mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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f54dd50203
Add an rtspclientsink element that accepts streams for which
there is a registered payloader and sends them to
an RTSP server using RECORD.
Sending is synchronised to the pipeline clock. Payload-types
are automatically selected. The 'new-payloader' signal is fired
for custom configuration of payloaders when they are created.
Can now stream a movie like this:
receiver:
./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
decodebin name=depay1 ! audioconvert ! autoaudiosink )"
sender:
gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
https://bugzilla.gnome.org/show_bug.cgi?id=758180
4885 lines
144 KiB
C
4885 lines
144 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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* <2015> Jan Schmidt <jan at centricular dot com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/*
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* 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-rtspclientsink
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*
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* Makes a connection to an RTSP server and send data via RTSP RECORD.
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* rtspclientsink strictly follows RFC 2326
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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 rtspclientsink 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 #GstRTSPClientSink:protocols property.
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*
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* rtspclientsink 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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* and packet reordering.
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* This feature is implemented using the gstrtpbin element.
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*
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* rtspclientsink accepts any stream for which there is an installed payloader,
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* creates the payloader and manages payload-types, as well as RTX setup.
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* The new-payloader signal is fired when a payloader is created, in case
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* an app wants to do custom configuration (such as for MTU).
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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-1.0 videotestsrc ! jpegenc ! rtspclientsink location=rtsp://some.server/url
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* ]| Establish a connection to an RTSP server and send JPEG encoded video packets
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* </refsect2>
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*/
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/* FIXMEs
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* - Handle EOS properly and shutdown. The problem with EOS is we don't know
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* when the server has received all data, so we don't know when to do teardown.
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* At the moment, we forward EOS to the app as soon as we stop sending. Is there
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* a way to know from the receiver that it's got all data? Some session timeout?
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* - Implement extension support for Real / WMS if they support RECORD?
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* - Add support for network clock synchronised streaming?
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* - Fix crypto key nego so SAVP/SAVPF profiles work.
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* - Test (&fix?) HTTP tunnel support
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* - Add an address pool object for GstRTSPStreams to use for multicast
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* - Test multicast UDP transport
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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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#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 <stdio.h>
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#include <stdarg.h>
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#include <gst/net/gstnet.h>
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#include <gst/sdp/gstsdpmessage.h>
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#include <gst/sdp/gstmikey.h>
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#include <gst/rtp/rtp.h>
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#include "gstrtspclientsink.h"
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GST_DEBUG_CATEGORY_STATIC (rtsp_client_sink_debug);
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#define GST_CAT_DEFAULT (rtsp_client_sink_debug)
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static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS_ANY); /* Actual caps come from available set of payloaders */
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enum
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{
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SIGNAL_HANDLE_REQUEST,
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SIGNAL_NEW_MANAGER,
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SIGNAL_NEW_PAYLOADER,
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SIGNAL_REQUEST_RTCP_KEY,
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LAST_SIGNAL
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};
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enum _GstRTSPClientSinkNtpTimeSource
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{
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NTP_TIME_SOURCE_NTP,
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NTP_TIME_SOURCE_UNIX,
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NTP_TIME_SOURCE_RUNNING_TIME,
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NTP_TIME_SOURCE_CLOCK_TIME
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};
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#define GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE (gst_rtsp_client_sink_ntp_time_source_get_type())
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static GType
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gst_rtsp_client_sink_ntp_time_source_get_type (void)
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{
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static GType ntp_time_source_type = 0;
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static const GEnumValue ntp_time_source_values[] = {
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{NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"},
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{NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"},
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{NTP_TIME_SOURCE_RUNNING_TIME,
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"Running time based on pipeline clock",
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"running-time"},
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{NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"},
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{0, NULL, NULL},
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};
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if (!ntp_time_source_type) {
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ntp_time_source_type =
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g_enum_register_static ("GstRTSPClientSinkNtpTimeSource",
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ntp_time_source_values);
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}
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return ntp_time_source_type;
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}
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#define AES_128_KEY_LEN 16
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#define AES_256_KEY_LEN 32
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#define HMAC_32_KEY_LEN 4
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#define HMAC_80_KEY_LEN 10
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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_DO_RTSP_KEEP_ALIVE 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_PORT_RANGE NULL
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#define DEFAULT_UDP_RECONNECT TRUE
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#define DEFAULT_MULTICAST_IFACE NULL
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#define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL
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#define DEFAULT_TLS_DATABASE NULL
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#define DEFAULT_TLS_INTERACTION NULL
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#define DEFAULT_NTP_TIME_SOURCE NTP_TIME_SOURCE_NTP
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#define DEFAULT_USER_AGENT "GStreamer/" PACKAGE_VERSION
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#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
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#define DEFAULT_RTX_TIME_MS 500
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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_RTX_TIME,
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PROP_DO_RTSP_KEEP_ALIVE,
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PROP_PROXY,
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PROP_PROXY_ID,
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PROP_PROXY_PW,
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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_PORT_RANGE,
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PROP_UDP_BUFFER_SIZE,
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PROP_UDP_RECONNECT,
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PROP_MULTICAST_IFACE,
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PROP_SDES,
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PROP_TLS_VALIDATION_FLAGS,
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PROP_TLS_DATABASE,
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PROP_TLS_INTERACTION,
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PROP_NTP_TIME_SOURCE,
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PROP_USER_AGENT,
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PROP_PROFILES
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};
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static void gst_rtsp_client_sink_finalize (GObject * object);
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static void gst_rtsp_client_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtsp_client_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstClock *gst_rtsp_client_sink_provide_clock (GstElement * element);
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static void gst_rtsp_client_sink_uri_handler_init (gpointer g_iface,
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gpointer iface_data);
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static gboolean gst_rtsp_client_sink_set_proxy (GstRTSPClientSink * rtsp,
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const gchar * proxy);
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static void gst_rtsp_client_sink_set_tcp_timeout (GstRTSPClientSink *
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rtsp_client_sink, guint64 timeout);
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static GstStateChangeReturn gst_rtsp_client_sink_change_state (GstElement *
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element, GstStateChange transition);
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static void gst_rtsp_client_sink_handle_message (GstBin * bin,
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GstMessage * message);
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static gboolean gst_rtsp_client_sink_setup_auth (GstRTSPClientSink * sink,
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GstRTSPMessage * response);
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static gboolean gst_rtsp_client_sink_loop_send_cmd (GstRTSPClientSink * sink,
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gint cmd, gint mask);
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static GstRTSPResult gst_rtsp_client_sink_open (GstRTSPClientSink * sink,
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gboolean async);
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static GstRTSPResult gst_rtsp_client_sink_record (GstRTSPClientSink * sink,
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gboolean async);
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static GstRTSPResult gst_rtsp_client_sink_pause (GstRTSPClientSink * sink,
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gboolean async);
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static GstRTSPResult gst_rtsp_client_sink_close (GstRTSPClientSink * sink,
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gboolean async, gboolean only_close);
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static gboolean gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink);
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static gboolean gst_rtsp_client_sink_uri_set_uri (GstURIHandler * handler,
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const gchar * uri, GError ** error);
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static gchar *gst_rtsp_client_sink_uri_get_uri (GstURIHandler * handler);
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static gboolean gst_rtsp_client_sink_loop (GstRTSPClientSink * sink);
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static void gst_rtsp_client_sink_connection_flush (GstRTSPClientSink * sink,
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gboolean flush);
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static GstPad *gst_rtsp_client_sink_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
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static void gst_rtsp_client_sink_release_pad (GstElement * element,
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GstPad * pad);
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/* commands we send to out loop to notify it of events */
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#define CMD_OPEN (1 << 0)
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#define CMD_RECORD (1 << 1)
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#define CMD_PAUSE (1 << 2)
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#define CMD_CLOSE (1 << 3)
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#define CMD_WAIT (1 << 4)
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#define CMD_RECONNECT (1 << 5)
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#define CMD_LOOP (1 << 6)
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/* mask for all commands */
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#define CMD_ALL ((CMD_LOOP << 1) - 1)
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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_rtsp_client_sink_signals[LAST_SIGNAL] = { 0 };
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#define gst_rtsp_client_sink_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstRTSPClientSink, gst_rtsp_client_sink, GST_TYPE_BIN,
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
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gst_rtsp_client_sink_uri_handler_init));
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#ifndef GST_DISABLE_GST_DEBUG
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static inline const gchar *
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cmd_to_string (guint cmd)
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{
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switch (cmd) {
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case CMD_OPEN:
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return "OPEN";
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case CMD_RECORD:
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return "RECORD";
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case CMD_PAUSE:
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return "PAUSE";
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case CMD_CLOSE:
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return "CLOSE";
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case CMD_WAIT:
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return "WAIT";
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case CMD_RECONNECT:
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return "RECONNECT";
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case CMD_LOOP:
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return "LOOP";
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}
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return "unknown";
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}
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#endif
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static void
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gst_rtsp_client_sink_class_init (GstRTSPClientSinkClass * 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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GST_DEBUG_CATEGORY_INIT (rtsp_client_sink_debug, "rtspclientsink", 0,
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"RTSP sink element");
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gobject_class->set_property = gst_rtsp_client_sink_set_property;
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gobject_class->get_property = gst_rtsp_client_sink_get_property;
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gobject_class->finalize = gst_rtsp_client_sink_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_PROFILES,
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g_param_spec_flags ("profiles", "Profiles",
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"Allowed RTSP profiles", GST_TYPE_RTSP_PROFILE,
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DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_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_RTX_TIME,
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g_param_spec_uint ("rtx-time", "Retransmission buffer in ms",
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"Amount of ms to buffer for retransmission. 0 disables retransmission",
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0, G_MAXUINT, DEFAULT_RTX_TIME_MS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPClientSink:do-rtsp-keep-alive:
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*
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* Enable RTSP keep alive support. Some old server don't like RTSP
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* keep alive and then this property needs to be set to FALSE.
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*/
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g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
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g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
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"Send RTSP keep alive packets, disable for old incompatible server.",
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DEFAULT_DO_RTSP_KEEP_ALIVE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPClientSink: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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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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* GstRTSPClientSink:proxy-id:
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*
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* Sets the proxy URI user id for authentication. If the URI set via the
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* "proxy" property contains a user-id already, that will take precedence.
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*
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*/
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g_object_class_install_property (gobject_class, PROP_PROXY_ID,
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g_param_spec_string ("proxy-id", "proxy-id",
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"HTTP proxy URI user id for authentication", "",
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
|
|
* GstRTSPClientSink:proxy-pw:
|
|
*
|
|
* Sets the proxy URI password for authentication. If the URI set via the
|
|
* "proxy" property contains a password already, that will take precedence.
|
|
*
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_PROXY_PW,
|
|
g_param_spec_string ("proxy-pw", "proxy-pw",
|
|
"HTTP proxy URI user password for authentication", "",
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPClientSink:rtp-blocksize:
|
|
*
|
|
* RTP package size to suggest to server.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
|
|
g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
|
|
"RTP package size to suggest to server (0 = disabled)",
|
|
0, 65536, DEFAULT_RTP_BLOCKSIZE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_USER_ID,
|
|
g_param_spec_string ("user-id", "user-id",
|
|
"RTSP location URI user id for authentication", DEFAULT_USER_ID,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
g_object_class_install_property (gobject_class, PROP_USER_PW,
|
|
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));
|
|
|
|
/**
|
|
* GstRTSPClientSink:port-range:
|
|
*
|
|
* Configure the client port numbers that can be used to receive
|
|
* RTCP.
|
|
*/
|
|
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 RTCP data, "
|
|
"eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPClientSink:udp-buffer-size:
|
|
*
|
|
* Size of the kernel UDP receive buffer in bytes.
|
|
*/
|
|
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));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
|
|
g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
|
|
"Reconnect to the server if RTSP connection is closed when doing UDP",
|
|
DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
|
|
g_param_spec_string ("multicast-iface", "Multicast Interface",
|
|
"The network interface on which to join the multicast group",
|
|
DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_SDES,
|
|
g_param_spec_boxed ("sdes", "SDES",
|
|
"The SDES items of this session",
|
|
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPClientSink::tls-validation-flags:
|
|
*
|
|
* TLS certificate validation flags used to validate server
|
|
* certificate.
|
|
*
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
|
|
g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
|
|
"TLS certificate validation flags used to validate the server certificate",
|
|
G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPClientSink::tls-database:
|
|
*
|
|
* TLS database with anchor certificate authorities used to validate
|
|
* the server certificate.
|
|
*
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
|
|
g_param_spec_object ("tls-database", "TLS database",
|
|
"TLS database with anchor certificate authorities used to validate the server certificate",
|
|
G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPClientSink::tls-interaction:
|
|
*
|
|
* A #GTlsInteraction object to be used when the connection or certificate
|
|
* database need to interact with the user. This will be used to prompt the
|
|
* user for passwords where necessary.
|
|
*
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_TLS_INTERACTION,
|
|
g_param_spec_object ("tls-interaction", "TLS interaction",
|
|
"A GTlsInteraction object to prompt the user for password or certificate",
|
|
G_TYPE_TLS_INTERACTION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPClientSink::ntp-time-source:
|
|
*
|
|
* allows to select the time source that should be used
|
|
* for the NTP time in outgoing packets
|
|
*
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
|
|
g_param_spec_enum ("ntp-time-source", "NTP Time Source",
|
|
"NTP time source for RTCP packets",
|
|
GST_TYPE_RTSP_CLIENT_SINK_NTP_TIME_SOURCE, DEFAULT_NTP_TIME_SOURCE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPClientSink::user-agent:
|
|
*
|
|
* The string to set in the User-Agent header.
|
|
*
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_USER_AGENT,
|
|
g_param_spec_string ("user-agent", "User Agent",
|
|
"The User-Agent string to send to the server",
|
|
DEFAULT_USER_AGENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRTSPClientSink::handle-request:
|
|
* @rtsp_client_sink: a #GstRTSPClientSink
|
|
* @request: a #GstRTSPMessage
|
|
* @response: a #GstRTSPMessage
|
|
*
|
|
* Handle a server request in @request and prepare @response.
|
|
*
|
|
* This signal is called from the streaming thread, you should therefore not
|
|
* do any state changes on @rtsp_client_sink because this might deadlock. If you want
|
|
* to modify the state as a result of this signal, post a
|
|
* #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
|
|
* in some other way.
|
|
*
|
|
*/
|
|
gst_rtsp_client_sink_signals[SIGNAL_HANDLE_REQUEST] =
|
|
g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
|
|
0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
|
|
G_TYPE_POINTER, G_TYPE_POINTER);
|
|
|
|
/**
|
|
* GstRTSPClientSink::new-manager:
|
|
* @rtsp_client_sink: a #GstRTSPClientSink
|
|
* @manager: a #GstElement
|
|
*
|
|
* Emitted after a new manager (like rtpbin) was created and the default
|
|
* properties were configured.
|
|
*
|
|
*/
|
|
gst_rtsp_client_sink_signals[SIGNAL_NEW_MANAGER] =
|
|
g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
|
|
g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
|
|
|
|
/**
|
|
* GstRTSPClientSink::new-payloader:
|
|
* @rtsp_client_sink: a #GstRTSPClientSink
|
|
* @payloader: a #GstElement
|
|
*
|
|
* Emitted after a new RTP payloader was created and the default
|
|
* properties were configured.
|
|
*
|
|
*/
|
|
gst_rtsp_client_sink_signals[SIGNAL_NEW_PAYLOADER] =
|
|
g_signal_new_class_handler ("new-payloader", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
|
|
g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
|
|
|
|
/**
|
|
* GstRTSPClientSink::request-rtcp-key:
|
|
* @rtsp_client_sink: a #GstRTSPClientSink
|
|
* @num: the stream number
|
|
*
|
|
* Signal emitted to get the crypto parameters relevant to the RTCP
|
|
* stream. User should provide the key and the RTCP encryption ciphers
|
|
* and authentication, and return them wrapped in a GstCaps.
|
|
*
|
|
*/
|
|
gst_rtsp_client_sink_signals[SIGNAL_REQUEST_RTCP_KEY] =
|
|
g_signal_new ("request-rtcp-key", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
|
|
|
|
gstelement_class->provide_clock = gst_rtsp_client_sink_provide_clock;
|
|
gstelement_class->change_state = gst_rtsp_client_sink_change_state;
|
|
gstelement_class->request_new_pad =
|
|
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_request_new_pad);
|
|
gstelement_class->release_pad =
|
|
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_release_pad);
|
|
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&rtptemplate));
|
|
|
|
gst_element_class_set_static_metadata (gstelement_class,
|
|
"RTSP RECORD client", "Sink/Network",
|
|
"Send data over the network via RTSP RECORD(RFC 2326)",
|
|
"Jan Schmidt <jan@centricular.com>");
|
|
|
|
gstbin_class->handle_message = gst_rtsp_client_sink_handle_message;
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_init (GstRTSPClientSink * sink)
|
|
{
|
|
sink->conninfo.location = g_strdup (DEFAULT_LOCATION);
|
|
sink->protocols = DEFAULT_PROTOCOLS;
|
|
sink->debug = DEFAULT_DEBUG;
|
|
sink->retry = DEFAULT_RETRY;
|
|
sink->udp_timeout = DEFAULT_TIMEOUT;
|
|
gst_rtsp_client_sink_set_tcp_timeout (sink, DEFAULT_TCP_TIMEOUT);
|
|
sink->latency = DEFAULT_LATENCY_MS;
|
|
sink->rtx_time = DEFAULT_RTX_TIME_MS;
|
|
sink->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
|
|
gst_rtsp_client_sink_set_proxy (sink, DEFAULT_PROXY);
|
|
sink->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
|
|
sink->user_id = g_strdup (DEFAULT_USER_ID);
|
|
sink->user_pw = g_strdup (DEFAULT_USER_PW);
|
|
sink->client_port_range.min = 0;
|
|
sink->client_port_range.max = 0;
|
|
sink->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
|
|
sink->udp_reconnect = DEFAULT_UDP_RECONNECT;
|
|
sink->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
|
|
sink->sdes = NULL;
|
|
sink->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
|
|
sink->tls_database = DEFAULT_TLS_DATABASE;
|
|
sink->tls_interaction = DEFAULT_TLS_INTERACTION;
|
|
sink->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
|
|
sink->user_agent = g_strdup (DEFAULT_USER_AGENT);
|
|
|
|
sink->profiles = DEFAULT_PROFILES;
|
|
|
|
/* protects the streaming thread in interleaved mode or the polling
|
|
* thread in UDP mode. */
|
|
g_rec_mutex_init (&sink->stream_rec_lock);
|
|
|
|
/* protects our state changes from multiple invocations */
|
|
g_rec_mutex_init (&sink->state_rec_lock);
|
|
|
|
g_mutex_init (&sink->send_lock);
|
|
|
|
g_mutex_init (&sink->preroll_lock);
|
|
g_cond_init (&sink->preroll_cond);
|
|
|
|
sink->state = GST_RTSP_STATE_INVALID;
|
|
|
|
sink->internal_bin = (GstBin *) gst_bin_new ("rtspbin");
|
|
gst_element_set_locked_state (GST_ELEMENT_CAST (sink->internal_bin), TRUE);
|
|
gst_bin_add (GST_BIN (sink), GST_ELEMENT_CAST (sink->internal_bin));
|
|
|
|
sink->next_dyn_pt = 96;
|
|
|
|
gst_sdp_message_init (&sink->cursdp);
|
|
|
|
GST_OBJECT_FLAG_SET (sink, GST_ELEMENT_FLAG_SINK);
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_finalize (GObject * object)
|
|
{
|
|
GstRTSPClientSink *rtsp_client_sink;
|
|
|
|
rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
|
|
|
|
gst_sdp_message_uninit (&rtsp_client_sink->cursdp);
|
|
|
|
g_free (rtsp_client_sink->conninfo.location);
|
|
gst_rtsp_url_free (rtsp_client_sink->conninfo.url);
|
|
g_free (rtsp_client_sink->conninfo.url_str);
|
|
g_free (rtsp_client_sink->user_id);
|
|
g_free (rtsp_client_sink->user_pw);
|
|
g_free (rtsp_client_sink->multi_iface);
|
|
g_free (rtsp_client_sink->user_agent);
|
|
|
|
if (rtsp_client_sink->uri_sdp) {
|
|
gst_sdp_message_free (rtsp_client_sink->uri_sdp);
|
|
rtsp_client_sink->uri_sdp = NULL;
|
|
}
|
|
if (rtsp_client_sink->provided_clock)
|
|
gst_object_unref (rtsp_client_sink->provided_clock);
|
|
|
|
if (rtsp_client_sink->sdes)
|
|
gst_structure_free (rtsp_client_sink->sdes);
|
|
|
|
if (rtsp_client_sink->tls_database)
|
|
g_object_unref (rtsp_client_sink->tls_database);
|
|
|
|
if (rtsp_client_sink->tls_interaction)
|
|
g_object_unref (rtsp_client_sink->tls_interaction);
|
|
|
|
/* free locks */
|
|
g_rec_mutex_clear (&rtsp_client_sink->stream_rec_lock);
|
|
g_rec_mutex_clear (&rtsp_client_sink->state_rec_lock);
|
|
|
|
g_mutex_clear (&rtsp_client_sink->send_lock);
|
|
|
|
g_mutex_clear (&rtsp_client_sink->preroll_lock);
|
|
g_cond_clear (&rtsp_client_sink->preroll_cond);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_payloader_filter_func (GstPluginFeature * feature, gpointer user_data)
|
|
{
|
|
GstElementFactory *factory = NULL;
|
|
const gchar *klass;
|
|
|
|
if (!GST_IS_ELEMENT_FACTORY (feature))
|
|
return FALSE;
|
|
|
|
factory = GST_ELEMENT_FACTORY (feature);
|
|
|
|
if (gst_plugin_feature_get_rank (feature) == GST_RANK_NONE)
|
|
return FALSE;
|
|
|
|
if (!gst_element_factory_list_is_type (factory,
|
|
GST_ELEMENT_FACTORY_TYPE_PAYLOADER))
|
|
return FALSE;
|
|
|
|
klass =
|
|
gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_KLASS);
|
|
if (strstr (klass, "Codec") == NULL)
|
|
return FALSE;
|
|
if (strstr (klass, "RTP") == NULL)
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gint
|
|
compare_ranks (GstPluginFeature * f1, GstPluginFeature * f2)
|
|
{
|
|
gint diff;
|
|
const gchar *rname1, *rname2;
|
|
GstRank rank1, rank2;
|
|
|
|
rname1 = gst_plugin_feature_get_name (f1);
|
|
rname2 = gst_plugin_feature_get_name (f2);
|
|
|
|
rank1 = gst_plugin_feature_get_rank (f1);
|
|
rank2 = gst_plugin_feature_get_rank (f2);
|
|
|
|
/* HACK: Prefer rtpmp4apay over rtpmp4gpay */
|
|
if (g_str_equal (rname1, "rtpmp4apay"))
|
|
rank1 = GST_RANK_SECONDARY + 1;
|
|
if (g_str_equal (rname2, "rtpmp4apay"))
|
|
rank2 = GST_RANK_SECONDARY + 1;
|
|
|
|
diff = rank2 - rank1;
|
|
if (diff != 0)
|
|
return diff;
|
|
|
|
diff = strcmp (rname2, rname1);
|
|
|
|
return diff;
|
|
}
|
|
|
|
static GList *
|
|
gst_rtsp_client_sink_get_factories (void)
|
|
{
|
|
static GList *payloader_factories = NULL;
|
|
|
|
if (g_once_init_enter (&payloader_factories)) {
|
|
GList *all_factories;
|
|
|
|
all_factories =
|
|
gst_registry_feature_filter (gst_registry_get (),
|
|
gst_rtp_payloader_filter_func, FALSE, NULL);
|
|
|
|
all_factories = g_list_sort (all_factories, (GCompareFunc) compare_ranks);
|
|
|
|
g_once_init_leave (&payloader_factories, all_factories);
|
|
}
|
|
|
|
return payloader_factories;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_rtsp_client_sink_get_payloader_caps (void)
|
|
{
|
|
/* Cached caps result */
|
|
static GstCaps *ret;
|
|
|
|
if (g_once_init_enter (&ret)) {
|
|
GList *factories, *cur;
|
|
GstCaps *caps = gst_caps_new_empty ();
|
|
|
|
factories = gst_rtsp_client_sink_get_factories ();
|
|
for (cur = factories; cur != NULL; cur = g_list_next (cur)) {
|
|
GstElementFactory *factory = GST_ELEMENT_FACTORY (cur->data);
|
|
const GList *tmp;
|
|
|
|
for (tmp = gst_element_factory_get_static_pad_templates (factory);
|
|
tmp; tmp = g_list_next (tmp)) {
|
|
GstStaticPadTemplate *template = tmp->data;
|
|
|
|
if (template->direction == GST_PAD_SINK) {
|
|
GstCaps *static_caps = gst_static_pad_template_get_caps (template);
|
|
|
|
GST_LOG ("Found pad template %s on factory %s",
|
|
template->name_template, gst_plugin_feature_get_name (factory));
|
|
|
|
if (static_caps)
|
|
caps = gst_caps_merge (caps, static_caps);
|
|
|
|
/* Early out, any is absorbing */
|
|
if (gst_caps_is_any (caps))
|
|
goto out;
|
|
}
|
|
}
|
|
}
|
|
out:
|
|
g_once_init_leave (&ret, caps);
|
|
}
|
|
|
|
/* Return cached result */
|
|
return gst_caps_ref (ret);
|
|
}
|
|
|
|
static GstElement *
|
|
gst_rtsp_client_sink_make_payloader (GstCaps * caps)
|
|
{
|
|
GList *factories, *cur;
|
|
|
|
factories = gst_rtsp_client_sink_get_factories ();
|
|
for (cur = factories; cur != NULL; cur = g_list_next (cur)) {
|
|
GstElementFactory *factory = GST_ELEMENT_FACTORY (cur->data);
|
|
const GList *tmp;
|
|
|
|
for (tmp = gst_element_factory_get_static_pad_templates (factory);
|
|
tmp; tmp = g_list_next (tmp)) {
|
|
GstStaticPadTemplate *template = tmp->data;
|
|
|
|
if (template->direction == GST_PAD_SINK) {
|
|
GstCaps *static_caps = gst_static_pad_template_get_caps (template);
|
|
GstElement *payloader = NULL;
|
|
|
|
if (gst_caps_can_intersect (static_caps, caps)) {
|
|
GST_DEBUG ("caps %" GST_PTR_FORMAT " intersects with template %"
|
|
GST_PTR_FORMAT " for payloader %s", caps, static_caps,
|
|
gst_plugin_feature_get_name (factory));
|
|
payloader = gst_element_factory_create (factory, NULL);
|
|
}
|
|
|
|
gst_caps_unref (static_caps);
|
|
|
|
if (payloader)
|
|
return payloader;
|
|
}
|
|
}
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static GstRTSPStream *
|
|
gst_rtsp_client_sink_create_stream (GstRTSPClientSink * sink,
|
|
GstRTSPStreamContext * context, GstElement * payloader, GstPad * pad)
|
|
{
|
|
GstRTSPStream *stream = NULL;
|
|
guint pt, aux_pt;
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
|
|
g_object_get (G_OBJECT (payloader), "pt", &pt, NULL);
|
|
if (pt >= 96 && pt <= sink->next_dyn_pt) {
|
|
/* Payloader has a dynamic PT, but one that's already used */
|
|
/* FIXME: Create a caps->ptmap instead? */
|
|
pt = sink->next_dyn_pt;
|
|
|
|
if (pt > 127)
|
|
goto no_free_pt;
|
|
|
|
GST_DEBUG_OBJECT (sink, "Assigning pt %u to stream %d", pt, context->index);
|
|
|
|
sink->next_dyn_pt++;
|
|
} else {
|
|
GST_DEBUG_OBJECT (sink, "Keeping existing pt %u for stream %d",
|
|
pt, context->index);
|
|
}
|
|
|
|
aux_pt = sink->next_dyn_pt;
|
|
if (aux_pt > 127)
|
|
goto no_free_pt;
|
|
sink->next_dyn_pt++;
|
|
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
|
|
g_object_set (G_OBJECT (payloader), "pt", pt, NULL);
|
|
|
|
stream = gst_rtsp_stream_new (context->index, payloader, pad);
|
|
|
|
gst_rtsp_stream_set_client_side (stream, TRUE);
|
|
gst_rtsp_stream_set_retransmission_time (stream,
|
|
(GstClockTime) (sink->rtx_time) * GST_MSECOND);
|
|
gst_rtsp_stream_set_protocols (stream, sink->protocols);
|
|
gst_rtsp_stream_set_profiles (stream, sink->profiles);
|
|
gst_rtsp_stream_set_retransmission_pt (stream, aux_pt);
|
|
gst_rtsp_stream_set_buffer_size (stream, sink->udp_buffer_size);
|
|
if (sink->rtp_blocksize > 0)
|
|
gst_rtsp_stream_set_mtu (stream, sink->rtp_blocksize);
|
|
|
|
#if 0
|
|
if (priv->pool)
|
|
gst_rtsp_stream_set_address_pool (stream, priv->pool);
|
|
#endif
|
|
|
|
return stream;
|
|
no_free_pt:
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, NO_SPACE_LEFT, (NULL),
|
|
("Ran out of dynamic payload types."));
|
|
|
|
if (stream)
|
|
g_object_unref (stream);
|
|
return NULL;
|
|
}
|
|
|
|
static GstPadProbeReturn
|
|
handle_payloader_block (GstPad * pad, GstPadProbeInfo * info,
|
|
GstRTSPStreamContext * context)
|
|
{
|
|
GstRTSPClientSink *sink = context->parent;
|
|
|
|
GST_INFO_OBJECT (sink, "Block on pad %" GST_PTR_FORMAT, pad);
|
|
|
|
g_mutex_lock (&sink->preroll_lock);
|
|
context->prerolled = TRUE;
|
|
g_cond_broadcast (&sink->preroll_cond);
|
|
g_mutex_unlock (&sink->preroll_lock);
|
|
|
|
GST_INFO_OBJECT (sink, "Announced preroll on pad %" GST_PTR_FORMAT, pad);
|
|
|
|
return GST_PAD_PROBE_OK;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_client_sink_setup_payloader (GstRTSPClientSink * sink, GstPad * pad,
|
|
GstCaps * caps)
|
|
{
|
|
GstRTSPStreamContext *context;
|
|
|
|
GstElement *payloader;
|
|
GstPad *sinkpad, *srcpad, *ghostsink;
|
|
|
|
context = gst_pad_get_element_private (pad);
|
|
|
|
/* Find the payloader. FIXME: Allow user to provide payloader via pad property */
|
|
payloader = gst_rtsp_client_sink_make_payloader (caps);
|
|
if (payloader == NULL)
|
|
return FALSE;
|
|
|
|
GST_DEBUG_OBJECT (sink, "Configuring payloader %" GST_PTR_FORMAT
|
|
" for pad %" GST_PTR_FORMAT, payloader, pad);
|
|
|
|
sinkpad = gst_element_get_static_pad (payloader, "sink");
|
|
if (sinkpad == NULL)
|
|
goto no_sinkpad;
|
|
|
|
srcpad = gst_element_get_static_pad (payloader, "src");
|
|
if (srcpad == NULL)
|
|
goto no_srcpad;
|
|
|
|
gst_bin_add (GST_BIN (sink->internal_bin), payloader);
|
|
ghostsink = gst_ghost_pad_new (NULL, sinkpad);
|
|
gst_pad_set_active (ghostsink, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT (sink->internal_bin), ghostsink);
|
|
|
|
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_NEW_PAYLOADER], 0,
|
|
payloader);
|
|
|
|
GST_RTSP_STATE_LOCK (sink);
|
|
context->payloader_block_id =
|
|
gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM,
|
|
(GstPadProbeCallback) handle_payloader_block, context, NULL);
|
|
context->payloader = payloader;
|
|
|
|
payloader = gst_object_ref (payloader);
|
|
|
|
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), ghostsink);
|
|
gst_object_unref (GST_OBJECT (sinkpad));
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
|
|
gst_element_sync_state_with_parent (payloader);
|
|
|
|
gst_object_unref (payloader);
|
|
gst_object_unref (GST_OBJECT (srcpad));
|
|
|
|
return TRUE;
|
|
|
|
no_sinkpad:
|
|
GST_ERROR_OBJECT (sink,
|
|
"Could not find sink pad on payloader %" GST_PTR_FORMAT, payloader);
|
|
gst_object_unref (payloader);
|
|
return FALSE;
|
|
|
|
no_srcpad:
|
|
GST_ERROR_OBJECT (sink,
|
|
"Could not find src pad on payloader %" GST_PTR_FORMAT, payloader);
|
|
gst_object_unref (GST_OBJECT (sinkpad));
|
|
gst_object_unref (payloader);
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_client_sink_sinkpad_event (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
|
|
GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
|
|
if (target == NULL) {
|
|
GstCaps *caps;
|
|
|
|
/* No target yet - choose a payloader and configure it */
|
|
gst_event_parse_caps (event, &caps);
|
|
|
|
GST_DEBUG_OBJECT (parent,
|
|
"Have set caps event on pad %" GST_PTR_FORMAT
|
|
" caps %" GST_PTR_FORMAT, pad, caps);
|
|
|
|
if (!gst_rtsp_client_sink_setup_payloader (GST_RTSP_CLIENT_SINK (parent),
|
|
pad, caps)) {
|
|
gst_event_unref (event);
|
|
return FALSE;
|
|
}
|
|
}
|
|
}
|
|
|
|
return gst_pad_event_default (pad, parent, event);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_client_sink_sinkpad_query (GstPad * pad, GstObject * parent,
|
|
GstQuery * query)
|
|
{
|
|
if (GST_QUERY_TYPE (query) == GST_QUERY_CAPS) {
|
|
GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
|
|
if (target == NULL) {
|
|
/* No target yet - return the union of all payloader caps */
|
|
GstCaps *caps = gst_rtsp_client_sink_get_payloader_caps ();
|
|
|
|
GST_TRACE_OBJECT (parent, "Returning payloader caps %" GST_PTR_FORMAT,
|
|
caps);
|
|
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
return gst_pad_query_default (pad, parent, query);
|
|
}
|
|
|
|
static GstPad *
|
|
gst_rtsp_client_sink_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
|
|
{
|
|
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
|
|
GstPad *pad;
|
|
GstRTSPStreamContext *context;
|
|
guint idx = (guint) - 1;
|
|
gchar *tmpname;
|
|
|
|
g_mutex_lock (&sink->preroll_lock);
|
|
if (sink->streams_collected) {
|
|
GST_WARNING_OBJECT (element, "Can't add streams to a running session");
|
|
g_mutex_unlock (&sink->preroll_lock);
|
|
return NULL;
|
|
}
|
|
g_mutex_unlock (&sink->preroll_lock);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
if (name) {
|
|
if (!sscanf (name, "sink_%u", &idx)) {
|
|
GST_OBJECT_UNLOCK (sink);
|
|
GST_ERROR_OBJECT (element, "Invalid sink pad name %s", name);
|
|
return NULL;
|
|
}
|
|
|
|
if (idx >= sink->next_pad_id)
|
|
sink->next_pad_id = idx + 1;
|
|
}
|
|
if (idx == (guint) - 1) {
|
|
idx = sink->next_pad_id;
|
|
sink->next_pad_id++;
|
|
}
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
tmpname = g_strdup_printf ("sink_%u", idx);
|
|
pad = gst_ghost_pad_new_no_target_from_template (tmpname, templ);
|
|
g_free (tmpname);
|
|
|
|
GST_DEBUG_OBJECT (element, "Creating request pad %" GST_PTR_FORMAT, pad);
|
|
|
|
gst_pad_set_event_function (pad,
|
|
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_sinkpad_event));
|
|
gst_pad_set_query_function (pad,
|
|
GST_DEBUG_FUNCPTR (gst_rtsp_client_sink_sinkpad_query));
|
|
|
|
context = g_new0 (GstRTSPStreamContext, 1);
|
|
context->parent = sink;
|
|
context->index = idx;
|
|
|
|
gst_pad_set_element_private (pad, context);
|
|
|
|
/* The rest of the context is configured on a caps set */
|
|
gst_pad_set_active (pad, TRUE);
|
|
gst_element_add_pad (element, pad);
|
|
|
|
(void) gst_rtsp_client_sink_get_factories ();
|
|
|
|
GST_RTSP_STATE_LOCK (sink);
|
|
sink->contexts = g_list_prepend (sink->contexts, context);
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
|
|
return pad;
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
|
|
GstRTSPStreamContext *context;
|
|
|
|
context = gst_pad_get_element_private (pad);
|
|
|
|
GST_RTSP_STATE_LOCK (sink);
|
|
sink->contexts = g_list_remove (sink->contexts, context);
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
|
|
/* FIXME: Shut down and clean up streaming on this pad,
|
|
* do teardown if needed */
|
|
GST_LOG_OBJECT (sink,
|
|
"Cleaning up payloader and stream for released pad %" GST_PTR_FORMAT,
|
|
pad);
|
|
|
|
if (context->stream_transport) {
|
|
gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
|
|
gst_object_unref (context->stream_transport);
|
|
context->stream_transport = NULL;
|
|
}
|
|
if (context->stream) {
|
|
if (context->joined) {
|
|
gst_rtsp_stream_leave_bin (context->stream,
|
|
GST_BIN (sink->internal_bin), sink->rtpbin);
|
|
context->joined = FALSE;
|
|
}
|
|
gst_object_unref (context->stream);
|
|
context->stream = NULL;
|
|
}
|
|
if (context->srtcpparams)
|
|
gst_caps_unref (context->srtcpparams);
|
|
|
|
g_free (context->conninfo.location);
|
|
context->conninfo.location = NULL;
|
|
|
|
g_free (context);
|
|
|
|
gst_element_remove_pad (element, pad);
|
|
}
|
|
|
|
static GstClock *
|
|
gst_rtsp_client_sink_provide_clock (GstElement * element)
|
|
{
|
|
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (element);
|
|
GstClock *clock;
|
|
|
|
if ((clock = sink->provided_clock) != NULL)
|
|
gst_object_ref (clock);
|
|
|
|
return clock;
|
|
}
|
|
|
|
/* a proxy string of the format [user:passwd@]host[:port] */
|
|
static gboolean
|
|
gst_rtsp_client_sink_set_proxy (GstRTSPClientSink * 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;
|
|
} else {
|
|
if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
|
|
rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
|
|
if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
|
|
rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
|
|
if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
|
|
GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
|
|
GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
|
|
}
|
|
}
|
|
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_rtsp_client_sink_set_tcp_timeout (GstRTSPClientSink * rtsp_client_sink,
|
|
guint64 timeout)
|
|
{
|
|
rtsp_client_sink->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
|
|
rtsp_client_sink->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
|
|
|
|
if (timeout != 0)
|
|
rtsp_client_sink->ptcp_timeout = &rtsp_client_sink->tcp_timeout;
|
|
else
|
|
rtsp_client_sink->ptcp_timeout = NULL;
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTSPClientSink *rtsp_client_sink;
|
|
|
|
rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LOCATION:
|
|
gst_rtsp_client_sink_uri_set_uri (GST_URI_HANDLER (rtsp_client_sink),
|
|
g_value_get_string (value), NULL);
|
|
break;
|
|
case PROP_PROTOCOLS:
|
|
rtsp_client_sink->protocols = g_value_get_flags (value);
|
|
break;
|
|
case PROP_PROFILES:
|
|
rtsp_client_sink->profiles = g_value_get_flags (value);
|
|
break;
|
|
case PROP_DEBUG:
|
|
rtsp_client_sink->debug = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_RETRY:
|
|
rtsp_client_sink->retry = g_value_get_uint (value);
|
|
break;
|
|
case PROP_TIMEOUT:
|
|
rtsp_client_sink->udp_timeout = g_value_get_uint64 (value);
|
|
break;
|
|
case PROP_TCP_TIMEOUT:
|
|
gst_rtsp_client_sink_set_tcp_timeout (rtsp_client_sink,
|
|
g_value_get_uint64 (value));
|
|
break;
|
|
case PROP_LATENCY:
|
|
rtsp_client_sink->latency = g_value_get_uint (value);
|
|
break;
|
|
case PROP_RTX_TIME:
|
|
rtsp_client_sink->rtx_time = g_value_get_uint (value);
|
|
break;
|
|
case PROP_DO_RTSP_KEEP_ALIVE:
|
|
rtsp_client_sink->do_rtsp_keep_alive = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_PROXY:
|
|
gst_rtsp_client_sink_set_proxy (rtsp_client_sink,
|
|
g_value_get_string (value));
|
|
break;
|
|
case PROP_PROXY_ID:
|
|
if (rtsp_client_sink->prop_proxy_id)
|
|
g_free (rtsp_client_sink->prop_proxy_id);
|
|
rtsp_client_sink->prop_proxy_id = g_value_dup_string (value);
|
|
break;
|
|
case PROP_PROXY_PW:
|
|
if (rtsp_client_sink->prop_proxy_pw)
|
|
g_free (rtsp_client_sink->prop_proxy_pw);
|
|
rtsp_client_sink->prop_proxy_pw = g_value_dup_string (value);
|
|
break;
|
|
case PROP_RTP_BLOCKSIZE:
|
|
rtsp_client_sink->rtp_blocksize = g_value_get_uint (value);
|
|
break;
|
|
case PROP_USER_ID:
|
|
if (rtsp_client_sink->user_id)
|
|
g_free (rtsp_client_sink->user_id);
|
|
rtsp_client_sink->user_id = g_value_dup_string (value);
|
|
break;
|
|
case PROP_USER_PW:
|
|
if (rtsp_client_sink->user_pw)
|
|
g_free (rtsp_client_sink->user_pw);
|
|
rtsp_client_sink->user_pw = g_value_dup_string (value);
|
|
break;
|
|
case PROP_PORT_RANGE:
|
|
{
|
|
const gchar *str;
|
|
|
|
str = g_value_get_string (value);
|
|
if (str) {
|
|
sscanf (str, "%u-%u",
|
|
&rtsp_client_sink->client_port_range.min,
|
|
&rtsp_client_sink->client_port_range.max);
|
|
} else {
|
|
rtsp_client_sink->client_port_range.min = 0;
|
|
rtsp_client_sink->client_port_range.max = 0;
|
|
}
|
|
break;
|
|
}
|
|
case PROP_UDP_BUFFER_SIZE:
|
|
rtsp_client_sink->udp_buffer_size = g_value_get_int (value);
|
|
break;
|
|
case PROP_UDP_RECONNECT:
|
|
rtsp_client_sink->udp_reconnect = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_MULTICAST_IFACE:
|
|
g_free (rtsp_client_sink->multi_iface);
|
|
|
|
if (g_value_get_string (value) == NULL)
|
|
rtsp_client_sink->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
|
|
else
|
|
rtsp_client_sink->multi_iface = g_value_dup_string (value);
|
|
break;
|
|
case PROP_SDES:
|
|
rtsp_client_sink->sdes = g_value_dup_boxed (value);
|
|
break;
|
|
case PROP_TLS_VALIDATION_FLAGS:
|
|
rtsp_client_sink->tls_validation_flags = g_value_get_flags (value);
|
|
break;
|
|
case PROP_TLS_DATABASE:
|
|
g_clear_object (&rtsp_client_sink->tls_database);
|
|
rtsp_client_sink->tls_database = g_value_dup_object (value);
|
|
break;
|
|
case PROP_TLS_INTERACTION:
|
|
g_clear_object (&rtsp_client_sink->tls_interaction);
|
|
rtsp_client_sink->tls_interaction = g_value_dup_object (value);
|
|
break;
|
|
case PROP_NTP_TIME_SOURCE:
|
|
rtsp_client_sink->ntp_time_source = g_value_get_enum (value);
|
|
break;
|
|
case PROP_USER_AGENT:
|
|
g_free (rtsp_client_sink->user_agent);
|
|
rtsp_client_sink->user_agent = g_value_dup_string (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTSPClientSink *rtsp_client_sink;
|
|
|
|
rtsp_client_sink = GST_RTSP_CLIENT_SINK (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LOCATION:
|
|
g_value_set_string (value, rtsp_client_sink->conninfo.location);
|
|
break;
|
|
case PROP_PROTOCOLS:
|
|
g_value_set_flags (value, rtsp_client_sink->protocols);
|
|
break;
|
|
case PROP_PROFILES:
|
|
g_value_set_flags (value, rtsp_client_sink->profiles);
|
|
break;
|
|
case PROP_DEBUG:
|
|
g_value_set_boolean (value, rtsp_client_sink->debug);
|
|
break;
|
|
case PROP_RETRY:
|
|
g_value_set_uint (value, rtsp_client_sink->retry);
|
|
break;
|
|
case PROP_TIMEOUT:
|
|
g_value_set_uint64 (value, rtsp_client_sink->udp_timeout);
|
|
break;
|
|
case PROP_TCP_TIMEOUT:
|
|
{
|
|
guint64 timeout;
|
|
|
|
timeout = rtsp_client_sink->tcp_timeout.tv_sec * G_USEC_PER_SEC +
|
|
rtsp_client_sink->tcp_timeout.tv_usec;
|
|
g_value_set_uint64 (value, timeout);
|
|
break;
|
|
}
|
|
case PROP_LATENCY:
|
|
g_value_set_uint (value, rtsp_client_sink->latency);
|
|
break;
|
|
case PROP_RTX_TIME:
|
|
g_value_set_uint (value, rtsp_client_sink->rtx_time);
|
|
break;
|
|
case PROP_DO_RTSP_KEEP_ALIVE:
|
|
g_value_set_boolean (value, rtsp_client_sink->do_rtsp_keep_alive);
|
|
break;
|
|
case PROP_PROXY:
|
|
{
|
|
gchar *str;
|
|
|
|
if (rtsp_client_sink->proxy_host) {
|
|
str =
|
|
g_strdup_printf ("%s:%d", rtsp_client_sink->proxy_host,
|
|
rtsp_client_sink->proxy_port);
|
|
} else {
|
|
str = NULL;
|
|
}
|
|
g_value_take_string (value, str);
|
|
break;
|
|
}
|
|
case PROP_PROXY_ID:
|
|
g_value_set_string (value, rtsp_client_sink->prop_proxy_id);
|
|
break;
|
|
case PROP_PROXY_PW:
|
|
g_value_set_string (value, rtsp_client_sink->prop_proxy_pw);
|
|
break;
|
|
case PROP_RTP_BLOCKSIZE:
|
|
g_value_set_uint (value, rtsp_client_sink->rtp_blocksize);
|
|
break;
|
|
case PROP_USER_ID:
|
|
g_value_set_string (value, rtsp_client_sink->user_id);
|
|
break;
|
|
case PROP_USER_PW:
|
|
g_value_set_string (value, rtsp_client_sink->user_pw);
|
|
break;
|
|
case PROP_PORT_RANGE:
|
|
{
|
|
gchar *str;
|
|
|
|
if (rtsp_client_sink->client_port_range.min != 0) {
|
|
str = g_strdup_printf ("%u-%u", rtsp_client_sink->client_port_range.min,
|
|
rtsp_client_sink->client_port_range.max);
|
|
} else {
|
|
str = NULL;
|
|
}
|
|
g_value_take_string (value, str);
|
|
break;
|
|
}
|
|
case PROP_UDP_BUFFER_SIZE:
|
|
g_value_set_int (value, rtsp_client_sink->udp_buffer_size);
|
|
break;
|
|
case PROP_UDP_RECONNECT:
|
|
g_value_set_boolean (value, rtsp_client_sink->udp_reconnect);
|
|
break;
|
|
case PROP_MULTICAST_IFACE:
|
|
g_value_set_string (value, rtsp_client_sink->multi_iface);
|
|
break;
|
|
case PROP_SDES:
|
|
g_value_set_boxed (value, rtsp_client_sink->sdes);
|
|
break;
|
|
case PROP_TLS_VALIDATION_FLAGS:
|
|
g_value_set_flags (value, rtsp_client_sink->tls_validation_flags);
|
|
break;
|
|
case PROP_TLS_DATABASE:
|
|
g_value_set_object (value, rtsp_client_sink->tls_database);
|
|
break;
|
|
case PROP_TLS_INTERACTION:
|
|
g_value_set_object (value, rtsp_client_sink->tls_interaction);
|
|
break;
|
|
case PROP_NTP_TIME_SOURCE:
|
|
g_value_set_enum (value, rtsp_client_sink->ntp_time_source);
|
|
break;
|
|
case PROP_USER_AGENT:
|
|
g_value_set_string (value, rtsp_client_sink->user_agent);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static const gchar *
|
|
get_aggregate_control (GstRTSPClientSink * sink)
|
|
{
|
|
const gchar *base;
|
|
|
|
if (sink->control)
|
|
base = sink->control;
|
|
else if (sink->content_base)
|
|
base = sink->content_base;
|
|
else if (sink->conninfo.url_str)
|
|
base = sink->conninfo.url_str;
|
|
else
|
|
base = "/";
|
|
|
|
return base;
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_cleanup (GstRTSPClientSink * sink)
|
|
{
|
|
GList *walk;
|
|
|
|
GST_DEBUG_OBJECT (sink, "cleanup");
|
|
|
|
gst_element_set_state (GST_ELEMENT (sink->internal_bin), GST_STATE_NULL);
|
|
|
|
/* Clean up any left over stream objects */
|
|
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamContext *context = (GstRTSPStreamContext *) (walk->data);
|
|
if (context->stream_transport) {
|
|
gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
|
|
gst_object_unref (context->stream_transport);
|
|
context->stream_transport = NULL;
|
|
}
|
|
|
|
if (context->stream) {
|
|
if (context->joined) {
|
|
gst_rtsp_stream_leave_bin (context->stream,
|
|
GST_BIN (sink->internal_bin), sink->rtpbin);
|
|
context->joined = FALSE;
|
|
}
|
|
gst_object_unref (context->stream);
|
|
context->stream = NULL;
|
|
}
|
|
|
|
if (context->srtcpparams)
|
|
gst_caps_unref (context->srtcpparams);
|
|
g_free (context->conninfo.location);
|
|
context->conninfo.location = NULL;
|
|
}
|
|
|
|
if (sink->rtpbin) {
|
|
gst_element_set_state (sink->rtpbin, GST_STATE_NULL);
|
|
gst_bin_remove (GST_BIN_CAST (sink->internal_bin), sink->rtpbin);
|
|
sink->rtpbin = NULL;
|
|
}
|
|
|
|
g_free (sink->content_base);
|
|
sink->content_base = NULL;
|
|
|
|
g_free (sink->control);
|
|
sink->control = NULL;
|
|
|
|
if (sink->range)
|
|
gst_rtsp_range_free (sink->range);
|
|
sink->range = NULL;
|
|
|
|
/* don't clear the SDP when it was used in the url */
|
|
if (sink->uri_sdp && !sink->from_sdp) {
|
|
gst_sdp_message_free (sink->uri_sdp);
|
|
sink->uri_sdp = NULL;
|
|
}
|
|
|
|
if (sink->provided_clock) {
|
|
gst_object_unref (sink->provided_clock);
|
|
sink->provided_clock = NULL;
|
|
}
|
|
|
|
g_free (sink->server_ip);
|
|
sink->server_ip = NULL;
|
|
|
|
sink->next_pad_id = 0;
|
|
sink->next_dyn_pt = 96;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_connection_send (GstRTSPClientSink * sink,
|
|
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_rtsp_client_sink_connection_receive (GstRTSPClientSink * sink,
|
|
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 GstRTSPResult
|
|
gst_rtsp_conninfo_connect (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
|
|
gboolean async)
|
|
{
|
|
GstRTSPResult res;
|
|
|
|
if (info->connection == NULL) {
|
|
if (info->url == NULL) {
|
|
GST_DEBUG_OBJECT (sink, "parsing uri (%s)...", info->location);
|
|
if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
|
|
goto parse_error;
|
|
}
|
|
|
|
/* create connection */
|
|
GST_DEBUG_OBJECT (sink, "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 (sink, "sanitized uri %s", info->url_str);
|
|
|
|
if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
|
|
if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
|
|
sink->tls_validation_flags))
|
|
GST_WARNING_OBJECT (sink, "Unable to set TLS validation flags");
|
|
|
|
if (sink->tls_database)
|
|
gst_rtsp_connection_set_tls_database (info->connection,
|
|
sink->tls_database);
|
|
|
|
if (sink->tls_interaction)
|
|
gst_rtsp_connection_set_tls_interaction (info->connection,
|
|
sink->tls_interaction);
|
|
}
|
|
|
|
if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
|
|
gst_rtsp_connection_set_tunneled (info->connection, TRUE);
|
|
|
|
if (sink->proxy_host) {
|
|
GST_DEBUG_OBJECT (sink, "setting proxy %s:%d", sink->proxy_host,
|
|
sink->proxy_port);
|
|
gst_rtsp_connection_set_proxy (info->connection, sink->proxy_host,
|
|
sink->proxy_port);
|
|
}
|
|
}
|
|
|
|
if (!info->connected) {
|
|
/* connect */
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (sink, CONTINUE, "connect",
|
|
("Connecting to %s", info->location));
|
|
GST_DEBUG_OBJECT (sink, "connecting (%s)...", info->location);
|
|
if ((res =
|
|
gst_rtsp_connection_connect (info->connection,
|
|
sink->ptcp_timeout)) < 0)
|
|
goto could_not_connect;
|
|
|
|
info->connected = TRUE;
|
|
}
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
parse_error:
|
|
{
|
|
GST_ERROR_OBJECT (sink, "No valid RTSP URL was provided");
|
|
return res;
|
|
}
|
|
could_not_create:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
GST_ERROR_OBJECT (sink, "Could not create connection. (%s)", str);
|
|
g_free (str);
|
|
return res;
|
|
}
|
|
could_not_connect:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
GST_ERROR_OBJECT (sink, "Could not connect to server. (%s)", str);
|
|
g_free (str);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_conninfo_close (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
|
|
gboolean free)
|
|
{
|
|
GST_RTSP_STATE_LOCK (sink);
|
|
if (info->connected) {
|
|
GST_DEBUG_OBJECT (sink, "closing connection...");
|
|
gst_rtsp_connection_close (info->connection);
|
|
info->connected = FALSE;
|
|
}
|
|
if (free && info->connection) {
|
|
/* free connection */
|
|
GST_DEBUG_OBJECT (sink, "freeing connection...");
|
|
gst_rtsp_connection_free (info->connection);
|
|
info->connection = NULL;
|
|
}
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_conninfo_reconnect (GstRTSPClientSink * sink, GstRTSPConnInfo * info,
|
|
gboolean async)
|
|
{
|
|
GstRTSPResult res;
|
|
|
|
GST_DEBUG_OBJECT (sink, "reconnecting connection...");
|
|
gst_rtsp_conninfo_close (sink, info, FALSE);
|
|
res = gst_rtsp_conninfo_connect (sink, info, async);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_connection_flush (GstRTSPClientSink * sink, gboolean flush)
|
|
{
|
|
GList *walk;
|
|
|
|
GST_DEBUG_OBJECT (sink, "set flushing %d", flush);
|
|
g_mutex_lock (&sink->preroll_lock);
|
|
if (sink->conninfo.connection && sink->conninfo.flushing != flush) {
|
|
GST_DEBUG_OBJECT (sink, "connection flush");
|
|
gst_rtsp_connection_flush (sink->conninfo.connection, flush);
|
|
sink->conninfo.flushing = flush;
|
|
}
|
|
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
|
|
if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
|
|
GST_DEBUG_OBJECT (sink, "stream %p flush", stream);
|
|
gst_rtsp_connection_flush (stream->conninfo.connection, flush);
|
|
stream->conninfo.flushing = flush;
|
|
}
|
|
}
|
|
g_cond_broadcast (&sink->preroll_cond);
|
|
g_mutex_unlock (&sink->preroll_lock);
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_init_request (GstRTSPClientSink * sink,
|
|
GstRTSPMessage * msg, GstRTSPMethod method, const gchar * uri)
|
|
{
|
|
GstRTSPResult res;
|
|
|
|
res = gst_rtsp_message_init_request (msg, method, uri);
|
|
if (res < 0)
|
|
return res;
|
|
|
|
/* set user-agent */
|
|
if (sink->user_agent)
|
|
gst_rtsp_message_add_header (msg, GST_RTSP_HDR_USER_AGENT,
|
|
sink->user_agent);
|
|
|
|
return res;
|
|
}
|
|
|
|
/* FIXME, handle server request, reply with OK, for now */
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_handle_request (GstRTSPClientSink * sink,
|
|
GstRTSPConnection * conn, GstRTSPMessage * request)
|
|
{
|
|
GstRTSPMessage response = { 0 };
|
|
GstRTSPResult res;
|
|
|
|
GST_DEBUG_OBJECT (sink, "got server request message");
|
|
|
|
if (sink->debug)
|
|
gst_rtsp_message_dump (request);
|
|
|
|
/* default implementation, send OK */
|
|
GST_DEBUG_OBJECT (sink, "prepare OK reply");
|
|
res =
|
|
gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
|
|
request);
|
|
if (res < 0)
|
|
goto send_error;
|
|
|
|
/* let app parse and reply */
|
|
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_HANDLE_REQUEST],
|
|
0, request, &response);
|
|
|
|
if (sink->debug)
|
|
gst_rtsp_message_dump (&response);
|
|
|
|
res = gst_rtsp_client_sink_connection_send (sink, conn, &response, NULL);
|
|
if (res < 0)
|
|
goto send_error;
|
|
|
|
gst_rtsp_message_unset (&response);
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
send_error:
|
|
{
|
|
gst_rtsp_message_unset (&response);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
/* send server keep-alive */
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_send_keep_alive (GstRTSPClientSink * sink)
|
|
{
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPResult res;
|
|
GstRTSPMethod method;
|
|
const gchar *control;
|
|
|
|
if (sink->do_rtsp_keep_alive == FALSE) {
|
|
GST_DEBUG_OBJECT (sink, "do-rtsp-keep-alive is FALSE, not sending.");
|
|
gst_rtsp_connection_reset_timeout (sink->conninfo.connection);
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (sink, "creating server keep-alive");
|
|
|
|
/* find a method to use for keep-alive */
|
|
if (sink->methods & GST_RTSP_GET_PARAMETER)
|
|
method = GST_RTSP_GET_PARAMETER;
|
|
else
|
|
method = GST_RTSP_OPTIONS;
|
|
|
|
control = get_aggregate_control (sink);
|
|
if (control == NULL)
|
|
goto no_control;
|
|
|
|
res = gst_rtsp_client_sink_init_request (sink, &request, method, control);
|
|
if (res < 0)
|
|
goto send_error;
|
|
|
|
if (sink->debug)
|
|
gst_rtsp_message_dump (&request);
|
|
|
|
res =
|
|
gst_rtsp_client_sink_connection_send (sink, sink->conninfo.connection,
|
|
&request, NULL);
|
|
if (res < 0)
|
|
goto send_error;
|
|
|
|
gst_rtsp_connection_reset_timeout (sink->conninfo.connection);
|
|
gst_rtsp_message_unset (&request);
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
no_control:
|
|
{
|
|
GST_WARNING_OBJECT (sink, "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 (sink, RESOURCE, WRITE, (NULL),
|
|
("Could not send keep-alive. (%s)", str));
|
|
g_free (str);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtsp_client_sink_loop_rx (GstRTSPClientSink * sink)
|
|
{
|
|
GstRTSPResult res;
|
|
GstRTSPMessage message = { 0 };
|
|
gint retry = 0;
|
|
|
|
while (TRUE) {
|
|
GTimeVal tv_timeout;
|
|
|
|
/* get the next timeout interval */
|
|
gst_rtsp_connection_next_timeout (sink->conninfo.connection, &tv_timeout);
|
|
|
|
GST_DEBUG_OBJECT (sink, "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_rtsp_client_sink_connection_receive (sink,
|
|
sink->conninfo.connection, &message, &tv_timeout);
|
|
|
|
switch (res) {
|
|
case GST_RTSP_OK:
|
|
GST_DEBUG_OBJECT (sink, "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 (sink, "timeout, sending keep-alive");
|
|
if ((res =
|
|
gst_rtsp_client_sink_send_keep_alive (sink)) == 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 (sink, RESOURCE, READ, (NULL),
|
|
("The server closed the connection."));
|
|
if (sink->udp_reconnect) {
|
|
if ((res =
|
|
gst_rtsp_conninfo_reconnect (sink, &sink->conninfo,
|
|
FALSE)) < 0)
|
|
goto connect_error;
|
|
} else {
|
|
goto server_eof;
|
|
}
|
|
continue;
|
|
case GST_RTSP_ENET:
|
|
GST_DEBUG_OBJECT (sink, "An ethernet problem occured.");
|
|
default:
|
|
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
|
|
("Unhandled return value %d.", res));
|
|
goto receive_error;
|
|
}
|
|
|
|
switch (message.type) {
|
|
case GST_RTSP_MESSAGE_REQUEST:
|
|
/* server sends us a request message, handle it */
|
|
res =
|
|
gst_rtsp_client_sink_handle_request (sink,
|
|
sink->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 (sink, "ignoring response message");
|
|
if (sink->debug)
|
|
gst_rtsp_message_dump (&message);
|
|
if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
|
|
GST_DEBUG_OBJECT (sink, "but is Unauthorized response ...");
|
|
if (gst_rtsp_client_sink_setup_auth (sink, &message) && !(retry++)) {
|
|
GST_DEBUG_OBJECT (sink, "so retrying keep-alive");
|
|
if ((res =
|
|
gst_rtsp_client_sink_send_keep_alive (sink)) ==
|
|
GST_RTSP_EINTR)
|
|
goto interrupt;
|
|
}
|
|
} else {
|
|
retry = 0;
|
|
}
|
|
break;
|
|
case GST_RTSP_MESSAGE_DATA:
|
|
/* we ignore response and data messages */
|
|
GST_DEBUG_OBJECT (sink, "ignoring data message");
|
|
break;
|
|
default:
|
|
GST_WARNING_OBJECT (sink, "ignoring unknown message type %d",
|
|
message.type);
|
|
break;
|
|
}
|
|
}
|
|
g_assert_not_reached ();
|
|
|
|
/* we get here when the connection got interrupted */
|
|
interrupt:
|
|
{
|
|
gst_rtsp_message_unset (&message);
|
|
GST_DEBUG_OBJECT (sink, "got interrupted");
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
connect_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
GstFlowReturn ret;
|
|
|
|
sink->conninfo.connected = FALSE;
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ_WRITE, (NULL),
|
|
("Could not connect to server. (%s)", str));
|
|
g_free (str);
|
|
ret = GST_FLOW_ERROR;
|
|
} else {
|
|
ret = GST_FLOW_FLUSHING;
|
|
}
|
|
return ret;
|
|
}
|
|
receive_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (sink, 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 (sink, RESOURCE, WRITE, (NULL),
|
|
("Could not handle server message. (%s)", str));
|
|
g_free (str);
|
|
ret = GST_FLOW_ERROR;
|
|
} else {
|
|
ret = GST_FLOW_FLUSHING;
|
|
}
|
|
return ret;
|
|
}
|
|
server_eof:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "we got an eof from the server");
|
|
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
|
|
("The server closed the connection."));
|
|
sink->conninfo.connected = FALSE;
|
|
gst_rtsp_message_unset (&message);
|
|
return GST_FLOW_EOS;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_reconnect (GstRTSPClientSink * sink, gboolean async)
|
|
{
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
gboolean restart = FALSE;
|
|
|
|
GST_DEBUG_OBJECT (sink, "doing reconnect");
|
|
|
|
GST_FIXME_OBJECT (sink, "Reconnection is not yet implemented");
|
|
|
|
/* no need to restart, we're done */
|
|
if (!restart)
|
|
goto done;
|
|
|
|
/* we can try only TCP now */
|
|
sink->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
|
|
|
|
/* close and cleanup our state */
|
|
if ((res = gst_rtsp_client_sink_close (sink, 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 (!(sink->protocols & GST_RTSP_LOWER_TRANS_TCP) || sink->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 (sink, 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 (sink->udp_timeout / 1000000.0)));
|
|
|
|
/* open new connection using tcp */
|
|
if (gst_rtsp_client_sink_open (sink, async) < 0)
|
|
goto open_failed;
|
|
|
|
/* start recording */
|
|
if (gst_rtsp_client_sink_record (sink, async) < 0)
|
|
goto play_failed;
|
|
|
|
done:
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
no_protocols:
|
|
{
|
|
sink->cur_protocols = 0;
|
|
/* no transport possible, post an error and stop */
|
|
GST_ELEMENT_ERROR (sink, 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 (sink->udp_timeout / 1000000.0)));
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
open_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "open failed");
|
|
return GST_RTSP_OK;
|
|
}
|
|
play_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "play failed");
|
|
return GST_RTSP_OK;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_loop_start_cmd (GstRTSPClientSink * sink, gint cmd)
|
|
{
|
|
switch (cmd) {
|
|
case CMD_OPEN:
|
|
GST_ELEMENT_PROGRESS (sink, START, "open", ("Opening Stream"));
|
|
break;
|
|
case CMD_RECORD:
|
|
GST_ELEMENT_PROGRESS (sink, START, "request", ("Sending RECORD request"));
|
|
break;
|
|
case CMD_PAUSE:
|
|
GST_ELEMENT_PROGRESS (sink, START, "request", ("Sending PAUSE request"));
|
|
break;
|
|
case CMD_CLOSE:
|
|
GST_ELEMENT_PROGRESS (sink, START, "close", ("Closing Stream"));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_loop_complete_cmd (GstRTSPClientSink * sink, gint cmd)
|
|
{
|
|
switch (cmd) {
|
|
case CMD_OPEN:
|
|
GST_ELEMENT_PROGRESS (sink, COMPLETE, "open", ("Opened Stream"));
|
|
break;
|
|
case CMD_RECORD:
|
|
GST_ELEMENT_PROGRESS (sink, COMPLETE, "request", ("Sent RECORD request"));
|
|
break;
|
|
case CMD_PAUSE:
|
|
GST_ELEMENT_PROGRESS (sink, COMPLETE, "request", ("Sent PAUSE request"));
|
|
break;
|
|
case CMD_CLOSE:
|
|
GST_ELEMENT_PROGRESS (sink, COMPLETE, "close", ("Closed Stream"));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_loop_cancel_cmd (GstRTSPClientSink * sink, gint cmd)
|
|
{
|
|
switch (cmd) {
|
|
case CMD_OPEN:
|
|
GST_ELEMENT_PROGRESS (sink, CANCELED, "open", ("Open canceled"));
|
|
break;
|
|
case CMD_RECORD:
|
|
GST_ELEMENT_PROGRESS (sink, CANCELED, "request", ("RECORD canceled"));
|
|
break;
|
|
case CMD_PAUSE:
|
|
GST_ELEMENT_PROGRESS (sink, CANCELED, "request", ("PAUSE canceled"));
|
|
break;
|
|
case CMD_CLOSE:
|
|
GST_ELEMENT_PROGRESS (sink, CANCELED, "close", ("Close canceled"));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_loop_error_cmd (GstRTSPClientSink * sink, gint cmd)
|
|
{
|
|
switch (cmd) {
|
|
case CMD_OPEN:
|
|
GST_ELEMENT_PROGRESS (sink, ERROR, "open", ("Open failed"));
|
|
break;
|
|
case CMD_RECORD:
|
|
GST_ELEMENT_PROGRESS (sink, ERROR, "request", ("RECORD failed"));
|
|
break;
|
|
case CMD_PAUSE:
|
|
GST_ELEMENT_PROGRESS (sink, ERROR, "request", ("PAUSE failed"));
|
|
break;
|
|
case CMD_CLOSE:
|
|
GST_ELEMENT_PROGRESS (sink, ERROR, "close", ("Close failed"));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_loop_end_cmd (GstRTSPClientSink * sink, gint cmd,
|
|
GstRTSPResult ret)
|
|
{
|
|
if (ret == GST_RTSP_OK)
|
|
gst_rtsp_client_sink_loop_complete_cmd (sink, cmd);
|
|
else if (ret == GST_RTSP_EINTR)
|
|
gst_rtsp_client_sink_loop_cancel_cmd (sink, cmd);
|
|
else
|
|
gst_rtsp_client_sink_loop_error_cmd (sink, cmd);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_client_sink_loop_send_cmd (GstRTSPClientSink * sink, gint cmd,
|
|
gint mask)
|
|
{
|
|
gint old;
|
|
gboolean flushed = FALSE;
|
|
|
|
/* start new request */
|
|
gst_rtsp_client_sink_loop_start_cmd (sink, cmd);
|
|
|
|
GST_DEBUG_OBJECT (sink, "sending cmd %s", cmd_to_string (cmd));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
old = sink->pending_cmd;
|
|
if (old == CMD_RECONNECT) {
|
|
GST_DEBUG_OBJECT (sink, "ignore, we were reconnecting");
|
|
cmd = CMD_RECONNECT;
|
|
}
|
|
if (old != CMD_WAIT) {
|
|
sink->pending_cmd = CMD_WAIT;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
/* cancel previous request */
|
|
GST_DEBUG_OBJECT (sink, "cancel previous request %s", cmd_to_string (old));
|
|
gst_rtsp_client_sink_loop_cancel_cmd (sink, old);
|
|
GST_OBJECT_LOCK (sink);
|
|
}
|
|
sink->pending_cmd = cmd;
|
|
/* interrupt if allowed */
|
|
if (sink->busy_cmd & mask) {
|
|
GST_DEBUG_OBJECT (sink, "connection flush busy %s",
|
|
cmd_to_string (sink->busy_cmd));
|
|
gst_rtsp_client_sink_connection_flush (sink, TRUE);
|
|
flushed = TRUE;
|
|
} else {
|
|
GST_DEBUG_OBJECT (sink, "not interrupting busy cmd %s",
|
|
cmd_to_string (sink->busy_cmd));
|
|
}
|
|
if (sink->task)
|
|
gst_task_start (sink->task);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return flushed;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_client_sink_loop (GstRTSPClientSink * sink)
|
|
{
|
|
GstFlowReturn ret;
|
|
|
|
if (!sink->conninfo.connection || !sink->conninfo.connected)
|
|
goto no_connection;
|
|
|
|
ret = gst_rtsp_client_sink_loop_rx (sink);
|
|
if (ret != GST_FLOW_OK)
|
|
goto pause;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_connection:
|
|
{
|
|
GST_WARNING_OBJECT (sink, "we are not connected");
|
|
ret = GST_FLOW_FLUSHING;
|
|
goto pause;
|
|
}
|
|
pause:
|
|
{
|
|
const gchar *reason = gst_flow_get_name (ret);
|
|
|
|
GST_DEBUG_OBJECT (sink, "pausing task, reason %s", reason);
|
|
gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT, CMD_LOOP);
|
|
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_rtsp_client_sink_skip_lws (const gchar * s)
|
|
{
|
|
while (g_ascii_isspace (*s))
|
|
s++;
|
|
return s;
|
|
}
|
|
|
|
static const gchar *
|
|
gst_rtsp_client_sink_unskip_lws (const gchar * s, const gchar * start)
|
|
{
|
|
while (s > start && g_ascii_isspace (*(s - 1)))
|
|
s--;
|
|
return s;
|
|
}
|
|
|
|
static const gchar *
|
|
gst_rtsp_client_sink_skip_commas (const gchar * s)
|
|
{
|
|
/* The grammar allows for multiple commas */
|
|
while (g_ascii_isspace (*s) || *s == ',')
|
|
s++;
|
|
return s;
|
|
}
|
|
|
|
static const gchar *
|
|
gst_rtsp_client_sink_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_rtsp_client_sink_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_rtsp_client_sink_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_rtsp_client_sink_skip_commas (header);
|
|
while (*header) {
|
|
end = gst_rtsp_client_sink_skip_item (header);
|
|
list = g_slist_prepend (list, g_strndup (header, end - header));
|
|
header = gst_rtsp_client_sink_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_rtsp_client_sink_unskip_lws (eq, item);
|
|
if (name_end == item) {
|
|
/* That's no good... */
|
|
g_free (item);
|
|
continue;
|
|
}
|
|
|
|
*name_end = '\0';
|
|
|
|
value = (gchar *) gst_rtsp_client_sink_skip_lws (eq + 1);
|
|
if (*value == '"')
|
|
gst_rtsp_decode_quoted_string (value);
|
|
} else
|
|
value = NULL;
|
|
|
|
if (value && strcmp (item, "stale") == 0 && 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_rtsp_client_sink_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_rtsp_client_sink_parse_digest_challenge (conn, &start[7], stale);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_sink_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_rtsp_client_sink_setup_auth (GstRTSPClientSink * sink,
|
|
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 = sink->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_rtsp_client_sink_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)
|
|
sink->tried_url_auth = FALSE;
|
|
|
|
url = gst_rtsp_connection_get_url (conn);
|
|
|
|
/* Do we have username and password available? */
|
|
if (url != NULL && !sink->tried_url_auth && url->user != NULL
|
|
&& url->passwd != NULL) {
|
|
user = url->user;
|
|
pass = url->passwd;
|
|
sink->tried_url_auth = TRUE;
|
|
GST_DEBUG_OBJECT (sink,
|
|
"Attempting authentication using credentials from the URL");
|
|
} else {
|
|
user = sink->user_id;
|
|
pass = sink->user_pw;
|
|
GST_DEBUG_OBJECT (sink,
|
|
"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 (sink, "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 (sink, 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_rtsp_client_sink_try_send (GstRTSPClientSink * sink,
|
|
GstRTSPConnection * conn, GstRTSPMessage * request,
|
|
GstRTSPMessage * response, GstRTSPStatusCode * code)
|
|
{
|
|
GstRTSPResult res;
|
|
GstRTSPStatusCode thecode;
|
|
gchar *content_base = NULL;
|
|
gint try = 0;
|
|
|
|
again:
|
|
GST_DEBUG_OBJECT (sink, "sending message");
|
|
|
|
if (sink->debug)
|
|
gst_rtsp_message_dump (request);
|
|
|
|
g_mutex_lock (&sink->send_lock);
|
|
|
|
res =
|
|
gst_rtsp_client_sink_connection_send (sink, conn, request,
|
|
sink->ptcp_timeout);
|
|
if (res < 0) {
|
|
g_mutex_unlock (&sink->send_lock);
|
|
goto send_error;
|
|
}
|
|
|
|
gst_rtsp_connection_reset_timeout (conn);
|
|
|
|
/* See if we should handle the response */
|
|
if (response == NULL) {
|
|
g_mutex_unlock (&sink->send_lock);
|
|
return GST_RTSP_OK;
|
|
}
|
|
next:
|
|
res =
|
|
gst_rtsp_client_sink_connection_receive (sink, conn, response,
|
|
sink->ptcp_timeout);
|
|
|
|
g_mutex_unlock (&sink->send_lock);
|
|
|
|
if (res < 0)
|
|
goto receive_error;
|
|
|
|
if (sink->debug)
|
|
gst_rtsp_message_dump (response);
|
|
|
|
|
|
switch (response->type) {
|
|
case GST_RTSP_MESSAGE_REQUEST:
|
|
res = gst_rtsp_client_sink_handle_request (sink, conn, response);
|
|
if (res == GST_RTSP_EEOF)
|
|
goto server_eof;
|
|
else if (res < 0)
|
|
goto handle_request_failed;
|
|
g_mutex_lock (&sink->send_lock);
|
|
goto next;
|
|
case GST_RTSP_MESSAGE_RESPONSE:
|
|
/* ok, a response is good */
|
|
GST_DEBUG_OBJECT (sink, "received response message");
|
|
break;
|
|
case GST_RTSP_MESSAGE_DATA:
|
|
/* we ignore data messages */
|
|
GST_DEBUG_OBJECT (sink, "ignoring data message");
|
|
g_mutex_lock (&sink->send_lock);
|
|
goto next;
|
|
default:
|
|
GST_WARNING_OBJECT (sink, "ignoring unknown message type %d",
|
|
response->type);
|
|
g_mutex_lock (&sink->send_lock);
|
|
goto next;
|
|
}
|
|
|
|
thecode = response->type_data.response.code;
|
|
|
|
GST_DEBUG_OBJECT (sink, "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 (sink->content_base);
|
|
sink->content_base = g_strdup (content_base);
|
|
}
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
send_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (sink, "send interrupted");
|
|
}
|
|
g_free (str);
|
|
return res;
|
|
}
|
|
receive_error:
|
|
{
|
|
switch (res) {
|
|
case GST_RTSP_EEOF:
|
|
GST_WARNING_OBJECT (sink, "server closed connection");
|
|
if ((try == 0) && !sink->interleaved && sink->udp_reconnect) {
|
|
try++;
|
|
/* if reconnect succeeds, try again */
|
|
if ((res =
|
|
gst_rtsp_conninfo_reconnect (sink, &sink->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 (sink, RESOURCE, READ, (NULL),
|
|
("Could not receive message. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (sink, "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 (sink, "we got an eof from the server");
|
|
GST_ELEMENT_WARNING (sink, RESOURCE, READ, (NULL),
|
|
("The server closed the connection."));
|
|
gst_rtsp_message_unset (response);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_set_state (GstRTSPClientSink * sink, GstState state)
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "Setting internal state to %s",
|
|
gst_element_state_get_name (state));
|
|
gst_element_set_state (GST_ELEMENT (sink->internal_bin), state);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_sink_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_rtsp_client_sink_setup_auth and retry
|
|
* the request.
|
|
*
|
|
* Returns: #GST_RTSP_OK if the processing was successful.
|
|
*/
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_send (GstRTSPClientSink * sink, 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_rtsp_client_sink_try_send (sink, conn, request, response,
|
|
&int_code)) < 0)
|
|
goto error;
|
|
|
|
switch (int_code) {
|
|
case GST_RTSP_STS_UNAUTHORIZED:
|
|
if (gst_rtsp_client_sink_setup_auth (sink, 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 (sink, "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 (sink, RESOURCE, NOT_FOUND, (NULL), ("%s",
|
|
response->type_data.response.reason));
|
|
break;
|
|
case GST_RTSP_STS_UNAUTHORIZED:
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_AUTHORIZED, (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 (sink, "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 (sink, "redirection to %s", new_location);
|
|
|
|
/* save current transports */
|
|
if (sink->conninfo.url)
|
|
transports = sink->conninfo.url->transports;
|
|
else
|
|
transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
|
|
|
|
gst_rtsp_client_sink_uri_set_uri (GST_URI_HANDLER (sink), new_location,
|
|
NULL);
|
|
|
|
/* set old transports */
|
|
if (sink->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
|
|
sink->conninfo.url->transports = transports;
|
|
|
|
sink->need_redirect = TRUE;
|
|
sink->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 (sink, "got NOT IMPLEMENTED, disable method %s",
|
|
gst_rtsp_method_as_text (method));
|
|
sink->methods &= ~method;
|
|
res = GST_RTSP_OK;
|
|
break;
|
|
default:
|
|
GST_ELEMENT_ERROR (sink, 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;
|
|
}
|
|
}
|
|
|
|
/* 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_rtsp_client_sink_parse_methods (GstRTSPClientSink * sink,
|
|
GstRTSPMessage * response)
|
|
{
|
|
GstRTSPHeaderField field;
|
|
gchar *respoptions;
|
|
gint indx = 0;
|
|
|
|
/* reset supported methods */
|
|
sink->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;
|
|
|
|
sink->methods |= gst_rtsp_options_from_text (respoptions);
|
|
|
|
indx++;
|
|
}
|
|
|
|
if (sink->methods == 0) {
|
|
/* neither Allow nor Public are required, assume the server supports
|
|
* at least SETUP. */
|
|
GST_DEBUG_OBJECT (sink, "could not get OPTIONS");
|
|
sink->methods = GST_RTSP_SETUP;
|
|
}
|
|
|
|
/* Even if the server replied, and didn't say it supports
|
|
* RECORD|ANNOUNCE, try anyway by assuming it does */
|
|
sink->methods |= GST_RTSP_ANNOUNCE | GST_RTSP_RECORD;
|
|
|
|
if (!(sink->methods & GST_RTSP_SETUP))
|
|
goto no_setup;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_setup:
|
|
{
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_READ, (NULL),
|
|
("Server does not support SETUP."));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_connect_to_server (GstRTSPClientSink * sink,
|
|
gboolean async)
|
|
{
|
|
GstRTSPResult res;
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPMessage response = { 0 };
|
|
GSocket *conn_socket;
|
|
GSocketAddress *sa;
|
|
GInetAddress *ia;
|
|
|
|
sink->need_redirect = FALSE;
|
|
|
|
/* can't continue without a valid url */
|
|
if (G_UNLIKELY (sink->conninfo.url == NULL)) {
|
|
res = GST_RTSP_EINVAL;
|
|
goto no_url;
|
|
}
|
|
sink->tried_url_auth = FALSE;
|
|
|
|
if ((res = gst_rtsp_conninfo_connect (sink, &sink->conninfo, async)) < 0)
|
|
goto connect_failed;
|
|
|
|
conn_socket = gst_rtsp_connection_get_read_socket (sink->conninfo.connection);
|
|
sa = g_socket_get_remote_address (conn_socket, NULL);
|
|
ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
|
|
|
|
sink->server_ip = g_inet_address_to_string (ia);
|
|
|
|
g_object_unref (sa);
|
|
|
|
/* create OPTIONS */
|
|
GST_DEBUG_OBJECT (sink, "create options...");
|
|
res =
|
|
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_OPTIONS,
|
|
sink->conninfo.url_str);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
/* send OPTIONS */
|
|
GST_DEBUG_OBJECT (sink, "send options...");
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (sink, CONTINUE, "open",
|
|
("Retrieving server options"));
|
|
|
|
if ((res =
|
|
gst_rtsp_client_sink_send (sink, sink->conninfo.connection, &request,
|
|
&response, NULL)) < 0)
|
|
goto send_error;
|
|
|
|
/* parse OPTIONS */
|
|
if (!gst_rtsp_client_sink_parse_methods (sink, &response))
|
|
goto methods_error;
|
|
|
|
/* FIXME: Do we need to handle REDIRECT responses for OPTIONS? */
|
|
|
|
/* clean up any messages */
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
no_url:
|
|
{
|
|
GST_ELEMENT_ERROR (sink, 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 (sink, RESOURCE, OPEN_READ_WRITE, (NULL),
|
|
("Failed to connect. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (sink, "connect interrupted");
|
|
}
|
|
g_free (str);
|
|
goto cleanup_error;
|
|
}
|
|
create_request_failed:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (sink, 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;
|
|
}
|
|
cleanup_error:
|
|
{
|
|
if (sink->conninfo.connection) {
|
|
GST_DEBUG_OBJECT (sink, "free connection");
|
|
gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
|
|
}
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_open (GstRTSPClientSink * sink, gboolean async)
|
|
{
|
|
GstRTSPResult ret;
|
|
|
|
sink->methods =
|
|
GST_RTSP_SETUP | GST_RTSP_RECORD | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
|
|
|
|
if ((ret = gst_rtsp_client_sink_connect_to_server (sink, async)) < 0)
|
|
goto open_failed;
|
|
|
|
if (async)
|
|
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_OPEN, ret);
|
|
|
|
/* Collect all our input streams and create
|
|
* stream objects before actually returning */
|
|
gst_rtsp_client_sink_collect_streams (sink);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
open_failed:
|
|
{
|
|
GST_WARNING_OBJECT (sink, "Failed to connect to server");
|
|
sink->open_error = TRUE;
|
|
if (async)
|
|
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_OPEN, ret);
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_close (GstRTSPClientSink * sink, gboolean async,
|
|
gboolean only_close)
|
|
{
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPMessage response = { 0 };
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
GList *walk;
|
|
const gchar *control;
|
|
|
|
GST_DEBUG_OBJECT (sink, "TEARDOWN...");
|
|
|
|
gst_rtsp_client_sink_set_state (sink, GST_STATE_NULL);
|
|
|
|
if (sink->state < GST_RTSP_STATE_READY) {
|
|
GST_DEBUG_OBJECT (sink, "not ready, doing cleanup");
|
|
goto close;
|
|
}
|
|
|
|
if (only_close)
|
|
goto close;
|
|
|
|
/* construct a control url */
|
|
control = get_aggregate_control (sink);
|
|
|
|
if (!(sink->methods & (GST_RTSP_RECORD | GST_RTSP_TEARDOWN)))
|
|
goto not_supported;
|
|
|
|
/* stop streaming */
|
|
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
|
|
|
|
if (context->stream_transport)
|
|
gst_rtsp_stream_transport_set_active (context->stream_transport, FALSE);
|
|
|
|
if (context->joined) {
|
|
gst_rtsp_stream_leave_bin (context->stream, GST_BIN (sink->internal_bin),
|
|
sink->rtpbin);
|
|
context->joined = FALSE;
|
|
}
|
|
}
|
|
|
|
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
|
|
const gchar *setup_url;
|
|
GstRTSPConnInfo *info;
|
|
|
|
GST_DEBUG_OBJECT (sink, "Looking at stream %p for teardown",
|
|
context->stream);
|
|
|
|
/* try aggregate control first but do non-aggregate control otherwise */
|
|
if (control)
|
|
setup_url = control;
|
|
else if ((setup_url = context->conninfo.location) == NULL) {
|
|
GST_DEBUG_OBJECT (sink, "Skipping TEARDOWN stream %p - no setup URL",
|
|
context->stream);
|
|
continue;
|
|
}
|
|
|
|
if (sink->conninfo.connection) {
|
|
info = &sink->conninfo;
|
|
} else if (context->conninfo.connection) {
|
|
info = &context->conninfo;
|
|
} else {
|
|
continue;
|
|
}
|
|
if (!info->connected)
|
|
goto next;
|
|
|
|
/* do TEARDOWN */
|
|
GST_DEBUG_OBJECT (sink, "Sending teardown for stream %p at URL %s",
|
|
context->stream, setup_url);
|
|
res =
|
|
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_TEARDOWN,
|
|
setup_url);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (sink, CONTINUE, "close", ("Closing stream"));
|
|
|
|
if ((res =
|
|
gst_rtsp_client_sink_send (sink, 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 (sink, "closing connection...");
|
|
gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
|
|
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
|
|
gst_rtsp_conninfo_close (sink, &stream->conninfo, TRUE);
|
|
}
|
|
|
|
/* cleanup */
|
|
gst_rtsp_client_sink_cleanup (sink);
|
|
|
|
sink->state = GST_RTSP_STATE_INVALID;
|
|
|
|
if (async)
|
|
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_CLOSE, res);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
create_request_failed:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (sink, 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 (sink, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (sink, "TEARDOWN interrupted");
|
|
}
|
|
g_free (str);
|
|
goto close;
|
|
}
|
|
not_supported:
|
|
{
|
|
GST_DEBUG_OBJECT (sink,
|
|
"TEARDOWN and PLAY not supported, can't do TEARDOWN");
|
|
goto close;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_client_sink_configure_manager (GstRTSPClientSink * sink)
|
|
{
|
|
GstElement *rtpbin;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtpbin = sink->rtpbin;
|
|
|
|
if (rtpbin == NULL) {
|
|
GObjectClass *klass;
|
|
|
|
rtpbin = gst_element_factory_make ("rtpbin", NULL);
|
|
if (rtpbin == NULL)
|
|
goto no_rtpbin;
|
|
|
|
gst_bin_add (GST_BIN_CAST (sink->internal_bin), rtpbin);
|
|
|
|
sink->rtpbin = rtpbin;
|
|
|
|
/* Any more settings we should configure on rtpbin here? */
|
|
g_object_set (sink->rtpbin, "latency", sink->latency, NULL);
|
|
|
|
klass = G_OBJECT_GET_CLASS (G_OBJECT (rtpbin));
|
|
|
|
if (g_object_class_find_property (klass, "ntp-time-source")) {
|
|
g_object_set (sink->rtpbin, "ntp-time-source", sink->ntp_time_source,
|
|
NULL);
|
|
}
|
|
|
|
if (sink->sdes && g_object_class_find_property (klass, "sdes")) {
|
|
g_object_set (sink->rtpbin, "sdes", sink->sdes, NULL);
|
|
}
|
|
|
|
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_NEW_MANAGER], 0,
|
|
sink->rtpbin);
|
|
}
|
|
|
|
ret = gst_element_set_state (rtpbin, GST_STATE_PAUSED);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
goto start_manager_failure;
|
|
|
|
return TRUE;
|
|
|
|
no_rtpbin:
|
|
{
|
|
GST_WARNING ("no rtpbin element");
|
|
g_warning ("failed to create element 'rtpbin', check your installation");
|
|
return FALSE;
|
|
}
|
|
start_manager_failure:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "could not start session manager");
|
|
gst_bin_remove (GST_BIN_CAST (sink->internal_bin), rtpbin);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstElement *
|
|
request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPClientSink * sink)
|
|
{
|
|
GstRTSPStream *stream = NULL;
|
|
GstElement *ret = NULL;
|
|
GList *walk;
|
|
|
|
GST_RTSP_STATE_LOCK (sink);
|
|
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
|
|
|
|
if (sessid == gst_rtsp_stream_get_index (context->stream)) {
|
|
stream = context->stream;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (stream != NULL) {
|
|
GST_DEBUG_OBJECT (sink, "Creating aux sender for stream %u", sessid);
|
|
ret = gst_rtsp_stream_request_aux_sender (stream, sessid);
|
|
}
|
|
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink)
|
|
{
|
|
GstRTSPStreamContext *context;
|
|
GList *walk;
|
|
const gchar *base;
|
|
gboolean has_slash;
|
|
|
|
GST_DEBUG_OBJECT (sink, "Collecting stream information");
|
|
|
|
if (!gst_rtsp_client_sink_configure_manager (sink))
|
|
return FALSE;
|
|
|
|
base = get_aggregate_control (sink);
|
|
/* check if the base ends with / */
|
|
has_slash = g_str_has_suffix (base, "/");
|
|
|
|
g_mutex_lock (&sink->preroll_lock);
|
|
while (sink->contexts == NULL && !sink->conninfo.flushing) {
|
|
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
|
|
}
|
|
g_mutex_unlock (&sink->preroll_lock);
|
|
|
|
/* FIXME: Need different locking - need to protect against pad releases
|
|
* and potential state changes ruining things here */
|
|
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
|
|
GstPad *srcpad;
|
|
|
|
context = (GstRTSPStreamContext *) walk->data;
|
|
if (context->stream)
|
|
continue;
|
|
|
|
g_mutex_lock (&sink->preroll_lock);
|
|
while (!context->prerolled && !sink->conninfo.flushing) {
|
|
GST_DEBUG_OBJECT (sink, "Waiting for caps on stream %d", context->index);
|
|
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
|
|
}
|
|
if (sink->conninfo.flushing) {
|
|
g_mutex_unlock (&sink->preroll_lock);
|
|
break;
|
|
}
|
|
g_mutex_unlock (&sink->preroll_lock);
|
|
|
|
if (context->payloader == NULL)
|
|
continue;
|
|
|
|
srcpad = gst_element_get_static_pad (context->payloader, "src");
|
|
|
|
GST_DEBUG_OBJECT (sink, "Creating stream object for stream %d",
|
|
context->index);
|
|
context->stream =
|
|
gst_rtsp_client_sink_create_stream (sink, context, context->payloader,
|
|
srcpad);
|
|
|
|
/* concatenate the two strings, insert / when not present */
|
|
g_free (context->conninfo.location);
|
|
context->conninfo.location =
|
|
g_strdup_printf ("%s%sstream=%d", base, has_slash ? "" : "/",
|
|
context->index);
|
|
|
|
if (sink->rtx_time > 0) {
|
|
/* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
|
|
g_signal_connect (sink->rtpbin, "request-aux-sender",
|
|
(GCallback) request_aux_sender, sink);
|
|
}
|
|
|
|
if (!gst_rtsp_stream_join_bin (context->stream,
|
|
GST_BIN (sink->internal_bin), sink->rtpbin, GST_STATE_PAUSED)) {
|
|
goto join_bin_failed;
|
|
}
|
|
context->joined = TRUE;
|
|
|
|
/* Let the stream object receive data */
|
|
gst_pad_remove_probe (srcpad, context->payloader_block_id);
|
|
|
|
gst_object_unref (srcpad);
|
|
}
|
|
|
|
/* Now wait for the preroll of the rtp bin */
|
|
g_mutex_lock (&sink->preroll_lock);
|
|
while (!sink->prerolled && !sink->conninfo.flushing) {
|
|
GST_LOG_OBJECT (sink, "Waiting for preroll before continuing");
|
|
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
|
|
}
|
|
GST_LOG_OBJECT (sink, "Marking streams as collected");
|
|
sink->streams_collected = TRUE;
|
|
g_mutex_unlock (&sink->preroll_lock);
|
|
|
|
return TRUE;
|
|
|
|
join_bin_failed:
|
|
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
|
|
("Could not start stream %d", context->index));
|
|
return FALSE;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_create_transports_string (GstRTSPClientSink * sink,
|
|
GstRTSPStreamContext * context, GSocketFamily family,
|
|
GstRTSPLowerTrans protocols, GstRTSPProfile profiles, gchar ** transports)
|
|
{
|
|
GString *result;
|
|
GstRTSPStream *stream = context->stream;
|
|
gboolean first = TRUE;
|
|
|
|
/* the default RTSP transports */
|
|
result = g_string_new ("RTP");
|
|
|
|
while (profiles != 0) {
|
|
if (!first)
|
|
g_string_append (result, ",RTP");
|
|
|
|
if (profiles & GST_RTSP_PROFILE_SAVPF) {
|
|
g_string_append (result, "/SAVPF");
|
|
profiles &= ~GST_RTSP_PROFILE_SAVPF;
|
|
} else if (profiles & GST_RTSP_PROFILE_SAVP) {
|
|
g_string_append (result, "/SAVP");
|
|
profiles &= ~GST_RTSP_PROFILE_SAVP;
|
|
} else if (profiles & GST_RTSP_PROFILE_AVPF) {
|
|
g_string_append (result, "/AVPF");
|
|
profiles &= ~GST_RTSP_PROFILE_AVPF;
|
|
} else if (profiles & GST_RTSP_PROFILE_AVP) {
|
|
g_string_append (result, "/AVP");
|
|
profiles &= ~GST_RTSP_PROFILE_AVP;
|
|
} else {
|
|
GST_WARNING_OBJECT (sink, "Unimplemented profile(s) 0x%x", profiles);
|
|
break;
|
|
}
|
|
|
|
if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
|
|
GstRTSPRange ports;
|
|
|
|
GST_DEBUG_OBJECT (sink, "adding UDP unicast");
|
|
gst_rtsp_stream_get_server_port (stream, &ports, family);
|
|
|
|
g_string_append_printf (result, "/UDP;unicast;client_port=%d-%d",
|
|
ports.min, ports.max);
|
|
} else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
|
|
GstRTSPAddress *addr =
|
|
gst_rtsp_stream_get_multicast_address (stream, family);
|
|
if (addr) {
|
|
GST_DEBUG_OBJECT (sink, "adding UDP multicast");
|
|
g_string_append_printf (result, "/UDP;multicast;client_port=%d-%d",
|
|
addr->port, addr->port + addr->n_ports - 1);
|
|
gst_rtsp_address_free (addr);
|
|
}
|
|
} else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
|
|
GST_DEBUG_OBJECT (sink, "adding TCP");
|
|
g_string_append_printf (result, "/TCP;unicast;interleaved=%d-%d",
|
|
sink->free_channel, sink->free_channel + 1);
|
|
}
|
|
|
|
g_string_append (result, ";mode=RECORD");
|
|
/* FIXME: Support appending too:
|
|
if (sink->append)
|
|
g_string_append (result, ";append");
|
|
*/
|
|
|
|
first = FALSE;
|
|
}
|
|
|
|
if (first) {
|
|
/* No valid transport could be constructed */
|
|
GST_ERROR_OBJECT (sink, "No supported profiles configured");
|
|
goto fail;
|
|
}
|
|
|
|
*transports = g_string_free (result, FALSE);
|
|
|
|
GST_DEBUG_OBJECT (sink, "prepared transports %s", GST_STR_NULL (*transports));
|
|
|
|
return GST_RTSP_OK;
|
|
fail:
|
|
g_string_free (result, TRUE);
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
|
|
static guint8
|
|
enc_key_length_from_cipher_name (const gchar * cipher)
|
|
{
|
|
if (g_strcmp0 (cipher, "aes-128-icm") == 0)
|
|
return AES_128_KEY_LEN;
|
|
else if (g_strcmp0 (cipher, "aes-256-icm") == 0)
|
|
return AES_256_KEY_LEN;
|
|
else {
|
|
GST_ERROR ("encryption algorithm '%s' not supported", cipher);
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
static guint8
|
|
auth_key_length_from_auth_name (const gchar * auth)
|
|
{
|
|
if (g_strcmp0 (auth, "hmac-sha1-32") == 0)
|
|
return HMAC_32_KEY_LEN;
|
|
else if (g_strcmp0 (auth, "hmac-sha1-80") == 0)
|
|
return HMAC_80_KEY_LEN;
|
|
else {
|
|
GST_ERROR ("authentication algorithm '%s' not supported", auth);
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
static GstCaps *
|
|
signal_get_srtcp_params (GstRTSPClientSink * sink,
|
|
GstRTSPStreamContext * context)
|
|
{
|
|
GstCaps *caps = NULL;
|
|
|
|
g_signal_emit (sink, gst_rtsp_client_sink_signals[SIGNAL_REQUEST_RTCP_KEY], 0,
|
|
context->index, &caps);
|
|
|
|
if (caps != NULL)
|
|
GST_DEBUG_OBJECT (sink, "SRTP parameters received");
|
|
|
|
return caps;
|
|
}
|
|
|
|
static gchar *
|
|
gst_rtsp_client_sink_stream_make_keymgmt (GstRTSPClientSink * sink,
|
|
GstRTSPStreamContext * context)
|
|
{
|
|
GBytes *bytes;
|
|
gchar *result, *base64;
|
|
const guint8 *data;
|
|
gsize size;
|
|
GstMIKEYMessage *msg;
|
|
GstMIKEYPayload *payload, *pkd;
|
|
guint8 byte;
|
|
GstStructure *s;
|
|
GstMapInfo info;
|
|
GstBuffer *srtpkey;
|
|
const GValue *val;
|
|
const gchar *srtcpcipher, *srtcpauth;
|
|
guint send_ssrc;
|
|
|
|
context->srtcpparams = signal_get_srtcp_params (sink, context);
|
|
if (context->srtcpparams == NULL)
|
|
context->srtcpparams = gst_rtsp_stream_get_caps (context->stream);
|
|
|
|
s = gst_caps_get_structure (context->srtcpparams, 0);
|
|
|
|
srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
|
|
srtcpauth = gst_structure_get_string (s, "srtcp-auth");
|
|
val = gst_structure_get_value (s, "srtp-key");
|
|
|
|
if (srtcpcipher == NULL || srtcpauth == NULL || val == NULL) {
|
|
GST_ERROR_OBJECT (sink, "could not find the right SRTP parameters in caps");
|
|
return NULL;
|
|
}
|
|
|
|
srtpkey = gst_value_get_buffer (val);
|
|
|
|
gst_rtsp_stream_get_ssrc (context->stream, &send_ssrc);
|
|
GST_LOG_OBJECT (sink, "Stream %p ssrc %x", context->stream, send_ssrc);
|
|
|
|
msg = gst_mikey_message_new ();
|
|
/* unencrypted MIKEY message, we send this over TLS so this is allowed */
|
|
gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
|
|
FALSE, GST_MIKEY_PRF_MIKEY_1, g_random_int (), GST_MIKEY_MAP_TYPE_SRTP);
|
|
/* add policy '0' for our SSRC */
|
|
gst_mikey_message_add_cs_srtp (msg, 0, send_ssrc, 0);
|
|
/* timestamp is now */
|
|
gst_mikey_message_add_t_now_ntp_utc (msg);
|
|
/* add some random data */
|
|
gst_mikey_message_add_rand_len (msg, 16);
|
|
|
|
/* the policy '0' is SRTP */
|
|
payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
|
|
gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
|
|
|
|
/* only AES-CM is supported */
|
|
byte = 1;
|
|
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1, &byte);
|
|
/* encryption key length */
|
|
byte = enc_key_length_from_cipher_name (srtcpcipher);
|
|
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_KEY_LEN, 1,
|
|
&byte);
|
|
/* only HMAC-SHA1 */
|
|
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
|
|
&byte);
|
|
/* authentication key length */
|
|
byte = auth_key_length_from_auth_name (srtcpauth);
|
|
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_KEY_LEN, 1,
|
|
&byte);
|
|
/* we enable encryption on RTP and RTCP */
|
|
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
|
|
&byte);
|
|
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
|
|
&byte);
|
|
/* we enable authentication on RTP and RTCP */
|
|
gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
|
|
&byte);
|
|
gst_mikey_message_add_payload (msg, payload);
|
|
|
|
/* make unencrypted KEMAC */
|
|
payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC);
|
|
gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL, GST_MIKEY_MAC_NULL);
|
|
/* add the key in KEMAC */
|
|
pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA);
|
|
gst_buffer_map (srtpkey, &info, GST_MAP_READ);
|
|
gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size,
|
|
info.data);
|
|
gst_buffer_unmap (srtpkey, &info);
|
|
gst_mikey_payload_kemac_add_sub (payload, pkd);
|
|
gst_mikey_message_add_payload (msg, payload);
|
|
|
|
/* now serialize this to bytes */
|
|
bytes = gst_mikey_message_to_bytes (msg, NULL, NULL);
|
|
gst_mikey_message_unref (msg);
|
|
/* and make it into base64 */
|
|
data = g_bytes_get_data (bytes, &size);
|
|
base64 = g_base64_encode (data, size);
|
|
g_bytes_unref (bytes);
|
|
|
|
result = g_strdup_printf ("prot=mikey;uri=\"%s\";data=\"%s\"",
|
|
context->conninfo.location, base64);
|
|
g_free (base64);
|
|
|
|
return result;
|
|
}
|
|
|
|
/* masks to be kept in sync with the hardcoded protocol order of preference
|
|
* in code below */
|
|
static const guint protocol_masks[] = {
|
|
GST_RTSP_LOWER_TRANS_UDP,
|
|
GST_RTSP_LOWER_TRANS_UDP_MCAST,
|
|
GST_RTSP_LOWER_TRANS_TCP,
|
|
0
|
|
};
|
|
|
|
/* Same for profile_masks */
|
|
static const guint profile_masks[] = {
|
|
GST_RTSP_PROFILE_SAVPF,
|
|
GST_RTSP_PROFILE_SAVP,
|
|
GST_RTSP_PROFILE_AVPF,
|
|
GST_RTSP_PROFILE_AVP,
|
|
0
|
|
};
|
|
|
|
static gboolean
|
|
do_send_data (GstBuffer * buffer, guint8 channel,
|
|
GstRTSPStreamContext * context)
|
|
{
|
|
GstRTSPClientSink *sink = context->parent;
|
|
GstRTSPMessage message = { 0 };
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
GstMapInfo map_info;
|
|
guint8 *data;
|
|
guint usize;
|
|
|
|
gst_rtsp_message_init_data (&message, channel);
|
|
|
|
/* FIXME, need some sort of iovec RTSPMessage here */
|
|
if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
|
|
return FALSE;
|
|
|
|
gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
|
|
|
|
res =
|
|
gst_rtsp_client_sink_try_send (sink, sink->conninfo.connection, &message,
|
|
NULL, NULL);
|
|
|
|
gst_rtsp_message_steal_body (&message, &data, &usize);
|
|
gst_buffer_unmap (buffer, &map_info);
|
|
|
|
gst_rtsp_message_unset (&message);
|
|
|
|
return res == GST_RTSP_OK;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_setup_streams (GstRTSPClientSink * sink, gboolean async)
|
|
{
|
|
GstRTSPResult res = GST_RTSP_ERROR;
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPMessage response = { 0 };
|
|
GstRTSPLowerTrans protocols;
|
|
GstRTSPStatusCode code;
|
|
GSocketFamily family;
|
|
GSocketAddress *sa;
|
|
GSocket *conn_socket;
|
|
GstRTSPUrl *url;
|
|
GList *walk;
|
|
gchar *hval;
|
|
|
|
if (sink->conninfo.connection) {
|
|
url = gst_rtsp_connection_get_url (sink->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 & sink->cur_protocols;
|
|
} else {
|
|
url = NULL;
|
|
protocols = sink->cur_protocols;
|
|
}
|
|
|
|
if (protocols == 0)
|
|
goto no_protocols;
|
|
|
|
GST_RTSP_STATE_LOCK (sink);
|
|
|
|
if (G_UNLIKELY (sink->contexts == NULL))
|
|
goto no_streams;
|
|
|
|
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
|
|
GstRTSPStream *stream;
|
|
|
|
GstRTSPConnection *conn;
|
|
GstRTSPProfile profiles;
|
|
GstRTSPProfile cur_profile;
|
|
gchar *transports;
|
|
gint retry = 0;
|
|
guint profile_mask = 0;
|
|
guint mask = 0;
|
|
GstCaps *caps;
|
|
const GstSDPMedia *media;
|
|
|
|
stream = context->stream;
|
|
profiles = gst_rtsp_stream_get_profiles (stream);
|
|
|
|
caps = gst_rtsp_stream_get_caps (stream);
|
|
if (caps == NULL) {
|
|
GST_DEBUG_OBJECT (sink, "skipping stream %p, no caps", stream);
|
|
continue;
|
|
}
|
|
gst_caps_unref (caps);
|
|
media = gst_sdp_message_get_media (&sink->cursdp, context->sdp_index);
|
|
if (media == NULL) {
|
|
GST_DEBUG_OBJECT (sink, "skipping stream %p, no SDP info", stream);
|
|
continue;
|
|
}
|
|
|
|
/* skip setup if we have no URL for it */
|
|
if (context->conninfo.location == NULL) {
|
|
GST_DEBUG_OBJECT (sink, "skipping stream %p, no setup", stream);
|
|
continue;
|
|
}
|
|
|
|
if (sink->conninfo.connection == NULL) {
|
|
if (!gst_rtsp_conninfo_connect (sink, &context->conninfo, async)) {
|
|
GST_DEBUG_OBJECT (sink, "skipping stream %p, failed to connect",
|
|
stream);
|
|
continue;
|
|
}
|
|
conn = context->conninfo.connection;
|
|
} else {
|
|
conn = sink->conninfo.connection;
|
|
}
|
|
GST_DEBUG_OBJECT (sink, "doing setup of stream %p with %s", stream,
|
|
context->conninfo.location);
|
|
|
|
conn_socket = gst_rtsp_connection_get_read_socket (conn);
|
|
sa = g_socket_get_local_address (conn_socket, NULL);
|
|
family = g_socket_address_get_family (sa);
|
|
g_object_unref (sa);
|
|
|
|
next_protocol:
|
|
/* first selectable profile */
|
|
while (profile_masks[profile_mask]
|
|
&& !(profiles & profile_masks[profile_mask]))
|
|
profile_mask++;
|
|
if (!profile_masks[profile_mask])
|
|
goto no_profiles;
|
|
|
|
/* first selectable protocol */
|
|
while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
|
|
mask++;
|
|
if (!protocol_masks[mask])
|
|
goto no_protocols;
|
|
|
|
retry:
|
|
GST_DEBUG_OBJECT (sink, "protocols = 0x%x, protocol mask = 0x%x", protocols,
|
|
protocol_masks[mask]);
|
|
/* create a string with first transport in line */
|
|
transports = NULL;
|
|
cur_profile = profiles & profile_masks[profile_mask];
|
|
res = gst_rtsp_client_sink_create_transports_string (sink, context, family,
|
|
protocols & protocol_masks[mask], cur_profile, &transports);
|
|
if (res < 0 || transports == NULL)
|
|
goto setup_transport_failed;
|
|
|
|
if (strlen (transports) == 0) {
|
|
g_free (transports);
|
|
GST_DEBUG_OBJECT (sink, "no transports found");
|
|
mask++;
|
|
profile_mask = 0;
|
|
goto next_protocol;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (sink, "transport is %s", GST_STR_NULL (transports));
|
|
|
|
/* create SETUP request */
|
|
res =
|
|
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_SETUP,
|
|
context->conninfo.location);
|
|
if (res < 0) {
|
|
g_free (transports);
|
|
goto create_request_failed;
|
|
}
|
|
|
|
/* select transport */
|
|
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
|
|
|
|
/* set up keys */
|
|
if (cur_profile == GST_RTSP_PROFILE_SAVP ||
|
|
cur_profile == GST_RTSP_PROFILE_SAVPF) {
|
|
hval = gst_rtsp_client_sink_stream_make_keymgmt (sink, context);
|
|
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_KEYMGMT, hval);
|
|
}
|
|
|
|
/* if the user wants a non default RTP packet size we add the blocksize
|
|
* parameter */
|
|
if (sink->rtp_blocksize > 0) {
|
|
hval = g_strdup_printf ("%d", sink->rtp_blocksize);
|
|
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
|
|
}
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (sink, CONTINUE, "request", ("SETUP stream %d",
|
|
context->index));
|
|
|
|
/* handle the code ourselves */
|
|
res = gst_rtsp_client_sink_send (sink, conn, &request, &response, &code);
|
|
if (res < 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);
|
|
|
|
/* Try another profile. If no more, move to the next protocol */
|
|
profile_mask++;
|
|
while (profile_masks[profile_mask]
|
|
&& !(profiles & profile_masks[profile_mask]))
|
|
profile_mask++;
|
|
if (profile_masks[profile_mask])
|
|
goto retry;
|
|
|
|
/* select next available protocol, give up on this stream if none */
|
|
/* Reset profiles to try: */
|
|
profile_mask = 0;
|
|
|
|
mask++;
|
|
while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
|
|
mask++;
|
|
if (!protocol_masks[mask])
|
|
continue;
|
|
else
|
|
goto retry;
|
|
default:
|
|
goto response_error;
|
|
}
|
|
|
|
/* parse response transport */
|
|
{
|
|
gchar *resptrans = NULL;
|
|
GstRTSPTransport *transport;
|
|
|
|
gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
|
|
&resptrans, 0);
|
|
if (!resptrans) {
|
|
goto no_transport;
|
|
}
|
|
|
|
gst_rtsp_transport_new (&transport);
|
|
|
|
/* parse transport, go to next stream on parse error */
|
|
if (gst_rtsp_transport_parse (resptrans, transport) != GST_RTSP_OK) {
|
|
GST_WARNING_OBJECT (sink, "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 (sink, "stream %p as TCP interleaved", stream);
|
|
protocols = GST_RTSP_LOWER_TRANS_TCP;
|
|
sink->interleaved = TRUE;
|
|
/* update free channels */
|
|
sink->free_channel =
|
|
MAX (transport->interleaved.min, sink->free_channel);
|
|
sink->free_channel =
|
|
MAX (transport->interleaved.max, sink->free_channel);
|
|
sink->free_channel++;
|
|
break;
|
|
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
|
|
/* only allow multicast for other streams */
|
|
GST_DEBUG_OBJECT (sink, "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 (sink, "stream %p as UDP unicast", stream);
|
|
protocols = GST_RTSP_LOWER_TRANS_UDP;
|
|
/* Update transport with server destination if not provided by the server */
|
|
if (transport->destination == NULL) {
|
|
transport->destination = g_strdup (sink->server_ip);
|
|
}
|
|
break;
|
|
default:
|
|
GST_DEBUG_OBJECT (sink, "stream %p unknown transport %d", stream,
|
|
transport->lower_transport);
|
|
break;
|
|
}
|
|
|
|
if (!retry) {
|
|
GST_DEBUG ("Configuring the stream transport for stream %d",
|
|
context->index);
|
|
if (context->stream_transport == NULL)
|
|
context->stream_transport =
|
|
gst_rtsp_stream_transport_new (stream, transport);
|
|
else
|
|
gst_rtsp_stream_transport_set_transport (context->stream_transport,
|
|
transport);
|
|
|
|
if (transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
|
|
/* our callbacks to send data on this TCP connection */
|
|
gst_rtsp_stream_transport_set_callbacks (context->stream_transport,
|
|
(GstRTSPSendFunc) do_send_data,
|
|
(GstRTSPSendFunc) do_send_data, context, NULL);
|
|
}
|
|
|
|
/* The stream_transport now owns the transport */
|
|
transport = NULL;
|
|
|
|
gst_rtsp_stream_transport_set_active (context->stream_transport, TRUE);
|
|
}
|
|
next:
|
|
if (transport)
|
|
gst_rtsp_transport_free (transport);
|
|
/* clean up used RTSP messages */
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
}
|
|
}
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
|
|
/* store the transport protocol that was configured */
|
|
sink->cur_protocols = protocols;
|
|
|
|
return res;
|
|
|
|
no_streams:
|
|
{
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
|
|
("SDP contains no streams"));
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
setup_transport_failed:
|
|
{
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
|
|
("Could not setup transport."));
|
|
res = GST_RTSP_ERROR;
|
|
goto cleanup_error;
|
|
}
|
|
no_profiles:
|
|
{
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
/* no transport possible, post an error and stop */
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
|
|
("Could not connect to server, no profiles left"));
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
no_protocols:
|
|
{
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
/* no transport possible, post an error and stop */
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, READ, (NULL),
|
|
("Could not connect to server, no protocols left"));
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
no_transport:
|
|
{
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
|
|
("Server did not select transport."));
|
|
res = GST_RTSP_ERROR;
|
|
goto cleanup_error;
|
|
}
|
|
create_request_failed:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
GST_ELEMENT_ERROR (sink, LIBRARY, INIT, (NULL),
|
|
("Could not create request. (%s)", str));
|
|
g_free (str);
|
|
goto cleanup_error;
|
|
}
|
|
send_error:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
if (res != GST_RTSP_EINTR) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (sink, "send interrupted");
|
|
}
|
|
g_free (str);
|
|
goto cleanup_error;
|
|
}
|
|
response_error:
|
|
{
|
|
const gchar *str = gst_rtsp_status_as_text (code);
|
|
|
|
GST_RTSP_STATE_UNLOCK (sink);
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
|
|
("Error (%d): %s", code, GST_STR_NULL (str)));
|
|
res = GST_RTSP_ERROR;
|
|
goto cleanup_error;
|
|
}
|
|
cleanup_error:
|
|
{
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_ensure_open (GstRTSPClientSink * sink, gboolean async)
|
|
{
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
|
|
if (sink->state < GST_RTSP_STATE_READY) {
|
|
res = GST_RTSP_ERROR;
|
|
if (sink->open_error) {
|
|
GST_DEBUG_OBJECT (sink, "the stream was in error");
|
|
goto done;
|
|
}
|
|
if (async)
|
|
gst_rtsp_client_sink_loop_start_cmd (sink, CMD_OPEN);
|
|
|
|
if ((res = gst_rtsp_client_sink_open (sink, async)) < 0) {
|
|
GST_DEBUG_OBJECT (sink, "failed to open stream");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
done:
|
|
return res;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_record (GstRTSPClientSink * sink, gboolean async)
|
|
{
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPMessage response = { 0 };
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
GstSDPMessage *sdp;
|
|
guint sdp_index = 0;
|
|
GstSDPInfo info = { 0, };
|
|
|
|
const gchar *proto;
|
|
gchar *sess_id, *client_ip, *str;
|
|
GSocketAddress *sa;
|
|
GInetAddress *ia;
|
|
GSocket *conn_socket;
|
|
GList *walk;
|
|
|
|
/* Wait for streams to preroll */
|
|
g_mutex_lock (&sink->preroll_lock);
|
|
while (sink->in_async) {
|
|
GST_LOG_OBJECT (sink, "Waiting for ASYNC_DONE preroll");
|
|
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
|
|
}
|
|
g_mutex_unlock (&sink->preroll_lock);
|
|
|
|
if (sink->state == GST_RTSP_STATE_PLAYING) {
|
|
/* Already recording, don't send another request */
|
|
GST_LOG_OBJECT (sink, "Already in RECORD. Skipping duplicate request.");
|
|
goto done;
|
|
}
|
|
|
|
/* Send announce, then setup for all streams */
|
|
gst_sdp_message_init (&sink->cursdp);
|
|
sdp = &sink->cursdp;
|
|
|
|
/* some standard things first */
|
|
gst_sdp_message_set_version (sdp, "0");
|
|
|
|
/* session ID doesn't have to be super-unique in this case */
|
|
sess_id = g_strdup_printf ("%u", g_random_int ());
|
|
|
|
if (sink->conninfo.connection == NULL)
|
|
return GST_RTSP_ERROR;
|
|
|
|
conn_socket = gst_rtsp_connection_get_read_socket (sink->conninfo.connection);
|
|
|
|
sa = g_socket_get_local_address (conn_socket, NULL);
|
|
ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
|
|
client_ip = g_inet_address_to_string (ia);
|
|
if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV6) {
|
|
info.is_ipv6 = TRUE;
|
|
proto = "IP6";
|
|
} else if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV4)
|
|
proto = "IP4";
|
|
else
|
|
g_assert_not_reached ();
|
|
g_object_unref (sa);
|
|
|
|
/* FIXME: Should this actually be the server's IP or ours? */
|
|
info.server_ip = sink->server_ip;
|
|
|
|
gst_sdp_message_set_origin (sdp, "-", sess_id, "1", "IN", proto, client_ip);
|
|
|
|
gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
|
|
gst_sdp_message_set_information (sdp, "rtspclientsink");
|
|
gst_sdp_message_add_time (sdp, "0", "0", NULL);
|
|
gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
|
|
|
|
/* add stream */
|
|
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamContext *context = (GstRTSPStreamContext *) walk->data;
|
|
|
|
gst_rtsp_sdp_from_stream (sdp, &info, context->stream);
|
|
context->sdp_index = sdp_index++;
|
|
}
|
|
|
|
g_free (sess_id);
|
|
g_free (client_ip);
|
|
|
|
/* send ANNOUNCE request */
|
|
GST_DEBUG_OBJECT (sink, "create ANNOUNCE request...");
|
|
res =
|
|
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_ANNOUNCE,
|
|
sink->conninfo.url_str);
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CONTENT_TYPE,
|
|
"application/sdp");
|
|
|
|
/* add SDP to the request body */
|
|
str = gst_sdp_message_as_text (sdp);
|
|
gst_rtsp_message_take_body (&request, (guint8 *) str, strlen (str));
|
|
|
|
/* send ANNOUNCE */
|
|
GST_DEBUG_OBJECT (sink, "sending announce...");
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (sink, CONTINUE, "record",
|
|
("Sending server stream info"));
|
|
|
|
if ((res =
|
|
gst_rtsp_client_sink_send (sink, sink->conninfo.connection, &request,
|
|
&response, NULL)) < 0)
|
|
goto send_error;
|
|
|
|
/* send setup for all streams */
|
|
if ((res = gst_rtsp_client_sink_setup_streams (sink, async)) < 0)
|
|
goto setup_failed;
|
|
|
|
res = gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_RECORD,
|
|
sink->conninfo.url_str);
|
|
|
|
if (res < 0)
|
|
goto create_request_failed;
|
|
|
|
#if 0 /* FIXME: Configure a range based on input segments? */
|
|
if (src->need_range) {
|
|
hval = gen_range_header (src, segment);
|
|
|
|
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, 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);
|
|
}
|
|
#endif
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (sink, CONTINUE, "record", ("Starting recording"));
|
|
if ((res =
|
|
gst_rtsp_client_sink_send (sink, sink->conninfo.connection, &request,
|
|
&response, NULL)) < 0)
|
|
goto send_error;
|
|
|
|
#if 0 /* FIXME: Check if servers return these for record: */
|
|
/* 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);
|
|
#endif
|
|
|
|
gst_rtsp_client_sink_set_state (sink, GST_STATE_PLAYING);
|
|
sink->state = GST_RTSP_STATE_PLAYING;
|
|
|
|
/* clean up any messages */
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
|
|
done:
|
|
return res;
|
|
|
|
create_request_failed:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (sink, 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;
|
|
}
|
|
setup_failed:
|
|
{
|
|
GST_ERROR_OBJECT (sink, "setup failed");
|
|
goto cleanup_error;
|
|
}
|
|
cleanup_error:
|
|
{
|
|
if (sink->conninfo.connection) {
|
|
GST_DEBUG_OBJECT (sink, "free connection");
|
|
gst_rtsp_conninfo_close (sink, &sink->conninfo, TRUE);
|
|
}
|
|
gst_rtsp_message_unset (&request);
|
|
gst_rtsp_message_unset (&response);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
gst_rtsp_client_sink_pause (GstRTSPClientSink * sink, gboolean async)
|
|
{
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPMessage response = { 0 };
|
|
GList *walk;
|
|
const gchar *control;
|
|
|
|
GST_DEBUG_OBJECT (sink, "PAUSE...");
|
|
|
|
if ((res = gst_rtsp_client_sink_ensure_open (sink, async)) < 0)
|
|
goto open_failed;
|
|
|
|
if (!(sink->methods & GST_RTSP_PAUSE))
|
|
goto not_supported;
|
|
|
|
if (sink->state == GST_RTSP_STATE_READY)
|
|
goto was_paused;
|
|
|
|
if (!sink->conninfo.connection || !sink->conninfo.connected)
|
|
goto no_connection;
|
|
|
|
/* construct a control url */
|
|
control = get_aggregate_control (sink);
|
|
|
|
/* loop over the streams. We might exit the loop early when we could do an
|
|
* aggregate control */
|
|
for (walk = sink->contexts; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamContext *stream = (GstRTSPStreamContext *) walk->data;
|
|
GstRTSPConnection *conn;
|
|
const 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 (sink->conninfo.connection) {
|
|
conn = sink->conninfo.connection;
|
|
} else if (stream->conninfo.connection) {
|
|
conn = stream->conninfo.connection;
|
|
} else {
|
|
continue;
|
|
}
|
|
|
|
if (async)
|
|
GST_ELEMENT_PROGRESS (sink, CONTINUE, "request",
|
|
("Sending PAUSE request"));
|
|
|
|
if ((res =
|
|
gst_rtsp_client_sink_init_request (sink, &request, GST_RTSP_PAUSE,
|
|
setup_url)) < 0)
|
|
goto create_request_failed;
|
|
|
|
if ((res =
|
|
gst_rtsp_client_sink_send (sink, 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;
|
|
}
|
|
|
|
/* change element states now */
|
|
gst_rtsp_client_sink_set_state (sink, GST_STATE_PAUSED);
|
|
|
|
no_connection:
|
|
sink->state = GST_RTSP_STATE_READY;
|
|
|
|
done:
|
|
if (async)
|
|
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_PAUSE, res);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
open_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "failed to open stream");
|
|
goto done;
|
|
}
|
|
not_supported:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "PAUSE is not supported");
|
|
goto done;
|
|
}
|
|
was_paused:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "we were already PAUSED");
|
|
goto done;
|
|
}
|
|
create_request_failed:
|
|
{
|
|
gchar *str = gst_rtsp_strresult (res);
|
|
|
|
GST_ELEMENT_ERROR (sink, 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 (sink, RESOURCE, WRITE, (NULL),
|
|
("Could not send message. (%s)", str));
|
|
} else {
|
|
GST_WARNING_OBJECT (sink, "PAUSE interrupted");
|
|
}
|
|
g_free (str);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_handle_message (GstBin * bin, GstMessage * message)
|
|
{
|
|
GstRTSPClientSink *rtsp_client_sink;
|
|
|
|
rtsp_client_sink = GST_RTSP_CLIENT_SINK (bin);
|
|
|
|
switch (GST_MESSAGE_TYPE (message)) {
|
|
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 (rtsp_client_sink);
|
|
ignore_timeout = rtsp_client_sink->ignore_timeout;
|
|
rtsp_client_sink->ignore_timeout = TRUE;
|
|
GST_OBJECT_UNLOCK (rtsp_client_sink);
|
|
|
|
/* we only act on the first udp timeout message, others are irrelevant
|
|
* and can be ignored. */
|
|
if (!ignore_timeout)
|
|
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_RECONNECT,
|
|
CMD_LOOP);
|
|
/* eat and free */
|
|
gst_message_unref (message);
|
|
return;
|
|
} else if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
|
|
/* An RTSPStream has prerolled */
|
|
g_cond_broadcast (&rtsp_client_sink->preroll_cond);
|
|
}
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
case GST_MESSAGE_ASYNC_START:{
|
|
GstObject *sender;
|
|
|
|
sender = GST_MESSAGE_SRC (message);
|
|
|
|
GST_LOG_OBJECT (rtsp_client_sink,
|
|
"Have async-start from %" GST_PTR_FORMAT, sender);
|
|
if (sender == GST_OBJECT (rtsp_client_sink->internal_bin)) {
|
|
GST_LOG_OBJECT (rtsp_client_sink, "child bin is now ASYNC");
|
|
}
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
case GST_MESSAGE_ASYNC_DONE:
|
|
{
|
|
GstObject *sender;
|
|
gboolean need_async_done;
|
|
|
|
sender = GST_MESSAGE_SRC (message);
|
|
GST_LOG_OBJECT (rtsp_client_sink, "Have async-done from %" GST_PTR_FORMAT,
|
|
sender);
|
|
|
|
g_mutex_lock (&rtsp_client_sink->preroll_lock);
|
|
if (sender == GST_OBJECT_CAST (rtsp_client_sink->internal_bin)) {
|
|
GST_LOG_OBJECT (rtsp_client_sink, "child bin is no longer ASYNC");
|
|
}
|
|
need_async_done = rtsp_client_sink->in_async;
|
|
if (rtsp_client_sink->in_async) {
|
|
rtsp_client_sink->in_async = FALSE;
|
|
g_cond_broadcast (&rtsp_client_sink->preroll_cond);
|
|
}
|
|
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
|
|
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
|
|
if (need_async_done) {
|
|
GST_DEBUG_OBJECT (rtsp_client_sink, "Posting ASYNC-DONE");
|
|
gst_element_post_message (GST_ELEMENT_CAST (rtsp_client_sink),
|
|
gst_message_new_async_done (GST_OBJECT_CAST (rtsp_client_sink),
|
|
GST_CLOCK_TIME_NONE));
|
|
}
|
|
break;
|
|
}
|
|
case GST_MESSAGE_ERROR:
|
|
{
|
|
GstObject *sender;
|
|
|
|
sender = GST_MESSAGE_SRC (message);
|
|
|
|
GST_DEBUG_OBJECT (rtsp_client_sink, "got error from %s",
|
|
GST_ELEMENT_NAME (sender));
|
|
|
|
/* FIXME: Ignore errors on RTCP? */
|
|
/* fatal but not our message, forward */
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
case GST_MESSAGE_STATE_CHANGED:
|
|
{
|
|
if (GST_MESSAGE_SRC (message) ==
|
|
(GstObject *) rtsp_client_sink->internal_bin) {
|
|
GstState newstate, pending;
|
|
gst_message_parse_state_changed (message, NULL, &newstate, &pending);
|
|
g_mutex_lock (&rtsp_client_sink->preroll_lock);
|
|
rtsp_client_sink->prerolled = (newstate >= GST_STATE_PAUSED)
|
|
&& pending == GST_STATE_VOID_PENDING;
|
|
g_cond_broadcast (&rtsp_client_sink->preroll_cond);
|
|
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
|
|
GST_DEBUG_OBJECT (bin,
|
|
"Internal bin changed state to %s (pending %s). Prerolled now %d",
|
|
gst_element_state_get_name (newstate),
|
|
gst_element_state_get_name (pending), rtsp_client_sink->prerolled);
|
|
}
|
|
}
|
|
default:
|
|
{
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* the thread where everything happens */
|
|
static void
|
|
gst_rtsp_client_sink_thread (GstRTSPClientSink * sink)
|
|
{
|
|
gint cmd;
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
cmd = sink->pending_cmd;
|
|
if (cmd == CMD_RECONNECT || cmd == CMD_RECORD || cmd == CMD_PAUSE
|
|
|| cmd == CMD_LOOP || cmd == CMD_OPEN)
|
|
sink->pending_cmd = CMD_LOOP;
|
|
else
|
|
sink->pending_cmd = CMD_WAIT;
|
|
GST_DEBUG_OBJECT (sink, "got command %s", cmd_to_string (cmd));
|
|
|
|
/* we got the message command, so ensure communication is possible again */
|
|
gst_rtsp_client_sink_connection_flush (sink, FALSE);
|
|
|
|
sink->busy_cmd = cmd;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
switch (cmd) {
|
|
case CMD_OPEN:
|
|
gst_rtsp_client_sink_open (sink, TRUE);
|
|
break;
|
|
case CMD_RECORD:
|
|
gst_rtsp_client_sink_record (sink, TRUE);
|
|
break;
|
|
case CMD_PAUSE:
|
|
gst_rtsp_client_sink_pause (sink, TRUE);
|
|
break;
|
|
case CMD_CLOSE:
|
|
gst_rtsp_client_sink_close (sink, TRUE, FALSE);
|
|
break;
|
|
case CMD_LOOP:
|
|
gst_rtsp_client_sink_loop (sink);
|
|
break;
|
|
case CMD_RECONNECT:
|
|
gst_rtsp_client_sink_reconnect (sink, FALSE);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
/* and go back to sleep */
|
|
if (sink->pending_cmd == CMD_WAIT) {
|
|
if (sink->task)
|
|
gst_task_pause (sink->task);
|
|
}
|
|
/* reset waiting */
|
|
sink->busy_cmd = CMD_WAIT;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_client_sink_start (GstRTSPClientSink * sink)
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "starting");
|
|
|
|
sink->streams_collected = FALSE;
|
|
sink->in_async = TRUE;
|
|
gst_element_set_locked_state (GST_ELEMENT (sink->internal_bin), TRUE);
|
|
|
|
gst_rtsp_client_sink_set_state (sink, GST_STATE_READY);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->pending_cmd = CMD_WAIT;
|
|
|
|
if (sink->task == NULL) {
|
|
sink->task =
|
|
gst_task_new ((GstTaskFunction) gst_rtsp_client_sink_thread, sink,
|
|
NULL);
|
|
if (sink->task == NULL)
|
|
goto task_error;
|
|
|
|
gst_task_set_lock (sink->task, GST_RTSP_STREAM_GET_LOCK (sink));
|
|
}
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
task_error:
|
|
{
|
|
GST_OBJECT_UNLOCK (sink);
|
|
GST_ERROR_OBJECT (sink, "failed to create task");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_client_sink_stop (GstRTSPClientSink * sink)
|
|
{
|
|
GstTask *task;
|
|
|
|
GST_DEBUG_OBJECT (sink, "stopping");
|
|
|
|
/* also cancels pending task */
|
|
gst_rtsp_client_sink_loop_send_cmd (sink, CMD_WAIT, CMD_ALL & ~CMD_CLOSE);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
if ((task = sink->task)) {
|
|
sink->task = NULL;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
gst_task_stop (task);
|
|
|
|
/* make sure it is not running */
|
|
GST_RTSP_STREAM_LOCK (sink);
|
|
GST_RTSP_STREAM_UNLOCK (sink);
|
|
|
|
/* now wait for the task to finish */
|
|
gst_task_join (task);
|
|
|
|
/* and free the task */
|
|
gst_object_unref (GST_OBJECT (task));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
}
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
/* ensure synchronously all is closed and clean */
|
|
gst_rtsp_client_sink_close (sink, FALSE, TRUE);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtsp_client_sink_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstRTSPClientSink *rtsp_client_sink;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtsp_client_sink = GST_RTSP_CLIENT_SINK (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
if (!gst_rtsp_client_sink_start (rtsp_client_sink))
|
|
goto start_failed;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
/* init some state */
|
|
rtsp_client_sink->cur_protocols = rtsp_client_sink->protocols;
|
|
/* first attempt, don't ignore timeouts */
|
|
rtsp_client_sink->ignore_timeout = FALSE;
|
|
rtsp_client_sink->open_error = FALSE;
|
|
|
|
gst_rtsp_client_sink_set_state (rtsp_client_sink, GST_STATE_PAUSED);
|
|
|
|
g_mutex_lock (&rtsp_client_sink->preroll_lock);
|
|
if (rtsp_client_sink->in_async) {
|
|
GST_DEBUG_OBJECT (rtsp_client_sink, "Posting ASYNC-START");
|
|
gst_element_post_message (GST_ELEMENT_CAST (rtsp_client_sink),
|
|
gst_message_new_async_start (GST_OBJECT_CAST (rtsp_client_sink)));
|
|
}
|
|
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
|
|
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
/* fall-through */
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
/* unblock the tcp tasks and make the loop waiting */
|
|
if (gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_WAIT,
|
|
CMD_LOOP)) {
|
|
/* make sure it is waiting before we send PLAY below */
|
|
GST_RTSP_STREAM_LOCK (rtsp_client_sink);
|
|
GST_RTSP_STREAM_UNLOCK (rtsp_client_sink);
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_rtsp_client_sink_set_state (rtsp_client_sink, GST_STATE_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_NULL_TO_READY:
|
|
ret = GST_STATE_CHANGE_SUCCESS;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
/* Return ASYNC and preroll input streams */
|
|
g_mutex_lock (&rtsp_client_sink->preroll_lock);
|
|
if (rtsp_client_sink->in_async)
|
|
ret = GST_STATE_CHANGE_ASYNC;
|
|
g_mutex_unlock (&rtsp_client_sink->preroll_lock);
|
|
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_OPEN, 0);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
|
|
GST_DEBUG_OBJECT (rtsp_client_sink,
|
|
"Switching to playing -sending RECORD");
|
|
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_RECORD, 0);
|
|
ret = GST_STATE_CHANGE_SUCCESS;
|
|
break;
|
|
}
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
/* send pause request and keep the idle task around */
|
|
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_PAUSE,
|
|
CMD_LOOP);
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_rtsp_client_sink_loop_send_cmd (rtsp_client_sink, CMD_CLOSE,
|
|
CMD_PAUSE);
|
|
ret = GST_STATE_CHANGE_SUCCESS;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
gst_rtsp_client_sink_stop (rtsp_client_sink);
|
|
ret = GST_STATE_CHANGE_SUCCESS;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
|
|
start_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (rtsp_client_sink, "start failed");
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
}
|
|
|
|
/*** GSTURIHANDLER INTERFACE *************************************************/
|
|
|
|
static GstURIType
|
|
gst_rtsp_client_sink_uri_get_type (GType type)
|
|
{
|
|
return GST_URI_SINK;
|
|
}
|
|
|
|
static const gchar *const *
|
|
gst_rtsp_client_sink_uri_get_protocols (GType type)
|
|
{
|
|
static const gchar *protocols[] =
|
|
{ "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
|
|
"rtsps", "rtspsu", "rtspst", "rtspsh", NULL
|
|
};
|
|
|
|
return protocols;
|
|
}
|
|
|
|
static gchar *
|
|
gst_rtsp_client_sink_uri_get_uri (GstURIHandler * handler)
|
|
{
|
|
GstRTSPClientSink *sink = GST_RTSP_CLIENT_SINK (handler);
|
|
|
|
/* FIXME: make thread-safe */
|
|
return g_strdup (sink->conninfo.location);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtsp_client_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri,
|
|
GError ** error)
|
|
{
|
|
GstRTSPClientSink *sink;
|
|
GstRTSPResult res;
|
|
GstSDPResult sres;
|
|
GstRTSPUrl *newurl = NULL;
|
|
GstSDPMessage *sdp = NULL;
|
|
|
|
sink = GST_RTSP_CLIENT_SINK (handler);
|
|
|
|
/* same URI, we're fine */
|
|
if (sink->conninfo.location && uri && !strcmp (uri, sink->conninfo.location))
|
|
goto was_ok;
|
|
|
|
if (g_str_has_prefix (uri, "rtsp-sdp://")) {
|
|
sres = gst_sdp_message_new (&sdp);
|
|
if (sres < 0)
|
|
goto sdp_failed;
|
|
|
|
GST_DEBUG_OBJECT (sink, "parsing SDP message");
|
|
sres = gst_sdp_message_parse_uri (uri, sdp);
|
|
if (sres < 0)
|
|
goto invalid_sdp;
|
|
} else {
|
|
/* try to parse */
|
|
GST_DEBUG_OBJECT (sink, "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 (sink, "configuring URI");
|
|
g_free (sink->conninfo.location);
|
|
sink->conninfo.location = g_strdup (uri);
|
|
gst_rtsp_url_free (sink->conninfo.url);
|
|
sink->conninfo.url = newurl;
|
|
g_free (sink->conninfo.url_str);
|
|
if (newurl)
|
|
sink->conninfo.url_str = gst_rtsp_url_get_request_uri (sink->conninfo.url);
|
|
else
|
|
sink->conninfo.url_str = NULL;
|
|
|
|
if (sink->uri_sdp)
|
|
gst_sdp_message_free (sink->uri_sdp);
|
|
sink->uri_sdp = sdp;
|
|
sink->from_sdp = sdp != NULL;
|
|
|
|
GST_DEBUG_OBJECT (sink, "set uri: %s", GST_STR_NULL (uri));
|
|
GST_DEBUG_OBJECT (sink, "request uri is: %s",
|
|
GST_STR_NULL (sink->conninfo.url_str));
|
|
|
|
return TRUE;
|
|
|
|
/* Special cases */
|
|
was_ok:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "URI was ok: '%s'", GST_STR_NULL (uri));
|
|
return TRUE;
|
|
}
|
|
sdp_failed:
|
|
{
|
|
GST_ERROR_OBJECT (sink, "Could not create new SDP (%d)", sres);
|
|
g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
|
|
"Could not create SDP");
|
|
return FALSE;
|
|
}
|
|
invalid_sdp:
|
|
{
|
|
GST_ERROR_OBJECT (sink, "Not a valid SDP (%d) '%s'", sres,
|
|
GST_STR_NULL (uri));
|
|
gst_sdp_message_free (sdp);
|
|
g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
|
|
"Invalid SDP");
|
|
return FALSE;
|
|
}
|
|
parse_error:
|
|
{
|
|
GST_ERROR_OBJECT (sink, "Not a valid RTSP url '%s' (%d)",
|
|
GST_STR_NULL (uri), res);
|
|
g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
|
|
"Invalid RTSP URI");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_sink_uri_handler_init (gpointer g_iface, gpointer iface_data)
|
|
{
|
|
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
|
|
|
|
iface->get_type = gst_rtsp_client_sink_uri_get_type;
|
|
iface->get_protocols = gst_rtsp_client_sink_uri_get_protocols;
|
|
iface->get_uri = gst_rtsp_client_sink_uri_get_uri;
|
|
iface->set_uri = gst_rtsp_client_sink_uri_set_uri;
|
|
}
|