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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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653 lines
18 KiB
C
653 lines
18 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.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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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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/**
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* SECTION:rtsp-sdp
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* @short_description: Make SDP messages
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* @see_also: #GstRTSPMedia
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*
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* Last reviewed on 2013-07-11 (1.0.0)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/net/net.h>
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#include <gst/sdp/gstmikey.h>
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#include "rtsp-sdp.h"
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static gboolean
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get_info_from_tags (GstPad * pad, GstEvent ** event, gpointer user_data)
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{
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GstSDPMedia *media = (GstSDPMedia *) user_data;
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if (GST_EVENT_TYPE (*event) == GST_EVENT_TAG) {
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GstTagList *tags;
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guint bitrate = 0;
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gst_event_parse_tag (*event, &tags);
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if (gst_tag_list_get_scope (tags) != GST_TAG_SCOPE_STREAM)
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return TRUE;
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if (!gst_tag_list_get_uint (tags, GST_TAG_MAXIMUM_BITRATE,
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&bitrate) || bitrate == 0)
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if (!gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &bitrate) ||
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bitrate == 0)
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return TRUE;
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/* set bandwidth (kbits/s) */
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gst_sdp_media_add_bandwidth (media, GST_SDP_BWTYPE_AS, bitrate / 1000);
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return FALSE;
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}
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return TRUE;
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}
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static void
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update_sdp_from_tags (GstRTSPStream * stream, GstSDPMedia * stream_media)
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{
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GstPad *src_pad;
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src_pad = gst_rtsp_stream_get_srcpad (stream);
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if (!src_pad)
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return;
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gst_pad_sticky_events_foreach (src_pad, get_info_from_tags, stream_media);
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gst_object_unref (src_pad);
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}
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static guint
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get_roc_from_stats (GstStructure * stats, guint ssrc)
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{
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const GValue *va, *v;
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guint i, len;
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/* initialize roc to something different than 0, so if we don't get
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the proper ROC from the encoder, streaming should fail initially. */
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guint roc = -1;
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va = gst_structure_get_value (stats, "streams");
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if (!va || !G_VALUE_HOLDS (va, GST_TYPE_ARRAY)) {
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GST_WARNING ("stats doesn't have a valid 'streams' field");
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return 0;
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}
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len = gst_value_array_get_size (va);
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/* look if there's any SSRC that matches. */
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for (i = 0; i < len; i++) {
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GstStructure *stream;
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v = gst_value_array_get_value (va, i);
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if (v && (stream = g_value_get_boxed (v))) {
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guint stream_ssrc;
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gst_structure_get_uint (stream, "ssrc", &stream_ssrc);
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if (stream_ssrc == ssrc) {
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gst_structure_get_uint (stream, "roc", &roc);
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break;
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}
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}
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}
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return roc;
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}
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static gboolean
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mikey_add_crypto_sessions (GstRTSPStream * stream, GstMIKEYMessage * msg)
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{
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guint i;
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GObject *session;
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GstElement *encoder;
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GValueArray *sources;
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gboolean roc_found;
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encoder = gst_rtsp_stream_get_srtp_encoder (stream);
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if (encoder == NULL) {
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GST_ERROR ("unable to get SRTP encoder from stream %p", stream);
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return FALSE;
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}
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session = gst_rtsp_stream_get_rtpsession (stream);
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if (session == NULL) {
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GST_ERROR ("unable to get RTP session from stream %p", stream);
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gst_object_unref (encoder);
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return FALSE;
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}
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roc_found = FALSE;
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g_object_get (session, "sources", &sources, NULL);
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for (i = 0; sources && (i < sources->n_values); i++) {
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GValue *val;
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GObject *source;
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guint32 ssrc;
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gboolean is_sender;
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val = g_value_array_get_nth (sources, i);
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source = (GObject *) g_value_get_object (val);
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g_object_get (source, "ssrc", &ssrc, "is-sender", &is_sender, NULL);
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if (is_sender) {
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guint32 roc = -1;
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GstStructure *stats;
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g_object_get (encoder, "stats", &stats, NULL);
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if (stats) {
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roc = get_roc_from_stats (stats, ssrc);
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gst_structure_free (stats);
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}
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roc_found = !!(roc != -1);
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if (!roc_found) {
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GST_ERROR ("unable to obtain ROC for stream %p with SSRC %u",
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stream, ssrc);
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goto cleanup;
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}
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GST_INFO ("stream %p with SSRC %u has a ROC of %u", stream, ssrc, roc);
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gst_mikey_message_add_cs_srtp (msg, 0, ssrc, roc);
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}
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}
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cleanup:
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{
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g_value_array_free (sources);
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gst_object_unref (encoder);
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g_object_unref (session);
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return roc_found;
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}
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}
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/**
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* gst_rtsp_sdp_make_media:
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* @sdp: a #GstRTSPMessage
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* @info: a #GstSDPInfo
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* @stream: a #GstRTSPStream
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* @caps: a #GstCaps
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* @profile: a #GstRTSPProfile
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*
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* Creates a #GstSDPMedia from the parameters and stores it in @sdp.
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*
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* Returns: %TRUE on success
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*
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* Since: 1.14
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*/
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gboolean
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gst_rtsp_sdp_make_media (GstSDPMessage * sdp, GstSDPInfo * info,
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GstRTSPStream * stream, GstCaps * caps, GstRTSPProfile profile)
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{
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GstSDPMedia *smedia;
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gchar *tmp;
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GstRTSPLowerTrans ltrans;
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GSocketFamily family;
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const gchar *addrtype, *proto;
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gchar *address;
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guint ttl;
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GstClockTime rtx_time;
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gchar *base64;
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GstMIKEYMessage *mikey_msg;
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gst_sdp_media_new (&smedia);
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if (gst_sdp_media_set_media_from_caps (caps, smedia) != GST_SDP_OK) {
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goto caps_error;
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}
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gst_sdp_media_set_port_info (smedia, 0, 1);
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switch (profile) {
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case GST_RTSP_PROFILE_AVP:
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proto = "RTP/AVP";
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break;
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case GST_RTSP_PROFILE_AVPF:
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proto = "RTP/AVPF";
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break;
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case GST_RTSP_PROFILE_SAVP:
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proto = "RTP/SAVP";
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break;
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case GST_RTSP_PROFILE_SAVPF:
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proto = "RTP/SAVPF";
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break;
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default:
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proto = "udp";
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break;
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}
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gst_sdp_media_set_proto (smedia, proto);
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if (info->is_ipv6) {
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addrtype = "IP6";
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family = G_SOCKET_FAMILY_IPV6;
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} else {
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addrtype = "IP4";
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family = G_SOCKET_FAMILY_IPV4;
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}
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ltrans = gst_rtsp_stream_get_protocols (stream);
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if (ltrans == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
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GstRTSPAddress *addr;
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addr = gst_rtsp_stream_get_multicast_address (stream, family);
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if (addr == NULL)
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goto no_multicast;
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address = g_strdup (addr->address);
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ttl = addr->ttl;
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gst_rtsp_address_free (addr);
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} else {
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ttl = 16;
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if (info->is_ipv6)
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address = g_strdup ("::");
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else
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address = g_strdup ("0.0.0.0");
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}
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/* for the c= line */
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gst_sdp_media_add_connection (smedia, "IN", addrtype, address, ttl, 1);
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g_free (address);
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/* the config uri */
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tmp = gst_rtsp_stream_get_control (stream);
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gst_sdp_media_add_attribute (smedia, "control", tmp);
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g_free (tmp);
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/* check for srtp */
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mikey_msg = gst_mikey_message_new_from_caps (caps);
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if (mikey_msg) {
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/* add policy '0' for all sending SSRC */
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if (!mikey_add_crypto_sessions (stream, mikey_msg)) {
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gst_mikey_message_unref (mikey_msg);
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goto crypto_sessions_error;
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}
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base64 = gst_mikey_message_base64_encode (mikey_msg);
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if (base64) {
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tmp = g_strdup_printf ("mikey %s", base64);
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g_free (base64);
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gst_sdp_media_add_attribute (smedia, "key-mgmt", tmp);
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g_free (tmp);
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}
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gst_mikey_message_unref (mikey_msg);
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}
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/* RFC 7273 clock signalling */
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if (gst_rtsp_stream_is_sender (stream)) {
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GstBin *joined_bin = gst_rtsp_stream_get_joined_bin (stream);
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GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (joined_bin));
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gchar *ts_refclk = NULL;
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gchar *mediaclk = NULL;
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guint rtptime, clock_rate;
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GstClockTime running_time, base_time, clock_time;
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GstRTSPPublishClockMode publish_clock_mode =
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gst_rtsp_stream_get_publish_clock_mode (stream);
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if (!gst_rtsp_stream_get_rtpinfo (stream, &rtptime, NULL, &clock_rate,
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&running_time))
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goto clock_signalling_cleanup;
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base_time = gst_element_get_base_time (GST_ELEMENT_CAST (joined_bin));
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g_assert (base_time != GST_CLOCK_TIME_NONE);
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clock_time = running_time + base_time;
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if (publish_clock_mode != GST_RTSP_PUBLISH_CLOCK_MODE_NONE && clock) {
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if (GST_IS_NTP_CLOCK (clock) || GST_IS_PTP_CLOCK (clock)) {
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if (publish_clock_mode == GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
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guint32 mediaclk_offset;
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/* Calculate RTP time at the clock's epoch. That's the direct offset */
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clock_time =
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gst_util_uint64_scale (clock_time, clock_rate, GST_SECOND);
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clock_time &= 0xffffffff;
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mediaclk_offset =
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G_GUINT64_CONSTANT (0xffffffff) + rtptime - clock_time;
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mediaclk = g_strdup_printf ("direct=%u", (guint32) mediaclk_offset);
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}
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if (GST_IS_NTP_CLOCK (clock)) {
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gchar *ntp_address;
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guint ntp_port;
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g_object_get (clock, "address", &ntp_address, "port", &ntp_port,
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NULL);
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if (ntp_port == 123)
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ts_refclk = g_strdup_printf ("ntp=%s", ntp_address);
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else
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ts_refclk = g_strdup_printf ("ntp=%s:%u", ntp_address, ntp_port);
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g_free (ntp_address);
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} else {
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guint64 ptp_clock_id;
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guint ptp_domain;
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g_object_get (clock, "grandmaster-clock-id", &ptp_clock_id, "domain",
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&ptp_domain, NULL);
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if (ptp_domain != 0)
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ts_refclk =
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g_strdup_printf
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("ptp=IEEE1588-2008:%02X-%02X-%02X-%02X-%02X-%02X-%02X-%02X:%u",
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(guint) (ptp_clock_id >> 56) & 0xff,
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(guint) (ptp_clock_id >> 48) & 0xff,
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(guint) (ptp_clock_id >> 40) & 0xff,
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(guint) (ptp_clock_id >> 32) & 0xff,
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(guint) (ptp_clock_id >> 24) & 0xff,
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(guint) (ptp_clock_id >> 16) & 0xff,
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(guint) (ptp_clock_id >> 8) & 0xff,
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(guint) (ptp_clock_id >> 0) & 0xff, ptp_domain);
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else
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ts_refclk =
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g_strdup_printf
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("ptp=IEEE1588-2008:%02X-%02X-%02X-%02X-%02X-%02X-%02X-%02X",
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(guint) (ptp_clock_id >> 56) & 0xff,
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(guint) (ptp_clock_id >> 48) & 0xff,
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(guint) (ptp_clock_id >> 40) & 0xff,
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(guint) (ptp_clock_id >> 32) & 0xff,
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(guint) (ptp_clock_id >> 24) & 0xff,
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(guint) (ptp_clock_id >> 16) & 0xff,
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(guint) (ptp_clock_id >> 8) & 0xff,
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(guint) (ptp_clock_id >> 0) & 0xff);
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}
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}
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}
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clock_signalling_cleanup:
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if (clock)
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gst_object_unref (clock);
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if (!ts_refclk)
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ts_refclk = g_strdup ("local");
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if (!mediaclk)
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mediaclk = g_strdup ("sender");
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gst_sdp_media_add_attribute (smedia, "ts-refclk", ts_refclk);
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gst_sdp_media_add_attribute (smedia, "mediaclk", mediaclk);
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g_free (ts_refclk);
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g_free (mediaclk);
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gst_object_unref (joined_bin);
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}
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update_sdp_from_tags (stream, smedia);
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if (profile == GST_RTSP_PROFILE_AVPF || profile == GST_RTSP_PROFILE_SAVPF) {
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if ((rtx_time = gst_rtsp_stream_get_retransmission_time (stream))) {
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/* ssrc multiplexed retransmit functionality */
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guint rtx_pt = gst_rtsp_stream_get_retransmission_pt (stream);
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if (rtx_pt == 0) {
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g_warning ("failed to find an available dynamic payload type. "
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"Not adding retransmission");
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} else {
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gchar *tmp;
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GstStructure *s;
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gint caps_pt, caps_rate;
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s = gst_caps_get_structure (caps, 0);
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if (s == NULL)
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goto no_caps_info;
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/* get payload type and clock rate */
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gst_structure_get_int (s, "payload", &caps_pt);
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gst_structure_get_int (s, "clock-rate", &caps_rate);
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tmp = g_strdup_printf ("%d", rtx_pt);
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gst_sdp_media_add_format (smedia, tmp);
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g_free (tmp);
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tmp = g_strdup_printf ("%d rtx/%d", rtx_pt, caps_rate);
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gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
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g_free (tmp);
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tmp =
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g_strdup_printf ("%d apt=%d;rtx-time=%" G_GINT64_FORMAT, rtx_pt,
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caps_pt, GST_TIME_AS_MSECONDS (rtx_time));
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gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
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g_free (tmp);
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}
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}
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if (gst_rtsp_stream_get_ulpfec_percentage (stream)) {
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guint ulpfec_pt = gst_rtsp_stream_get_ulpfec_pt (stream);
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if (ulpfec_pt == 0) {
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g_warning ("failed to find an available dynamic payload type. "
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"Not adding ulpfec");
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} else {
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gchar *tmp;
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GstStructure *s;
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gint caps_pt, caps_rate;
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s = gst_caps_get_structure (caps, 0);
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if (s == NULL)
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goto no_caps_info;
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/* get payload type and clock rate */
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gst_structure_get_int (s, "payload", &caps_pt);
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gst_structure_get_int (s, "clock-rate", &caps_rate);
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tmp = g_strdup_printf ("%d", ulpfec_pt);
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gst_sdp_media_add_format (smedia, tmp);
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g_free (tmp);
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tmp = g_strdup_printf ("%d ulpfec/%d", ulpfec_pt, caps_rate);
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gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
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g_free (tmp);
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tmp = g_strdup_printf ("%d apt=%d", ulpfec_pt, caps_pt);
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gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
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g_free (tmp);
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}
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}
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}
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/* RFC5576 Source-specific media attributes */
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{
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GObject *session;
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guint ssrc;
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GstStructure *sdes;
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const gchar *cname;
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gchar *ssrc_cname;
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session = gst_rtsp_stream_get_rtpsession (stream);
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if (session) {
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g_object_get (session, "sdes", &sdes, NULL);
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cname = gst_structure_get_string (sdes, "cname");
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gst_rtsp_stream_get_ssrc (stream, &ssrc);
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if (cname) {
|
|
ssrc_cname = g_strdup_printf ("%u cname:%s", ssrc, cname);
|
|
gst_sdp_media_add_attribute (smedia, "ssrc", ssrc_cname);
|
|
g_free (ssrc_cname);
|
|
} else {
|
|
GST_ERROR ("unable to get CNAME for stream %p", stream);
|
|
}
|
|
gst_structure_free (sdes);
|
|
g_object_unref (session);
|
|
} else {
|
|
GST_ERROR ("unable to get RTP session from stream %p", stream);
|
|
}
|
|
}
|
|
|
|
gst_sdp_message_add_media (sdp, smedia);
|
|
gst_sdp_media_free (smedia);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
caps_error:
|
|
{
|
|
gst_sdp_media_free (smedia);
|
|
GST_ERROR ("unable to set media from caps for stream %d",
|
|
gst_rtsp_stream_get_index (stream));
|
|
return FALSE;
|
|
}
|
|
no_multicast:
|
|
{
|
|
gst_sdp_media_free (smedia);
|
|
GST_ERROR ("stream %d has no multicast address",
|
|
gst_rtsp_stream_get_index (stream));
|
|
return FALSE;
|
|
}
|
|
no_caps_info:
|
|
{
|
|
gst_sdp_media_free (smedia);
|
|
GST_ERROR ("caps for stream %d have no structure",
|
|
gst_rtsp_stream_get_index (stream));
|
|
return FALSE;
|
|
}
|
|
crypto_sessions_error:
|
|
{
|
|
gst_sdp_media_free (smedia);
|
|
GST_ERROR ("unable to add MIKEY crypto sessions for stream %d",
|
|
gst_rtsp_stream_get_index (stream));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_sdp_from_media:
|
|
* @sdp: a #GstSDPMessage
|
|
* @info: (transfer none): a #GstSDPInfo
|
|
* @media: (transfer none): a #GstRTSPMedia
|
|
*
|
|
* Add @media specific info to @sdp. @info is used to configure the connection
|
|
* information in the SDP.
|
|
*
|
|
* Returns: TRUE on success.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info,
|
|
GstRTSPMedia * media)
|
|
{
|
|
guint i, n_streams;
|
|
gchar *rangestr;
|
|
gboolean res;
|
|
|
|
n_streams = gst_rtsp_media_n_streams (media);
|
|
|
|
rangestr = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
|
|
if (rangestr == NULL)
|
|
goto not_prepared;
|
|
|
|
gst_sdp_message_add_attribute (sdp, "range", rangestr);
|
|
g_free (rangestr);
|
|
|
|
res = TRUE;
|
|
for (i = 0; res && (i < n_streams); i++) {
|
|
GstRTSPStream *stream;
|
|
|
|
stream = gst_rtsp_media_get_stream (media, i);
|
|
res = gst_rtsp_sdp_from_stream (sdp, info, stream);
|
|
if (!res) {
|
|
GST_ERROR ("could not get SDP from stream %p", stream);
|
|
goto sdp_error;
|
|
}
|
|
}
|
|
|
|
{
|
|
GstNetTimeProvider *provider;
|
|
|
|
if ((provider =
|
|
gst_rtsp_media_get_time_provider (media, info->server_ip, 0))) {
|
|
GstClock *clock;
|
|
gchar *address, *str;
|
|
gint port;
|
|
|
|
g_object_get (provider, "clock", &clock, "address", &address, "port",
|
|
&port, NULL);
|
|
|
|
str = g_strdup_printf ("GstNetTimeProvider %s %s:%d %" G_GUINT64_FORMAT,
|
|
g_type_name (G_TYPE_FROM_INSTANCE (clock)), address, port,
|
|
gst_clock_get_time (clock));
|
|
|
|
gst_sdp_message_add_attribute (sdp, "x-gst-clock", str);
|
|
g_free (str);
|
|
gst_object_unref (clock);
|
|
g_free (address);
|
|
gst_object_unref (provider);
|
|
}
|
|
}
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
not_prepared:
|
|
{
|
|
GST_ERROR ("media %p is not prepared", media);
|
|
return FALSE;
|
|
}
|
|
sdp_error:
|
|
{
|
|
GST_ERROR ("could not get SDP from media %p", media);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_sdp_from_stream:
|
|
* @sdp: a #GstSDPMessage
|
|
* @info: (transfer none): a #GstSDPInfo
|
|
* @stream: (transfer none): a #GstRTSPStream
|
|
*
|
|
* Add info from @stream to @sdp.
|
|
*
|
|
* Returns: TRUE on success.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_sdp_from_stream (GstSDPMessage * sdp, GstSDPInfo * info,
|
|
GstRTSPStream * stream)
|
|
{
|
|
GstCaps *caps;
|
|
GstRTSPProfile profiles;
|
|
guint mask;
|
|
gboolean res;
|
|
|
|
caps = gst_rtsp_stream_get_caps (stream);
|
|
|
|
if (caps == NULL) {
|
|
GST_ERROR ("stream %p has no caps", stream);
|
|
return FALSE;
|
|
}
|
|
|
|
/* make a new media for each profile */
|
|
profiles = gst_rtsp_stream_get_profiles (stream);
|
|
mask = 1;
|
|
res = TRUE;
|
|
while (res && (profiles >= mask)) {
|
|
GstRTSPProfile prof = profiles & mask;
|
|
|
|
if (prof)
|
|
res = gst_rtsp_sdp_make_media (sdp, info, stream, caps, prof);
|
|
|
|
mask <<= 1;
|
|
}
|
|
gst_caps_unref (caps);
|
|
|
|
return res;
|
|
}
|