mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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993bc8fc01
Local msid values are taken from sink pad property, or fallback to the previously used cname. The remote msid values are exposed on the relevant src pads. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3106>
250 lines
6 KiB
C
250 lines
6 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <stdlib.h>
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#include "utils.h"
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#include "gstwebrtcbin.h"
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GstPadTemplate *
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_find_pad_template (GstElement * element, GstPadDirection direction,
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GstPadPresence presence, const gchar * name)
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{
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GstElementClass *element_class = GST_ELEMENT_GET_CLASS (element);
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const GList *l = gst_element_class_get_pad_template_list (element_class);
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GstPadTemplate *templ = NULL;
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for (; l; l = l->next) {
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templ = l->data;
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if (templ->direction != direction)
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continue;
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if (templ->presence != presence)
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continue;
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if (g_strcmp0 (templ->name_template, name) == 0) {
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return templ;
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}
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}
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return NULL;
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}
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GstSDPMessage *
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_get_latest_offer (GstWebRTCBin * webrtc)
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{
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if (webrtc->current_local_description &&
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webrtc->current_local_description->type == GST_WEBRTC_SDP_TYPE_OFFER) {
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return webrtc->current_local_description->sdp;
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}
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if (webrtc->current_remote_description &&
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webrtc->current_remote_description->type == GST_WEBRTC_SDP_TYPE_OFFER) {
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return webrtc->current_remote_description->sdp;
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}
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return NULL;
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}
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GstSDPMessage *
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_get_latest_answer (GstWebRTCBin * webrtc)
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{
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if (webrtc->current_local_description &&
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webrtc->current_local_description->type == GST_WEBRTC_SDP_TYPE_ANSWER) {
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return webrtc->current_local_description->sdp;
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}
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if (webrtc->current_remote_description &&
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webrtc->current_remote_description->type == GST_WEBRTC_SDP_TYPE_ANSWER) {
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return webrtc->current_remote_description->sdp;
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}
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return NULL;
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}
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GstSDPMessage *
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_get_latest_sdp (GstWebRTCBin * webrtc)
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{
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GstSDPMessage *ret = NULL;
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if ((ret = _get_latest_answer (webrtc)))
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return ret;
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if ((ret = _get_latest_offer (webrtc)))
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return ret;
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return NULL;
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}
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GstSDPMessage *
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_get_latest_self_generated_sdp (GstWebRTCBin * webrtc)
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{
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if (webrtc->priv->last_generated_answer)
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return webrtc->priv->last_generated_answer->sdp;
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if (webrtc->priv->last_generated_offer)
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return webrtc->priv->last_generated_offer->sdp;
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return NULL;
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}
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struct pad_block *
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_create_pad_block (GstElement * element, GstPad * pad, gulong block_id,
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gpointer user_data, GDestroyNotify notify)
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{
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struct pad_block *ret = g_new0 (struct pad_block, 1);
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ret->element = gst_object_ref (element);
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ret->pad = gst_object_ref (pad);
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ret->block_id = block_id;
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ret->user_data = user_data;
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ret->notify = notify;
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return ret;
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}
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void
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_free_pad_block (struct pad_block *block)
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{
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if (!block)
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return;
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if (block->block_id)
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gst_pad_remove_probe (block->pad, block->block_id);
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gst_object_unref (block->element);
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gst_object_unref (block->pad);
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if (block->notify)
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block->notify (block->user_data);
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g_free (block);
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}
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const gchar *
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_enum_value_to_string (GType type, guint value)
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{
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GEnumClass *enum_class;
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GEnumValue *enum_value;
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const gchar *str = NULL;
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enum_class = g_type_class_ref (type);
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enum_value = g_enum_get_value (enum_class, value);
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if (enum_value)
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str = enum_value->value_nick;
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g_type_class_unref (enum_class);
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return str;
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}
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const gchar *
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_g_checksum_to_webrtc_string (GChecksumType type)
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{
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switch (type) {
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case G_CHECKSUM_SHA1:
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return "sha-1";
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case G_CHECKSUM_SHA256:
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return "sha-256";
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#ifdef G_CHECKSUM_SHA384
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case G_CHECKSUM_SHA384:
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return "sha-384";
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#endif
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case G_CHECKSUM_SHA512:
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return "sha-512";
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default:
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g_warning ("unknown GChecksumType!");
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return NULL;
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}
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}
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GstCaps *
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_rtp_caps_from_media (const GstSDPMedia * media)
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{
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GstCaps *ret;
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int i, j;
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ret = gst_caps_new_empty ();
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for (i = 0; i < gst_sdp_media_formats_len (media); i++) {
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guint pt = atoi (gst_sdp_media_get_format (media, i));
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GstCaps *caps;
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caps = gst_sdp_media_get_caps_from_media (media, pt);
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if (!caps)
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continue;
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/* gst_sdp_media_get_caps_from_media() produces caps with name
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* "application/x-unknown" which will fail intersection with
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* "application/x-rtp" caps so mangle the returns caps to have the
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* correct name here */
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for (j = 0; j < gst_caps_get_size (caps); j++) {
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GstStructure *s = gst_caps_get_structure (caps, j);
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gst_structure_set_name (s, "application/x-rtp");
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}
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gst_caps_append (ret, caps);
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}
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return ret;
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}
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GstWebRTCKind
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webrtc_kind_from_caps (const GstCaps * caps)
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{
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GstStructure *s;
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const gchar *media;
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if (!caps || gst_caps_get_size (caps) == 0)
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return GST_WEBRTC_KIND_UNKNOWN;
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s = gst_caps_get_structure (caps, 0);
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media = gst_structure_get_string (s, "media");
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if (media == NULL)
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return GST_WEBRTC_KIND_UNKNOWN;
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if (!g_strcmp0 (media, "audio"))
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return GST_WEBRTC_KIND_AUDIO;
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if (!g_strcmp0 (media, "video"))
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return GST_WEBRTC_KIND_VIDEO;
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return GST_WEBRTC_KIND_UNKNOWN;
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}
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char *
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_get_msid_from_media (const GstSDPMedia * media)
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{
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int i;
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for (i = 0; i < gst_sdp_media_attributes_len (media); i++) {
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const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, i);
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const char *start, *end;
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if (!attr->value)
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continue;
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start = strstr (attr->value, "msid:");
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if (!start)
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continue;
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start += strlen ("msid:");
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end = strstr (start, " ");
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if (end)
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return g_strndup (start, end - start);
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}
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return NULL;
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}
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