gstreamer/subprojects/gst-plugins-bad/ext/webrtc/utils.c
Matthew Waters 993bc8fc01 webrtc: implement support for msid values
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>
2022-12-14 12:23:32 +11:00

250 lines
6 KiB
C

/* GStreamer
* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include "utils.h"
#include "gstwebrtcbin.h"
GstPadTemplate *
_find_pad_template (GstElement * element, GstPadDirection direction,
GstPadPresence presence, const gchar * name)
{
GstElementClass *element_class = GST_ELEMENT_GET_CLASS (element);
const GList *l = gst_element_class_get_pad_template_list (element_class);
GstPadTemplate *templ = NULL;
for (; l; l = l->next) {
templ = l->data;
if (templ->direction != direction)
continue;
if (templ->presence != presence)
continue;
if (g_strcmp0 (templ->name_template, name) == 0) {
return templ;
}
}
return NULL;
}
GstSDPMessage *
_get_latest_offer (GstWebRTCBin * webrtc)
{
if (webrtc->current_local_description &&
webrtc->current_local_description->type == GST_WEBRTC_SDP_TYPE_OFFER) {
return webrtc->current_local_description->sdp;
}
if (webrtc->current_remote_description &&
webrtc->current_remote_description->type == GST_WEBRTC_SDP_TYPE_OFFER) {
return webrtc->current_remote_description->sdp;
}
return NULL;
}
GstSDPMessage *
_get_latest_answer (GstWebRTCBin * webrtc)
{
if (webrtc->current_local_description &&
webrtc->current_local_description->type == GST_WEBRTC_SDP_TYPE_ANSWER) {
return webrtc->current_local_description->sdp;
}
if (webrtc->current_remote_description &&
webrtc->current_remote_description->type == GST_WEBRTC_SDP_TYPE_ANSWER) {
return webrtc->current_remote_description->sdp;
}
return NULL;
}
GstSDPMessage *
_get_latest_sdp (GstWebRTCBin * webrtc)
{
GstSDPMessage *ret = NULL;
if ((ret = _get_latest_answer (webrtc)))
return ret;
if ((ret = _get_latest_offer (webrtc)))
return ret;
return NULL;
}
GstSDPMessage *
_get_latest_self_generated_sdp (GstWebRTCBin * webrtc)
{
if (webrtc->priv->last_generated_answer)
return webrtc->priv->last_generated_answer->sdp;
if (webrtc->priv->last_generated_offer)
return webrtc->priv->last_generated_offer->sdp;
return NULL;
}
struct pad_block *
_create_pad_block (GstElement * element, GstPad * pad, gulong block_id,
gpointer user_data, GDestroyNotify notify)
{
struct pad_block *ret = g_new0 (struct pad_block, 1);
ret->element = gst_object_ref (element);
ret->pad = gst_object_ref (pad);
ret->block_id = block_id;
ret->user_data = user_data;
ret->notify = notify;
return ret;
}
void
_free_pad_block (struct pad_block *block)
{
if (!block)
return;
if (block->block_id)
gst_pad_remove_probe (block->pad, block->block_id);
gst_object_unref (block->element);
gst_object_unref (block->pad);
if (block->notify)
block->notify (block->user_data);
g_free (block);
}
const gchar *
_enum_value_to_string (GType type, guint value)
{
GEnumClass *enum_class;
GEnumValue *enum_value;
const gchar *str = NULL;
enum_class = g_type_class_ref (type);
enum_value = g_enum_get_value (enum_class, value);
if (enum_value)
str = enum_value->value_nick;
g_type_class_unref (enum_class);
return str;
}
const gchar *
_g_checksum_to_webrtc_string (GChecksumType type)
{
switch (type) {
case G_CHECKSUM_SHA1:
return "sha-1";
case G_CHECKSUM_SHA256:
return "sha-256";
#ifdef G_CHECKSUM_SHA384
case G_CHECKSUM_SHA384:
return "sha-384";
#endif
case G_CHECKSUM_SHA512:
return "sha-512";
default:
g_warning ("unknown GChecksumType!");
return NULL;
}
}
GstCaps *
_rtp_caps_from_media (const GstSDPMedia * media)
{
GstCaps *ret;
int i, j;
ret = gst_caps_new_empty ();
for (i = 0; i < gst_sdp_media_formats_len (media); i++) {
guint pt = atoi (gst_sdp_media_get_format (media, i));
GstCaps *caps;
caps = gst_sdp_media_get_caps_from_media (media, pt);
if (!caps)
continue;
/* gst_sdp_media_get_caps_from_media() produces caps with name
* "application/x-unknown" which will fail intersection with
* "application/x-rtp" caps so mangle the returns caps to have the
* correct name here */
for (j = 0; j < gst_caps_get_size (caps); j++) {
GstStructure *s = gst_caps_get_structure (caps, j);
gst_structure_set_name (s, "application/x-rtp");
}
gst_caps_append (ret, caps);
}
return ret;
}
GstWebRTCKind
webrtc_kind_from_caps (const GstCaps * caps)
{
GstStructure *s;
const gchar *media;
if (!caps || gst_caps_get_size (caps) == 0)
return GST_WEBRTC_KIND_UNKNOWN;
s = gst_caps_get_structure (caps, 0);
media = gst_structure_get_string (s, "media");
if (media == NULL)
return GST_WEBRTC_KIND_UNKNOWN;
if (!g_strcmp0 (media, "audio"))
return GST_WEBRTC_KIND_AUDIO;
if (!g_strcmp0 (media, "video"))
return GST_WEBRTC_KIND_VIDEO;
return GST_WEBRTC_KIND_UNKNOWN;
}
char *
_get_msid_from_media (const GstSDPMedia * media)
{
int i;
for (i = 0; i < gst_sdp_media_attributes_len (media); i++) {
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, i);
const char *start, *end;
if (!attr->value)
continue;
start = strstr (attr->value, "msid:");
if (!start)
continue;
start += strlen ("msid:");
end = strstr (start, " ");
if (end)
return g_strndup (start, end - start);
}
return NULL;
}