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/* 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"
GQuark
gst_webrtc_bin_error_quark (void)
{
return g_quark_from_static_string ("gst-webrtc-bin-error-quark");
}
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);
}
gchar *
_enum_value_to_string (GType type, guint value)
{
GEnumClass *enum_class;
GEnumValue *enum_value;
gchar *str = NULL;
enum_class = g_type_class_ref (type);
enum_value = g_enum_get_value (enum_class, value);
if (enum_value)
str = g_strdup (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 (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;
}
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