mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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6820 lines
194 KiB
C
6820 lines
194 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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* 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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*
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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 #GstRTSPSrc:protocols property.
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*
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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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*
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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 implemented using the gstrtpbin element.
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*
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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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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
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* ]| Establish a connection to an RTSP server and send the raw RTP packets to a
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* fakesink.
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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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/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
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* with newer GLib versions (>= 2.31.0) */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#ifdef HAVE_UNISTD_H
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#include <unistd.h>
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#endif /* HAVE_UNISTD_H */
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#include <stdlib.h>
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#include <string.h>
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#include <locale.h>
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#include <stdio.h>
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#include <stdarg.h>
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#include <gst/sdp/gstsdpmessage.h>
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#include <gst/rtp/gstrtppayloads.h>
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#include "gst/gst-i18n-plugin.h"
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#include "gstrtspsrc.h"
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#ifdef G_OS_WIN32
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#include <winsock2.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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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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/* templates used internally */
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static GstStaticPadTemplate anysrctemplate =
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GST_STATIC_PAD_TEMPLATE ("internalsrc%d",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS_ANY);
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static GstStaticPadTemplate anysinktemplate =
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GST_STATIC_PAD_TEMPLATE ("internalsink%d",
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GST_PAD_SINK,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS_ANY);
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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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enum _GstRtspSrcRtcpSyncMode
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{
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RTCP_SYNC_ALWAYS,
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RTCP_SYNC_INITIAL,
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RTCP_SYNC_RTP
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};
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enum _GstRtspSrcBufferMode
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{
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BUFFER_MODE_NONE,
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BUFFER_MODE_SLAVE,
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BUFFER_MODE_BUFFER,
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BUFFER_MODE_AUTO
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};
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#define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
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static GType
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gst_rtsp_src_buffer_mode_get_type (void)
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{
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static GType buffer_mode_type = 0;
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static const GEnumValue buffer_modes[] = {
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{BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
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{BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
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{BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
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{BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
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{0, NULL, NULL},
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};
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if (!buffer_mode_type) {
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buffer_mode_type =
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g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
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}
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return buffer_mode_type;
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}
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#define DEFAULT_LOCATION NULL
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#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_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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#define DEFAULT_UDP_BUFFER_SIZE 0x80000
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#define DEFAULT_TCP_TIMEOUT 20000000
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#define DEFAULT_LATENCY_MS 2000
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#define DEFAULT_CONNECTION_SPEED 0
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#define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
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#define DEFAULT_DO_RTCP TRUE
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#define DEFAULT_PROXY NULL
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#define DEFAULT_RTP_BLOCKSIZE 0
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#define DEFAULT_USER_ID NULL
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#define DEFAULT_USER_PW NULL
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#define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
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#define DEFAULT_PORT_RANGE NULL
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#define DEFAULT_SHORT_HEADER FALSE
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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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PROP_TCP_TIMEOUT,
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PROP_LATENCY,
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PROP_CONNECTION_SPEED,
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PROP_NAT_METHOD,
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PROP_DO_RTCP,
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PROP_PROXY,
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PROP_RTP_BLOCKSIZE,
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PROP_USER_ID,
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PROP_USER_PW,
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PROP_BUFFER_MODE,
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PROP_PORT_RANGE,
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PROP_UDP_BUFFER_SIZE,
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PROP_SHORT_HEADER,
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PROP_LAST
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};
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#define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
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static GType
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gst_rtsp_nat_method_get_type (void)
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{
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static GType rtsp_nat_method_type = 0;
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static const GEnumValue rtsp_nat_method[] = {
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{GST_RTSP_NAT_NONE, "None", "none"},
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{GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
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{0, NULL, NULL},
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};
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if (!rtsp_nat_method_type) {
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rtsp_nat_method_type =
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g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
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}
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return rtsp_nat_method_type;
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}
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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 void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
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GstCaps * caps);
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static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
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static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
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static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
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static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
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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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GstRTSPMessage * response);
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static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
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gboolean flush);
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static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
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GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
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static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
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static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
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gboolean async);
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static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle,
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gboolean async);
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static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
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gboolean only_close);
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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 gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
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GstRTSPStream * stream, GstEvent * event, gboolean source);
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static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event,
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gboolean source);
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/* commands we send to out loop to notify it of events */
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#define CMD_OPEN 0
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#define CMD_PLAY 1
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#define CMD_PAUSE 2
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#define CMD_CLOSE 3
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#define CMD_WAIT 4
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#define CMD_RECONNECT 5
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#define CMD_LOOP 6
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#define GST_ELEMENT_PROGRESS(el, type, code, text) \
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G_STMT_START { \
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gchar *__txt = _gst_element_error_printf text; \
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gst_element_post_message (GST_ELEMENT_CAST (el), \
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gst_message_new_progress (GST_OBJECT_CAST (el), \
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GST_PROGRESS_TYPE_ ##type, code, __txt)); \
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g_free (__txt); \
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} G_STMT_END
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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_static_pad_template (element_class, &rtptemplate);
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gst_element_class_set_details_simple (element_class, "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>, "
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"Thijs Vermeir <thijs.vermeir@barco.com>, "
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"Lutz Mueller <lutz@topfrose.de>");
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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 | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
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g_param_spec_flags ("protocols", "Protocols",
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"Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
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DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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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_STATIC_STRINGS));
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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_STATIC_STRINGS));
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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 UDP timeout microseconds (0 = disabled)",
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0, G_MAXUINT64, DEFAULT_TIMEOUT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
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g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
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"Fail after timeout microseconds on TCP connections (0 = disabled)",
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0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_LATENCY,
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g_param_spec_uint ("latency", "Buffer latency in ms",
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"Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
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g_param_spec_uint ("connection-speed", "Connection Speed",
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"Network connection speed in kbps (0 = unknown)",
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0, G_MAXINT / 1000, DEFAULT_CONNECTION_SPEED,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
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g_param_spec_enum ("nat-method", "NAT Method",
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"Method to use for traversing firewalls and NAT",
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GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPSrc::do-rtcp
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*
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* Enable RTCP support. Some old server don't like RTCP and then this property
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* needs to be set to FALSE.
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*
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* Since: 0.10.15
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*/
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g_object_class_install_property (gobject_class, PROP_DO_RTCP,
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g_param_spec_boolean ("do-rtcp", "Do RTCP",
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"Send RTCP packets, disable for old incompatible server.",
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DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPSrc::proxy
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*
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* Set the proxy parameters. This has to be a string of the format
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* [http://][user:passwd@]host[:port].
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*
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* Since: 0.10.15
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*/
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g_object_class_install_property (gobject_class, PROP_PROXY,
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g_param_spec_string ("proxy", "Proxy",
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"Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
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DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPSrc::rtp_blocksize
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*
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* RTP package size to suggest to server.
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*
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* Since: 0.10.16
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*/
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g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
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g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
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"RTP package size to suggest to server (0 = disabled)",
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0, 65536, DEFAULT_RTP_BLOCKSIZE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_USER_ID,
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g_param_spec_string ("user-id", "user-id",
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"RTSP location URI user id for authentication", DEFAULT_USER_ID,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_USER_PW,
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g_param_spec_string ("user-pw", "user-pw",
|
|
"RTSP location URI user password for authentication", DEFAULT_USER_PW,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPSrc::buffer-mode:
|
|
*
|
|
* Control the buffering and timestamping mode used by the jitterbuffer.
|
|
*
|
|
* Since: 0.10.22
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
|
|
g_param_spec_enum ("buffer-mode", "Buffer Mode",
|
|
"Control the buffering algorithm in use",
|
|
GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPSrc::port-range:
|
|
*
|
|
* Configure the client port numbers that can be used to recieve RTP and
|
|
* RTCP.
|
|
*
|
|
* Since: 0.10.25
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
|
|
g_param_spec_string ("port-range", "Port range",
|
|
"Client port range that can be used to receive RTP and RTCP data, "
|
|
"eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPSrc::udp-buffer-size:
|
|
*
|
|
* Size of the kernel UDP receive buffer in bytes.
|
|
*
|
|
* Since: 0.10.26
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
|
|
g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
|
|
"Size of the kernel UDP receive buffer in bytes, 0=default",
|
|
0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPSrc::short-header:
|
|
*
|
|
* Only send the basic RTSP headers for broken encoders.
|
|
*
|
|
* Since: 0.10.31
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
|
|
g_param_spec_boolean ("short-header", "Short Header",
|
|
"Only send the basic RTSP headers for broken encoders",
|
|
DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
gstelement_class->send_event = gst_rtspsrc_send_event;
|
|
gstelement_class->change_state = gst_rtspsrc_change_state;
|
|
|
|
gstbin_class->handle_message = gst_rtspsrc_handle_message;
|
|
|
|
gst_rtsp_ext_list_init ();
|
|
}
|
|
|
|
|
|
static void
|
|
gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
|
|
{
|
|
#ifdef G_OS_WIN32
|
|
WSADATA wsa_data;
|
|
|
|
if (WSAStartup (MAKEWORD (2, 2), &wsa_data) != 0) {
|
|
GST_ERROR_OBJECT (src, "WSAStartup failed: 0x%08x", WSAGetLastError ());
|
|
}
|
|
#endif
|
|
|
|
src->conninfo.location = g_strdup (DEFAULT_LOCATION);
|
|
src->protocols = DEFAULT_PROTOCOLS;
|
|
src->debug = DEFAULT_DEBUG;
|
|
src->retry = DEFAULT_RETRY;
|
|
src->udp_timeout = DEFAULT_TIMEOUT;
|
|
gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
|
|
src->latency = DEFAULT_LATENCY_MS;
|
|
src->connection_speed = DEFAULT_CONNECTION_SPEED;
|
|
src->nat_method = DEFAULT_NAT_METHOD;
|
|
src->do_rtcp = DEFAULT_DO_RTCP;
|
|
gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
|
|
src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
|
|
src->user_id = g_strdup (DEFAULT_USER_ID);
|
|
src->user_pw = g_strdup (DEFAULT_USER_PW);
|
|
src->buffer_mode = DEFAULT_BUFFER_MODE;
|
|
src->client_port_range.min = 0;
|
|
src->client_port_range.max = 0;
|
|
src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
|
|
src->short_header = DEFAULT_SHORT_HEADER;
|
|
|
|
/* get a list of all extensions */
|
|
src->extensions = gst_rtsp_ext_list_get ();
|
|
|
|
/* connect to send signal */
|
|
gst_rtsp_ext_list_connect (src->extensions, "send",
|
|
(GCallback) gst_rtspsrc_send_cb, src);
|
|
|
|
/* protects the streaming thread in interleaved mode or the polling
|
|
* thread in UDP mode. */
|
|
src->stream_rec_lock = g_new (GStaticRecMutex, 1);
|
|
g_static_rec_mutex_init (src->stream_rec_lock);
|
|
|
|
/* protects our state changes from multiple invocations */
|
|
src->state_rec_lock = g_new (GStaticRecMutex, 1);
|
|
g_static_rec_mutex_init (src->state_rec_lock);
|
|
|
|
src->state = GST_RTSP_STATE_INVALID;
|
|
|
|
GST_OBJECT_FLAG_SET (src, GST_ELEMENT_IS_SOURCE);
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_finalize (GObject * object)
|
|
{
|
|
GstRTSPSrc *rtspsrc;
|
|
|
|
rtspsrc = GST_RTSPSRC (object);
|
|
|
|
gst_rtsp_ext_list_free (rtspsrc->extensions);
|
|
g_free (rtspsrc->conninfo.location);
|
|
gst_rtsp_url_free (rtspsrc->conninfo.url);
|
|
g_free (rtspsrc->conninfo.url_str);
|
|
g_free (rtspsrc->user_id);
|
|
g_free (rtspsrc->user_pw);
|
|
|
|
if (rtspsrc->sdp) {
|
|
gst_sdp_message_free (rtspsrc->sdp);
|
|
rtspsrc->sdp = NULL;
|
|
}
|
|
|
|
/* free locks */
|
|
g_static_rec_mutex_free (rtspsrc->stream_rec_lock);
|
|
g_free (rtspsrc->stream_rec_lock);
|
|
g_static_rec_mutex_free (rtspsrc->state_rec_lock);
|
|
g_free (rtspsrc->state_rec_lock);
|
|
|
|
#ifdef G_OS_WIN32
|
|
WSACleanup ();
|
|
#endif
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
/* a proxy string of the format [user:passwd@]host[:port] */
|
|
static gboolean
|
|
gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
|
|
{
|
|
gchar *p, *at, *col;
|
|
|
|
g_free (rtsp->proxy_user);
|
|
rtsp->proxy_user = NULL;
|
|
g_free (rtsp->proxy_passwd);
|
|
rtsp->proxy_passwd = NULL;
|
|
g_free (rtsp->proxy_host);
|
|
rtsp->proxy_host = NULL;
|
|
rtsp->proxy_port = 0;
|
|
|
|
p = (gchar *) proxy;
|
|
|
|
if (p == NULL)
|
|
return TRUE;
|
|
|
|
/* we allow http:// in front but ignore it */
|
|
if (g_str_has_prefix (p, "http://"))
|
|
p += 7;
|
|
|
|
at = strchr (p, '@');
|
|
if (at) {
|
|
/* look for user:passwd */
|
|
col = strchr (proxy, ':');
|
|
if (col == NULL || col > at)
|
|
return FALSE;
|
|
|
|
rtsp->proxy_user = g_strndup (p, col - p);
|
|
col++;
|
|
rtsp->proxy_passwd = g_strndup (col, at - col);
|
|
|
|
/* move to host */
|
|
p = at + 1;
|
|
}
|
|
col = strchr (p, ':');
|
|
|
|
if (col) {
|
|
/* everything before the colon is the hostname */
|
|
rtsp->proxy_host = g_strndup (p, col - p);
|
|
p = col + 1;
|
|
rtsp->proxy_port = strtoul (p, (char **) &p, 10);
|
|
} else {
|
|
rtsp->proxy_host = g_strdup (p);
|
|
rtsp->proxy_port = 8080;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
|
|
{
|
|
rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
|
|
rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
|
|
|
|
if (timeout != 0)
|
|
rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
|
|
else
|
|
rtspsrc->ptcp_timeout = NULL;
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstRTSPSrc *rtspsrc;
|
|
|
|
rtspsrc = GST_RTSPSRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LOCATION:
|
|
gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
|
|
g_value_get_string (value));
|
|
break;
|
|
case PROP_PROTOCOLS:
|
|
rtspsrc->protocols = g_value_get_flags (value);
|
|
break;
|
|
case PROP_DEBUG:
|
|
rtspsrc->debug = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_RETRY:
|
|
rtspsrc->retry = g_value_get_uint (value);
|
|
break;
|
|
case PROP_TIMEOUT:
|
|
rtspsrc->udp_timeout = g_value_get_uint64 (value);
|
|
break;
|
|
case PROP_TCP_TIMEOUT:
|
|
gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
|
|
break;
|
|
case PROP_LATENCY:
|
|
rtspsrc->latency = g_value_get_uint (value);
|
|
break;
|
|
case PROP_CONNECTION_SPEED:
|
|
rtspsrc->connection_speed = g_value_get_uint (value);
|
|
break;
|
|
case PROP_NAT_METHOD:
|
|
rtspsrc->nat_method = g_value_get_enum (value);
|
|
break;
|
|
case PROP_DO_RTCP:
|
|
rtspsrc->do_rtcp = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_PROXY:
|
|
gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
|
|
break;
|
|
case PROP_RTP_BLOCKSIZE:
|
|
rtspsrc->rtp_blocksize = g_value_get_uint (value);
|
|
break;
|
|
case PROP_USER_ID:
|
|
if (rtspsrc->user_id)
|
|
g_free (rtspsrc->user_id);
|
|
rtspsrc->user_id = g_value_dup_string (value);
|
|
break;
|
|
case PROP_USER_PW:
|
|
if (rtspsrc->user_pw)
|
|
g_free (rtspsrc->user_pw);
|
|
rtspsrc->user_pw = g_value_dup_string (value);
|
|
break;
|
|
case PROP_BUFFER_MODE:
|
|
rtspsrc->buffer_mode = g_value_get_enum (value);
|
|
break;
|
|
case PROP_PORT_RANGE:
|
|
{
|
|
const gchar *str;
|
|
|
|
str = g_value_get_string (value);
|
|
if (str) {
|
|
sscanf (str, "%u-%u",
|
|
&rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
|
|
} else {
|
|
rtspsrc->client_port_range.min = 0;
|
|
rtspsrc->client_port_range.max = 0;
|
|
}
|
|
break;
|
|
}
|
|
case PROP_UDP_BUFFER_SIZE:
|
|
rtspsrc->udp_buffer_size = g_value_get_int (value);
|
|
break;
|
|
case PROP_SHORT_HEADER:
|
|
rtspsrc->short_header = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstRTSPSrc *rtspsrc;
|
|
|
|
rtspsrc = GST_RTSPSRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LOCATION:
|
|
g_value_set_string (value, rtspsrc->conninfo.location);
|
|
break;
|
|
case PROP_PROTOCOLS:
|
|
g_value_set_flags (value, rtspsrc->protocols);
|
|
break;
|
|
case PROP_DEBUG:
|
|
g_value_set_boolean (value, rtspsrc->debug);
|
|
break;
|
|
case PROP_RETRY:
|
|
g_value_set_uint (value, rtspsrc->retry);
|
|
break;
|
|
case PROP_TIMEOUT:
|
|
g_value_set_uint64 (value, rtspsrc->udp_timeout);
|
|
break;
|
|
case PROP_TCP_TIMEOUT:
|
|
{
|
|
guint64 timeout;
|
|
|
|
timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
|
|
rtspsrc->tcp_timeout.tv_usec;
|
|
g_value_set_uint64 (value, timeout);
|
|
break;
|
|
}
|
|
case PROP_LATENCY:
|
|
g_value_set_uint (value, rtspsrc->latency);
|
|
break;
|
|
case PROP_CONNECTION_SPEED:
|
|
g_value_set_uint (value, rtspsrc->connection_speed);
|
|
break;
|
|
case PROP_NAT_METHOD:
|
|
g_value_set_enum (value, rtspsrc->nat_method);
|
|
break;
|
|
case PROP_DO_RTCP:
|
|
g_value_set_boolean (value, rtspsrc->do_rtcp);
|
|
break;
|
|
case PROP_PROXY:
|
|
{
|
|
gchar *str;
|
|
|
|
if (rtspsrc->proxy_host) {
|
|
str =
|
|
g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
|
|
} else {
|
|
str = NULL;
|
|
}
|
|
g_value_take_string (value, str);
|
|
break;
|
|
}
|
|
case PROP_RTP_BLOCKSIZE:
|
|
g_value_set_uint (value, rtspsrc->rtp_blocksize);
|
|
break;
|
|
case PROP_USER_ID:
|
|
g_value_set_string (value, rtspsrc->user_id);
|
|
break;
|
|
case PROP_USER_PW:
|
|
g_value_set_string (value, rtspsrc->user_pw);
|
|
break;
|
|
case PROP_BUFFER_MODE:
|
|
g_value_set_enum (value, rtspsrc->buffer_mode);
|
|
break;
|
|
case PROP_PORT_RANGE:
|
|
{
|
|
gchar *str;
|
|
|
|
if (rtspsrc->client_port_range.min != 0) {
|
|
str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
|
|
rtspsrc->client_port_range.max);
|
|
} else {
|
|
str = NULL;
|
|
}
|
|
g_value_take_string (value, str);
|
|
break;
|
|
}
|
|
case PROP_UDP_BUFFER_SIZE:
|
|
g_value_set_int (value, rtspsrc->udp_buffer_size);
|
|
break;
|
|
case PROP_SHORT_HEADER:
|
|
g_value_set_boolean (value, rtspsrc->short_header);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gint
|
|
find_stream_by_id (GstRTSPStream * stream, gint * id)
|
|
{
|
|
if (stream->id == *id)
|
|
return 0;
|
|
|
|
return -1;
|
|
}
|
|
|
|
static gint
|
|
find_stream_by_channel (GstRTSPStream * stream, gint * channel)
|
|
{
|
|
if (stream->channel[0] == *channel || stream->channel[1] == *channel)
|
|
return 0;
|
|
|
|
return -1;
|
|
}
|
|
|
|
static gint
|
|
find_stream_by_pt (GstRTSPStream * stream, gint * pt)
|
|
{
|
|
if (stream->pt == *pt)
|
|
return 0;
|
|
|
|
return -1;
|
|
}
|
|
|
|
static gint
|
|
find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
|
|
{
|
|
GstElement *src = (GstElement *) a;
|
|
|
|
if (stream->udpsrc[0] == src)
|
|
return 0;
|
|
if (stream->udpsrc[1] == src)
|
|
return 0;
|
|
|
|
return -1;
|
|
}
|
|
|
|
static gint
|
|
find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
|
|
{
|
|
/* check qualified setup_url */
|
|
if (!strcmp (stream->conninfo.location, (gchar *) a))
|
|
return 0;
|
|
/* check original control_url */
|
|
if (!strcmp (stream->control_url, (gchar *) a))
|
|
return 0;
|
|
|
|
/* check if qualified setup_url ends with string */
|
|
if (g_str_has_suffix (stream->control_url, (gchar *) a))
|
|
return 0;
|
|
|
|
return -1;
|
|
}
|
|
|
|
static GstRTSPStream *
|
|
find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
|
|
{
|
|
GList *lstream;
|
|
|
|
/* find and get stream */
|
|
if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
|
|
return (GstRTSPStream *) lstream->data;
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static const GstSDPBandwidth *
|
|
gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
|
|
const GstSDPMedia * media, const gchar * type)
|
|
{
|
|
guint i, len;
|
|
|
|
/* first look in the media specific section */
|
|
len = gst_sdp_media_bandwidths_len (media);
|
|
for (i = 0; i < len; i++) {
|
|
const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
|
|
|
|
if (strcmp (bw->bwtype, type) == 0)
|
|
return bw;
|
|
}
|
|
/* then look in the message specific section */
|
|
len = gst_sdp_message_bandwidths_len (sdp);
|
|
for (i = 0; i < len; i++) {
|
|
const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
|
|
|
|
if (strcmp (bw->bwtype, type) == 0)
|
|
return bw;
|
|
}
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
|
|
const GstSDPMedia * media, GstRTSPStream * stream)
|
|
{
|
|
const GstSDPBandwidth *bw;
|
|
|
|
if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
|
|
stream->as_bandwidth = bw->bandwidth;
|
|
else
|
|
stream->as_bandwidth = -1;
|
|
|
|
if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
|
|
stream->rr_bandwidth = bw->bandwidth;
|
|
else
|
|
stream->rr_bandwidth = -1;
|
|
|
|
if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
|
|
stream->rs_bandwidth = bw->bandwidth;
|
|
else
|
|
stream->rs_bandwidth = -1;
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
|
|
const GstSDPConnection * conn)
|
|
{
|
|
if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
|
|
return;
|
|
|
|
if (conn->addrtype == NULL)
|
|
return;
|
|
|
|
/* check for IPV6 */
|
|
if (strcmp (conn->addrtype, "IP4") == 0)
|
|
stream->is_ipv6 = FALSE;
|
|
else if (strcmp (conn->addrtype, "IP6") == 0)
|
|
stream->is_ipv6 = TRUE;
|
|
else
|
|
return;
|
|
|
|
/* save address */
|
|
g_free (stream->destination);
|
|
stream->destination = g_strdup (conn->address);
|
|
|
|
/* check for multicast */
|
|
stream->is_multicast =
|
|
gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
|
|
conn->address);
|
|
stream->ttl = conn->ttl;
|
|
}
|
|
|
|
/* Go over the connections for a stream.
|
|
* - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
|
|
* receiving.
|
|
* - If we are dealing with a localhost address, we disable multicast
|
|
*/
|
|
static void
|
|
gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
|
|
const GstSDPMedia * media, GstRTSPStream * stream)
|
|
{
|
|
const GstSDPConnection *conn;
|
|
guint i, len;
|
|
|
|
/* first look in the media specific section */
|
|
len = gst_sdp_media_connections_len (media);
|
|
for (i = 0; i < len; i++) {
|
|
conn = gst_sdp_media_get_connection (media, i);
|
|
|
|
gst_rtspsrc_do_stream_connection (src, stream, conn);
|
|
}
|
|
/* then look in the message specific section */
|
|
if ((conn = gst_sdp_message_get_connection (sdp))) {
|
|
gst_rtspsrc_do_stream_connection (src, stream, conn);
|
|
}
|
|
}
|
|
|
|
static GstRTSPStream *
|
|
gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
|
|
{
|
|
GstRTSPStream *stream;
|
|
const gchar *control_url;
|
|
const gchar *payload;
|
|
const GstSDPMedia *media;
|
|
|
|
/* get media, should not return NULL */
|
|
media = gst_sdp_message_get_media (sdp, idx);
|
|
if (media == NULL)
|
|
return NULL;
|
|
|
|
stream = g_new0 (GstRTSPStream, 1);
|
|
stream->parent = src;
|
|
/* we mark the pad as not linked, we will mark it as OK when we add the pad to
|
|
* the element. */
|
|
stream->last_ret = GST_FLOW_NOT_LINKED;
|
|
stream->added = FALSE;
|
|
stream->disabled = FALSE;
|
|
stream->id = src->numstreams++;
|
|
stream->eos = FALSE;
|
|
stream->discont = TRUE;
|
|
stream->seqbase = -1;
|
|
stream->timebase = -1;
|
|
|
|
/* collect bandwidth information for this steam. FIXME, configure in the RTP
|
|
* session manager to scale RTCP. */
|
|
gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
|
|
|
|
/* collect connection info */
|
|
gst_rtspsrc_collect_connections (src, sdp, media, stream);
|
|
|
|
/* we must have a payload. No payload means we cannot create caps */
|
|
/* FIXME, handle multiple formats. The problem here is that we just want to
|
|
* take the first available format that we can handle but in order to do that
|
|
* we need to scan for depayloader plugins. Scanning for payloader plugins is
|
|
* also suboptimal because the user maybe just wants to save the raw stream
|
|
* and then we don't care. */
|
|
if ((payload = gst_sdp_media_get_format (media, 0))) {
|
|
stream->pt = atoi (payload);
|
|
/* convert caps */
|
|
stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
|
|
|
|
GST_DEBUG ("mapping sdp session level attributes to caps");
|
|
gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, stream->caps);
|
|
GST_DEBUG ("mapping sdp media level attributes to caps");
|
|
gst_rtspsrc_sdp_attributes_to_caps (media->attributes, stream->caps);
|
|
|
|
if (stream->pt >= 96) {
|
|
/* If we have a dynamic payload type, see if we have a stream with the
|
|
* same payload number. If there is one, they are part of the same
|
|
* container and we only need to add one pad. */
|
|
if (find_stream (src, &stream->pt, (gpointer) find_stream_by_pt)) {
|
|
stream->container = TRUE;
|
|
GST_DEBUG ("found another stream with pt %d, marking as container",
|
|
stream->pt);
|
|
}
|
|
}
|
|
}
|
|
/* collect port number */
|
|
stream->port = gst_sdp_media_get_port (media);
|
|
|
|
/* get control url to construct the setup url. The setup url is used to
|
|
* configure the transport of the stream and is used to identity the stream in
|
|
* the RTP-Info header field returned from PLAY. */
|
|
control_url = gst_sdp_media_get_attribute_val (media, "control");
|
|
if (control_url == NULL)
|
|
control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
|
|
|
|
GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
|
|
GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
|
|
GST_DEBUG_OBJECT (src, " port: %d", stream->port);
|
|
GST_DEBUG_OBJECT (src, " container: %d", stream->container);
|
|
GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
|
|
GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
|
|
|
|
if (control_url != NULL) {
|
|
stream->control_url = g_strdup (control_url);
|
|
/* Build a fully qualified url using the content_base if any or by prefixing
|
|
* the original request.
|
|
* If the control_url starts with a '/' or a non rtsp: protocol we will most
|
|
* likely build a URL that the server will fail to understand, this is ok,
|
|
* we will fail then. */
|
|
if (g_str_has_prefix (control_url, "rtsp://"))
|
|
stream->conninfo.location = g_strdup (control_url);
|
|
else {
|
|
const gchar *base;
|
|
gboolean has_slash;
|
|
|
|
if (g_strcmp0 (control_url, "*") == 0)
|
|
control_url = "";
|
|
|
|
if (src->control)
|
|
base = src->control;
|
|
else if (src->content_base)
|
|
base = src->content_base;
|
|
else if (src->conninfo.url_str)
|
|
base = src->conninfo.url_str;
|
|
else
|
|
base = "/";
|
|
|
|
/* check if the base ends or control starts with / */
|
|
has_slash = g_str_has_prefix (control_url, "/");
|
|
has_slash = has_slash || g_str_has_suffix (base, "/");
|
|
|
|
/* concatenate the two strings, insert / when not present */
|
|
stream->conninfo.location =
|
|
g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
|
|
}
|
|
}
|
|
GST_DEBUG_OBJECT (src, " setup: %s",
|
|
GST_STR_NULL (stream->conninfo.location));
|
|
|
|
/* we keep track of all streams */
|
|
src->streams = g_list_append (src->streams, stream);
|
|
|
|
return stream;
|
|
|
|
/* ERRORS */
|
|
}
|
|
|
|
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->destination);
|
|
g_free (stream->control_url);
|
|
g_free (stream->conninfo.location);
|
|
|
|
for (i = 0; i < 2; i++) {
|
|
if (stream->udpsrc[i]) {
|
|
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
|
|
gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
|
|
gst_object_unref (stream->udpsrc[i]);
|
|
stream->udpsrc[i] = NULL;
|
|
}
|
|
if (stream->channelpad[i]) {
|
|
gst_object_unref (stream->channelpad[i]);
|
|
stream->channelpad[i] = NULL;
|
|
}
|
|
if (stream->udpsink[i]) {
|
|
gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
|
|
gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
|
|
gst_object_unref (stream->udpsink[i]);
|
|
stream->udpsink[i] = NULL;
|
|
}
|
|
}
|
|
if (stream->fakesrc) {
|
|
gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
|
|
gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
|
|
gst_object_unref (stream->fakesrc);
|
|
stream->fakesrc = 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;
|
|
}
|
|
if (stream->rtcppad) {
|
|
gst_object_unref (stream->rtcppad);
|
|
stream->rtcppad = NULL;
|
|
}
|
|
if (stream->session) {
|
|
g_object_unref (stream->session);
|
|
stream->session = 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;
|
|
if (src->manager) {
|
|
if (src->manager_sig_id) {
|
|
g_signal_handler_disconnect (src->manager, src->manager_sig_id);
|
|
src->manager_sig_id = 0;
|
|
}
|
|
gst_element_set_state (src->manager, GST_STATE_NULL);
|
|
gst_bin_remove (GST_BIN_CAST (src), src->manager);
|
|
src->manager = NULL;
|
|
}
|
|
src->numstreams = 0;
|
|
if (src->props)
|
|
gst_structure_free (src->props);
|
|
src->props = NULL;
|
|
|
|
g_free (src->content_base);
|
|
src->content_base = NULL;
|
|
|
|
g_free (src->control);
|
|
src->control = NULL;
|
|
|
|
if (src->range)
|
|
gst_rtsp_range_free (src->range);
|
|
src->range = NULL;
|
|
|
|
/* don't clear the SDP when it was used in the url */
|
|
if (src->sdp && !src->from_sdp) {
|
|
gst_sdp_message_free (src->sdp);
|
|
src->sdp = 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; \
|
|
p = t; \
|
|
} \
|
|
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 (const gchar * rtpmap, gint * payload, gchar ** name,
|
|
gint * rate, gchar ** params)
|
|
{
|
|
gchar *p, *t;
|
|
|
|
p = (gchar *) 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) {
|
|
GST_DEBUG ("no rate, name %s", p);
|
|
/* no rate, assume -1 then, this is not supposed to happen but RealMedia
|
|
* streams seem to omit the rate. */
|
|
*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 SDP attributes to caps
|
|
*
|
|
* prepend 'a-' to IANA registered sdp attributes names
|
|
* (ie: not prefixed with 'x-') in order to avoid
|
|
* collision with gstreamer standard caps properties names
|
|
*/
|
|
static void
|
|
gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
|
|
{
|
|
if (attributes->len > 0) {
|
|
GstStructure *s;
|
|
guint i;
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
|
|
for (i = 0; i < attributes->len; i++) {
|
|
GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
|
|
gchar *tofree, *key;
|
|
|
|
key = attr->key;
|
|
|
|
/* skip some of the attribute we already handle */
|
|
if (!strcmp (key, "fmtp"))
|
|
continue;
|
|
if (!strcmp (key, "rtpmap"))
|
|
continue;
|
|
if (!strcmp (key, "control"))
|
|
continue;
|
|
if (!strcmp (key, "range"))
|
|
continue;
|
|
|
|
/* string must be valid UTF8 */
|
|
if (!g_utf8_validate (attr->value, -1, NULL))
|
|
continue;
|
|
|
|
if (!g_str_has_prefix (key, "x-"))
|
|
tofree = key = g_strdup_printf ("a-%s", key);
|
|
else
|
|
tofree = NULL;
|
|
|
|
GST_DEBUG ("adding caps: %s=%s", key, attr->value);
|
|
gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
|
|
g_free (tofree);
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
* 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, const GstSDPMedia * media)
|
|
{
|
|
GstCaps *caps;
|
|
const gchar *rtpmap;
|
|
const gchar *fmtp;
|
|
gchar *name = NULL;
|
|
gint rate = -1;
|
|
gchar *params = NULL;
|
|
gchar *tmp;
|
|
GstStructure *s;
|
|
gint payload = 0;
|
|
gboolean ret;
|
|
|
|
/* get and parse rtpmap */
|
|
if ((rtpmap = gst_sdp_media_get_attribute_val (media, "rtpmap"))) {
|
|
ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
|
|
if (ret) {
|
|
if (payload != pt) {
|
|
/* we ignore the rtpmap if the payload type is different. */
|
|
g_warning ("rtpmap of wrong payload type, ignoring");
|
|
name = NULL;
|
|
rate = -1;
|
|
params = NULL;
|
|
}
|
|
} else {
|
|
/* if we failed to parse the rtpmap for a dynamic payload type, we have an
|
|
* error */
|
|
if (pt >= 96)
|
|
goto no_rtpmap;
|
|
/* else we can ignore */
|
|
g_warning ("error parsing rtpmap, ignoring");
|
|
}
|
|
} else {
|
|
/* dynamic payloads need rtpmap or we fail */
|
|
if (pt >= 96)
|
|
goto no_rtpmap;
|
|
}
|
|
/* check if we have a rate, if not, we need to look up the rate from the
|
|
* default rates based on the payload types. */
|
|
if (rate == -1) {
|
|
const GstRTPPayloadInfo *info;
|
|
|
|
if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
|
|
/* dynamic types, use media and encoding_name */
|
|
tmp = g_ascii_strdown (media->media, -1);
|
|
info = gst_rtp_payload_info_for_name (tmp, name);
|
|
g_free (tmp);
|
|
} else {
|
|
/* static types, use payload type */
|
|
info = gst_rtp_payload_info_for_pt (pt);
|
|
}
|
|
|
|
if (info) {
|
|
if ((rate = info->clock_rate) == 0)
|
|
rate = -1;
|
|
}
|
|
/* we fail if we cannot find one */
|
|
if (rate == -1)
|
|
goto no_rate;
|
|
}
|
|
|
|
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);
|
|
|
|
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 = gst_sdp_media_get_attribute_val (media, "fmtp"))) {
|
|
gchar *p;
|
|
gint payload = 0;
|
|
|
|
p = (gchar *) 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;
|
|
const 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 ';'. We split the pairs at ;
|
|
* boundaries so we can take the remainder of the value. Some servers
|
|
* put spaces around the value which we strip off here. Alternatively
|
|
* we could strip those spaces in the depayloaders should these spaces
|
|
* actually carry any meaning in the future. */
|
|
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]);
|
|
if (strlen (key) > 1) {
|
|
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;
|
|
}
|
|
no_rate:
|
|
{
|
|
g_warning ("rate unknown for payload type %d", pt);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
|
|
gint * rtpport, gint * rtcpport)
|
|
{
|
|
GstRTSPSrc *src;
|
|
GstStateChangeReturn ret;
|
|
GstElement *udpsrc0, *udpsrc1;
|
|
gint tmp_rtp, tmp_rtcp;
|
|
guint count;
|
|
const gchar *host;
|
|
|
|
src = stream->parent;
|
|
|
|
udpsrc0 = NULL;
|
|
udpsrc1 = NULL;
|
|
count = 0;
|
|
|
|
/* Start at next port */
|
|
tmp_rtp = src->next_port_num;
|
|
|
|
if (stream->is_ipv6)
|
|
host = "udp://[::0]";
|
|
else
|
|
host = "udp://0.0.0.0";
|
|
|
|
/* 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:
|
|
|
|
if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
|
|
tmp_rtp >= src->client_port_range.max)
|
|
goto no_ports;
|
|
|
|
udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
|
|
if (udpsrc0 == NULL)
|
|
goto no_udp_protocol;
|
|
g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
|
|
|
|
if (src->udp_buffer_size != 0)
|
|
g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
|
|
NULL);
|
|
|
|
ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
|
|
if (ret == GST_STATE_CHANGE_FAILURE) {
|
|
if (tmp_rtp != 0) {
|
|
GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
|
|
|
|
tmp_rtp += 2;
|
|
if (++count > src->retry)
|
|
goto no_ports;
|
|
|
|
GST_DEBUG_OBJECT (src, "free RTP udpsrc");
|
|
gst_element_set_state (udpsrc0, GST_STATE_NULL);
|
|
gst_object_unref (udpsrc0);
|
|
|
|
GST_DEBUG_OBJECT (src, "retry %d", count);
|
|
goto again;
|
|
}
|
|
goto no_udp_protocol;
|
|
}
|
|
|
|
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 */
|
|
if (++count > src->retry)
|
|
goto no_ports;
|
|
|
|
GST_DEBUG_OBJECT (src, "RTP port not even");
|
|
|
|
GST_DEBUG_OBJECT (src, "free RTP udpsrc");
|
|
gst_element_set_state (udpsrc0, GST_STATE_NULL);
|
|
gst_object_unref (udpsrc0);
|
|
|
|
GST_DEBUG_OBJECT (src, "retry %d", count);
|
|
tmp_rtp++;
|
|
goto again;
|
|
}
|
|
|
|
/* allocate port+1 for RTCP now */
|
|
udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
|
|
if (udpsrc1 == NULL)
|
|
goto no_udp_rtcp_protocol;
|
|
|
|
/* set port */
|
|
tmp_rtcp = tmp_rtp + 1;
|
|
if (src->client_port_range.max > 0 && tmp_rtcp >= src->client_port_range.max)
|
|
goto no_ports;
|
|
|
|
g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
|
|
|
|
GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
|
|
ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
|
|
/* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
|
|
if (ret == GST_STATE_CHANGE_FAILURE) {
|
|
GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
|
|
|
|
if (++count > src->retry)
|
|
goto no_ports;
|
|
|
|
GST_DEBUG_OBJECT (src, "free RTP udpsrc");
|
|
gst_element_set_state (udpsrc0, GST_STATE_NULL);
|
|
gst_object_unref (udpsrc0);
|
|
|
|
GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
|
|
gst_element_set_state (udpsrc1, GST_STATE_NULL);
|
|
gst_object_unref (udpsrc1);
|
|
udpsrc1 = NULL;
|
|
|
|
tmp_rtp += 2;
|
|
GST_DEBUG_OBJECT (src, "retry %d", count);
|
|
goto again;
|
|
}
|
|
|
|
/* all fine, do port check */
|
|
g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
|
|
g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
|
|
|
|
/* 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);
|
|
|
|
/* keep track of next available port number when we have a range
|
|
* configured */
|
|
if (src->next_port_num != 0)
|
|
src->next_port_num = tmp_rtcp + 1;
|
|
|
|
/* 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;
|
|
}
|
|
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;
|
|
}
|
|
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 (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
|
|
gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
|
|
{
|
|
GstEvent *event;
|
|
gint cmd, i;
|
|
GstState state;
|
|
GList *walk;
|
|
GstClock *clock;
|
|
GstClockTime base_time = GST_CLOCK_TIME_NONE;
|
|
|
|
if (flush) {
|
|
event = gst_event_new_flush_start ();
|
|
GST_DEBUG_OBJECT (src, "start flush");
|
|
cmd = CMD_WAIT;
|
|
state = GST_STATE_PAUSED;
|
|
} else {
|
|
event = gst_event_new_flush_stop ();
|
|
GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
|
|
cmd = CMD_LOOP;
|
|
if (playing)
|
|
state = GST_STATE_PLAYING;
|
|
else
|
|
state = GST_STATE_PAUSED;
|
|
clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
|
|
if (clock) {
|
|
base_time = gst_clock_get_time (clock);
|
|
gst_object_unref (clock);
|
|
}
|
|
}
|
|
gst_rtspsrc_push_event (src, event, FALSE);
|
|
gst_rtspsrc_loop_send_cmd (src, cmd, flush);
|
|
|
|
/* set up manager before data-flow resumes */
|
|
/* to manage jitterbuffer buffer mode */
|
|
if (src->manager) {
|
|
gst_element_set_base_time (GST_ELEMENT_CAST (src->manager), base_time);
|
|
/* and to have base_time trickle further down,
|
|
* e.g. to jitterbuffer for its timeout handling */
|
|
if (base_time != -1)
|
|
gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
|
|
}
|
|
|
|
/* make running time start start at 0 again */
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
|
|
for (i = 0; i < 2; i++) {
|
|
/* for udp case */
|
|
if (stream->udpsrc[i]) {
|
|
if (base_time != -1)
|
|
gst_element_set_base_time (stream->udpsrc[i], base_time);
|
|
gst_element_set_state (stream->udpsrc[i], state);
|
|
}
|
|
}
|
|
}
|
|
/* for tcp interleaved case */
|
|
if (base_time != -1)
|
|
gst_element_set_base_time (GST_ELEMENT_CAST (src), base_time);
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
|
|
GstRTSPMessage * message, GTimeVal * timeout)
|
|
{
|
|
GstRTSPResult ret;
|
|
|
|
if (conn)
|
|
ret = gst_rtsp_connection_send (conn, message, timeout);
|
|
else
|
|
ret = GST_RTSP_ERROR;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
|
|
GstRTSPMessage * message, GTimeVal * timeout)
|
|
{
|
|
GstRTSPResult ret;
|
|
|
|
if (conn)
|
|
ret = gst_rtsp_connection_receive (conn, message, timeout);
|
|
else
|
|
ret = GST_RTSP_ERROR;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_get_position (GstRTSPSrc * src)
|
|
{
|
|
GstQuery *query;
|
|
GList *walk;
|
|
|
|
query = gst_query_new_position (GST_FORMAT_TIME);
|
|
/* should be known somewhere down the stream (e.g. jitterbuffer) */
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
GstFormat fmt;
|
|
gint64 pos;
|
|
|
|
if (stream->srcpad) {
|
|
if (gst_pad_query (stream->srcpad, query)) {
|
|
gst_query_parse_position (query, &fmt, &pos);
|
|
GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (pos));
|
|
src->last_pos = pos;
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
src->last_pos = 0;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
|
|
{
|
|
src->state = GST_RTSP_STATE_SEEKING;
|
|
/* PLAY will add the range header now. */
|
|
src->need_range = TRUE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
|
|
{
|
|
gdouble rate;
|
|
GstFormat format;
|
|
GstSeekFlags flags;
|
|
GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
|
|
gint64 cur, stop;
|
|
gboolean flush, skip;
|
|
gboolean update;
|
|
gboolean playing;
|
|
GstSegment seeksegment = { 0, };
|
|
GList *walk;
|
|
|
|
if (event) {
|
|
GST_DEBUG_OBJECT (src, "doing seek with event");
|
|
|
|
gst_event_parse_seek (event, &rate, &format, &flags,
|
|
&cur_type, &cur, &stop_type, &stop);
|
|
|
|
/* no negative rates yet */
|
|
if (rate < 0.0)
|
|
goto negative_rate;
|
|
|
|
/* we need TIME format */
|
|
if (format != src->segment.format)
|
|
goto no_format;
|
|
} else {
|
|
GST_DEBUG_OBJECT (src, "doing seek without event");
|
|
flags = 0;
|
|
cur_type = GST_SEEK_TYPE_SET;
|
|
stop_type = GST_SEEK_TYPE_SET;
|
|
}
|
|
|
|
/* get flush flag */
|
|
flush = flags & GST_SEEK_FLAG_FLUSH;
|
|
skip = flags & GST_SEEK_FLAG_SKIP;
|
|
|
|
/* now we need to make sure the streaming thread is stopped. We do this by
|
|
* either sending a FLUSH_START event downstream which will cause the
|
|
* streaming thread to stop with a WRONG_STATE.
|
|
* For a non-flushing seek we simply pause the task, which will happen as soon
|
|
* as it completes one iteration (and thus might block when the sink is
|
|
* blocking in preroll). */
|
|
if (flush) {
|
|
GST_DEBUG_OBJECT (src, "starting flush");
|
|
gst_rtspsrc_flush (src, TRUE, FALSE);
|
|
} else {
|
|
if (src->task) {
|
|
gst_task_pause (src->task);
|
|
}
|
|
}
|
|
|
|
/* we should now be able to grab the streaming thread because we stopped it
|
|
* with the above flush/pause code */
|
|
GST_RTSP_STREAM_LOCK (src);
|
|
|
|
GST_DEBUG_OBJECT (src, "stopped streaming");
|
|
|
|
/* copy segment, we need this because we still need the old
|
|
* segment when we close the current segment. */
|
|
memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
|
|
|
|
/* configure the seek parameters in the seeksegment. We will then have the
|
|
* right values in the segment to perform the seek */
|
|
if (event) {
|
|
GST_DEBUG_OBJECT (src, "configuring seek");
|
|
gst_segment_set_seek (&seeksegment, rate, format, flags,
|
|
cur_type, cur, stop_type, stop, &update);
|
|
}
|
|
|
|
/* figure out the last position we need to play. If it's configured (stop !=
|
|
* -1), use that, else we play until the total duration of the file */
|
|
if ((stop = seeksegment.stop) == -1)
|
|
stop = seeksegment.duration;
|
|
|
|
playing = (src->state == GST_RTSP_STATE_PLAYING);
|
|
|
|
/* if we were playing, pause first */
|
|
if (playing) {
|
|
/* obtain current position in case seek fails */
|
|
gst_rtspsrc_get_position (src);
|
|
gst_rtspsrc_pause (src, FALSE, FALSE);
|
|
}
|
|
|
|
gst_rtspsrc_do_seek (src, &seeksegment);
|
|
|
|
/* and continue playing */
|
|
if (playing)
|
|
gst_rtspsrc_play (src, &seeksegment, FALSE);
|
|
|
|
/* prepare for streaming again */
|
|
if (flush) {
|
|
/* if we started flush, we stop now */
|
|
GST_DEBUG_OBJECT (src, "stopping flush");
|
|
gst_rtspsrc_flush (src, FALSE, playing);
|
|
} else if (src->running) {
|
|
/* re-engage loop */
|
|
gst_rtspsrc_loop_send_cmd (src, CMD_LOOP, FALSE);
|
|
|
|
/* we are running the current segment and doing a non-flushing seek,
|
|
* close the segment first based on the previous last_stop. */
|
|
GST_DEBUG_OBJECT (src, "closing running segment %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT, src->segment.accum, src->segment.last_stop);
|
|
|
|
/* queue the segment for sending in the stream thread */
|
|
if (src->close_segment)
|
|
gst_event_unref (src->close_segment);
|
|
src->close_segment = gst_event_new_new_segment (TRUE,
|
|
src->segment.rate, src->segment.format,
|
|
src->segment.accum, src->segment.last_stop, src->segment.accum);
|
|
|
|
/* keep track of our last_stop */
|
|
seeksegment.accum = src->segment.last_stop;
|
|
}
|
|
|
|
/* now we did the seek and can activate the new segment values */
|
|
memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
|
|
|
|
/* if we're doing a segment seek, post a SEGMENT_START message */
|
|
if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
|
|
gst_element_post_message (GST_ELEMENT_CAST (src),
|
|
gst_message_new_segment_start (GST_OBJECT_CAST (src),
|
|
src->segment.format, src->segment.last_stop));
|
|
}
|
|
|
|
/* now create the newsegment */
|
|
GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT, src->segment.last_stop, stop);
|
|
|
|
/* store the newsegment event so it can be sent from the streaming thread. */
|
|
if (src->start_segment)
|
|
gst_event_unref (src->start_segment);
|
|
src->start_segment =
|
|
gst_event_new_new_segment (FALSE, src->segment.rate,
|
|
src->segment.format, src->segment.last_stop, stop,
|
|
src->segment.last_stop);
|
|
|
|
/* mark discont */
|
|
GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
stream->discont = TRUE;
|
|
}
|
|
src->skip = skip;
|
|
|
|
GST_RTSP_STREAM_UNLOCK (src);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
negative_rate:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
|
|
return FALSE;
|
|
}
|
|
no_format:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_handle_src_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstRTSPSrc *src;
|
|
gboolean res = TRUE;
|
|
gboolean forward;
|
|
|
|
src = GST_RTSPSRC_CAST (gst_pad_get_parent (pad));
|
|
|
|
GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
|
|
GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
res = gst_rtspsrc_perform_seek (src, event);
|
|
forward = FALSE;
|
|
break;
|
|
case GST_EVENT_QOS:
|
|
case GST_EVENT_NAVIGATION:
|
|
case GST_EVENT_LATENCY:
|
|
default:
|
|
forward = TRUE;
|
|
break;
|
|
}
|
|
if (forward) {
|
|
GstPad *target;
|
|
|
|
if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
|
|
res = gst_pad_send_event (target, event);
|
|
gst_object_unref (target);
|
|
} else {
|
|
gst_event_unref (event);
|
|
}
|
|
} else {
|
|
gst_event_unref (event);
|
|
}
|
|
gst_object_unref (src);
|
|
|
|
return res;
|
|
}
|
|
|
|
/* this is the final event function we receive on the internal source pad when
|
|
* we deal with TCP connections */
|
|
static gboolean
|
|
gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
gboolean res;
|
|
|
|
GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
case GST_EVENT_QOS:
|
|
case GST_EVENT_NAVIGATION:
|
|
case GST_EVENT_LATENCY:
|
|
default:
|
|
gst_event_unref (event);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
/* this is the final query function we receive on the internal source pad when
|
|
* we deal with TCP connections */
|
|
static gboolean
|
|
gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
GstRTSPSrc *src;
|
|
gboolean res = TRUE;
|
|
|
|
src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
|
|
|
|
GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
|
|
GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
/* no idea */
|
|
break;
|
|
}
|
|
case GST_QUERY_DURATION:
|
|
{
|
|
GstFormat format;
|
|
|
|
gst_query_parse_duration (query, &format, NULL);
|
|
|
|
switch (format) {
|
|
case GST_FORMAT_TIME:
|
|
gst_query_set_duration (query, format, src->segment.duration);
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
/* we are live with a min latency of 0 and unlimited max latency, this
|
|
* result will be updated by the session manager if there is any. */
|
|
gst_query_set_latency (query, TRUE, 0, -1);
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
/* this query is executed on the ghost source pad exposed on rtspsrc. */
|
|
static gboolean
|
|
gst_rtspsrc_handle_src_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
GstRTSPSrc *src;
|
|
gboolean res = FALSE;
|
|
|
|
src = GST_RTSPSRC_CAST (gst_pad_get_parent (pad));
|
|
|
|
GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
|
|
GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_DURATION:
|
|
{
|
|
GstFormat format;
|
|
|
|
gst_query_parse_duration (query, &format, NULL);
|
|
|
|
switch (format) {
|
|
case GST_FORMAT_TIME:
|
|
gst_query_set_duration (query, format, src->segment.duration);
|
|
res = TRUE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_SEEKING:
|
|
{
|
|
GstFormat format;
|
|
|
|
gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
|
|
if (format == GST_FORMAT_TIME) {
|
|
gboolean seekable =
|
|
src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
|
|
|
|
/* seeking without duration is unlikely */
|
|
seekable = seekable && src->seekable && src->segment.duration &&
|
|
GST_CLOCK_TIME_IS_VALID (src->segment.duration);
|
|
|
|
/* FIXME ?? should we have 0 and segment.duration here; see demuxers */
|
|
gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
|
|
src->segment.start, src->segment.stop);
|
|
res = TRUE;
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
{
|
|
GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
|
|
|
|
/* forward the query to the proxy target pad */
|
|
if (target) {
|
|
res = gst_pad_query (target, query);
|
|
gst_object_unref (target);
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
gst_object_unref (src);
|
|
|
|
return res;
|
|
}
|
|
|
|
/* callback for RTCP messages to be sent to the server when operating in TCP
|
|
* mode. */
|
|
static GstFlowReturn
|
|
gst_rtspsrc_sink_chain (GstPad * pad, GstBuffer * buffer)
|
|
{
|
|
GstRTSPSrc *src;
|
|
GstRTSPStream *stream;
|
|
GstFlowReturn res = GST_FLOW_OK;
|
|
guint8 *data;
|
|
guint size;
|
|
GstRTSPResult ret;
|
|
GstRTSPMessage message = { 0 };
|
|
GstRTSPConnection *conn;
|
|
|
|
stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
|
|
src = stream->parent;
|
|
|
|
data = GST_BUFFER_DATA (buffer);
|
|
size = GST_BUFFER_SIZE (buffer);
|
|
|
|
gst_rtsp_message_init_data (&message, stream->channel[1]);
|
|
|
|
/* lend the body data to the message */
|
|
gst_rtsp_message_take_body (&message, data, size);
|
|
|
|
if (stream->conninfo.connection)
|
|
conn = stream->conninfo.connection;
|
|
else
|
|
conn = src->conninfo.connection;
|
|
|
|
GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
|
|
ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
|
|
GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
|
|
|
|
/* and steal it away again because we will free it when unreffing the
|
|
* buffer */
|
|
gst_rtsp_message_steal_body (&message, &data, &size);
|
|
gst_rtsp_message_unset (&message);
|
|
|
|
gst_buffer_unref (buffer);
|
|
|
|
return res;
|
|
}
|
|
|
|
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;
|
|
}
|
|
}
|
|
|
|
/* this callback is called when the session manager generated a new src pad with
|
|
* payloaded RTP packets. We simply ghost the pad here. */
|
|
static void
|
|
new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
|
|
{
|
|
gchar *name;
|
|
GstPadTemplate *template;
|
|
gint id, ssrc, pt;
|
|
GList *lstream;
|
|
GstRTSPStream *stream;
|
|
gboolean all_added;
|
|
|
|
GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
|
|
|
|
GST_RTSP_STATE_LOCK (src);
|
|
/* find stream */
|
|
name = gst_object_get_name (GST_OBJECT_CAST (pad));
|
|
if (sscanf (name, "recv_rtp_src_%d_%d_%d", &id, &ssrc, &pt) != 3)
|
|
goto unknown_stream;
|
|
|
|
GST_DEBUG_OBJECT (src, "stream: %u, SSRC %d, PT %d", id, ssrc, pt);
|
|
|
|
stream = find_stream (src, &id, (gpointer) find_stream_by_id);
|
|
if (stream == NULL)
|
|
goto unknown_stream;
|
|
|
|
/* create a new pad we will use to stream to */
|
|
template = gst_static_pad_template_get (&rtptemplate);
|
|
stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
|
|
gst_object_unref (template);
|
|
g_free (name);
|
|
|
|
stream->added = TRUE;
|
|
gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
|
|
gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
|
|
gst_pad_set_active (stream->srcpad, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
|
|
|
|
/* check if we added all streams */
|
|
all_added = TRUE;
|
|
for (lstream = src->streams; lstream; lstream = g_list_next (lstream)) {
|
|
stream = (GstRTSPStream *) lstream->data;
|
|
|
|
GST_DEBUG_OBJECT (src, "stream %p, container %d, disabled %d, added %d",
|
|
stream, stream->container, stream->disabled, stream->added);
|
|
|
|
/* a container stream only needs one pad added. Also disabled streams don't
|
|
* count */
|
|
if (!stream->container && !stream->disabled && !stream->added) {
|
|
all_added = FALSE;
|
|
break;
|
|
}
|
|
}
|
|
GST_RTSP_STATE_UNLOCK (src);
|
|
|
|
if (all_added) {
|
|
GST_DEBUG_OBJECT (src, "We added all streams");
|
|
/* when we get here, all stream are added and we can fire the no-more-pads
|
|
* signal. */
|
|
gst_element_no_more_pads (GST_ELEMENT_CAST (src));
|
|
}
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
unknown_stream:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "ignoring unknown stream");
|
|
GST_RTSP_STATE_UNLOCK (src);
|
|
g_free (name);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static GstCaps *
|
|
request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
|
|
{
|
|
GstRTSPStream *stream;
|
|
GstCaps *caps;
|
|
|
|
GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
|
|
|
|
GST_RTSP_STATE_LOCK (src);
|
|
stream = find_stream (src, &session, (gpointer) find_stream_by_id);
|
|
if (!stream)
|
|
goto unknown_stream;
|
|
|
|
caps = stream->caps;
|
|
if (caps)
|
|
gst_caps_ref (caps);
|
|
GST_RTSP_STATE_UNLOCK (src);
|
|
|
|
return caps;
|
|
|
|
unknown_stream:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "unknown stream %d", session);
|
|
GST_RTSP_STATE_UNLOCK (src);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
|
|
{
|
|
GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
|
|
|
|
if (stream->eos)
|
|
goto was_eos;
|
|
|
|
stream->eos = TRUE;
|
|
gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos (), TRUE);
|
|
return;
|
|
|
|
/* ERRORS */
|
|
was_eos:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void
|
|
on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
|
|
{
|
|
GstRTSPSrc *src = stream->parent;
|
|
|
|
GST_DEBUG_OBJECT (src, "source in session %u received BYE", stream->id);
|
|
|
|
gst_rtspsrc_do_stream_eos (src, stream);
|
|
}
|
|
|
|
static void
|
|
on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
|
|
{
|
|
GstRTSPSrc *src = stream->parent;
|
|
|
|
GST_DEBUG_OBJECT (src, "source in session %u timed out", stream->id);
|
|
|
|
gst_rtspsrc_do_stream_eos (src, stream);
|
|
}
|
|
|
|
static void
|
|
on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
|
|
{
|
|
GstRTSPStream *stream;
|
|
|
|
GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
|
|
|
|
/* get stream for session */
|
|
stream = find_stream (src, &session, (gpointer) find_stream_by_id);
|
|
if (stream) {
|
|
gst_rtspsrc_do_stream_eos (src, stream);
|
|
}
|
|
}
|
|
|
|
static void
|
|
on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
|
|
{
|
|
GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
|
|
stream->id);
|
|
}
|
|
|
|
/* try to get and configure a manager */
|
|
static gboolean
|
|
gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
|
|
GstRTSPTransport * transport)
|
|
{
|
|
const gchar *manager;
|
|
gchar *name;
|
|
GstStateChangeReturn ret;
|
|
|
|
/* find a manager */
|
|
if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
|
|
goto no_manager;
|
|
|
|
if (manager) {
|
|
GST_DEBUG_OBJECT (src, "using manager %s", manager);
|
|
|
|
/* configure the manager */
|
|
if (src->manager == NULL) {
|
|
GObjectClass *klass;
|
|
GstState target;
|
|
|
|
if (!(src->manager = gst_element_factory_make (manager, NULL))) {
|
|
/* fallback */
|
|
if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
|
|
goto no_manager;
|
|
|
|
if (!manager)
|
|
goto use_no_manager;
|
|
|
|
if (!(src->manager = gst_element_factory_make (manager, NULL)))
|
|
goto manager_failed;
|
|
}
|
|
|
|
/* we manage this element */
|
|
gst_bin_add (GST_BIN_CAST (src), src->manager);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
target = GST_STATE_TARGET (src);
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
ret = gst_element_set_state (src->manager, target);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
goto start_manager_failure;
|
|
|
|
g_object_set (src->manager, "latency", src->latency, NULL);
|
|
|
|
klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
|
|
if (g_object_class_find_property (klass, "buffer-mode")) {
|
|
if (src->buffer_mode != BUFFER_MODE_AUTO) {
|
|
g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
|
|
} else {
|
|
gboolean need_slave;
|
|
GstStructure *s;
|
|
const gchar *encoding;
|
|
|
|
/* buffer mode pauses are handled by adding offsets to buffer times,
|
|
* but some depayloaders may have a hard time syncing output times
|
|
* with such input times, e.g. container ones, most notably ASF */
|
|
/* TODO alternatives are having an event that indicates these shifts,
|
|
* or having rtsp extensions provide suggestion on buffer mode */
|
|
need_slave = stream->container;
|
|
if (stream->caps && (s = gst_caps_get_structure (stream->caps, 0)) &&
|
|
(encoding = gst_structure_get_string (s, "encoding-name")))
|
|
need_slave = need_slave || (strcmp (encoding, "X-ASF-PF") == 0);
|
|
GST_DEBUG_OBJECT (src, "auto buffering mode, need_slave %d",
|
|
need_slave);
|
|
/* valid duration implies not likely live pipeline,
|
|
* so slaving in jitterbuffer does not make much sense
|
|
* (and might mess things up due to bursts) */
|
|
if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
|
|
src->segment.duration && !need_slave) {
|
|
GST_DEBUG_OBJECT (src, "selected buffer");
|
|
g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER,
|
|
NULL);
|
|
} else {
|
|
GST_DEBUG_OBJECT (src, "selected slave");
|
|
g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* connect to signals if we did not already do so */
|
|
GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
|
|
stream);
|
|
src->manager_sig_id =
|
|
g_signal_connect (src->manager, "pad-added",
|
|
(GCallback) new_manager_pad, src);
|
|
src->manager_ptmap_id =
|
|
g_signal_connect (src->manager, "request-pt-map",
|
|
(GCallback) request_pt_map, src);
|
|
|
|
g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
|
|
src);
|
|
}
|
|
|
|
/* we stream directly to the manager, get some pads. Each RTSP stream goes
|
|
* into a separate RTP session. */
|
|
name = g_strdup_printf ("recv_rtp_sink_%d", stream->id);
|
|
stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
|
|
g_free (name);
|
|
name = g_strdup_printf ("recv_rtcp_sink_%d", stream->id);
|
|
stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
|
|
g_free (name);
|
|
|
|
/* now configure the bandwidth in the manager */
|
|
if (g_signal_lookup ("get-internal-session",
|
|
G_OBJECT_TYPE (src->manager)) != 0) {
|
|
GObject *rtpsession;
|
|
|
|
g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
|
|
&rtpsession);
|
|
if (rtpsession) {
|
|
GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
|
|
|
|
stream->session = rtpsession;
|
|
|
|
if (stream->as_bandwidth != -1) {
|
|
GST_INFO_OBJECT (src, "setting AS: %f",
|
|
(gdouble) (stream->as_bandwidth * 1000));
|
|
g_object_set (rtpsession, "bandwidth",
|
|
(gdouble) (stream->as_bandwidth * 1000), NULL);
|
|
}
|
|
if (stream->rr_bandwidth != -1) {
|
|
GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
|
|
g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
|
|
NULL);
|
|
}
|
|
if (stream->rs_bandwidth != -1) {
|
|
GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
|
|
g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
|
|
NULL);
|
|
}
|
|
g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
|
|
stream);
|
|
g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
|
|
stream);
|
|
g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
|
|
stream);
|
|
g_signal_connect (rtpsession, "on-ssrc-active",
|
|
(GCallback) on_ssrc_active, stream);
|
|
}
|
|
}
|
|
}
|
|
|
|
use_no_manager:
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_manager:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "cannot get a session manager");
|
|
return FALSE;
|
|
}
|
|
manager_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
|
|
return FALSE;
|
|
}
|
|
start_manager_failure:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "could not start session manager");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* free the UDP sources allocated when negotiating a transport.
|
|
* This function is called when the server negotiated to a transport where the
|
|
* UDP sources are not needed anymore, such as TCP or multicast. */
|
|
static void
|
|
gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
|
|
{
|
|
gint i;
|
|
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* for TCP, create pads to send and receive data to and from the manager and to
|
|
* intercept various events and queries
|
|
*/
|
|
static gboolean
|
|
gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
|
|
GstRTSPTransport * transport, GstPad ** outpad)
|
|
{
|
|
gchar *name;
|
|
GstPadTemplate *template;
|
|
GstPad *pad0, *pad1;
|
|
|
|
/* 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 */
|
|
gst_rtspsrc_stream_free_udp (stream);
|
|
|
|
/* 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_active (stream->channelpad[0], TRUE);
|
|
|
|
*outpad = gst_object_ref (stream->channelpad[0]);
|
|
} else {
|
|
GST_DEBUG_OBJECT (src, "using manager source pad");
|
|
|
|
template = gst_static_pad_template_get (&anysrctemplate);
|
|
|
|
/* allocate pads for sending the channel data into the manager */
|
|
pad0 = gst_pad_new_from_template (template, "internalsrc0");
|
|
gst_pad_link (pad0, stream->channelpad[0]);
|
|
gst_object_unref (stream->channelpad[0]);
|
|
stream->channelpad[0] = pad0;
|
|
gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
|
|
gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
|
|
gst_pad_set_element_private (pad0, src);
|
|
gst_pad_set_active (pad0, TRUE);
|
|
|
|
if (stream->channelpad[1]) {
|
|
/* if we have a sinkpad for the other channel, create a pad and link to the
|
|
* manager. */
|
|
pad1 = gst_pad_new_from_template (template, "internalsrc1");
|
|
gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
|
|
gst_pad_link (pad1, stream->channelpad[1]);
|
|
gst_object_unref (stream->channelpad[1]);
|
|
stream->channelpad[1] = pad1;
|
|
gst_pad_set_active (pad1, TRUE);
|
|
}
|
|
gst_object_unref (template);
|
|
}
|
|
/* setup RTCP transport back to the server if we have to. */
|
|
if (src->manager && src->do_rtcp) {
|
|
GstPad *pad;
|
|
|
|
template = gst_static_pad_template_get (&anysinktemplate);
|
|
|
|
stream->rtcppad = gst_pad_new_from_template (template, "internalsink0");
|
|
gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
|
|
gst_pad_set_element_private (stream->rtcppad, stream);
|
|
gst_pad_set_active (stream->rtcppad, TRUE);
|
|
|
|
/* get session RTCP pad */
|
|
name = g_strdup_printf ("send_rtcp_src_%d", stream->id);
|
|
pad = gst_element_get_request_pad (src->manager, name);
|
|
g_free (name);
|
|
|
|
/* and link */
|
|
if (pad) {
|
|
gst_pad_link (pad, stream->rtcppad);
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
gst_object_unref (template);
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
|
|
GstRTSPTransport * transport, const gchar ** destination, gint * min,
|
|
gint * max, guint * ttl)
|
|
{
|
|
if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
|
|
if (destination) {
|
|
if (!(*destination = transport->destination))
|
|
*destination = stream->destination;
|
|
}
|
|
if (min && max) {
|
|
/* transport first */
|
|
*min = transport->port.min;
|
|
*max = transport->port.max;
|
|
if (*min == -1 && *max == -1) {
|
|
/* then try from SDP */
|
|
if (stream->port != 0) {
|
|
*min = stream->port;
|
|
*max = stream->port + 1;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (ttl) {
|
|
if (!(*ttl = transport->ttl))
|
|
*ttl = stream->ttl;
|
|
}
|
|
} else {
|
|
if (destination) {
|
|
/* first take the source, then the endpoint to figure out where to send
|
|
* the RTCP. */
|
|
if (!(*destination = transport->source)) {
|
|
if (src->conninfo.connection)
|
|
*destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
|
|
else if (stream->conninfo.connection)
|
|
*destination =
|
|
gst_rtsp_connection_get_ip (stream->conninfo.connection);
|
|
}
|
|
}
|
|
if (min && max) {
|
|
/* for unicast we only expect the ports here */
|
|
*min = transport->server_port.min;
|
|
*max = transport->server_port.max;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* For multicast create UDP sources and join the multicast group. */
|
|
static gboolean
|
|
gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
|
|
GstRTSPTransport * transport, GstPad ** outpad)
|
|
{
|
|
gchar *uri;
|
|
const gchar *destination;
|
|
gint min, max;
|
|
|
|
GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
|
|
|
|
/* we can remove the allocated UDP ports now */
|
|
gst_rtspsrc_stream_free_udp (stream);
|
|
|
|
gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
|
|
&max, NULL);
|
|
|
|
/* we need a destination now */
|
|
if (destination == NULL)
|
|
goto no_destination;
|
|
|
|
/* we really need ports now or we won't be able to receive anything at all */
|
|
if (min == -1 && max == -1)
|
|
goto no_ports;
|
|
|
|
GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
|
|
destination, min, max);
|
|
|
|
/* creating UDP source for RTP */
|
|
if (min != -1) {
|
|
uri = g_strdup_printf ("udp://%s:%d", destination, 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_PAUSED);
|
|
}
|
|
|
|
/* creating another UDP source for RTCP */
|
|
if (max != -1) {
|
|
uri = g_strdup_printf ("udp://%s:%d", destination, 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[1]);
|
|
gst_object_sink (stream->udpsrc[1]);
|
|
|
|
gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
|
|
}
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_element:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no UDP source element found");
|
|
return FALSE;
|
|
}
|
|
no_destination:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no destination found");
|
|
return FALSE;
|
|
}
|
|
no_ports:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no ports found");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* configure the remainder of the UDP ports */
|
|
static gboolean
|
|
gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
|
|
GstRTSPTransport * transport, GstPad ** outpad)
|
|
{
|
|
/* 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");
|
|
|
|
/* 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->udp_timeout,
|
|
NULL);
|
|
|
|
/* get output pad of the UDP source. */
|
|
*outpad = gst_element_get_static_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 (stream->blockedpad, 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);
|
|
*outpad = NULL;
|
|
/* we connected to pad-added signal to get pads from the manager */
|
|
} else {
|
|
GST_DEBUG_OBJECT (src, "using UDP src pad as output");
|
|
}
|
|
}
|
|
|
|
/* RTCP port */
|
|
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_static_pad (stream->udpsrc[1], "src");
|
|
gst_pad_link (pad, stream->channelpad[1]);
|
|
gst_object_unref (pad);
|
|
} else {
|
|
/* leave unlinked */
|
|
}
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
/* configure the UDP sink back to the server for status reports */
|
|
static gboolean
|
|
gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
|
|
GstRTSPStream * stream, GstRTSPTransport * transport)
|
|
{
|
|
GstPad *pad;
|
|
gint rtp_port, rtcp_port, sockfd = -1;
|
|
gboolean do_rtp, do_rtcp;
|
|
const gchar *destination;
|
|
gchar *uri, *name;
|
|
guint ttl = 0;
|
|
|
|
/* get transport info */
|
|
gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
|
|
&rtp_port, &rtcp_port, &ttl);
|
|
|
|
/* see what we need to do */
|
|
do_rtp = (rtp_port != -1);
|
|
/* it's possible that the server does not want us to send RTCP in which case
|
|
* the port is -1 */
|
|
do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
|
|
|
|
/* we need a destination when we have RTP or RTCP ports */
|
|
if (destination == NULL && (do_rtp || do_rtcp))
|
|
goto no_destination;
|
|
|
|
/* try to construct the fakesrc to the RTP port of the server to open up any
|
|
* NAT firewalls */
|
|
if (do_rtp) {
|
|
GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
|
|
rtp_port);
|
|
|
|
uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
|
|
stream->udpsink[0] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
|
|
g_free (uri);
|
|
if (stream->udpsink[0] == NULL)
|
|
goto no_sink_element;
|
|
|
|
/* don't join multicast group, we will have the source socket do that */
|
|
/* no sync or async state changes needed */
|
|
g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
|
|
"loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
|
|
if (ttl > 0)
|
|
g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
|
|
|
|
if (stream->udpsrc[0]) {
|
|
/* configure socket, we give it the same UDP socket as the udpsrc for RTP
|
|
* so that NAT firewalls will open a hole for us */
|
|
g_object_get (G_OBJECT (stream->udpsrc[0]), "sock", &sockfd, NULL);
|
|
GST_DEBUG_OBJECT (src, "RTP UDP src has sock %d", sockfd);
|
|
/* configure socket and make sure udpsink does not close it when shutting
|
|
* down, it belongs to udpsrc after all. */
|
|
g_object_set (G_OBJECT (stream->udpsink[0]), "sockfd", sockfd,
|
|
"closefd", FALSE, NULL);
|
|
}
|
|
|
|
/* the source for the dummy packets to open up NAT */
|
|
stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
|
|
if (stream->fakesrc == NULL)
|
|
goto no_fakesrc_element;
|
|
|
|
/* random data in 5 buffers, a size of 200 bytes should be fine */
|
|
g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
|
|
"sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
|
|
|
|
/* we don't want to consider this a sink */
|
|
GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_IS_SINK);
|
|
|
|
/* keep everything locked */
|
|
gst_element_set_locked_state (stream->udpsink[0], TRUE);
|
|
gst_element_set_locked_state (stream->fakesrc, TRUE);
|
|
|
|
gst_object_ref (stream->udpsink[0]);
|
|
gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
|
|
gst_object_ref (stream->fakesrc);
|
|
gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
|
|
|
|
gst_element_link (stream->fakesrc, stream->udpsink[0]);
|
|
}
|
|
if (do_rtcp) {
|
|
GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
|
|
rtcp_port);
|
|
|
|
uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
|
|
stream->udpsink[1] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
|
|
g_free (uri);
|
|
if (stream->udpsink[1] == NULL)
|
|
goto no_sink_element;
|
|
|
|
/* don't join multicast group, we will have the source socket do that */
|
|
/* no sync or async state changes needed */
|
|
g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
|
|
"loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
|
|
if (ttl > 0)
|
|
g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
|
|
|
|
if (stream->udpsrc[1]) {
|
|
/* configure socket, we give it the same UDP socket as the udpsrc for RTCP
|
|
* because some servers check the port number of where it sends RTCP to identify
|
|
* the RTCP packets it receives */
|
|
g_object_get (G_OBJECT (stream->udpsrc[1]), "sock", &sockfd, NULL);
|
|
GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %d", sockfd);
|
|
/* configure socket and make sure udpsink does not close it when shutting
|
|
* down, it belongs to udpsrc after all. */
|
|
g_object_set (G_OBJECT (stream->udpsink[1]), "sockfd", sockfd,
|
|
"closefd", FALSE, NULL);
|
|
}
|
|
|
|
/* we don't want to consider this a sink */
|
|
GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_IS_SINK);
|
|
|
|
/* we keep this playing always */
|
|
gst_element_set_locked_state (stream->udpsink[1], TRUE);
|
|
gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
|
|
|
|
gst_object_ref (stream->udpsink[1]);
|
|
gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
|
|
|
|
stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
|
|
|
|
/* get session RTCP pad */
|
|
name = g_strdup_printf ("send_rtcp_src_%d", stream->id);
|
|
pad = gst_element_get_request_pad (src->manager, name);
|
|
g_free (name);
|
|
|
|
/* and link */
|
|
if (pad) {
|
|
gst_pad_link (pad, stream->rtcppad);
|
|
gst_object_unref (pad);
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_destination:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no destination address specified");
|
|
return FALSE;
|
|
}
|
|
no_sink_element:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no UDP sink element found");
|
|
return FALSE;
|
|
}
|
|
no_fakesrc_element:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no fakesrc element found");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* 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,
|
|
GstRTSPTransport * transport)
|
|
{
|
|
GstRTSPSrc *src;
|
|
GstPad *outpad = NULL;
|
|
GstPadTemplate *template;
|
|
gchar *name;
|
|
GstStructure *s;
|
|
const gchar *mime;
|
|
|
|
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 (gst_rtsp_transport_get_mime (transport->trans, &mime) < 0)
|
|
goto unknown_transport;
|
|
if (!mime)
|
|
goto unknown_transport;
|
|
|
|
/* configure the final mime type */
|
|
GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
|
|
gst_structure_set_name (s, mime);
|
|
|
|
/* try to get and configure a manager, channelpad[0-1] will be configured with
|
|
* the pads for the manager, or NULL when no manager is needed. */
|
|
if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
|
|
goto no_manager;
|
|
|
|
switch (transport->lower_transport) {
|
|
case GST_RTSP_LOWER_TRANS_TCP:
|
|
if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
|
|
goto transport_failed;
|
|
break;
|
|
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
|
|
if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
|
|
goto transport_failed;
|
|
/* fallthrough, the rest is the same for UDP and MCAST */
|
|
case GST_RTSP_LOWER_TRANS_UDP:
|
|
if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
|
|
goto transport_failed;
|
|
/* configure udpsinks back to the server for RTCP messages and for the
|
|
* dummy RTP messages to open NAT. */
|
|
if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
|
|
goto transport_failed;
|
|
break;
|
|
default:
|
|
goto unknown_transport;
|
|
}
|
|
|
|
if (outpad) {
|
|
GST_DEBUG_OBJECT (src, "creating ghostpad");
|
|
|
|
gst_pad_use_fixed_caps (outpad);
|
|
|
|
/* 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_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
|
|
gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
|
|
gst_object_unref (template);
|
|
g_free (name);
|
|
|
|
gst_object_unref (outpad);
|
|
}
|
|
/* mark pad as ok */
|
|
stream->last_ret = GST_FLOW_OK;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
transport_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "failed to configure transport");
|
|
return FALSE;
|
|
}
|
|
unknown_transport:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "unknown transport");
|
|
return FALSE;
|
|
}
|
|
no_manager:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "cannot get a session manager");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* send a couple of dummy random packets on the receiver RTP port to the server,
|
|
* this should make a firewall think we initiated the data transfer and
|
|
* hopefully allow packets to go from the sender port to our RTP receiver port */
|
|
static gboolean
|
|
gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
|
|
{
|
|
GList *walk;
|
|
|
|
if (src->nat_method != GST_RTSP_NAT_DUMMY)
|
|
return TRUE;
|
|
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
|
|
if (stream->fakesrc && stream->udpsink[0]) {
|
|
GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
|
|
gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
|
|
gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
|
|
gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
|
|
gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
|
|
}
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
/* 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;
|
|
|
|
GST_DEBUG_OBJECT (src, "activating streams");
|
|
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
|
|
if (stream->udpsrc[0]) {
|
|
/* remove timeout, we are streaming now and timeouts will be handled by
|
|
* the session manager and jitter buffer */
|
|
g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
|
|
}
|
|
if (stream->srcpad) {
|
|
/* if we don't have a session manager, set the caps now. If we have a
|
|
* session, we will get a notification of the pad and the caps. */
|
|
if (!src->manager) {
|
|
GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
|
|
gst_pad_set_caps (stream->srcpad, stream->caps);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
|
|
gst_pad_set_active (stream->srcpad, TRUE);
|
|
/* add the pad */
|
|
if (!stream->added) {
|
|
GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
|
|
stream->added = TRUE;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* unblock all pads */
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
|
|
if (stream->blockedpad) {
|
|
GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
|
|
gst_pad_set_blocked_async (stream->blockedpad, FALSE,
|
|
(GstPadBlockCallback) pad_unblocked, src);
|
|
stream->blockedpad = NULL;
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment)
|
|
{
|
|
GList *walk;
|
|
guint64 start, stop;
|
|
gdouble play_speed, play_scale;
|
|
|
|
GST_DEBUG_OBJECT (src, "configuring stream caps");
|
|
|
|
start = segment->last_stop;
|
|
stop = segment->duration;
|
|
play_speed = segment->rate;
|
|
play_scale = segment->applied_rate;
|
|
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
GstCaps *caps;
|
|
|
|
if ((caps = stream->caps)) {
|
|
caps = gst_caps_make_writable (caps);
|
|
/* update caps */
|
|
if (stream->timebase != -1)
|
|
gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
|
|
(guint) stream->timebase, NULL);
|
|
if (stream->seqbase != -1)
|
|
gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
|
|
(guint) stream->seqbase, NULL);
|
|
gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
|
|
if (stop != -1)
|
|
gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
|
|
gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
|
|
gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
|
|
|
|
stream->caps = caps;
|
|
}
|
|
GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
|
|
}
|
|
if (src->manager) {
|
|
GST_DEBUG_OBJECT (src, "clear session");
|
|
g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
|
|
}
|
|
}
|
|
|
|
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 (ret == GST_FLOW_OK)
|
|
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 gboolean
|
|
gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
|
|
GstEvent * event, gboolean source)
|
|
{
|
|
gboolean res = TRUE;
|
|
|
|
/* only streams that have a connection to the outside world */
|
|
if (stream->srcpad == NULL)
|
|
goto done;
|
|
|
|
if (source && stream->udpsrc[0]) {
|
|
gst_event_ref (event);
|
|
res = gst_element_send_event (stream->udpsrc[0], event);
|
|
} else if (stream->channelpad[0]) {
|
|
gst_event_ref (event);
|
|
if (GST_PAD_IS_SRC (stream->channelpad[0]))
|
|
res = gst_pad_push_event (stream->channelpad[0], event);
|
|
else
|
|
res = gst_pad_send_event (stream->channelpad[0], event);
|
|
}
|
|
|
|
if (source && stream->udpsrc[1]) {
|
|
gst_event_ref (event);
|
|
res &= gst_element_send_event (stream->udpsrc[1], event);
|
|
} else if (stream->channelpad[1]) {
|
|
gst_event_ref (event);
|
|
if (GST_PAD_IS_SRC (stream->channelpad[1]))
|
|
res &= gst_pad_push_event (stream->channelpad[1], event);
|
|
else
|
|
res &= gst_pad_send_event (stream->channelpad[1], event);
|
|
}
|
|
|
|
done:
|
|
gst_event_unref (event);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event, gboolean source)
|
|
{
|
|
GList *streams;
|
|
gboolean res = TRUE;
|
|
|
|
for (streams = src->streams; streams; streams = g_list_next (streams)) {
|
|
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
|
|
|
|
gst_event_ref (event);
|
|
res &= gst_rtspsrc_stream_push_event (src, ostream, event, source);
|
|
}
|
|
gst_event_unref (event);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
|
|
gboolean async)
|
|
{
|
|
GstRTSPResult res;
|
|
|
|
if (info->connection == NULL) {
|
|
if (info->url == NULL) {
|
|
GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
|
|
if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
|
|
goto parse_error;
|
|
}
|
|
|
|
/* create connection */
|
|
GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
|
|
if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
|
|
goto could_not_create;
|
|
|
|
if (info->url_str)
|
|
g_free (info->url_str);
|
|
info->url_str = gst_rtsp_url_get_request_uri (info->url);
|
|
|
|
GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
|
|
|
|
if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
|
|
gst_rtsp_connection_set_tunneled (info->connection, TRUE);
|
|
|
|
if (src->proxy_host) {
|
|
GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
|
|
src->proxy_port);
|
|
gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
|
|
src->proxy_port);
|
|
}
|
|
}
|
|
|
|
if (!info->connected) {
|
|
/* connect */
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
|
|
("Connecting to %s", info->location));
|
|
GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
|
|
if ((res =
|
|
gst_rtsp_connection_connect (info->connection,
|
|
src->ptcp_timeout)) < 0)
|
|
goto could_not_connect;
|
|
|
|
info->connected = TRUE;
|
|
}
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
parse_error:
|
|
{
|
|
GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
|
|
return res;
|
|
}
|
|
could_not_create:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
|
|
g_free (str);
|
|
return res;
|
|
}
|
|
could_not_connect:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
|
|
g_free (str);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
|
|
gboolean free)
|
|
{
|
|
if (info->connected) {
|
|
GST_DEBUG_OBJECT (src, "closing connection...");
|
|
gst_rtsp_connection_close (info->connection);
|
|
info->connected = FALSE;
|
|
}
|
|
if (free && info->connection) {
|
|
/* free connection */
|
|
GST_DEBUG_OBJECT (src, "freeing connection...");
|
|
gst_rtsp_connection_free (info->connection);
|
|
info->connection = NULL;
|
|
}
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
|
|
gboolean async)
|
|
{
|
|
GstRTSPResult res;
|
|
|
|
GST_DEBUG_OBJECT (src, "reconnecting connection...");
|
|
gst_rtsp_conninfo_close (src, info, FALSE);
|
|
res = gst_rtsp_conninfo_connect (src, info, async);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
|
|
{
|
|
GList *walk;
|
|
|
|
GST_DEBUG_OBJECT (src, "set flushing %d", flush);
|
|
if (src->conninfo.connection) {
|
|
GST_DEBUG_OBJECT (src, "connection flush");
|
|
gst_rtsp_connection_flush (src->conninfo.connection, flush);
|
|
}
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
GST_DEBUG_OBJECT (src, "stream %p flush", stream);
|
|
if (stream->conninfo.connection)
|
|
gst_rtsp_connection_flush (stream->conninfo.connection, flush);
|
|
}
|
|
}
|
|
|
|
/* FIXME, handle server request, reply with OK, for now */
|
|
static GstRTSPResult
|
|
gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
|
|
GstRTSPMessage * request)
|
|
{
|
|
GstRTSPMessage response = { 0 };
|
|
GstRTSPResult res;
|
|
|
|
GST_DEBUG_OBJECT (src, "got server request message");
|
|
|
|
if (src->debug)
|
|
gst_rtsp_message_dump (request);
|
|
|
|
res = gst_rtsp_ext_list_receive_request (src->extensions, request);
|
|
|
|
if (res == GST_RTSP_ENOTIMPL) {
|
|
/* default implementation, send OK */
|
|
res =
|
|
gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
|
|
request);
|
|
if (res < 0)
|
|
goto send_error;
|
|
|
|
GST_DEBUG_OBJECT (src, "replying with OK");
|
|
|
|
if (src->debug)
|
|
gst_rtsp_message_dump (&response);
|
|
|
|
res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
|
|
if (res < 0)
|
|
goto send_error;
|
|
|
|
gst_rtsp_message_unset (&response);
|
|
} else if (res == GST_RTSP_EEOF)
|
|
return res;
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
send_error:
|
|
{
|
|
gst_rtsp_message_unset (&response);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
/* send server keep-alive */
|
|
static GstRTSPResult
|
|
gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
|
|
{
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPResult res;
|
|
GstRTSPMethod method;
|
|
gchar *control;
|
|
|
|
GST_DEBUG_OBJECT (src, "creating server keep-alive");
|
|
|
|
/* find a method to use for keep-alive */
|
|
if (src->methods & GST_RTSP_GET_PARAMETER)
|
|
method = GST_RTSP_GET_PARAMETER;
|
|
else
|
|
method = GST_RTSP_OPTIONS;
|
|
|
|
if (src->control)
|
|
control = src->control;
|
|
else
|
|
control = src->conninfo.url_str;
|
|
|
|
if (control == NULL)
|
|
goto no_control;
|
|
|
|
res = gst_rtsp_message_init_request (&request, method, control);
|
|
if (res < 0)
|
|
goto send_error;
|
|
|
|
if (src->debug)
|
|
gst_rtsp_message_dump (&request);
|
|
|
|
res =
|
|
gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
|
|
NULL);
|
|
if (res < 0)
|
|
goto send_error;
|
|
|
|
gst_rtsp_connection_reset_timeout (src->conninfo.connection);
|
|
gst_rtsp_message_unset (&request);
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
no_control:
|
|
{
|
|
GST_WARNING_OBJECT (src, "no control url to send keepalive");
|
|
return GST_RTSP_OK;
|
|
}
|
|
send_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
gst_rtsp_message_unset (&request);
|
|
GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send keep-alive. (%s)", str));
|
|
g_free (str);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
|
|
{
|
|
GstRTSPMessage message = { 0 };
|
|
GstRTSPResult res;
|
|
gint channel;
|
|
GstRTSPStream *stream;
|
|
GstPad *outpad = NULL;
|
|
guint8 *data;
|
|
guint size;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstBuffer *buf;
|
|
gboolean is_rtcp, have_data;
|
|
|
|
/* here we are only interested in data messages */
|
|
have_data = FALSE;
|
|
do {
|
|
GTimeVal tv_timeout;
|
|
|
|
/* get the next timeout interval */
|
|
gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
|
|
|
|
/* see if the timeout period expired */
|
|
if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
|
|
GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
|
|
/* send keep-alive, only act on interrupt, a warning will be posted for
|
|
* other errors. */
|
|
if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
|
|
goto interrupt;
|
|
/* get new timeout */
|
|
gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
|
|
tv_timeout.tv_sec, tv_timeout.tv_usec);
|
|
|
|
/* protect the connection with the connection lock so that we can see when
|
|
* we are finished doing server communication */
|
|
res =
|
|
gst_rtspsrc_connection_receive (src, src->conninfo.connection,
|
|
&message, src->ptcp_timeout);
|
|
|
|
switch (res) {
|
|
case GST_RTSP_OK:
|
|
GST_DEBUG_OBJECT (src, "we received a server message");
|
|
break;
|
|
case GST_RTSP_EINTR:
|
|
/* we got interrupted this means we need to stop */
|
|
goto interrupt;
|
|
case GST_RTSP_ETIMEOUT:
|
|
/* no reply, send keep alive */
|
|
GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
|
|
if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
|
|
goto interrupt;
|
|
continue;
|
|
case GST_RTSP_EEOF:
|
|
/* go EOS when the server closed the connection */
|
|
goto server_eof;
|
|
default:
|
|
goto receive_error;
|
|
}
|
|
|
|
switch (message.type) {
|
|
case GST_RTSP_MESSAGE_REQUEST:
|
|
/* server sends us a request message, handle it */
|
|
res =
|
|
gst_rtspsrc_handle_request (src, src->conninfo.connection,
|
|
&message);
|
|
if (res == GST_RTSP_EEOF)
|
|
goto server_eof;
|
|
else if (res < 0)
|
|
goto handle_request_failed;
|
|
break;
|
|
case GST_RTSP_MESSAGE_RESPONSE:
|
|
/* we ignore response messages */
|
|
GST_DEBUG_OBJECT (src, "ignoring response message");
|
|
if (src->debug)
|
|
gst_rtsp_message_dump (&message);
|
|
break;
|
|
case GST_RTSP_MESSAGE_DATA:
|
|
GST_DEBUG_OBJECT (src, "got data message");
|
|
have_data = TRUE;
|
|
break;
|
|
default:
|
|
GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
|
|
message.type);
|
|
break;
|
|
}
|
|
}
|
|
while (!have_data);
|
|
|
|
channel = message.type_data.data.channel;
|
|
|
|
stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
|
|
if (!stream)
|
|
goto unknown_stream;
|
|
|
|
if (channel == stream->channel[0]) {
|
|
outpad = stream->channelpad[0];
|
|
is_rtcp = FALSE;
|
|
} else if (channel == stream->channel[1]) {
|
|
outpad = stream->channelpad[1];
|
|
is_rtcp = TRUE;
|
|
} else {
|
|
is_rtcp = FALSE;
|
|
}
|
|
|
|
/* take a look at the body to figure out what we have */
|
|
gst_rtsp_message_get_body (&message, &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];
|
|
is_rtcp = TRUE;
|
|
}
|
|
|
|
/* we have no clue what this is, just ignore then. */
|
|
if (outpad == NULL)
|
|
goto unknown_stream;
|
|
|
|
/* take the message body for further processing */
|
|
gst_rtsp_message_steal_body (&message, &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 */
|
|
gst_rtsp_message_unset (&message);
|
|
|
|
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;
|
|
}
|
|
|
|
if (!src->manager) {
|
|
/* set stream caps on buffer when we don't have a session manager to do it
|
|
* for us */
|
|
gst_buffer_set_caps (buf, stream->caps);
|
|
}
|
|
|
|
if (src->base_time == -1) {
|
|
/* Take current running_time. This timestamp will be put on
|
|
* the first buffer of each stream because we are a live source and so we
|
|
* timestamp with the running_time. When we are dealing with TCP, we also
|
|
* only timestamp the first buffer (using the DISCONT flag) because a server
|
|
* typically bursts data, for which we don't want to compensate by speeding
|
|
* up the media. The other timestamps will be interpollated from this one
|
|
* using the RTP timestamps. */
|
|
GST_OBJECT_LOCK (src);
|
|
if (GST_ELEMENT_CLOCK (src)) {
|
|
GstClockTime now;
|
|
GstClockTime base_time;
|
|
|
|
now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
|
|
base_time = GST_ELEMENT_CAST (src)->base_time;
|
|
|
|
src->base_time = now - base_time;
|
|
|
|
GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
|
|
}
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
if (stream->discont && !is_rtcp) {
|
|
/* mark first RTP buffer as discont */
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
|
|
stream->discont = FALSE;
|
|
/* first buffer gets the timestamp, other buffers are not timestamped and
|
|
* their presentation time will be interpollated from the rtp timestamps. */
|
|
GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (src->base_time));
|
|
|
|
GST_BUFFER_TIMESTAMP (buf) = src->base_time;
|
|
}
|
|
|
|
/* chain to the peer pad */
|
|
if (GST_PAD_IS_SINK (outpad))
|
|
ret = gst_pad_chain (outpad, buf);
|
|
else
|
|
ret = gst_pad_push (outpad, buf);
|
|
|
|
if (!is_rtcp) {
|
|
/* combine all stream flows for the data transport */
|
|
ret = gst_rtspsrc_combine_flows (src, stream, ret);
|
|
}
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
unknown_stream:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
|
|
gst_rtsp_message_unset (&message);
|
|
return GST_FLOW_OK;
|
|
}
|
|
server_eof:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we got an eof from the server");
|
|
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
|
|
("The server closed the connection."));
|
|
src->conninfo.connected = FALSE;
|
|
gst_rtsp_message_unset (&message);
|
|
return GST_FLOW_UNEXPECTED;
|
|
}
|
|
interrupt:
|
|
{
|
|
gst_rtsp_message_unset (&message);
|
|
GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
|
|
gst_rtspsrc_connection_flush (src, FALSE);
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
receive_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
|
|
("Could not receive message. (%s)", str));
|
|
g_free (str);
|
|
|
|
gst_rtsp_message_unset (&message);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
handle_request_failed:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not handle server message. (%s)", str));
|
|
g_free (str);
|
|
gst_rtsp_message_unset (&message);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
invalid_length:
|
|
{
|
|
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
|
|
("Short message received, ignoring."));
|
|
gst_rtsp_message_unset (&message);
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtspsrc_loop_udp (GstRTSPSrc * src)
|
|
{
|
|
GstRTSPResult res;
|
|
GstRTSPMessage message = { 0 };
|
|
gint retry = 0;
|
|
|
|
while (TRUE) {
|
|
GTimeVal tv_timeout;
|
|
|
|
/* get the next timeout interval */
|
|
gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
|
|
|
|
GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
|
|
(gint) tv_timeout.tv_sec);
|
|
|
|
gst_rtsp_message_unset (&message);
|
|
|
|
/* we should continue reading the TCP socket because the server might
|
|
* send us requests. When the session timeout expires, we need to send a
|
|
* keep-alive request to keep the session open. */
|
|
res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
|
|
&message, &tv_timeout);
|
|
|
|
switch (res) {
|
|
case GST_RTSP_OK:
|
|
GST_DEBUG_OBJECT (src, "we received a server message");
|
|
break;
|
|
case GST_RTSP_EINTR:
|
|
/* we got interrupted, see what we have to do */
|
|
goto interrupt;
|
|
case GST_RTSP_ETIMEOUT:
|
|
/* send keep-alive, ignore the result, a warning will be posted. */
|
|
GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
|
|
if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
|
|
goto interrupt;
|
|
continue;
|
|
case GST_RTSP_EEOF:
|
|
/* server closed the connection. not very fatal for UDP, reconnect and
|
|
* see what happens. */
|
|
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
|
|
("The server closed the connection."));
|
|
if ((res =
|
|
gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
|
|
goto connect_error;
|
|
|
|
continue;
|
|
default:
|
|
goto receive_error;
|
|
}
|
|
|
|
switch (message.type) {
|
|
case GST_RTSP_MESSAGE_REQUEST:
|
|
/* server sends us a request message, handle it */
|
|
res =
|
|
gst_rtspsrc_handle_request (src, src->conninfo.connection,
|
|
&message);
|
|
if (res == GST_RTSP_EEOF)
|
|
goto server_eof;
|
|
else if (res < 0)
|
|
goto handle_request_failed;
|
|
break;
|
|
case GST_RTSP_MESSAGE_RESPONSE:
|
|
/* we ignore response and data messages */
|
|
GST_DEBUG_OBJECT (src, "ignoring response message");
|
|
if (src->debug)
|
|
gst_rtsp_message_dump (&message);
|
|
if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
|
|
GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
|
|
if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
|
|
GST_DEBUG_OBJECT (src, "so retrying keep-alive");
|
|
if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
|
|
goto interrupt;
|
|
}
|
|
} else {
|
|
retry = 0;
|
|
}
|
|
break;
|
|
case GST_RTSP_MESSAGE_DATA:
|
|
/* we ignore response and data messages */
|
|
GST_DEBUG_OBJECT (src, "ignoring data message");
|
|
break;
|
|
default:
|
|
GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
|
|
message.type);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* we get here when the connection got interrupted */
|
|
interrupt:
|
|
{
|
|
gst_rtsp_message_unset (&message);
|
|
GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
|
|
gst_rtspsrc_connection_flush (src, FALSE);
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
connect_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
GstFlowReturn ret;
|
|
|
|
src->conninfo.connected = FALSE;
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
|
|
("Could not connect to server. (%s)", str));
|
|
g_free (str);
|
|
ret = GST_FLOW_ERROR;
|
|
} else {
|
|
ret = GST_FLOW_WRONG_STATE;
|
|
}
|
|
return ret;
|
|
}
|
|
receive_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
|
|
("Could not receive message. (%s)", str));
|
|
g_free (str);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
handle_request_failed:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
GstFlowReturn ret;
|
|
|
|
gst_rtsp_message_unset (&message);
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not handle server message. (%s)", str));
|
|
g_free (str);
|
|
ret = GST_FLOW_ERROR;
|
|
} else {
|
|
ret = GST_FLOW_WRONG_STATE;
|
|
}
|
|
return ret;
|
|
}
|
|
server_eof:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we got an eof from the server");
|
|
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
|
|
("The server closed the connection."));
|
|
src->conninfo.connected = FALSE;
|
|
gst_rtsp_message_unset (&message);
|
|
return GST_FLOW_UNEXPECTED;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
|
|
{
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
gboolean restart;
|
|
|
|
GST_DEBUG_OBJECT (src, "doing reconnect");
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
/* 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 can try only TCP now */
|
|
src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
|
|
|
|
/* close and cleanup our state */
|
|
if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
|
|
goto done;
|
|
|
|
/* see if we have TCP left to try. Also don't try TCP when we were configured
|
|
* with an SDP. */
|
|
if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
|
|
goto no_protocols;
|
|
|
|
/* 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->udp_timeout / 1000000.0)));
|
|
|
|
/* open new connection using tcp */
|
|
if (gst_rtspsrc_open (src, async) < 0)
|
|
goto open_failed;
|
|
|
|
/* start playback */
|
|
if (gst_rtspsrc_play (src, &src->segment, async) < 0)
|
|
goto play_failed;
|
|
|
|
done:
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
no_protocols:
|
|
{
|
|
src->cur_protocols = 0;
|
|
/* no transport possible, post an error and stop */
|
|
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
|
|
("Could not receive any UDP packets for %.4f seconds, maybe your "
|
|
"firewall is blocking it. No other protocols to try.",
|
|
gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
open_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "open failed");
|
|
return GST_FLOW_OK;
|
|
}
|
|
play_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "play failed");
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
|
|
{
|
|
switch (cmd) {
|
|
case CMD_OPEN:
|
|
GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
|
|
break;
|
|
case CMD_PLAY:
|
|
GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
|
|
break;
|
|
case CMD_PAUSE:
|
|
GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
|
|
break;
|
|
case CMD_CLOSE:
|
|
GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
|
|
{
|
|
switch (cmd) {
|
|
case CMD_OPEN:
|
|
GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
|
|
break;
|
|
case CMD_PLAY:
|
|
GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
|
|
break;
|
|
case CMD_PAUSE:
|
|
GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
|
|
break;
|
|
case CMD_CLOSE:
|
|
GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
|
|
{
|
|
switch (cmd) {
|
|
case CMD_OPEN:
|
|
GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
|
|
break;
|
|
case CMD_PLAY:
|
|
GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
|
|
break;
|
|
case CMD_PAUSE:
|
|
GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
|
|
break;
|
|
case CMD_CLOSE:
|
|
GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
|
|
{
|
|
switch (cmd) {
|
|
case CMD_OPEN:
|
|
GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
|
|
break;
|
|
case CMD_PLAY:
|
|
GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
|
|
break;
|
|
case CMD_PAUSE:
|
|
GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
|
|
break;
|
|
case CMD_CLOSE:
|
|
GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
|
|
{
|
|
if (ret == GST_RTSP_OK)
|
|
gst_rtspsrc_loop_complete_cmd (src, cmd);
|
|
else if (ret == GST_RTSP_EINTR)
|
|
gst_rtspsrc_loop_cancel_cmd (src, cmd);
|
|
else
|
|
gst_rtspsrc_loop_error_cmd (src, cmd);
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gboolean flush)
|
|
{
|
|
gint old;
|
|
|
|
/* FIXME flush param mute; remove at discretion */
|
|
|
|
/* start new request */
|
|
gst_rtspsrc_loop_start_cmd (src, cmd);
|
|
|
|
GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
old = src->loop_cmd;
|
|
if (old != CMD_WAIT) {
|
|
src->loop_cmd = CMD_WAIT;
|
|
GST_OBJECT_UNLOCK (src);
|
|
/* cancel previous request */
|
|
gst_rtspsrc_loop_cancel_cmd (src, old);
|
|
GST_OBJECT_LOCK (src);
|
|
}
|
|
src->loop_cmd = cmd;
|
|
/* interrupt if allowed */
|
|
if (src->waiting) {
|
|
GST_DEBUG_OBJECT (src, "start connection flush");
|
|
gst_rtspsrc_connection_flush (src, TRUE);
|
|
}
|
|
if (src->task)
|
|
gst_task_start (src->task);
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_loop (GstRTSPSrc * src)
|
|
{
|
|
GstFlowReturn ret;
|
|
|
|
if (!src->conninfo.connection || !src->conninfo.connected)
|
|
goto no_connection;
|
|
|
|
if (src->interleaved)
|
|
ret = gst_rtspsrc_loop_interleaved (src);
|
|
else
|
|
ret = gst_rtspsrc_loop_udp (src);
|
|
|
|
if (ret != GST_FLOW_OK)
|
|
goto pause;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_connection:
|
|
{
|
|
GST_WARNING_OBJECT (src, "we are not connected");
|
|
ret = GST_FLOW_WRONG_STATE;
|
|
goto pause;
|
|
}
|
|
pause:
|
|
{
|
|
const gchar *reason = gst_flow_get_name (ret);
|
|
|
|
GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
|
|
src->running = FALSE;
|
|
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 (), FALSE);
|
|
}
|
|
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_UNEXPECTED) {
|
|
/* for fatal errors we post an error message, post the error before the
|
|
* EOS 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 (), FALSE);
|
|
}
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
#ifndef GST_DISABLE_GST_DEBUG
|
|
static const gchar *
|
|
gst_rtsp_auth_method_to_string (GstRTSPAuthMethod 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
|
|
|
|
static const gchar *
|
|
gst_rtspsrc_skip_lws (const gchar * s)
|
|
{
|
|
while (g_ascii_isspace (*s))
|
|
s++;
|
|
return s;
|
|
}
|
|
|
|
static const gchar *
|
|
gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
|
|
{
|
|
while (s > start && g_ascii_isspace (*(s - 1)))
|
|
s--;
|
|
return s;
|
|
}
|
|
|
|
static const gchar *
|
|
gst_rtspsrc_skip_commas (const gchar * s)
|
|
{
|
|
/* The grammar allows for multiple commas */
|
|
while (g_ascii_isspace (*s) || *s == ',')
|
|
s++;
|
|
return s;
|
|
}
|
|
|
|
static const gchar *
|
|
gst_rtspsrc_skip_item (const gchar * s)
|
|
{
|
|
gboolean quoted = FALSE;
|
|
const gchar *start = s;
|
|
|
|
/* A list item ends at the last non-whitespace character
|
|
* before a comma which is not inside a quoted-string. Or at
|
|
* the end of the string.
|
|
*/
|
|
while (*s) {
|
|
if (*s == '"')
|
|
quoted = !quoted;
|
|
else if (quoted) {
|
|
if (*s == '\\' && *(s + 1))
|
|
s++;
|
|
} else {
|
|
if (*s == ',')
|
|
break;
|
|
}
|
|
s++;
|
|
}
|
|
|
|
return gst_rtspsrc_unskip_lws (s, start);
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_decode_quoted_string (gchar * quoted_string)
|
|
{
|
|
gchar *src, *dst;
|
|
|
|
src = quoted_string + 1;
|
|
dst = quoted_string;
|
|
while (*src && *src != '"') {
|
|
if (*src == '\\' && *(src + 1))
|
|
src++;
|
|
*dst++ = *src++;
|
|
}
|
|
*dst = '\0';
|
|
}
|
|
|
|
/* Extract the authentication tokens that the server provided for each method
|
|
* into an array of structures and give those to the connection object.
|
|
*/
|
|
static void
|
|
gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
|
|
const gchar * header, gboolean * stale)
|
|
{
|
|
GSList *list = NULL, *iter;
|
|
const gchar *end;
|
|
gchar *item, *eq, *name_end, *value;
|
|
|
|
g_return_if_fail (stale != NULL);
|
|
|
|
gst_rtsp_connection_clear_auth_params (conn);
|
|
*stale = FALSE;
|
|
|
|
/* Parse a header whose content is described by RFC2616 as
|
|
* "#something", where "something" does not itself contain commas,
|
|
* except as part of quoted-strings, into a list of allocated strings.
|
|
*/
|
|
header = gst_rtspsrc_skip_commas (header);
|
|
while (*header) {
|
|
end = gst_rtspsrc_skip_item (header);
|
|
list = g_slist_prepend (list, g_strndup (header, end - header));
|
|
header = gst_rtspsrc_skip_commas (end);
|
|
}
|
|
if (!list)
|
|
return;
|
|
|
|
list = g_slist_reverse (list);
|
|
for (iter = list; iter; iter = iter->next) {
|
|
item = iter->data;
|
|
|
|
eq = strchr (item, '=');
|
|
if (eq) {
|
|
name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
|
|
if (name_end == item) {
|
|
/* That's no good... */
|
|
g_free (item);
|
|
continue;
|
|
}
|
|
|
|
*name_end = '\0';
|
|
|
|
value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
|
|
if (*value == '"')
|
|
gst_rtsp_decode_quoted_string (value);
|
|
} else
|
|
value = NULL;
|
|
|
|
if (item && (strcmp (item, "stale") == 0) &&
|
|
value && (strcmp (value, "TRUE") == 0))
|
|
*stale = TRUE;
|
|
gst_rtsp_connection_set_auth_param (conn, item, value);
|
|
g_free (item);
|
|
}
|
|
|
|
g_slist_free (list);
|
|
}
|
|
|
|
/* Parse a WWW-Authenticate Response header and determine the
|
|
* available authentication methods
|
|
*
|
|
* This code should also cope with the fact that each WWW-Authenticate
|
|
* header can contain multiple challenge methods + tokens
|
|
*
|
|
* At the moment, for Basic auth, we just do a minimal check and don't
|
|
* even parse out the realm */
|
|
static void
|
|
gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
|
|
GstRTSPConnection * conn, gboolean * stale)
|
|
{
|
|
gchar *start;
|
|
|
|
g_return_if_fail (hdr != NULL);
|
|
g_return_if_fail (methods != NULL);
|
|
g_return_if_fail (stale != 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 |= GST_RTSP_AUTH_BASIC;
|
|
else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
|
|
*methods |= GST_RTSP_AUTH_DIGEST;
|
|
gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* 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, GstRTSPMessage * response)
|
|
{
|
|
gchar *user = NULL;
|
|
gchar *pass = NULL;
|
|
GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
|
|
GstRTSPAuthMethod method;
|
|
GstRTSPResult auth_result;
|
|
GstRTSPUrl *url;
|
|
GstRTSPConnection *conn;
|
|
gchar *hdr;
|
|
gboolean stale = FALSE;
|
|
|
|
conn = src->conninfo.connection;
|
|
|
|
/* Identify the available auth methods and see if any are supported */
|
|
if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
|
|
&hdr, 0) == GST_RTSP_OK) {
|
|
gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
|
|
}
|
|
|
|
if (avail_methods == GST_RTSP_AUTH_NONE)
|
|
goto no_auth_available;
|
|
|
|
/* 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 */
|
|
if (stale)
|
|
src->tried_url_auth = FALSE;
|
|
|
|
url = gst_rtsp_connection_get_url (conn);
|
|
|
|
/* Do we have username and password available? */
|
|
if (url != NULL && !src->tried_url_auth && url->user != NULL
|
|
&& url->passwd != NULL) {
|
|
user = url->user;
|
|
pass = url->passwd;
|
|
src->tried_url_auth = TRUE;
|
|
GST_DEBUG_OBJECT (src,
|
|
"Attempting authentication using credentials from the URL");
|
|
} else {
|
|
user = src->user_id;
|
|
pass = src->user_pw;
|
|
GST_DEBUG_OBJECT (src,
|
|
"Attempting authentication using credentials from the properties");
|
|
}
|
|
|
|
/* 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 = GST_RTSP_AUTH_MAX; method != GST_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 = gst_rtsp_connection_set_auth (conn, 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 == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
|
|
GST_DEBUG_OBJECT (src, "Attempting %s authentication",
|
|
gst_rtsp_auth_method_to_string (method));
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (method == GST_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;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
|
|
GstRTSPMessage * request, GstRTSPMessage * response,
|
|
GstRTSPStatusCode * code)
|
|
{
|
|
GstRTSPResult res;
|
|
GstRTSPStatusCode thecode;
|
|
gchar *content_base = NULL;
|
|
gint try = 0;
|
|
|
|
again:
|
|
if (!src->short_header)
|
|
gst_rtsp_ext_list_before_send (src->extensions, request);
|
|
|
|
GST_DEBUG_OBJECT (src, "sending message");
|
|
|
|
if (src->debug)
|
|
gst_rtsp_message_dump (request);
|
|
|
|
res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
|
|
if (res < 0)
|
|
goto send_error;
|
|
|
|
gst_rtsp_connection_reset_timeout (conn);
|
|
|
|
next:
|
|
res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
|
|
if (res < 0)
|
|
goto receive_error;
|
|
|
|
if (src->debug)
|
|
gst_rtsp_message_dump (response);
|
|
|
|
switch (response->type) {
|
|
case GST_RTSP_MESSAGE_REQUEST:
|
|
res = gst_rtspsrc_handle_request (src, conn, response);
|
|
if (res == GST_RTSP_EEOF)
|
|
goto server_eof;
|
|
else if (res < 0)
|
|
goto handle_request_failed;
|
|
goto next;
|
|
case GST_RTSP_MESSAGE_RESPONSE:
|
|
/* ok, a response is good */
|
|
GST_DEBUG_OBJECT (src, "received response message");
|
|
break;
|
|
case GST_RTSP_MESSAGE_DATA:
|
|
/* get next response */
|
|
GST_DEBUG_OBJECT (src, "ignoring data response message");
|
|
goto next;
|
|
default:
|
|
GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
|
|
response->type);
|
|
goto next;
|
|
}
|
|
|
|
thecode = response->type_data.response.code;
|
|
|
|
GST_DEBUG_OBJECT (src, "got response message %d", thecode);
|
|
|
|
/* 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 != GST_RTSP_STS_OK)
|
|
return GST_RTSP_OK;
|
|
|
|
/* store new content base if any */
|
|
gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
|
|
&content_base, 0);
|
|
if (content_base) {
|
|
g_free (src->content_base);
|
|
src->content_base = g_strdup (content_base);
|
|
}
|
|
gst_rtsp_ext_list_after_send (src->extensions, request, response);
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
send_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (src, "send interrupted");
|
|
}
|
|
g_free (str);
|
|
return res;
|
|
}
|
|
receive_error:
|
|
{
|
|
switch (res) {
|
|
case GST_RTSP_EEOF:
|
|
GST_WARNING_OBJECT (src, "server closed connection, doing reconnect");
|
|
if (try == 0) {
|
|
try++;
|
|
/* if reconnect succeeds, try again */
|
|
if ((res =
|
|
gst_rtsp_conninfo_reconnect (src, &src->conninfo,
|
|
FALSE)) == 0)
|
|
goto again;
|
|
}
|
|
/* only try once after reconnect, then fallthrough and error out */
|
|
default:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
|
|
("Could not receive message. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (src, "receive interrupted");
|
|
}
|
|
g_free (str);
|
|
break;
|
|
}
|
|
}
|
|
return res;
|
|
}
|
|
handle_request_failed:
|
|
{
|
|
/* ERROR was posted */
|
|
gst_rtsp_message_unset (response);
|
|
return res;
|
|
}
|
|
server_eof:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we got an eof from the server");
|
|
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
|
|
("The server closed the connection."));
|
|
gst_rtsp_message_unset (response);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtspsrc_send:
|
|
* @src: the rtsp source
|
|
* @conn: the connection to send on
|
|
* @request: must point to a valid request
|
|
* @response: must point to an empty #GstRTSPMessage
|
|
* @code: an optional code result
|
|
*
|
|
* 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 #GST_RTSP_OK, @response will contain a valid response
|
|
* message that should be cleaned with gst_rtsp_message_unset() after usage.
|
|
*
|
|
* If @code is NULL, this function will return #GST_RTSP_ERROR (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: #GST_RTSP_OK if the processing was successful.
|
|
*/
|
|
static GstRTSPResult
|
|
gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
|
|
GstRTSPMessage * request, GstRTSPMessage * response,
|
|
GstRTSPStatusCode * code)
|
|
{
|
|
GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
|
|
GstRTSPResult res = GST_RTSP_ERROR;
|
|
gint count;
|
|
gboolean retry;
|
|
GstRTSPMethod method = GST_RTSP_INVALID;
|
|
|
|
count = 0;
|
|
do {
|
|
retry = FALSE;
|
|
|
|
/* make sure we don't loop forever */
|
|
if (count++ > 8)
|
|
break;
|
|
|
|
/* save method so we can disable it when the server complains */
|
|
method = request->type_data.request.method;
|
|
|
|
if ((res =
|
|
gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
|
|
goto error;
|
|
|
|
switch (int_code) {
|
|
case GST_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;
|
|
}
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
} 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 != GST_RTSP_STS_OK)
|
|
goto error_response;
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "got error %d", res);
|
|
return res;
|
|
}
|
|
error_response:
|
|
{
|
|
res = GST_RTSP_ERROR;
|
|
|
|
switch (response->type_data.response.code) {
|
|
case GST_RTSP_STS_NOT_FOUND:
|
|
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
|
|
response->type_data.response.reason));
|
|
break;
|
|
case GST_RTSP_STS_MOVED_PERMANENTLY:
|
|
case GST_RTSP_STS_MOVE_TEMPORARILY:
|
|
{
|
|
gchar *new_location;
|
|
GstRTSPLowerTrans transports;
|
|
|
|
GST_DEBUG_OBJECT (src, "got redirection");
|
|
/* if we don't have a Location Header, we must error */
|
|
if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
|
|
&new_location, 0) < 0)
|
|
break;
|
|
|
|
/* When we receive a redirect result, we go back to the INIT state after
|
|
* parsing the new URI. The caller should do the needed steps to issue
|
|
* a new setup when it detects this state change. */
|
|
GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
|
|
|
|
/* save current transports */
|
|
if (src->conninfo.url)
|
|
transports = src->conninfo.url->transports;
|
|
else
|
|
transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
|
|
|
|
gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location);
|
|
|
|
/* set old transports */
|
|
if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
|
|
src->conninfo.url->transports = transports;
|
|
|
|
src->need_redirect = TRUE;
|
|
src->state = GST_RTSP_STATE_INIT;
|
|
res = GST_RTSP_OK;
|
|
break;
|
|
}
|
|
case GST_RTSP_STS_NOT_ACCEPTABLE:
|
|
case GST_RTSP_STS_NOT_IMPLEMENTED:
|
|
case GST_RTSP_STS_METHOD_NOT_ALLOWED:
|
|
GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
|
|
gst_rtsp_method_as_text (method));
|
|
src->methods &= ~method;
|
|
res = GST_RTSP_OK;
|
|
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;
|
|
}
|
|
/* if we return ERROR we should unset the response ourselves */
|
|
if (res == GST_RTSP_ERROR)
|
|
gst_rtsp_message_unset (response);
|
|
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
|
|
GstRTSPMessage * response, GstRTSPSrc * src)
|
|
{
|
|
return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
|
|
NULL);
|
|
}
|
|
|
|
|
|
/* 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, GstRTSPMessage * response)
|
|
{
|
|
GstRTSPHeaderField field;
|
|
gchar *respoptions;
|
|
gchar **options;
|
|
gint indx = 0;
|
|
gint i;
|
|
|
|
/* reset supported methods */
|
|
src->methods = 0;
|
|
|
|
/* Try Allow Header first */
|
|
field = GST_RTSP_HDR_ALLOW;
|
|
while (TRUE) {
|
|
respoptions = NULL;
|
|
gst_rtsp_message_get_header (response, field, &respoptions, indx);
|
|
if (indx == 0 && !respoptions) {
|
|
/* if no Allow header was found then try the Public header... */
|
|
field = GST_RTSP_HDR_PUBLIC;
|
|
gst_rtsp_message_get_header (response, field, &respoptions, indx);
|
|
}
|
|
if (!respoptions)
|
|
break;
|
|
|
|
/* 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 = gst_rtsp_find_method (stripped);
|
|
|
|
/* keep bitfield of supported methods */
|
|
if (method != GST_RTSP_INVALID)
|
|
src->methods |= method;
|
|
}
|
|
g_strfreev (options);
|
|
|
|
indx++;
|
|
}
|
|
|
|
if (src->methods == 0) {
|
|
/* neither Allow nor Public are required, assume the server supports
|
|
* at least DESCRIBE, SETUP, we always assume it supports PLAY as
|
|
* well. */
|
|
GST_DEBUG_OBJECT (src, "could not get OPTIONS");
|
|
src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
|
|
}
|
|
/* always assume PLAY, FIXME, extensions should be able to override
|
|
* this */
|
|
src->methods |= GST_RTSP_PLAY;
|
|
/* also assume it will support Range */
|
|
src->seekable = TRUE;
|
|
|
|
/* we need describe and setup */
|
|
if (!(src->methods & GST_RTSP_DESCRIBE))
|
|
goto no_describe;
|
|
if (!(src->methods & GST_RTSP_SETUP))
|
|
goto no_setup;
|
|
|
|
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;
|
|
}
|
|
}
|
|
|
|
/* masks to be kept in sync with the hardcoded protocol order of preference
|
|
* in code below */
|
|
static guint protocol_masks[] = {
|
|
GST_RTSP_LOWER_TRANS_UDP,
|
|
GST_RTSP_LOWER_TRANS_UDP_MCAST,
|
|
GST_RTSP_LOWER_TRANS_TCP,
|
|
0
|
|
};
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
|
|
GstRTSPLowerTrans protocols, gchar ** transports)
|
|
{
|
|
GstRTSPResult res;
|
|
GString *result;
|
|
gboolean add_udp_str;
|
|
|
|
*transports = NULL;
|
|
|
|
res =
|
|
gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
|
|
|
|
if (res < 0)
|
|
goto failed;
|
|
|
|
GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
|
|
|
|
/* extension listed transports, use those */
|
|
if (*transports != NULL)
|
|
return GST_RTSP_OK;
|
|
|
|
/* it's the default */
|
|
add_udp_str = FALSE;
|
|
|
|
/* the default RTSP transports */
|
|
result = g_string_new ("");
|
|
if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
|
|
GST_DEBUG_OBJECT (src, "adding UDP unicast");
|
|
|
|
g_string_append (result, "RTP/AVP");
|
|
if (add_udp_str)
|
|
g_string_append (result, "/UDP");
|
|
g_string_append (result, ";unicast;client_port=%%u1-%%u2");
|
|
} else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
|
|
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 */
|
|
if (result->len > 0)
|
|
g_string_append (result, ",");
|
|
g_string_append (result, "RTP/AVP");
|
|
if (add_udp_str)
|
|
g_string_append (result, "/UDP");
|
|
g_string_append (result, ";multicast");
|
|
} else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
|
|
GST_DEBUG_OBJECT (src, "adding TCP");
|
|
|
|
if (result->len > 0)
|
|
g_string_append (result, ",");
|
|
g_string_append (result, "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2");
|
|
}
|
|
*transports = g_string_free (result, FALSE);
|
|
|
|
GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
failed:
|
|
{
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
|
|
gint orig_rtpport, gint orig_rtcpport)
|
|
{
|
|
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 (!orig_rtpport || !orig_rtcpport) {
|
|
if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
|
|
goto failed;
|
|
} else {
|
|
rtpport = orig_rtpport;
|
|
rtcpport = orig_rtcpport;
|
|
}
|
|
}
|
|
|
|
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;
|
|
}
|
|
/* append final part */
|
|
g_string_append (str, p);
|
|
|
|
g_free (*transports);
|
|
*transports = g_string_free (str, FALSE);
|
|
|
|
done:
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
failed:
|
|
{
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_stream_is_real_media (GstRTSPStream * stream)
|
|
{
|
|
gboolean res = FALSE;
|
|
|
|
if (stream->caps) {
|
|
GstStructure *s;
|
|
const gchar *enc = NULL;
|
|
|
|
s = gst_caps_get_structure (stream->caps, 0);
|
|
if ((enc = gst_structure_get_string (s, "encoding-name"))) {
|
|
res = (strstr (enc, "-REAL") != NULL);
|
|
}
|
|
}
|
|
return res;
|
|
}
|
|
|
|
/* Perform the SETUP request for all the streams.
|
|
*
|
|
* We ask the server for a specific transport, which initially includes all the
|
|
* ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
|
|
* two local UDP ports that we send to the server.
|
|
*
|
|
* Once the server replied with a transport, we configure the other streams
|
|
* with the same transport.
|
|
*
|
|
* This function will also configure the stream for the selected transport,
|
|
* which basically means creating the pipeline.
|
|
*/
|
|
static GstRTSPResult
|
|
gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
|
|
{
|
|
GList *walk;
|
|
GstRTSPResult res = GST_RTSP_ERROR;
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPMessage response = { 0 };
|
|
GstRTSPStream *stream = NULL;
|
|
GstRTSPLowerTrans protocols;
|
|
GstRTSPStatusCode code;
|
|
gboolean unsupported_real = FALSE;
|
|
gint rtpport, rtcpport;
|
|
GstRTSPUrl *url;
|
|
gchar *hval;
|
|
|
|
if (src->conninfo.connection) {
|
|
url = gst_rtsp_connection_get_url (src->conninfo.connection);
|
|
/* we initially allow all configured lower transports. based on the URL
|
|
* transports and the replies from the server we narrow them down. */
|
|
protocols = url->transports & src->cur_protocols;
|
|
} else {
|
|
url = NULL;
|
|
protocols = src->cur_protocols;
|
|
}
|
|
|
|
if (protocols == 0)
|
|
goto no_protocols;
|
|
|
|
/* reset some state */
|
|
src->free_channel = 0;
|
|
src->interleaved = FALSE;
|
|
src->need_activate = FALSE;
|
|
/* keep track of next port number, 0 is random */
|
|
src->next_port_num = src->client_port_range.min;
|
|
rtpport = rtcpport = 0;
|
|
|
|
if (G_UNLIKELY (src->streams == NULL))
|
|
goto no_streams;
|
|
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPConnection *conn;
|
|
gchar *transports;
|
|
gint retry = 0;
|
|
guint mask = 0;
|
|
|
|
stream = (GstRTSPStream *) walk->data;
|
|
|
|
/* see if we need to configure this stream */
|
|
if (!gst_rtsp_ext_list_configure_stream (src->extensions, stream->caps)) {
|
|
GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
|
|
stream);
|
|
stream->disabled = TRUE;
|
|
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->conninfo.location == NULL) {
|
|
GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
|
|
continue;
|
|
}
|
|
|
|
if (src->conninfo.connection == NULL) {
|
|
if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
|
|
GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
|
|
continue;
|
|
}
|
|
conn = stream->conninfo.connection;
|
|
} else {
|
|
conn = src->conninfo.connection;
|
|
}
|
|
GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
|
|
stream->conninfo.location);
|
|
|
|
/* if we have a multicast connection, only suggest multicast from now on */
|
|
if (stream->is_multicast)
|
|
protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
|
|
|
|
next_protocol:
|
|
/* first selectable protocol */
|
|
while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
|
|
mask++;
|
|
if (!protocol_masks[mask])
|
|
goto no_protocols;
|
|
|
|
retry:
|
|
GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
|
|
protocol_masks[mask]);
|
|
/* create a string with first transport in line */
|
|
transports = NULL;
|
|
res = gst_rtspsrc_create_transports_string (src,
|
|
protocols & protocol_masks[mask], &transports);
|
|
if (res < 0 || transports == NULL)
|
|
goto setup_transport_failed;
|
|
|
|
if (strlen (transports) == 0) {
|
|
g_free (transports);
|
|
GST_DEBUG_OBJECT (src, "no transports found");
|
|
mask++;
|
|
goto next_protocol;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
|
|
|
|
/* 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,
|
|
retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
|
|
if (res < 0) {
|
|
g_free (transports);
|
|
goto setup_transport_failed;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
|
|
|
|
/* create SETUP request */
|
|
res =
|
|
gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
|
|
stream->conninfo.location);
|
|
if (res < 0) {
|
|
g_free (transports);
|
|
goto create_request_failed;
|
|
}
|
|
|
|
/* select transport, copy is made when adding to header so we can free it. */
|
|
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
|
|
g_free (transports);
|
|
|
|
/* if the user wants a non default RTP packet size we add the blocksize
|
|
* parameter */
|
|
if (src->rtp_blocksize > 0) {
|
|
hval = g_strdup_printf ("%d", src->rtp_blocksize);
|
|
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
|
|
g_free (hval);
|
|
}
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
|
|
stream->id));
|
|
|
|
/* handle the code ourselves */
|
|
if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 0))
|
|
goto send_error;
|
|
|
|
switch (code) {
|
|
case GST_RTSP_STS_OK:
|
|
break;
|
|
case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
/* cleanup of leftover transport */
|
|
gst_rtspsrc_stream_free_udp (stream);
|
|
/* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
|
|
* we might be in this case */
|
|
if (stream->container && rtpport && rtcpport && !retry) {
|
|
GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
|
|
rtpport, rtcpport);
|
|
retry++;
|
|
goto retry;
|
|
}
|
|
/* this transport did not go down well, but we may have others to try
|
|
* that we did not send yet, try those and only give up then
|
|
* but not without checking for lost cause/extension so we can
|
|
* post a nicer/more useful error message later */
|
|
if (!unsupported_real)
|
|
unsupported_real = gst_rtspsrc_stream_is_real_media (stream);
|
|
/* select next available protocol, give up on this stream if none */
|
|
mask++;
|
|
while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
|
|
mask++;
|
|
if (!protocol_masks[mask] || unsupported_real)
|
|
continue;
|
|
else
|
|
goto retry;
|
|
default:
|
|
/* cleanup of leftover transport and move to the next stream */
|
|
gst_rtspsrc_stream_free_udp (stream);
|
|
goto response_error;
|
|
}
|
|
|
|
/* parse response transport */
|
|
{
|
|
gchar *resptrans = NULL;
|
|
GstRTSPTransport transport = { 0 };
|
|
|
|
gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
|
|
&resptrans, 0);
|
|
if (!resptrans) {
|
|
gst_rtspsrc_stream_free_udp (stream);
|
|
goto no_transport;
|
|
}
|
|
|
|
/* parse transport, go to next stream on parse error */
|
|
if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
|
|
GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
|
|
goto next;
|
|
}
|
|
|
|
/* 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 GST_RTSP_LOWER_TRANS_TCP:
|
|
GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
|
|
protocols = GST_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 GST_RTSP_LOWER_TRANS_UDP_MCAST:
|
|
/* only allow multicast for other streams */
|
|
GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
|
|
protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
|
|
break;
|
|
case GST_RTSP_LOWER_TRANS_UDP:
|
|
/* only allow unicast for other streams */
|
|
GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
|
|
protocols = GST_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 && !retry)) {
|
|
/* 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);
|
|
goto next;
|
|
} else if (stream->udpsrc[0] && stream->udpsrc[1]) {
|
|
/* retain the first allocated UDP port pair */
|
|
g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
|
|
g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
|
|
}
|
|
}
|
|
/* we need to activate at least one streams when we detect activity */
|
|
src->need_activate = TRUE;
|
|
next:
|
|
/* clean up our transport struct */
|
|
gst_rtsp_transport_init (&transport);
|
|
/* clean up used RTSP messages */
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
}
|
|
}
|
|
|
|
/* store the transport protocol that was configured */
|
|
src->cur_protocols = protocols;
|
|
|
|
gst_rtsp_ext_list_stream_select (src->extensions, url);
|
|
|
|
/* if there is nothing to activate, error out */
|
|
if (!src->need_activate)
|
|
goto nothing_to_activate;
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
no_protocols:
|
|
{
|
|
/* no transport possible, post an error and stop */
|
|
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
|
|
("Could not connect to server, no protocols left"));
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
no_streams:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
("SDP contains no streams"));
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
create_request_failed:
|
|
{
|
|
gchar *str = gst_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."));
|
|
res = GST_RTSP_ERROR;
|
|
goto cleanup_error;
|
|
}
|
|
response_error:
|
|
{
|
|
const gchar *str = gst_rtsp_status_as_text (code);
|
|
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Error (%d): %s", code, GST_STR_NULL (str)));
|
|
res = GST_RTSP_ERROR;
|
|
goto cleanup_error;
|
|
}
|
|
send_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (src, "send interrupted");
|
|
}
|
|
g_free (str);
|
|
goto cleanup_error;
|
|
}
|
|
no_transport:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
("Server did not select transport."));
|
|
res = GST_RTSP_ERROR;
|
|
goto cleanup_error;
|
|
}
|
|
nothing_to_activate:
|
|
{
|
|
/* none of the available error codes is really right .. */
|
|
if (unsupported_real) {
|
|
GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
|
|
(_("No supported stream was found. You might need to install a "
|
|
"GStreamer RTSP extension plugin for Real media streams.")),
|
|
(NULL));
|
|
} else {
|
|
GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
|
|
(_("No supported stream was found. You might need to allow "
|
|
"more transport protocols or may otherwise be missing "
|
|
"the right GStreamer RTSP extension plugin.")), (NULL));
|
|
}
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
cleanup_error:
|
|
{
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
|
|
GstSegment * segment)
|
|
{
|
|
gint64 seconds;
|
|
GstRTSPTimeRange *therange;
|
|
|
|
if (src->range)
|
|
gst_rtsp_range_free (src->range);
|
|
|
|
if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
|
|
GST_DEBUG_OBJECT (src, "parsed range %s", range);
|
|
src->range = therange;
|
|
} else {
|
|
GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
|
|
src->range = NULL;
|
|
gst_segment_init (segment, GST_FORMAT_TIME);
|
|
return FALSE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
|
|
therange->min.type, therange->min.seconds, therange->max.type,
|
|
therange->max.seconds);
|
|
|
|
if (therange->min.type == GST_RTSP_TIME_NOW)
|
|
seconds = 0;
|
|
else if (therange->min.type == GST_RTSP_TIME_END)
|
|
seconds = 0;
|
|
else
|
|
seconds = therange->min.seconds * GST_SECOND;
|
|
|
|
GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (seconds));
|
|
|
|
/* we need to start playback without clipping from the position reported by
|
|
* the server */
|
|
segment->start = seconds;
|
|
segment->last_stop = seconds;
|
|
|
|
if (therange->max.type == GST_RTSP_TIME_NOW)
|
|
seconds = -1;
|
|
else if (therange->max.type == GST_RTSP_TIME_END)
|
|
seconds = -1;
|
|
else
|
|
seconds = therange->max.seconds * GST_SECOND;
|
|
|
|
GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (seconds));
|
|
|
|
/* live (WMS) server might send overflowed large max as its idea of infinity,
|
|
* compensate to prevent problems later on */
|
|
if (seconds != -1 && seconds < 0) {
|
|
seconds = -1;
|
|
GST_DEBUG_OBJECT (src, "insane range, set to NONE");
|
|
}
|
|
|
|
/* live (WMS) might send min == max, which is not worth recording */
|
|
if (segment->duration == -1 && seconds == segment->start)
|
|
seconds = -1;
|
|
|
|
/* don't change duration with unknown value, we might have a valid value
|
|
* there that we want to keep. */
|
|
if (seconds != -1)
|
|
gst_segment_set_duration (segment, GST_FORMAT_TIME, seconds);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* must be called with the RTSP state lock */
|
|
static GstRTSPResult
|
|
gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
|
|
gboolean async)
|
|
{
|
|
GstRTSPResult res;
|
|
gint i, n_streams;
|
|
|
|
/* prepare global stream caps properties */
|
|
if (src->props)
|
|
gst_structure_remove_all_fields (src->props);
|
|
else
|
|
src->props = gst_structure_empty_new ("RTSPProperties");
|
|
|
|
if (src->debug)
|
|
gst_sdp_message_dump (sdp);
|
|
|
|
gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
|
|
|
|
gst_segment_init (&src->segment, GST_FORMAT_TIME);
|
|
|
|
/* parse range for duration reporting. */
|
|
{
|
|
const gchar *range;
|
|
|
|
for (i = 0;; i++) {
|
|
range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
|
|
if (range == NULL)
|
|
break;
|
|
|
|
/* keep track of the range and configure it in the segment */
|
|
if (gst_rtspsrc_parse_range (src, range, &src->segment))
|
|
break;
|
|
}
|
|
}
|
|
/* try to find a global control attribute. Note that a '*' means that we should
|
|
* do aggregate control with the current url (so we don't do anything and
|
|
* leave the current connection as is) */
|
|
{
|
|
const gchar *control;
|
|
|
|
for (i = 0;; i++) {
|
|
control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
|
|
if (control == NULL)
|
|
break;
|
|
|
|
/* only take fully qualified urls */
|
|
if (g_str_has_prefix (control, "rtsp://"))
|
|
break;
|
|
}
|
|
if (control) {
|
|
g_free (src->conninfo.location);
|
|
src->conninfo.location = g_strdup (control);
|
|
/* make a connection for this, if there was a connection already, nothing
|
|
* happens. */
|
|
if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
|
|
GST_ERROR_OBJECT (src, "could not connect");
|
|
}
|
|
}
|
|
/* we need to keep the control url separate from the connection url because
|
|
* the rules for constructing the media control url need it */
|
|
g_free (src->control);
|
|
src->control = g_strdup (control);
|
|
}
|
|
|
|
/* create streams */
|
|
n_streams = gst_sdp_message_medias_len (sdp);
|
|
for (i = 0; i < n_streams; i++) {
|
|
gst_rtspsrc_create_stream (src, sdp, i);
|
|
}
|
|
|
|
src->state = GST_RTSP_STATE_INIT;
|
|
|
|
/* setup streams */
|
|
if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
|
|
goto setup_failed;
|
|
|
|
/* reset our state */
|
|
src->need_range = TRUE;
|
|
src->skip = FALSE;
|
|
|
|
src->state = GST_RTSP_STATE_READY;
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
setup_failed:
|
|
{
|
|
GST_ERROR_OBJECT (src, "setup failed");
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
|
|
gboolean async)
|
|
{
|
|
GstRTSPResult res;
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPMessage response = { 0 };
|
|
guint8 *data;
|
|
guint size;
|
|
gchar *respcont = NULL;
|
|
|
|
restart:
|
|
src->need_redirect = FALSE;
|
|
|
|
/* can't continue without a valid url */
|
|
if (G_UNLIKELY (src->conninfo.url == NULL)) {
|
|
res = GST_RTSP_EINVAL;
|
|
goto no_url;
|
|
}
|
|
src->tried_url_auth = FALSE;
|
|
|
|
if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
|
|
goto connect_failed;
|
|
|
|
/* create OPTIONS */
|
|
GST_DEBUG_OBJECT (src, "create options...");
|
|
res =
|
|
gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
|
|
src->conninfo.url_str);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
/* send OPTIONS */
|
|
GST_DEBUG_OBJECT (src, "send options...");
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
|
|
|
|
if ((res =
|
|
gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
|
|
NULL)) < 0)
|
|
goto send_error;
|
|
|
|
/* parse OPTIONS */
|
|
if (!gst_rtspsrc_parse_methods (src, &response))
|
|
goto methods_error;
|
|
|
|
/* create DESCRIBE */
|
|
GST_DEBUG_OBJECT (src, "create describe...");
|
|
res =
|
|
gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
|
|
src->conninfo.url_str);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
/* we only accept SDP for now */
|
|
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
|
|
"application/sdp");
|
|
|
|
/* send DESCRIBE */
|
|
GST_DEBUG_OBJECT (src, "send describe...");
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
|
|
|
|
if ((res =
|
|
gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
|
|
NULL)) < 0)
|
|
goto send_error;
|
|
|
|
/* we only perform redirect for the describe, currently */
|
|
if (src->need_redirect) {
|
|
/* close connection, we don't have to send a TEARDOWN yet, ignore the
|
|
* result. */
|
|
gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
|
|
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
|
|
/* and now retry */
|
|
goto restart;
|
|
}
|
|
|
|
/* it could be that the DESCRIBE method was not implemented */
|
|
if (!src->methods & GST_RTSP_DESCRIBE)
|
|
goto no_describe;
|
|
|
|
/* check if reply is SDP */
|
|
gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
|
|
0);
|
|
/* 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 */
|
|
gst_rtsp_message_get_body (&response, &data, &size);
|
|
if (data == NULL || size == 0)
|
|
goto no_describe;
|
|
|
|
GST_DEBUG_OBJECT (src, "parse SDP...");
|
|
gst_sdp_message_new (sdp);
|
|
gst_sdp_message_parse_buffer (data, size, *sdp);
|
|
|
|
/* clean up any messages */
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
no_url:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
|
|
("No valid RTSP URL was provided"));
|
|
goto cleanup_error;
|
|
}
|
|
connect_failed:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
|
|
("Failed to connect. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (src, "connect interrupted");
|
|
}
|
|
g_free (str);
|
|
goto cleanup_error;
|
|
}
|
|
create_request_failed:
|
|
{
|
|
gchar *str = gst_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 */
|
|
res = GST_RTSP_ERROR;
|
|
goto cleanup_error;
|
|
}
|
|
wrong_content_type:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
("Server does not support SDP, got %s.", respcont));
|
|
res = GST_RTSP_ERROR;
|
|
goto cleanup_error;
|
|
}
|
|
no_describe:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
("Server can not provide an SDP."));
|
|
res = GST_RTSP_ERROR;
|
|
goto cleanup_error;
|
|
}
|
|
cleanup_error:
|
|
{
|
|
if (src->conninfo.connection) {
|
|
GST_DEBUG_OBJECT (src, "free connection");
|
|
gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
|
|
}
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
|
|
{
|
|
GstRTSPResult ret;
|
|
|
|
src->methods =
|
|
GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
|
|
|
|
if (src->sdp == NULL) {
|
|
if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
|
|
goto no_sdp;
|
|
}
|
|
|
|
if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
|
|
goto open_failed;
|
|
|
|
done:
|
|
if (async)
|
|
gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
no_sdp:
|
|
{
|
|
GST_WARNING_OBJECT (src, "can't get sdp");
|
|
src->open_error = TRUE;
|
|
goto done;
|
|
}
|
|
open_failed:
|
|
{
|
|
GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
|
|
src->open_error = TRUE;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
|
|
{
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPMessage response = { 0 };
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
GList *walk;
|
|
gchar *control;
|
|
|
|
GST_DEBUG_OBJECT (src, "TEARDOWN...");
|
|
|
|
if (src->state < GST_RTSP_STATE_READY) {
|
|
GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
|
|
goto close;
|
|
}
|
|
|
|
if (only_close)
|
|
goto close;
|
|
|
|
/* construct a control url */
|
|
if (src->control)
|
|
control = src->control;
|
|
else
|
|
control = src->conninfo.url_str;
|
|
|
|
if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
|
|
goto not_supported;
|
|
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
gchar *setup_url;
|
|
GstRTSPConnInfo *info;
|
|
|
|
/* try aggregate control first but do non-aggregate control otherwise */
|
|
if (control)
|
|
setup_url = control;
|
|
else if ((setup_url = stream->conninfo.location) == NULL)
|
|
continue;
|
|
|
|
if (src->conninfo.connection) {
|
|
info = &src->conninfo;
|
|
} else if (stream->conninfo.connection) {
|
|
info = &stream->conninfo;
|
|
} else {
|
|
continue;
|
|
}
|
|
if (!info->connected)
|
|
goto next;
|
|
|
|
/* do TEARDOWN */
|
|
res =
|
|
gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
|
|
|
|
if ((res =
|
|
gst_rtspsrc_send (src, info->connection, &request, &response,
|
|
NULL)) < 0)
|
|
goto send_error;
|
|
|
|
/* FIXME, parse result? */
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
|
|
next:
|
|
/* early exit when we did aggregate control */
|
|
if (control)
|
|
break;
|
|
}
|
|
|
|
close:
|
|
/* close connections */
|
|
GST_DEBUG_OBJECT (src, "closing connection...");
|
|
gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
|
|
}
|
|
|
|
/* cleanup */
|
|
gst_rtspsrc_cleanup (src);
|
|
|
|
src->state = GST_RTSP_STATE_INVALID;
|
|
|
|
if (async)
|
|
gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
create_request_failed:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
|
|
("Could not create request. (%s)", str));
|
|
g_free (str);
|
|
goto close;
|
|
}
|
|
send_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
gst_rtsp_message_unset (&request);
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
|
|
}
|
|
g_free (str);
|
|
goto close;
|
|
}
|
|
not_supported:
|
|
{
|
|
GST_DEBUG_OBJECT (src,
|
|
"TEARDOWN and PLAY not supported, can't do TEARDOWN");
|
|
goto close;
|
|
}
|
|
}
|
|
|
|
/* RTP-Info is of the format:
|
|
*
|
|
* url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
|
|
*
|
|
* rtptime corresponds to the timestamp for the NPT time given in the header
|
|
* seqbase corresponds to the next sequence number we received. This number
|
|
* indicates the first seqnum after the seek and should be used to discard
|
|
* packets that are from before the seek.
|
|
*/
|
|
static gboolean
|
|
gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
|
|
{
|
|
gchar **infos;
|
|
gint i, j;
|
|
|
|
GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
|
|
|
|
infos = g_strsplit (rtpinfo, ",", 0);
|
|
for (i = 0; infos[i]; i++) {
|
|
gchar **fields;
|
|
GstRTSPStream *stream;
|
|
gint32 seqbase;
|
|
gint64 timebase;
|
|
|
|
GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
|
|
|
|
/* init values, types of seqbase and timebase are bigger than needed so we
|
|
* can store -1 as uninitialized values */
|
|
stream = NULL;
|
|
seqbase = -1;
|
|
timebase = -1;
|
|
|
|
/* 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. */
|
|
fields = g_strsplit (infos[i], ";", 0);
|
|
for (j = 0; fields[j]; j++) {
|
|
GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
|
|
/* remove leading whitespace */
|
|
fields[j] = g_strchug (fields[j]);
|
|
if (g_str_has_prefix (fields[j], "url=")) {
|
|
/* get the url and the stream */
|
|
stream =
|
|
find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
|
|
} else if (g_str_has_prefix (fields[j], "seq=")) {
|
|
seqbase = atoi (fields[j] + 4);
|
|
} else if (g_str_has_prefix (fields[j], "rtptime=")) {
|
|
timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
|
|
}
|
|
}
|
|
g_strfreev (fields);
|
|
/* now we need to store the values for the caps of the stream */
|
|
if (stream != NULL) {
|
|
GST_DEBUG_OBJECT (src,
|
|
"found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
|
|
stream, seqbase, timebase);
|
|
|
|
/* we have a stream, configure detected params */
|
|
stream->seqbase = seqbase;
|
|
stream->timebase = timebase;
|
|
}
|
|
}
|
|
g_strfreev (infos);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
|
|
{
|
|
guint64 interval;
|
|
GList *walk;
|
|
|
|
interval = strtoul (rtcp, NULL, 10);
|
|
GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
|
|
|
|
if (!interval)
|
|
return;
|
|
|
|
interval *= GST_MSECOND;
|
|
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
|
|
/* already (optionally) retrieved this when configuring manager */
|
|
if (stream->session) {
|
|
GObject *rtpsession = stream->session;
|
|
|
|
GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
|
|
rtpsession);
|
|
g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
|
|
}
|
|
}
|
|
|
|
/* now it happens that (Xenon) server sending this may also provide bogus
|
|
* RTCP SR sync data (i.e. with quite some jitter), so never mind those
|
|
* and just use RTP-Info to sync */
|
|
if (src->manager) {
|
|
GObjectClass *klass;
|
|
|
|
klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
|
|
if (g_object_class_find_property (klass, "rtcp-sync")) {
|
|
GST_DEBUG_OBJECT (src, "configuring rtp sync method");
|
|
g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
|
|
}
|
|
}
|
|
}
|
|
|
|
static gdouble
|
|
gst_rtspsrc_get_float (const gchar * dstr)
|
|
{
|
|
gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
|
|
|
|
/* canonicalise floating point string so we can handle float strings
|
|
* in the form "24.930" or "24,930" irrespective of the current locale */
|
|
g_strlcpy (s, dstr, sizeof (s));
|
|
g_strdelimit (s, ",", '.');
|
|
return g_ascii_strtod (s, NULL);
|
|
}
|
|
|
|
static gchar *
|
|
gen_range_header (GstRTSPSrc * src, GstSegment * segment)
|
|
{
|
|
gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
|
|
|
|
if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
|
|
g_strlcpy (val_str, "now", sizeof (val_str));
|
|
} else {
|
|
if (segment->last_stop == 0) {
|
|
g_strlcpy (val_str, "0", sizeof (val_str));
|
|
} else {
|
|
g_ascii_dtostr (val_str, sizeof (val_str),
|
|
((gdouble) segment->last_stop) / GST_SECOND);
|
|
}
|
|
}
|
|
return g_strdup_printf ("npt=%s-", val_str);
|
|
}
|
|
|
|
static void
|
|
clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
|
|
{
|
|
stream->timebase = -1;
|
|
stream->seqbase = -1;
|
|
if (stream->caps) {
|
|
GstStructure *s;
|
|
|
|
stream->caps = gst_caps_make_writable (stream->caps);
|
|
s = gst_caps_get_structure (stream->caps, 0);
|
|
gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
|
|
{
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
|
|
if (src->state < GST_RTSP_STATE_READY) {
|
|
res = GST_RTSP_ERROR;
|
|
if (src->open_error) {
|
|
GST_DEBUG_OBJECT (src, "the stream was in error");
|
|
goto done;
|
|
}
|
|
if (async)
|
|
gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
|
|
|
|
if ((res = gst_rtspsrc_open (src, async)) < 0) {
|
|
GST_DEBUG_OBJECT (src, "failed to open stream");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
done:
|
|
return res;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
|
|
{
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPMessage response = { 0 };
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
GList *walk;
|
|
gchar *hval;
|
|
gint hval_idx;
|
|
gchar *control;
|
|
|
|
GST_DEBUG_OBJECT (src, "PLAY...");
|
|
|
|
if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
|
|
goto open_failed;
|
|
|
|
if (!(src->methods & GST_RTSP_PLAY))
|
|
goto not_supported;
|
|
|
|
if (src->state == GST_RTSP_STATE_PLAYING)
|
|
goto was_playing;
|
|
|
|
if (!src->conninfo.connection || !src->conninfo.connected)
|
|
goto done;
|
|
|
|
/* send some dummy packets before we activate the receive in the
|
|
* udp sources */
|
|
gst_rtspsrc_send_dummy_packets (src);
|
|
|
|
/* activate receive elements;
|
|
* only in async case, since receive elements may not have been affected
|
|
* by overall state change (e.g. not around yet),
|
|
* do not mess with state in sync case (e.g. seeking) */
|
|
if (async)
|
|
gst_element_set_state (GST_ELEMENT_CAST (src), GST_STATE_PLAYING);
|
|
|
|
/* construct a control url */
|
|
if (src->control)
|
|
control = src->control;
|
|
else
|
|
control = src->conninfo.url_str;
|
|
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
gchar *setup_url;
|
|
GstRTSPConnection *conn;
|
|
|
|
/* try aggregate control first but do non-aggregate control otherwise */
|
|
if (control)
|
|
setup_url = control;
|
|
else if ((setup_url = stream->conninfo.location) == NULL)
|
|
continue;
|
|
|
|
if (src->conninfo.connection) {
|
|
conn = src->conninfo.connection;
|
|
} else if (stream->conninfo.connection) {
|
|
conn = stream->conninfo.connection;
|
|
} else {
|
|
continue;
|
|
}
|
|
|
|
/* do play */
|
|
res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
if (src->need_range) {
|
|
hval = gen_range_header (src, segment);
|
|
|
|
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
|
|
g_free (hval);
|
|
}
|
|
|
|
if (segment->rate != 1.0) {
|
|
gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
|
|
|
|
g_ascii_dtostr (hval, sizeof (hval), segment->rate);
|
|
if (src->skip)
|
|
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
|
|
else
|
|
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
|
|
}
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
|
|
|
|
if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
|
|
goto send_error;
|
|
|
|
/* seek may have silently failed as it is not supported */
|
|
if (!(src->methods & GST_RTSP_PLAY)) {
|
|
GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
|
|
/* obviously it is supported as we made it here */
|
|
src->methods |= GST_RTSP_PLAY;
|
|
src->seekable = FALSE;
|
|
/* but there is nothing to parse in the response,
|
|
* so convey we have no idea and not to expect anything particular */
|
|
clear_rtp_base (src, stream);
|
|
if (control) {
|
|
GList *run;
|
|
|
|
/* need to do for all streams */
|
|
for (run = src->streams; run; run = g_list_next (run))
|
|
clear_rtp_base (src, (GstRTSPStream *) run->data);
|
|
}
|
|
/* NOTE the above also disables npt based eos detection */
|
|
/* and below forces position to 0,
|
|
* which is visible feedback we lost the plot */
|
|
segment->start = segment->last_stop = src->last_pos;
|
|
}
|
|
|
|
gst_rtsp_message_unset (&request);
|
|
|
|
/* parse RTP npt field. This is the current position in the stream (Normal
|
|
* Play Time) and should be put in the NEWSEGMENT position field. */
|
|
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
|
|
0) == GST_RTSP_OK)
|
|
gst_rtspsrc_parse_range (src, hval, segment);
|
|
|
|
/* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
|
|
segment->rate = 1.0;
|
|
|
|
/* parse Speed header. This is the intended playback rate of the stream
|
|
* and should be put in the NEWSEGMENT rate field. */
|
|
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
|
|
0) == GST_RTSP_OK) {
|
|
segment->rate = gst_rtspsrc_get_float (hval);
|
|
} else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
|
|
&hval, 0) == GST_RTSP_OK) {
|
|
segment->rate = gst_rtspsrc_get_float (hval);
|
|
}
|
|
|
|
/* 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...
|
|
* This is info for the RTP session manager that we pass to it in caps. */
|
|
hval_idx = 0;
|
|
while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
|
|
&hval, hval_idx++) == GST_RTSP_OK)
|
|
gst_rtspsrc_parse_rtpinfo (src, hval);
|
|
|
|
/* some servers indicate RTCP parameters in PLAY response,
|
|
* rather than properly in SDP */
|
|
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
|
|
&hval, 0) == GST_RTSP_OK)
|
|
gst_rtspsrc_handle_rtcp_interval (src, hval);
|
|
|
|
gst_rtsp_message_unset (&response);
|
|
|
|
/* early exit when we did aggregate control */
|
|
if (control)
|
|
break;
|
|
}
|
|
/* set again when needed */
|
|
src->need_range = FALSE;
|
|
|
|
/* configure the caps of the streams after we parsed all headers. */
|
|
gst_rtspsrc_configure_caps (src, segment);
|
|
|
|
src->running = TRUE;
|
|
src->base_time = -1;
|
|
src->state = GST_RTSP_STATE_PLAYING;
|
|
|
|
/* mark discont */
|
|
GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
stream->discont = TRUE;
|
|
}
|
|
|
|
done:
|
|
if (async)
|
|
gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
open_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "failed to open stream");
|
|
goto done;
|
|
}
|
|
not_supported:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "PLAY is not supported");
|
|
goto done;
|
|
}
|
|
was_playing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we were already PLAYING");
|
|
goto done;
|
|
}
|
|
create_request_failed:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
|
|
("Could not create request. (%s)", str));
|
|
g_free (str);
|
|
goto done;
|
|
}
|
|
send_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
gst_rtsp_message_unset (&request);
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (src, "PLAY interrupted");
|
|
}
|
|
g_free (str);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle, gboolean async)
|
|
{
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPMessage response = { 0 };
|
|
GList *walk;
|
|
gchar *control;
|
|
|
|
GST_DEBUG_OBJECT (src, "PAUSE...");
|
|
|
|
if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
|
|
goto open_failed;
|
|
|
|
if (!(src->methods & GST_RTSP_PAUSE))
|
|
goto not_supported;
|
|
|
|
if (src->state == GST_RTSP_STATE_READY)
|
|
goto was_paused;
|
|
|
|
if (!src->conninfo.connection || !src->conninfo.connected)
|
|
goto no_connection;
|
|
|
|
/* construct a control url */
|
|
if (src->control)
|
|
control = src->control;
|
|
else
|
|
control = src->conninfo.url_str;
|
|
|
|
/* loop over the streams. We might exit the loop early when we could do an
|
|
* aggregate control */
|
|
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
|
GstRTSPConnection *conn;
|
|
gchar *setup_url;
|
|
|
|
/* try aggregate control first but do non-aggregate control otherwise */
|
|
if (control)
|
|
setup_url = control;
|
|
else if ((setup_url = stream->conninfo.location) == NULL)
|
|
continue;
|
|
|
|
if (src->conninfo.connection) {
|
|
conn = src->conninfo.connection;
|
|
} else if (stream->conninfo.connection) {
|
|
conn = stream->conninfo.connection;
|
|
} else {
|
|
continue;
|
|
}
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
|
|
("Sending PAUSE request"));
|
|
|
|
if ((res =
|
|
gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
|
|
setup_url)) < 0)
|
|
goto create_request_failed;
|
|
|
|
if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
|
|
goto send_error;
|
|
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
|
|
/* exit early when we did agregate control */
|
|
if (control)
|
|
break;
|
|
}
|
|
|
|
no_connection:
|
|
src->state = GST_RTSP_STATE_READY;
|
|
|
|
done:
|
|
if (async)
|
|
gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
open_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "failed to open stream");
|
|
goto done;
|
|
}
|
|
not_supported:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "PAUSE is not supported");
|
|
goto done;
|
|
}
|
|
was_paused:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we were already PAUSED");
|
|
goto done;
|
|
}
|
|
create_request_failed:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
|
|
("Could not create request. (%s)", str));
|
|
g_free (str);
|
|
goto done;
|
|
}
|
|
send_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
gst_rtsp_message_unset (&request);
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (src, "PAUSE interrupted");
|
|
}
|
|
g_free (str);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
|
|
{
|
|
GstRTSPSrc *rtspsrc;
|
|
|
|
rtspsrc = GST_RTSPSRC (bin);
|
|
|
|
switch (GST_MESSAGE_TYPE (message)) {
|
|
case GST_MESSAGE_EOS:
|
|
gst_message_unref (message);
|
|
break;
|
|
case GST_MESSAGE_ELEMENT:
|
|
{
|
|
const GstStructure *s = gst_message_get_structure (message);
|
|
|
|
if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
|
|
gboolean ignore_timeout;
|
|
|
|
GST_DEBUG_OBJECT (bin, "timeout on UDP port");
|
|
|
|
GST_OBJECT_LOCK (rtspsrc);
|
|
ignore_timeout = rtspsrc->ignore_timeout;
|
|
rtspsrc->ignore_timeout = TRUE;
|
|
GST_OBJECT_UNLOCK (rtspsrc);
|
|
|
|
/* we only act on the first udp timeout message, others are irrelevant
|
|
* and can be ignored. */
|
|
if (!ignore_timeout)
|
|
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, TRUE);
|
|
/* eat and free */
|
|
gst_message_unref (message);
|
|
return;
|
|
}
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
case GST_MESSAGE_ERROR:
|
|
{
|
|
GstObject *udpsrc;
|
|
GstRTSPStream *stream;
|
|
GstFlowReturn ret;
|
|
|
|
udpsrc = GST_MESSAGE_SRC (message);
|
|
|
|
GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
|
|
GST_ELEMENT_NAME (udpsrc));
|
|
|
|
stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
|
|
if (!stream)
|
|
goto forward;
|
|
|
|
/* we ignore the RTCP udpsrc */
|
|
if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
|
|
goto done;
|
|
|
|
/* if we get error messages from the udp sources, that's not a problem as
|
|
* long as not all of them error out. We also don't really know what the
|
|
* problem is, the message does not give enough detail... */
|
|
ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
|
|
GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
|
|
if (ret != GST_FLOW_OK)
|
|
goto forward;
|
|
|
|
done:
|
|
gst_message_unref (message);
|
|
break;
|
|
|
|
forward:
|
|
/* fatal but not our message, forward */
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
default:
|
|
{
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* the thread where everything happens */
|
|
static void
|
|
gst_rtspsrc_thread (GstRTSPSrc * src)
|
|
{
|
|
gint cmd;
|
|
GstRTSPResult ret;
|
|
gboolean running = FALSE;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
cmd = src->loop_cmd;
|
|
src->loop_cmd = CMD_WAIT;
|
|
GST_DEBUG_OBJECT (src, "got command %d", cmd);
|
|
|
|
/* we got the message command, so ensure communication is possible again */
|
|
gst_rtspsrc_connection_flush (src, FALSE);
|
|
|
|
/* we allow these to be interrupted */
|
|
if (cmd == CMD_LOOP || cmd == CMD_CLOSE || cmd == CMD_PAUSE)
|
|
src->waiting = TRUE;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
switch (cmd) {
|
|
case CMD_OPEN:
|
|
ret = gst_rtspsrc_open (src, TRUE);
|
|
break;
|
|
case CMD_PLAY:
|
|
ret = gst_rtspsrc_play (src, &src->segment, TRUE);
|
|
if (ret == GST_RTSP_OK)
|
|
running = TRUE;
|
|
break;
|
|
case CMD_PAUSE:
|
|
ret = gst_rtspsrc_pause (src, TRUE, TRUE);
|
|
if (ret == GST_RTSP_OK)
|
|
running = TRUE;
|
|
break;
|
|
case CMD_CLOSE:
|
|
ret = gst_rtspsrc_close (src, TRUE, FALSE);
|
|
break;
|
|
case CMD_LOOP:
|
|
running = gst_rtspsrc_loop (src);
|
|
break;
|
|
case CMD_RECONNECT:
|
|
ret = gst_rtspsrc_reconnect (src, FALSE);
|
|
if (ret == GST_RTSP_OK)
|
|
running = TRUE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
/* and go back to sleep */
|
|
if (src->loop_cmd == CMD_WAIT) {
|
|
if (running)
|
|
src->loop_cmd = CMD_LOOP;
|
|
else if (src->task)
|
|
gst_task_pause (src->task);
|
|
}
|
|
/* reset waiting */
|
|
src->waiting = FALSE;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_start (GstRTSPSrc * src)
|
|
{
|
|
GST_DEBUG_OBJECT (src, "starting");
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
|
|
src->loop_cmd = CMD_WAIT;
|
|
|
|
if (src->task == NULL) {
|
|
src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_thread, src);
|
|
if (src->task == NULL)
|
|
goto task_error;
|
|
|
|
gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
|
|
}
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
task_error:
|
|
{
|
|
GST_ERROR_OBJECT (src, "failed to create task");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_stop (GstRTSPSrc * src)
|
|
{
|
|
GstTask *task;
|
|
|
|
GST_DEBUG_OBJECT (src, "stopping");
|
|
|
|
/* also cancels pending task */
|
|
gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, TRUE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
if ((task = src->task)) {
|
|
src->task = NULL;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
gst_task_stop (task);
|
|
|
|
/* make sure it is not running */
|
|
GST_RTSP_STREAM_LOCK (src);
|
|
GST_RTSP_STREAM_UNLOCK (src);
|
|
|
|
/* now wait for the task to finish */
|
|
gst_task_join (task);
|
|
|
|
/* and free the task */
|
|
gst_object_unref (GST_OBJECT (task));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
}
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
/* ensure synchronously all is closed and clean */
|
|
gst_rtspsrc_close (src, FALSE, TRUE);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
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:
|
|
if (!gst_rtspsrc_start (rtspsrc))
|
|
goto start_failed;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
/* init some state */
|
|
rtspsrc->cur_protocols = rtspsrc->protocols;
|
|
/* first attempt, don't ignore timeouts */
|
|
rtspsrc->ignore_timeout = FALSE;
|
|
rtspsrc->open_error = FALSE;
|
|
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, FALSE);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
/* unblock the tcp tasks and make the loop waiting */
|
|
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, TRUE);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
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_PAUSED_TO_PLAYING:
|
|
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, FALSE);
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
/* send pause request and keep the idle task around */
|
|
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, FALSE);
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, FALSE);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
gst_rtspsrc_stop (rtspsrc);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
|
|
start_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (rtspsrc, "start failed");
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
|
|
{
|
|
gboolean res;
|
|
GstRTSPSrc *rtspsrc;
|
|
|
|
rtspsrc = GST_RTSPSRC (element);
|
|
|
|
if (GST_EVENT_IS_DOWNSTREAM (event)) {
|
|
res = gst_rtspsrc_push_event (rtspsrc, event, TRUE);
|
|
} else {
|
|
res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
|
|
/*** GSTURIHANDLER INTERFACE *************************************************/
|
|
|
|
static GstURIType
|
|
gst_rtspsrc_uri_get_type (void)
|
|
{
|
|
return GST_URI_SRC;
|
|
}
|
|
|
|
static gchar **
|
|
gst_rtspsrc_uri_get_protocols (void)
|
|
{
|
|
static const gchar *protocols[] =
|
|
{ "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp", NULL };
|
|
|
|
return (gchar **) protocols;
|
|
}
|
|
|
|
static const gchar *
|
|
gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
|
|
{
|
|
GstRTSPSrc *src = GST_RTSPSRC (handler);
|
|
|
|
/* should not dup */
|
|
return src->conninfo.location;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri)
|
|
{
|
|
GstRTSPSrc *src;
|
|
GstRTSPResult res;
|
|
GstRTSPUrl *newurl = NULL;
|
|
GstSDPMessage *sdp = NULL;
|
|
|
|
src = GST_RTSPSRC (handler);
|
|
|
|
/* same URI, we're fine */
|
|
if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
|
|
goto was_ok;
|
|
|
|
if (g_str_has_prefix (uri, "rtsp-sdp://")) {
|
|
if ((res = gst_sdp_message_new (&sdp) < 0))
|
|
goto sdp_failed;
|
|
|
|
GST_DEBUG_OBJECT (src, "parsing SDP message");
|
|
if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
|
|
goto invalid_sdp;
|
|
} else {
|
|
/* try to parse */
|
|
GST_DEBUG_OBJECT (src, "parsing URI");
|
|
if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
|
|
goto parse_error;
|
|
}
|
|
|
|
/* if worked, free previous and store new url object along with the original
|
|
* location. */
|
|
GST_DEBUG_OBJECT (src, "configuring URI");
|
|
g_free (src->conninfo.location);
|
|
src->conninfo.location = g_strdup (uri);
|
|
gst_rtsp_url_free (src->conninfo.url);
|
|
src->conninfo.url = newurl;
|
|
g_free (src->conninfo.url_str);
|
|
if (newurl)
|
|
src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
|
|
else
|
|
src->conninfo.url_str = NULL;
|
|
|
|
if (src->sdp)
|
|
gst_sdp_message_free (src->sdp);
|
|
src->sdp = sdp;
|
|
src->from_sdp = sdp != NULL;
|
|
|
|
GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
|
|
GST_DEBUG_OBJECT (src, "request uri is: %s",
|
|
GST_STR_NULL (src->conninfo.url_str));
|
|
|
|
return TRUE;
|
|
|
|
/* Special cases */
|
|
was_ok:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
|
|
return TRUE;
|
|
}
|
|
sdp_failed:
|
|
{
|
|
GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
|
|
return FALSE;
|
|
}
|
|
invalid_sdp:
|
|
{
|
|
GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
|
|
GST_STR_NULL (uri));
|
|
gst_sdp_message_free (sdp);
|
|
return FALSE;
|
|
}
|
|
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;
|
|
}
|