|
}
|
|
src->running = TRUE;
|
|
gst_rtspsrc_loop_send_cmd (src, CMD_WAIT);
|
|
gst_task_start (src->task);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
create_request_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
|
|
("Could not create request."));
|
|
return FALSE;
|
|
}
|
|
send_error:
|
|
{
|
|
rtsp_message_unset (&request);
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send message."));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_pause (GstRTSPSrc * src)
|
|
{
|
|
RTSPMessage request = { 0 };
|
|
RTSPMessage response = { 0 };
|
|
RTSPResult res;
|
|
|
|
if (!(src->methods & RTSP_PAUSE))
|
|
return TRUE;
|
|
|
|
GST_DEBUG_OBJECT (src, "PAUSE...");
|
|
/* do pause */
|
|
res = rtsp_message_init_request (&request, RTSP_PAUSE, src->req_location);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
if (!gst_rtspsrc_send (src, &request, &response, NULL))
|
|
goto send_error;
|
|
|
|
rtsp_message_unset (&request);
|
|
rtsp_message_unset (&response);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
create_request_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
|
|
("Could not create request."));
|
|
return FALSE;
|
|
}
|
|
send_error:
|
|
{
|
|
rtsp_message_unset (&request);
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send message."));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
|
|
{
|
|
switch (GST_MESSAGE_TYPE (message)) {
|
|
case GST_MESSAGE_ELEMENT:
|
|
{
|
|
GstRTSPSrc *rtspsrc;
|
|
const GstStructure *s = gst_message_get_structure (message);
|
|
|
|
rtspsrc = GST_RTSPSRC (bin);
|
|
|
|
if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
|
|
GST_DEBUG_OBJECT (bin, "timeout on UDP port");
|
|
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
|
|
return;
|
|
}
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
case GST_MESSAGE_ERROR:
|
|
{
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
default:
|
|
{
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstRTSPSrc *rtspsrc;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtspsrc = GST_RTSPSRC (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
rtspsrc->cur_protocols = rtspsrc->protocols;
|
|
if (!gst_rtspsrc_open (rtspsrc))
|
|
goto open_failed;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
rtsp_connection_flush (rtspsrc->connection, FALSE);
|
|
gst_rtspsrc_play (rtspsrc);
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
rtsp_connection_flush (rtspsrc->connection, TRUE);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
goto done;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
gst_rtspsrc_pause (rtspsrc);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_rtspsrc_close (rtspsrc);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
|
|
open_failed:
|
|
{
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
}
|
|
|
|
/*** GSTURIHANDLER INTERFACE *************************************************/
|
|
|
|
static GstURIType
|
|
gst_rtspsrc_uri_get_type (void)
|
|
{
|
|
return GST_URI_SRC;
|
|
}
|
|
static gchar **
|
|
gst_rtspsrc_uri_get_protocols (void)
|
|
{
|
|
static gchar *protocols[] = { "rtsp", "rtspu", "rtspt", NULL };
|
|
|
|
return protocols;
|
|
}
|
|
|
|
static const gchar *
|
|
gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
|
|
{
|
|
GstRTSPSrc *src = GST_RTSPSRC (handler);
|
|
|
|
/* should not dup */
|
|
return src->location;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri)
|
|
{
|
|
GstRTSPSrc *src;
|
|
RTSPResult res;
|
|
RTSPUrl *newurl;
|
|
|
|
src = GST_RTSPSRC (handler);
|
|
|
|
/* same URI, we're fine */
|
|
if (src->location && uri && !strcmp (uri, src->location))
|
|
goto was_ok;
|
|
|
|
/* try to parse */
|
|
if ((res = rtsp_url_parse (uri, &newurl)) < 0)
|
|
goto parse_error;
|
|
|
|
/* if worked, free previous and store new url object along with the original
|
|
* location. */
|
|
rtsp_url_free (src->url);
|
|
src->url = newurl;
|
|
g_free (src->location);
|
|
g_free (src->req_location);
|
|
src->location = g_strdup (uri);
|
|
src->req_location = rtsp_url_get_request_uri (src->url);
|
|
|
|
GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
|
|
GST_DEBUG_OBJECT (src, "request uri is: %s",
|
|
GST_STR_NULL (src->req_location));
|
|
|
|
return TRUE;
|
|
|
|
/* Special cases */
|
|
was_ok:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
|
|
return TRUE;
|
|
}
|
|
parse_error:
|
|
{
|
|
GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
|
|
GST_STR_NULL (uri), res);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
|
|
{
|
|
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
|
|
|
|
iface->get_type = gst_rtspsrc_uri_get_type;
|
|
iface->get_protocols = gst_rtspsrc_uri_get_protocols;
|
|
iface->get_uri = gst_rtspsrc_uri_get_uri;
|
|
iface->set_uri = gst_rtspsrc_uri_set_uri;
|
|
}
|