mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Conflicts: ext/chromaprint/gstchromaprint.c ext/mpeg2enc/Makefile.am ext/voaacenc/gstvoaacenc.c gst/dvbsuboverlay/gstdvbsuboverlay.c gst/mpegtsdemux/mpegtsbase.c gst/sdp/gstsdpdemux.c gst/videoparsers/gsth264parse.c sys/d3dvideosink/d3dvideosink.c tests/examples/camerabin/gst-camera-perf.c tests/examples/camerabin/gst-camerabin-test.c tests/examples/camerabin2/gst-camerabin2-test.c tests/examples/mxf/mxfdemux-structure.c tests/examples/scaletempo/demo-main.c
1539 lines
43 KiB
C
1539 lines
43 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim dot taymans at gmail 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-sdpdemux
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*
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* sdpdemux currently understands SDP as the input format of the session description.
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* For each stream listed in the SDP a new stream_%u pad will be created
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* with caps derived from the SDP media description. This is a caps of mime type
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* "application/x-rtp" that can be connected to any available RTP depayloader
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* element.
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*
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* sdpdemux will internally instantiate an RTP session manager element
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* that will handle the RTCP messages to and from the server, jitter removal,
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* packet reordering along with providing a clock for the pipeline.
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*
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* sdpdemux acts like a live element and will therefore only generate data in the
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* PLAYING state.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch gnomevfssrc location=http://some.server/session.sdp ! sdpdemux ! fakesink
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* ]| Establish a connection to an HTTP server that contains an SDP session description
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* that gets parsed by sdpdemux and send the raw RTP packets to a fakesink.
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* </refsect2>
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*
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* Last reviewed on 2007-10-01 (0.10.6)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
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* with newer GLib versions (>= 2.31.0) */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#include "gstsdpdemux.h"
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#include <gst/rtp/gstrtppayloads.h>
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#include <gst/sdp/gstsdpmessage.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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GST_DEBUG_CATEGORY_STATIC (sdpdemux_debug);
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#define GST_CAT_DEFAULT (sdpdemux_debug)
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static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/sdp"));
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static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtp"));
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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#define DEFAULT_DEBUG FALSE
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#define DEFAULT_TIMEOUT 10000000
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#define DEFAULT_LATENCY_MS 200
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#define DEFAULT_REDIRECT TRUE
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enum
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{
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PROP_0,
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PROP_DEBUG,
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PROP_TIMEOUT,
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PROP_LATENCY,
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PROP_REDIRECT,
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PROP_LAST
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};
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static void gst_sdp_demux_finalize (GObject * object);
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static void gst_sdp_demux_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_sdp_demux_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstCaps *gst_sdp_demux_media_to_caps (gint pt,
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const GstSDPMedia * media);
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static GstStateChangeReturn gst_sdp_demux_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_sdp_demux_handle_message (GstBin * bin, GstMessage * message);
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static void gst_sdp_demux_stream_push_event (GstSDPDemux * demux,
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GstSDPStream * stream, GstEvent * event);
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static gboolean gst_sdp_demux_sink_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static GstFlowReturn gst_sdp_demux_sink_chain (GstPad * pad, GstObject * parent,
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GstBuffer * buffer);
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/*static guint gst_sdp_demux_signals[LAST_SIGNAL] = { 0 }; */
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#define gst_sdp_demux_parent_class parent_class
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G_DEFINE_TYPE (GstSDPDemux, gst_sdp_demux, GST_TYPE_BIN);
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static void
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gst_sdp_demux_class_init (GstSDPDemuxClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBinClass *gstbin_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbin_class = (GstBinClass *) klass;
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gobject_class->set_property = gst_sdp_demux_set_property;
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gobject_class->get_property = gst_sdp_demux_get_property;
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gobject_class->finalize = gst_sdp_demux_finalize;
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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,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | 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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"Fail 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_CONSTRUCT | 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_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_REDIRECT,
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g_param_spec_boolean ("redirect", "Redirect",
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"Sends a redirection message instead of using a custom session element",
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DEFAULT_REDIRECT,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&sinktemplate));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&rtptemplate));
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gst_element_class_set_details_simple (gstelement_class, "SDP session setup",
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"Codec/Demuxer/Network/RTP",
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"Receive data over the network via SDP",
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"Wim Taymans <wim.taymans@gmail.com>");
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gstelement_class->change_state = gst_sdp_demux_change_state;
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gstbin_class->handle_message = gst_sdp_demux_handle_message;
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GST_DEBUG_CATEGORY_INIT (sdpdemux_debug, "sdpdemux", 0, "SDP demux");
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}
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static void
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gst_sdp_demux_init (GstSDPDemux * demux)
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{
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demux->sinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
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gst_pad_set_event_function (demux->sinkpad,
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GST_DEBUG_FUNCPTR (gst_sdp_demux_sink_event));
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gst_pad_set_chain_function (demux->sinkpad,
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GST_DEBUG_FUNCPTR (gst_sdp_demux_sink_chain));
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gst_element_add_pad (GST_ELEMENT (demux), demux->sinkpad);
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/* protects the streaming thread in interleaved mode or the polling
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* thread in UDP mode. */
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g_rec_mutex_init (&demux->stream_rec_lock);
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demux->adapter = gst_adapter_new ();
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}
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static void
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gst_sdp_demux_finalize (GObject * object)
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{
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GstSDPDemux *demux;
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demux = GST_SDP_DEMUX (object);
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/* free locks */
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g_rec_mutex_clear (&demux->stream_rec_lock);
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g_object_unref (demux->adapter);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_sdp_demux_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstSDPDemux *demux;
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demux = GST_SDP_DEMUX (object);
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switch (prop_id) {
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case PROP_DEBUG:
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demux->debug = g_value_get_boolean (value);
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break;
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case PROP_TIMEOUT:
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demux->udp_timeout = g_value_get_uint64 (value);
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break;
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case PROP_LATENCY:
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demux->latency = g_value_get_uint (value);
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break;
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case PROP_REDIRECT:
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demux->redirect = g_value_get_boolean (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_sdp_demux_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstSDPDemux *demux;
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demux = GST_SDP_DEMUX (object);
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switch (prop_id) {
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case PROP_DEBUG:
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g_value_set_boolean (value, demux->debug);
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break;
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case PROP_TIMEOUT:
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g_value_set_uint64 (value, demux->udp_timeout);
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break;
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case PROP_LATENCY:
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g_value_set_uint (value, demux->latency);
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break;
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case PROP_REDIRECT:
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g_value_set_boolean (value, demux->redirect);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static gint
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find_stream_by_id (GstSDPStream * stream, gconstpointer a)
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{
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gint id = GPOINTER_TO_INT (a);
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if (stream->id == id)
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return 0;
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return -1;
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}
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static gint
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find_stream_by_pt (GstSDPStream * stream, gconstpointer a)
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{
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gint pt = GPOINTER_TO_INT (a);
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if (stream->pt == pt)
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return 0;
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return -1;
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}
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static gint
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find_stream_by_udpsrc (GstSDPStream * stream, gconstpointer a)
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{
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GstElement *src = (GstElement *) a;
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if (stream->udpsrc[0] == src)
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return 0;
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if (stream->udpsrc[1] == src)
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return 0;
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return -1;
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}
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static GstSDPStream *
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find_stream (GstSDPDemux * demux, gconstpointer data, gconstpointer func)
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{
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GList *lstream;
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/* find and get stream */
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if ((lstream =
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g_list_find_custom (demux->streams, data, (GCompareFunc) func)))
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return (GstSDPStream *) lstream->data;
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return NULL;
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}
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static void
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gst_sdp_demux_stream_free (GstSDPDemux * demux, GstSDPStream * stream)
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{
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gint i;
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GST_DEBUG_OBJECT (demux, "free stream %p", stream);
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if (stream->caps)
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gst_caps_unref (stream->caps);
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for (i = 0; i < 2; i++) {
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GstElement *udpsrc = stream->udpsrc[i];
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if (udpsrc) {
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gst_element_set_state (udpsrc, GST_STATE_NULL);
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gst_bin_remove (GST_BIN_CAST (demux), udpsrc);
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stream->udpsrc[i] = NULL;
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}
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}
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if (stream->udpsink) {
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gst_element_set_state (stream->udpsink, GST_STATE_NULL);
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gst_bin_remove (GST_BIN_CAST (demux), stream->udpsink);
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stream->udpsink = NULL;
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}
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if (stream->srcpad) {
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gst_pad_set_active (stream->srcpad, FALSE);
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if (stream->added) {
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gst_element_remove_pad (GST_ELEMENT_CAST (demux), stream->srcpad);
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stream->added = FALSE;
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}
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stream->srcpad = NULL;
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}
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g_free (stream);
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}
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static gboolean
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is_multicast_address (const gchar * host_name)
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{
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GInetAddress *addr;
|
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GResolver *resolver = NULL;
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gboolean ret = FALSE;
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addr = g_inet_address_new_from_string (host_name);
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if (!addr) {
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GList *results;
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resolver = g_resolver_get_default ();
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results = g_resolver_lookup_by_name (resolver, host_name, NULL, NULL);
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if (!results)
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goto out;
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addr = G_INET_ADDRESS (g_object_ref (results->data));
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g_resolver_free_addresses (results);
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}
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g_assert (addr != NULL);
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ret = g_inet_address_get_is_multicast (addr);
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out:
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if (resolver)
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g_object_unref (resolver);
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if (addr)
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g_object_unref (addr);
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return ret;
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}
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|
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static GstSDPStream *
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gst_sdp_demux_create_stream (GstSDPDemux * demux, GstSDPMessage * sdp, gint idx)
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{
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GstSDPStream *stream;
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const gchar *payload;
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const GstSDPMedia *media;
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const GstSDPConnection *conn;
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|
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/* get media, should not return NULL */
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media = gst_sdp_message_get_media (sdp, idx);
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if (media == NULL)
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return NULL;
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|
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stream = g_new0 (GstSDPStream, 1);
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stream->parent = demux;
|
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/* we mark the pad as not linked, we will mark it as OK when we add the pad to
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* the element. */
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stream->last_ret = GST_FLOW_OK;
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stream->added = FALSE;
|
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stream->disabled = FALSE;
|
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stream->id = demux->numstreams++;
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stream->eos = FALSE;
|
|
|
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/* we must have a payload. No payload means we cannot create caps */
|
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/* FIXME, handle multiple formats. */
|
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if ((payload = gst_sdp_media_get_format (media, 0))) {
|
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stream->pt = atoi (payload);
|
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/* convert caps */
|
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stream->caps = gst_sdp_demux_media_to_caps (stream->pt, media);
|
|
|
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if (stream->pt >= 96) {
|
|
/* If we have a dynamic payload type, see if we have a stream with the
|
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* same payload number. If there is one, they are part of the same
|
|
* container and we only need to add one pad. */
|
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if (find_stream (demux, GINT_TO_POINTER (stream->pt),
|
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(gpointer) find_stream_by_pt)) {
|
|
stream->container = TRUE;
|
|
}
|
|
}
|
|
}
|
|
if (!(conn = gst_sdp_media_get_connection (media, 0))) {
|
|
if (!(conn = gst_sdp_message_get_connection (sdp)))
|
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goto no_connection;
|
|
}
|
|
|
|
if (!conn->address)
|
|
goto no_connection;
|
|
|
|
stream->destination = conn->address;
|
|
stream->ttl = conn->ttl;
|
|
stream->multicast = is_multicast_address (stream->destination);
|
|
|
|
stream->rtp_port = gst_sdp_media_get_port (media);
|
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if (gst_sdp_media_get_attribute_val (media, "rtcp")) {
|
|
/* FIXME, RFC 3605 */
|
|
stream->rtcp_port = stream->rtp_port + 1;
|
|
} else {
|
|
stream->rtcp_port = stream->rtp_port + 1;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (demux, "stream %d, (%p)", stream->id, stream);
|
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GST_DEBUG_OBJECT (demux, " pt: %d", stream->pt);
|
|
GST_DEBUG_OBJECT (demux, " container: %d", stream->container);
|
|
GST_DEBUG_OBJECT (demux, " caps: %" GST_PTR_FORMAT, stream->caps);
|
|
|
|
/* we keep track of all streams */
|
|
demux->streams = g_list_append (demux->streams, stream);
|
|
|
|
return stream;
|
|
|
|
/* ERRORS */
|
|
no_connection:
|
|
{
|
|
gst_sdp_demux_stream_free (demux, stream);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_sdp_demux_cleanup (GstSDPDemux * demux)
|
|
{
|
|
GList *walk;
|
|
|
|
GST_DEBUG_OBJECT (demux, "cleanup");
|
|
|
|
for (walk = demux->streams; walk; walk = g_list_next (walk)) {
|
|
GstSDPStream *stream = (GstSDPStream *) walk->data;
|
|
|
|
gst_sdp_demux_stream_free (demux, stream);
|
|
}
|
|
g_list_free (demux->streams);
|
|
demux->streams = NULL;
|
|
if (demux->session) {
|
|
if (demux->session_sig_id) {
|
|
g_signal_handler_disconnect (demux->session, demux->session_sig_id);
|
|
demux->session_sig_id = 0;
|
|
}
|
|
if (demux->session_nmp_id) {
|
|
g_signal_handler_disconnect (demux->session, demux->session_nmp_id);
|
|
demux->session_nmp_id = 0;
|
|
}
|
|
if (demux->session_ptmap_id) {
|
|
g_signal_handler_disconnect (demux->session, demux->session_ptmap_id);
|
|
demux->session_ptmap_id = 0;
|
|
}
|
|
gst_element_set_state (demux->session, GST_STATE_NULL);
|
|
gst_bin_remove (GST_BIN_CAST (demux), demux->session);
|
|
demux->session = NULL;
|
|
}
|
|
demux->numstreams = 0;
|
|
}
|
|
|
|
#define PARSE_INT(p, del, res) \
|
|
G_STMT_START { \
|
|
gchar *t = p; \
|
|
p = strstr (p, del); \
|
|
if (p == NULL) \
|
|
res = -1; \
|
|
else { \
|
|
*p = '\0'; \
|
|
p++; \
|
|
res = atoi (t); \
|
|
} \
|
|
} G_STMT_END
|
|
|
|
#define PARSE_STRING(p, del, res) \
|
|
G_STMT_START { \
|
|
gchar *t = p; \
|
|
p = strstr (p, del); \
|
|
if (p == NULL) { \
|
|
res = NULL; \
|
|
p = t; \
|
|
} \
|
|
else { \
|
|
*p = '\0'; \
|
|
p++; \
|
|
res = t; \
|
|
} \
|
|
} G_STMT_END
|
|
|
|
#define SKIP_SPACES(p) \
|
|
while (*p && g_ascii_isspace (*p)) \
|
|
p++;
|
|
|
|
/* rtpmap contains:
|
|
*
|
|
* <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
|
|
*/
|
|
static gboolean
|
|
gst_sdp_demux_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
|
|
gint * rate, gchar ** params)
|
|
{
|
|
gchar *p, *t;
|
|
|
|
t = p = (gchar *) rtpmap;
|
|
|
|
PARSE_INT (p, " ", *payload);
|
|
if (*payload == -1)
|
|
return FALSE;
|
|
|
|
SKIP_SPACES (p);
|
|
if (*p == '\0')
|
|
return FALSE;
|
|
|
|
PARSE_STRING (p, "/", *name);
|
|
if (*name == NULL) {
|
|
GST_DEBUG ("no rate, name %s", p);
|
|
/* no rate, assume -1 then */
|
|
*name = p;
|
|
*rate = -1;
|
|
return TRUE;
|
|
}
|
|
|
|
t = p;
|
|
p = strstr (p, "/");
|
|
if (p == NULL) {
|
|
*rate = atoi (t);
|
|
return TRUE;
|
|
}
|
|
*p = '\0';
|
|
p++;
|
|
*rate = atoi (t);
|
|
|
|
t = p;
|
|
if (*p == '\0')
|
|
return TRUE;
|
|
*params = t;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/*
|
|
* Mapping of caps to and from SDP fields:
|
|
*
|
|
* m=<media> <UDP port> RTP/AVP <payload>
|
|
* a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
|
|
* a=fmtp:<payload> <param>[=<value>];...
|
|
*/
|
|
static GstCaps *
|
|
gst_sdp_demux_media_to_caps (gint pt, const GstSDPMedia * media)
|
|
{
|
|
GstCaps *caps;
|
|
const gchar *rtpmap;
|
|
const gchar *fmtp;
|
|
gchar *name = NULL;
|
|
gint rate = -1;
|
|
gchar *params = NULL;
|
|
gchar *tmp;
|
|
GstStructure *s;
|
|
gint payload = 0;
|
|
gboolean ret;
|
|
|
|
/* get and parse rtpmap */
|
|
if ((rtpmap = gst_sdp_media_get_attribute_val (media, "rtpmap"))) {
|
|
ret = gst_sdp_demux_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
|
|
if (ret) {
|
|
if (payload != pt) {
|
|
/* we ignore the rtpmap if the payload type is different. */
|
|
g_warning ("rtpmap of wrong payload type, ignoring");
|
|
name = NULL;
|
|
rate = -1;
|
|
params = NULL;
|
|
}
|
|
} else {
|
|
/* if we failed to parse the rtpmap for a dynamic payload type, we have an
|
|
* error */
|
|
if (pt >= 96)
|
|
goto no_rtpmap;
|
|
/* else we can ignore */
|
|
g_warning ("error parsing rtpmap, ignoring");
|
|
}
|
|
} else {
|
|
/* dynamic payloads need rtpmap or we fail */
|
|
if (pt >= 96)
|
|
goto no_rtpmap;
|
|
}
|
|
/* check if we have a rate, if not, we need to look up the rate from the
|
|
* default rates based on the payload types. */
|
|
if (rate == -1) {
|
|
const GstRTPPayloadInfo *info;
|
|
|
|
if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
|
|
/* dynamic types, use media and encoding_name */
|
|
tmp = g_ascii_strdown (media->media, -1);
|
|
info = gst_rtp_payload_info_for_name (tmp, name);
|
|
g_free (tmp);
|
|
} else {
|
|
/* static types, use payload type */
|
|
info = gst_rtp_payload_info_for_pt (pt);
|
|
}
|
|
|
|
if (info) {
|
|
if ((rate = info->clock_rate) == 0)
|
|
rate = -1;
|
|
}
|
|
/* we fail if we cannot find one */
|
|
if (rate == -1)
|
|
goto no_rate;
|
|
}
|
|
|
|
tmp = g_ascii_strdown (media->media, -1);
|
|
caps = gst_caps_new_simple ("application/x-rtp",
|
|
"media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
|
|
g_free (tmp);
|
|
s = gst_caps_get_structure (caps, 0);
|
|
|
|
gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
|
|
|
|
/* encoding name must be upper case */
|
|
if (name != NULL) {
|
|
tmp = g_ascii_strup (name, -1);
|
|
gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
|
|
g_free (tmp);
|
|
}
|
|
|
|
/* params must be lower case */
|
|
if (params != NULL) {
|
|
tmp = g_ascii_strdown (params, -1);
|
|
gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
|
|
g_free (tmp);
|
|
}
|
|
|
|
/* parse optional fmtp: field */
|
|
if ((fmtp = gst_sdp_media_get_attribute_val (media, "fmtp"))) {
|
|
gchar *p;
|
|
gint payload = 0;
|
|
|
|
p = (gchar *) fmtp;
|
|
|
|
/* p is now of the format <payload> <param>[=<value>];... */
|
|
PARSE_INT (p, " ", payload);
|
|
if (payload != -1 && payload == pt) {
|
|
gchar **pairs;
|
|
gint i;
|
|
|
|
/* <param>[=<value>] are separated with ';' */
|
|
pairs = g_strsplit (p, ";", 0);
|
|
for (i = 0; pairs[i]; i++) {
|
|
gchar *valpos, *key;
|
|
const gchar *val;
|
|
|
|
/* the key may not have a '=', the value can have other '='s */
|
|
valpos = strstr (pairs[i], "=");
|
|
if (valpos) {
|
|
/* we have a '=' and thus a value, remove the '=' with \0 */
|
|
*valpos = '\0';
|
|
/* value is everything between '=' and ';'. FIXME, strip? */
|
|
val = g_strstrip (valpos + 1);
|
|
} else {
|
|
/* simple <param>;.. is translated into <param>=1;... */
|
|
val = "1";
|
|
}
|
|
/* strip the key of spaces, convert key to lowercase but not the value. */
|
|
key = g_strstrip (pairs[i]);
|
|
if (strlen (key) > 1) {
|
|
tmp = g_ascii_strdown (key, -1);
|
|
gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
|
|
g_free (tmp);
|
|
}
|
|
}
|
|
g_strfreev (pairs);
|
|
}
|
|
}
|
|
return caps;
|
|
|
|
/* ERRORS */
|
|
no_rtpmap:
|
|
{
|
|
g_warning ("rtpmap type not given for dynamic payload %d", pt);
|
|
return NULL;
|
|
}
|
|
no_rate:
|
|
{
|
|
g_warning ("rate unknown for payload type %d", pt);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* this callback is called when the session manager generated a new src pad with
|
|
* payloaded RTP packets. We simply ghost the pad here. */
|
|
static void
|
|
new_session_pad (GstElement * session, GstPad * pad, GstSDPDemux * demux)
|
|
{
|
|
gchar *name;
|
|
GstPadTemplate *template;
|
|
gint id, ssrc, pt;
|
|
GList *lstream;
|
|
GstSDPStream *stream;
|
|
gboolean all_added;
|
|
|
|
GST_DEBUG_OBJECT (demux, "got new session pad %" GST_PTR_FORMAT, pad);
|
|
|
|
GST_SDP_STREAM_LOCK (demux);
|
|
/* find stream */
|
|
name = gst_object_get_name (GST_OBJECT_CAST (pad));
|
|
if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
|
|
goto unknown_stream;
|
|
|
|
GST_DEBUG_OBJECT (demux, "stream: %u, SSRC %d, PT %d", id, ssrc, pt);
|
|
|
|
stream =
|
|
find_stream (demux, GINT_TO_POINTER (id), (gpointer) find_stream_by_id);
|
|
if (stream == NULL)
|
|
goto unknown_stream;
|
|
|
|
/* no need for a timeout anymore now */
|
|
g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
|
|
|
|
/* create a new pad we will use to stream to */
|
|
template = gst_static_pad_template_get (&rtptemplate);
|
|
stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
|
|
gst_object_unref (template);
|
|
g_free (name);
|
|
|
|
stream->added = TRUE;
|
|
gst_pad_set_active (stream->srcpad, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (demux), stream->srcpad);
|
|
|
|
/* check if we added all streams */
|
|
all_added = TRUE;
|
|
for (lstream = demux->streams; lstream; lstream = g_list_next (lstream)) {
|
|
stream = (GstSDPStream *) lstream->data;
|
|
/* a container stream only needs one pad added. Also disabled streams don't
|
|
* count */
|
|
if (!stream->container && !stream->disabled && !stream->added) {
|
|
all_added = FALSE;
|
|
break;
|
|
}
|
|
}
|
|
GST_SDP_STREAM_UNLOCK (demux);
|
|
|
|
if (all_added) {
|
|
GST_DEBUG_OBJECT (demux, "We added all streams");
|
|
/* when we get here, all stream are added and we can fire the no-more-pads
|
|
* signal. */
|
|
gst_element_no_more_pads (GST_ELEMENT_CAST (demux));
|
|
}
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
unknown_stream:
|
|
{
|
|
GST_DEBUG_OBJECT (demux, "ignoring unknown stream");
|
|
GST_SDP_STREAM_UNLOCK (demux);
|
|
g_free (name);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void
|
|
rtsp_session_pad_added (GstElement * session, GstPad * pad, GstSDPDemux * demux)
|
|
{
|
|
GstPad *srcpad = NULL;
|
|
gchar *name;
|
|
|
|
GST_DEBUG_OBJECT (demux, "got new session pad %" GST_PTR_FORMAT, pad);
|
|
|
|
name = gst_pad_get_name (pad);
|
|
srcpad = gst_ghost_pad_new (name, pad);
|
|
g_free (name);
|
|
|
|
gst_pad_set_active (srcpad, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (demux), srcpad);
|
|
}
|
|
|
|
static void
|
|
rtsp_session_no_more_pads (GstElement * session, GstSDPDemux * demux)
|
|
{
|
|
GST_DEBUG_OBJECT (demux, "got no-more-pads");
|
|
gst_element_no_more_pads (GST_ELEMENT_CAST (demux));
|
|
}
|
|
|
|
static GstCaps *
|
|
request_pt_map (GstElement * sess, guint session, guint pt, GstSDPDemux * demux)
|
|
{
|
|
GstSDPStream *stream;
|
|
GstCaps *caps;
|
|
|
|
GST_DEBUG_OBJECT (demux, "getting pt map for pt %d in session %d", pt,
|
|
session);
|
|
|
|
GST_SDP_STREAM_LOCK (demux);
|
|
stream =
|
|
find_stream (demux, GINT_TO_POINTER (session),
|
|
(gpointer) find_stream_by_id);
|
|
if (!stream)
|
|
goto unknown_stream;
|
|
|
|
caps = stream->caps;
|
|
if (caps)
|
|
gst_caps_ref (caps);
|
|
GST_SDP_STREAM_UNLOCK (demux);
|
|
|
|
return caps;
|
|
|
|
unknown_stream:
|
|
{
|
|
GST_DEBUG_OBJECT (demux, "unknown stream %d", session);
|
|
GST_SDP_STREAM_UNLOCK (demux);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_sdp_demux_do_stream_eos (GstSDPDemux * demux, guint session)
|
|
{
|
|
GstSDPStream *stream;
|
|
|
|
GST_DEBUG_OBJECT (demux, "setting stream for session %u to EOS", session);
|
|
|
|
/* get stream for session */
|
|
stream =
|
|
find_stream (demux, GINT_TO_POINTER (session),
|
|
(gpointer) find_stream_by_id);
|
|
if (!stream)
|
|
goto unknown_stream;
|
|
|
|
if (stream->eos)
|
|
goto was_eos;
|
|
|
|
stream->eos = TRUE;
|
|
gst_sdp_demux_stream_push_event (demux, stream, gst_event_new_eos ());
|
|
return;
|
|
|
|
/* ERRORS */
|
|
unknown_stream:
|
|
{
|
|
GST_DEBUG_OBJECT (demux, "unknown stream for session %u", session);
|
|
return;
|
|
}
|
|
was_eos:
|
|
{
|
|
GST_DEBUG_OBJECT (demux, "stream for session %u was already EOS", session);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void
|
|
on_bye_ssrc (GstElement * manager, guint session, guint32 ssrc,
|
|
GstSDPDemux * demux)
|
|
{
|
|
GST_DEBUG_OBJECT (demux, "SSRC %08x in session %u received BYE", ssrc,
|
|
session);
|
|
|
|
gst_sdp_demux_do_stream_eos (demux, session);
|
|
}
|
|
|
|
static void
|
|
on_timeout (GstElement * manager, guint session, guint32 ssrc,
|
|
GstSDPDemux * demux)
|
|
{
|
|
GST_DEBUG_OBJECT (demux, "SSRC %08x in session %u timed out", ssrc, session);
|
|
|
|
gst_sdp_demux_do_stream_eos (demux, session);
|
|
}
|
|
|
|
/* try to get and configure a manager */
|
|
static gboolean
|
|
gst_sdp_demux_configure_manager (GstSDPDemux * demux, char *rtsp_sdp)
|
|
{
|
|
/* configure the session manager */
|
|
if (rtsp_sdp != NULL) {
|
|
if (!(demux->session = gst_element_factory_make ("rtspsrc", NULL)))
|
|
goto rtspsrc_failed;
|
|
|
|
g_object_set (demux->session, "location", rtsp_sdp, NULL);
|
|
|
|
GST_DEBUG_OBJECT (demux, "connect to signals on rtspsrc");
|
|
demux->session_sig_id =
|
|
g_signal_connect (demux->session, "pad-added",
|
|
(GCallback) rtsp_session_pad_added, demux);
|
|
demux->session_nmp_id =
|
|
g_signal_connect (demux->session, "no-more-pads",
|
|
(GCallback) rtsp_session_no_more_pads, demux);
|
|
} else {
|
|
if (!(demux->session = gst_element_factory_make ("rtpbin", NULL)))
|
|
goto manager_failed;
|
|
|
|
/* connect to signals if we did not already do so */
|
|
GST_DEBUG_OBJECT (demux, "connect to signals on session manager");
|
|
demux->session_sig_id =
|
|
g_signal_connect (demux->session, "pad-added",
|
|
(GCallback) new_session_pad, demux);
|
|
demux->session_ptmap_id =
|
|
g_signal_connect (demux->session, "request-pt-map",
|
|
(GCallback) request_pt_map, demux);
|
|
g_signal_connect (demux->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
|
|
demux);
|
|
g_signal_connect (demux->session, "on-bye-timeout", (GCallback) on_timeout,
|
|
demux);
|
|
g_signal_connect (demux->session, "on-timeout", (GCallback) on_timeout,
|
|
demux);
|
|
}
|
|
|
|
g_object_set (demux->session, "latency", demux->latency, NULL);
|
|
|
|
/* we manage this element */
|
|
gst_bin_add (GST_BIN_CAST (demux), demux->session);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
manager_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (demux, "no session manager element gstrtpbin found");
|
|
return FALSE;
|
|
}
|
|
rtspsrc_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (demux, "no manager element rtspsrc found");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_sdp_demux_stream_configure_udp (GstSDPDemux * demux, GstSDPStream * stream)
|
|
{
|
|
gchar *uri, *name;
|
|
const gchar *destination;
|
|
GstPad *pad;
|
|
|
|
GST_DEBUG_OBJECT (demux, "creating UDP sources for multicast");
|
|
|
|
/* if the destination is not a multicast address, we just want to listen on
|
|
* our local ports */
|
|
if (!stream->multicast)
|
|
destination = "0.0.0.0";
|
|
else
|
|
destination = stream->destination;
|
|
|
|
/* creating UDP source */
|
|
if (stream->rtp_port != -1) {
|
|
GST_DEBUG_OBJECT (demux, "receiving RTP from %s:%d", destination,
|
|
stream->rtp_port);
|
|
|
|
uri = g_strdup_printf ("udp://%s:%d", destination, stream->rtp_port);
|
|
stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
|
|
g_free (uri);
|
|
if (stream->udpsrc[0] == NULL)
|
|
goto no_element;
|
|
|
|
/* take ownership */
|
|
gst_bin_add (GST_BIN_CAST (demux), stream->udpsrc[0]);
|
|
|
|
GST_DEBUG_OBJECT (demux,
|
|
"setting up UDP source with timeout %" G_GINT64_FORMAT,
|
|
demux->udp_timeout);
|
|
|
|
/* configure a timeout on the UDP port. When the timeout message is
|
|
* posted, we assume UDP transport is not possible. */
|
|
g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", demux->udp_timeout,
|
|
NULL);
|
|
|
|
/* get output pad of the UDP source. */
|
|
pad = gst_element_get_static_pad (stream->udpsrc[0], "src");
|
|
|
|
name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
|
|
stream->channelpad[0] = gst_element_get_request_pad (demux->session, name);
|
|
g_free (name);
|
|
|
|
GST_DEBUG_OBJECT (demux, "connecting RTP source 0 to manager");
|
|
/* configure for UDP delivery, we need to connect the UDP pads to
|
|
* the session plugin. */
|
|
gst_pad_link (pad, stream->channelpad[0]);
|
|
gst_object_unref (pad);
|
|
|
|
/* change state */
|
|
gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
|
|
}
|
|
|
|
/* creating another UDP source */
|
|
if (stream->rtcp_port != -1) {
|
|
GST_DEBUG_OBJECT (demux, "receiving RTCP from %s:%d", destination,
|
|
stream->rtcp_port);
|
|
uri = g_strdup_printf ("udp://%s:%d", destination, stream->rtcp_port);
|
|
stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
|
|
g_free (uri);
|
|
if (stream->udpsrc[1] == NULL)
|
|
goto no_element;
|
|
|
|
/* take ownership */
|
|
gst_bin_add (GST_BIN_CAST (demux), stream->udpsrc[1]);
|
|
|
|
GST_DEBUG_OBJECT (demux, "connecting RTCP source to manager");
|
|
|
|
name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
|
|
stream->channelpad[1] = gst_element_get_request_pad (demux->session, name);
|
|
g_free (name);
|
|
|
|
pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
|
|
gst_pad_link (pad, stream->channelpad[1]);
|
|
gst_object_unref (pad);
|
|
|
|
gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
|
|
}
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_element:
|
|
{
|
|
GST_DEBUG_OBJECT (demux, "no UDP source element found");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* configure the UDP sink back to the server for status reports */
|
|
static gboolean
|
|
gst_sdp_demux_stream_configure_udp_sink (GstSDPDemux * demux,
|
|
GstSDPStream * stream)
|
|
{
|
|
GstPad *pad, *sinkpad;
|
|
gint port;
|
|
GSocket *socket;
|
|
gchar *destination, *uri, *name;
|
|
|
|
/* get destination and port */
|
|
port = stream->rtcp_port;
|
|
destination = stream->destination;
|
|
|
|
GST_DEBUG_OBJECT (demux, "configure UDP sink for %s:%d", destination, port);
|
|
|
|
uri = g_strdup_printf ("udp://%s:%d", destination, port);
|
|
stream->udpsink = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
|
|
g_free (uri);
|
|
if (stream->udpsink == NULL)
|
|
goto no_sink_element;
|
|
|
|
/* we clear all destinations because we don't really know where to send the
|
|
* RTCP to and we want to avoid sending it to our own ports.
|
|
* FIXME when we get an RTCP packet from the sender, we could look at its
|
|
* source port and address and try to send RTCP there. */
|
|
if (!stream->multicast)
|
|
g_signal_emit_by_name (stream->udpsink, "clear");
|
|
|
|
g_object_set (G_OBJECT (stream->udpsink), "auto-multicast", FALSE, NULL);
|
|
g_object_set (G_OBJECT (stream->udpsink), "loop", FALSE, NULL);
|
|
/* no sync needed */
|
|
g_object_set (G_OBJECT (stream->udpsink), "sync", FALSE, NULL);
|
|
/* no async state changes needed */
|
|
g_object_set (G_OBJECT (stream->udpsink), "async", FALSE, NULL);
|
|
|
|
if (stream->udpsrc[1]) {
|
|
/* configure socket, we give it the same UDP socket as the udpsrc for RTCP
|
|
* because some servers check the port number of where it sends RTCP to identify
|
|
* the RTCP packets it receives */
|
|
g_object_get (G_OBJECT (stream->udpsrc[1]), "used_socket", &socket, NULL);
|
|
GST_DEBUG_OBJECT (demux, "UDP src has socket %p", socket);
|
|
/* configure socket and make sure udpsink does not close it when shutting
|
|
* down, it belongs to udpsrc after all. */
|
|
g_object_set (G_OBJECT (stream->udpsink), "socket", socket, NULL);
|
|
g_object_set (G_OBJECT (stream->udpsink), "close-socket", FALSE, NULL);
|
|
g_object_unref (socket);
|
|
}
|
|
|
|
/* we keep this playing always */
|
|
gst_element_set_locked_state (stream->udpsink, TRUE);
|
|
gst_element_set_state (stream->udpsink, GST_STATE_PLAYING);
|
|
|
|
gst_bin_add (GST_BIN_CAST (demux), stream->udpsink);
|
|
|
|
/* get session RTCP pad */
|
|
name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
|
|
pad = gst_element_get_request_pad (demux->session, name);
|
|
g_free (name);
|
|
|
|
/* and link */
|
|
if (pad) {
|
|
sinkpad = gst_element_get_static_pad (stream->udpsink, "sink");
|
|
gst_pad_link (pad, sinkpad);
|
|
gst_object_unref (sinkpad);
|
|
} else {
|
|
/* not very fatal, we just won't be able to send RTCP */
|
|
GST_WARNING_OBJECT (demux, "could not get session RTCP pad");
|
|
}
|
|
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_sink_element:
|
|
{
|
|
GST_DEBUG_OBJECT (demux, "no UDP sink element found");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_sdp_demux_combine_flows (GstSDPDemux * demux, GstSDPStream * stream,
|
|
GstFlowReturn ret)
|
|
{
|
|
GList *streams;
|
|
|
|
/* store the value */
|
|
stream->last_ret = ret;
|
|
|
|
/* if it's success we can return the value right away */
|
|
if (ret == GST_FLOW_OK)
|
|
goto done;
|
|
|
|
/* any other error that is not-linked can be returned right
|
|
* away */
|
|
if (ret != GST_FLOW_NOT_LINKED)
|
|
goto done;
|
|
|
|
/* only return NOT_LINKED if all other pads returned NOT_LINKED */
|
|
for (streams = demux->streams; streams; streams = g_list_next (streams)) {
|
|
GstSDPStream *ostream = (GstSDPStream *) streams->data;
|
|
|
|
ret = ostream->last_ret;
|
|
/* some other return value (must be SUCCESS but we can return
|
|
* other values as well) */
|
|
if (ret != GST_FLOW_NOT_LINKED)
|
|
goto done;
|
|
}
|
|
/* if we get here, all other pads were unlinked and we return
|
|
* NOT_LINKED then */
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_sdp_demux_stream_push_event (GstSDPDemux * demux, GstSDPStream * stream,
|
|
GstEvent * event)
|
|
{
|
|
/* only streams that have a connection to the outside world */
|
|
if (stream->srcpad == NULL)
|
|
goto done;
|
|
|
|
if (stream->channelpad[0]) {
|
|
gst_event_ref (event);
|
|
gst_pad_send_event (stream->channelpad[0], event);
|
|
}
|
|
|
|
if (stream->channelpad[1]) {
|
|
gst_event_ref (event);
|
|
gst_pad_send_event (stream->channelpad[1], event);
|
|
}
|
|
|
|
done:
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
static void
|
|
gst_sdp_demux_handle_message (GstBin * bin, GstMessage * message)
|
|
{
|
|
GstSDPDemux *demux;
|
|
|
|
demux = GST_SDP_DEMUX (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 (demux);
|
|
ignore_timeout = demux->ignore_timeout;
|
|
demux->ignore_timeout = TRUE;
|
|
GST_OBJECT_UNLOCK (demux);
|
|
|
|
/* we only act on the first udp timeout message, others are irrelevant
|
|
* and can be ignored. */
|
|
if (ignore_timeout)
|
|
gst_message_unref (message);
|
|
else {
|
|
GST_ELEMENT_ERROR (demux, RESOURCE, READ, (NULL),
|
|
("Could not receive any UDP packets for %.4f seconds, maybe your "
|
|
"firewall is blocking it.",
|
|
gst_guint64_to_gdouble (demux->udp_timeout / 1000000.0)));
|
|
}
|
|
return;
|
|
}
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
case GST_MESSAGE_ERROR:
|
|
{
|
|
GstObject *udpsrc;
|
|
GstSDPStream *stream;
|
|
GstFlowReturn ret;
|
|
|
|
udpsrc = GST_MESSAGE_SRC (message);
|
|
|
|
GST_DEBUG_OBJECT (demux, "got error from %s", GST_ELEMENT_NAME (udpsrc));
|
|
|
|
stream = find_stream (demux, udpsrc, (gpointer) find_stream_by_udpsrc);
|
|
/* fatal but not our message, forward */
|
|
if (!stream)
|
|
goto forward;
|
|
|
|
/* we ignore the RTCP udpsrc */
|
|
if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
|
|
goto done;
|
|
|
|
/* if we get error messages from the udp sources, that's not a problem as
|
|
* long as not all of them error out. We also don't really know what the
|
|
* problem is, the message does not give enough detail... */
|
|
ret = gst_sdp_demux_combine_flows (demux, stream, GST_FLOW_NOT_LINKED);
|
|
GST_DEBUG_OBJECT (demux, "combined flows: %s", gst_flow_get_name (ret));
|
|
if (ret != GST_FLOW_OK)
|
|
goto forward;
|
|
|
|
done:
|
|
gst_message_unref (message);
|
|
break;
|
|
|
|
forward:
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
default:
|
|
{
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_sdp_demux_start (GstSDPDemux * demux)
|
|
{
|
|
guint8 *data;
|
|
guint size;
|
|
gint i, n_streams;
|
|
GstSDPMessage sdp = { 0 };
|
|
GstSDPStream *stream = NULL;
|
|
GList *walk;
|
|
gchar *uri = NULL;
|
|
GstStateChangeReturn ret;
|
|
|
|
/* grab the lock so that no state change can interfere */
|
|
GST_SDP_STREAM_LOCK (demux);
|
|
|
|
GST_DEBUG_OBJECT (demux, "parse SDP...");
|
|
|
|
size = gst_adapter_available (demux->adapter);
|
|
data = gst_adapter_take (demux->adapter, size);
|
|
|
|
gst_sdp_message_init (&sdp);
|
|
if (gst_sdp_message_parse_buffer (data, size, &sdp) != GST_SDP_OK)
|
|
goto could_not_parse;
|
|
|
|
if (demux->debug)
|
|
gst_sdp_message_dump (&sdp);
|
|
|
|
/* maybe this is plain RTSP DESCRIBE rtsp and we should redirect */
|
|
/* look for rtsp control url */
|
|
{
|
|
const gchar *control;
|
|
|
|
for (i = 0;; i++) {
|
|
control = gst_sdp_message_get_attribute_val_n (&sdp, "control", i);
|
|
if (control == NULL)
|
|
break;
|
|
|
|
/* only take fully qualified urls */
|
|
if (g_str_has_prefix (control, "rtsp://"))
|
|
break;
|
|
}
|
|
if (!control) {
|
|
gint idx;
|
|
|
|
/* try to find non-aggragate control */
|
|
n_streams = gst_sdp_message_medias_len (&sdp);
|
|
|
|
for (idx = 0; idx < n_streams; idx++) {
|
|
const GstSDPMedia *media;
|
|
|
|
/* get media, should not return NULL */
|
|
media = gst_sdp_message_get_media (&sdp, idx);
|
|
if (media == NULL)
|
|
break;
|
|
|
|
for (i = 0;; i++) {
|
|
control = gst_sdp_media_get_attribute_val_n (media, "control", i);
|
|
if (control == NULL)
|
|
break;
|
|
|
|
/* only take fully qualified urls */
|
|
if (g_str_has_prefix (control, "rtsp://"))
|
|
break;
|
|
}
|
|
/* this media has no control, exit */
|
|
if (!control)
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (control) {
|
|
/* we have RTSP now */
|
|
uri = gst_sdp_message_as_uri ("rtsp-sdp", &sdp);
|
|
|
|
if (demux->redirect) {
|
|
GST_INFO_OBJECT (demux, "redirect to %s", uri);
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (demux),
|
|
gst_message_new_element (GST_OBJECT_CAST (demux),
|
|
gst_structure_new ("redirect",
|
|
"new-location", G_TYPE_STRING, uri, NULL)));
|
|
goto sent_redirect;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* we get here when we didn't do a redirect */
|
|
|
|
/* try to get and configure a manager */
|
|
if (!gst_sdp_demux_configure_manager (demux, uri))
|
|
goto no_manager;
|
|
if (!uri) {
|
|
/* create streams with UDP sources and sinks */
|
|
n_streams = gst_sdp_message_medias_len (&sdp);
|
|
for (i = 0; i < n_streams; i++) {
|
|
stream = gst_sdp_demux_create_stream (demux, &sdp, i);
|
|
|
|
if (!stream)
|
|
continue;
|
|
|
|
GST_DEBUG_OBJECT (demux, "configuring transport for stream %p", stream);
|
|
|
|
if (!gst_sdp_demux_stream_configure_udp (demux, stream))
|
|
goto transport_failed;
|
|
if (!gst_sdp_demux_stream_configure_udp_sink (demux, stream))
|
|
goto transport_failed;
|
|
}
|
|
|
|
if (!demux->streams)
|
|
goto no_streams;
|
|
}
|
|
|
|
/* set target state on session manager */
|
|
/* setting rtspsrc to PLAYING may cause it to loose it that target state
|
|
* along the way due to no-preroll udpsrc elements, so ...
|
|
* do it in two stages here (similar to other elements) */
|
|
if (demux->target > GST_STATE_PAUSED) {
|
|
ret = gst_element_set_state (demux->session, GST_STATE_PAUSED);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
goto start_session_failure;
|
|
}
|
|
ret = gst_element_set_state (demux->session, demux->target);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
goto start_session_failure;
|
|
|
|
if (!uri) {
|
|
/* activate all streams */
|
|
for (walk = demux->streams; walk; walk = g_list_next (walk)) {
|
|
stream = (GstSDPStream *) walk->data;
|
|
|
|
/* configure target state on udp sources */
|
|
gst_element_set_state (stream->udpsrc[0], demux->target);
|
|
gst_element_set_state (stream->udpsrc[1], demux->target);
|
|
}
|
|
}
|
|
GST_SDP_STREAM_UNLOCK (demux);
|
|
gst_sdp_message_uninit (&sdp);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
done:
|
|
{
|
|
GST_SDP_STREAM_UNLOCK (demux);
|
|
gst_sdp_message_uninit (&sdp);
|
|
return FALSE;
|
|
}
|
|
transport_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL),
|
|
("Could not create RTP stream transport."));
|
|
goto done;
|
|
}
|
|
no_manager:
|
|
{
|
|
GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL),
|
|
("Could not create RTP session manager."));
|
|
goto done;
|
|
}
|
|
could_not_parse:
|
|
{
|
|
GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL),
|
|
("Could not parse SDP message."));
|
|
goto done;
|
|
}
|
|
no_streams:
|
|
{
|
|
GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL),
|
|
("No streams in SDP message."));
|
|
goto done;
|
|
}
|
|
sent_redirect:
|
|
{
|
|
/* avoid hanging if redirect not handled */
|
|
GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL),
|
|
("Sent RTSP redirect."));
|
|
goto done;
|
|
}
|
|
start_session_failure:
|
|
{
|
|
GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL),
|
|
("Could not start RTP session manager."));
|
|
gst_element_set_state (demux->session, GST_STATE_NULL);
|
|
gst_bin_remove (GST_BIN_CAST (demux), demux->session);
|
|
demux->session = NULL;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_sdp_demux_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
|
|
{
|
|
GstSDPDemux *demux;
|
|
gboolean res = TRUE;
|
|
|
|
demux = GST_SDP_DEMUX (parent);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
/* when we get EOS, start parsing the SDP */
|
|
res = gst_sdp_demux_start (demux);
|
|
gst_event_unref (event);
|
|
break;
|
|
default:
|
|
gst_event_unref (event);
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_sdp_demux_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
|
|
{
|
|
GstSDPDemux *demux;
|
|
|
|
demux = GST_SDP_DEMUX (parent);
|
|
|
|
/* push the SDP message in an adapter, we start doing something with it when
|
|
* we receive EOS */
|
|
gst_adapter_push (demux->adapter, buffer);
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_sdp_demux_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstSDPDemux *demux;
|
|
GstStateChangeReturn ret;
|
|
|
|
demux = GST_SDP_DEMUX (element);
|
|
|
|
GST_SDP_STREAM_LOCK (demux);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
/* first attempt, don't ignore timeouts */
|
|
gst_adapter_clear (demux->adapter);
|
|
demux->ignore_timeout = FALSE;
|
|
demux->target = GST_STATE_PAUSED;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
demux->target = GST_STATE_PLAYING;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
goto done;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
demux->target = GST_STATE_PAUSED;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_sdp_demux_cleanup (demux);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
done:
|
|
GST_SDP_STREAM_UNLOCK (demux);
|
|
|
|
return ret;
|
|
}
|