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Purism-Calls/plugins/sip/calls-sip-media-manager.c

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/*
* Copyright (C) 2021 Purism SPC
*
* This file is part of Calls.
*
* Calls is free software: you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* Calls 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
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with Calls. If not, see <http://www.gnu.org/licenses/>.
*
* Author: Evangelos Ribeiro Tzaras <evangelos.tzaras@puri.sm>
*
* SPDX-License-Identifier: GPL-3.0-or-later
*
*/
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#define G_LOG_DOMAIN "CallsSipMediaManager"
#include "calls-sip-media-manager.h"
#include "gst-rfc3551.h"
#include <gst/gst.h>
/**
* SECTION:sip-media-manager
* @short_description: The media manager singleton
* @Title: CallsSipMediaManager
*
* #CallsSipMediaManager is mainly responsible for generating appropriate
* SDP messages for the set of supported codecs. In the future it
* shall also manage the #CallsSipMediaPipeline objects that are in use.
*/
typedef struct _CallsSipMediaManager
{
GObject parent;
GList *supported_codecs;
} CallsSipMediaManager;
G_DEFINE_TYPE (CallsSipMediaManager, calls_sip_media_manager, G_TYPE_OBJECT);
static void
calls_sip_media_manager_finalize (GObject *object)
{
gst_deinit ();
g_list_free (CALLS_SIP_MEDIA_MANAGER (object)->supported_codecs);
G_OBJECT_CLASS (calls_sip_media_manager_parent_class)->finalize (object);
}
static void
calls_sip_media_manager_class_init (CallsSipMediaManagerClass *klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
object_class->finalize = calls_sip_media_manager_finalize;
}
static void
calls_sip_media_manager_init (CallsSipMediaManager *self)
{
gst_init (NULL, NULL);
self->supported_codecs = media_codecs_get_candidates ();
}
/* Public functions */
CallsSipMediaManager *
calls_sip_media_manager_default (void)
{
static CallsSipMediaManager *instance = NULL;
if (instance == NULL) {
g_debug ("Creating CallsSipMediaManager");
instance = g_object_new (CALLS_TYPE_SIP_MEDIA_MANAGER, NULL);
g_object_add_weak_pointer (G_OBJECT (instance), (gpointer *) &instance);
}
return instance;
}
/* calls_sip_media_manager_get_capabilities:
*
* @self: A #CallsSipMediaManager
* @port: Should eventually come from the ICE stack
* @use_srtp: Whether to use srtp (not really handled)
* @supported_codecs: A #GList of #MediaCodecInfo
*
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* Returns: (transfer full): string describing capabilities
* to be used in the session description (SDP)
*/
char *
calls_sip_media_manager_get_capabilities (CallsSipMediaManager *self,
guint port,
gboolean use_srtp,
GList *supported_codecs)
{
char *payload_type = use_srtp ? "SAVP" : "AVP";
g_autoptr (GString) media_line = NULL;
g_autoptr (GString) attribute_lines = NULL;
GList *node;
g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
media_line = g_string_new (NULL);
attribute_lines = g_string_new (NULL);
if (supported_codecs == NULL) {
g_warning ("No supported codecs found. Can't build meaningful SDP message");
g_string_append_printf (media_line, "m=audio 0 RTP/AVP");
goto done;
}
/* media lines look f.e like "audio 31337 RTP/AVP 9 8 0" */
g_string_append_printf (media_line,
"m=audio %d RTP/%s", port, payload_type);
for (node = supported_codecs; node != NULL; node = node->next) {
MediaCodecInfo *codec = node->data;
g_string_append_printf (media_line, " %u", codec->payload_id);
g_string_append_printf (attribute_lines,
"a=rtpmap:%u %s/%u%s",
codec->payload_id,
codec->name,
codec->clock_rate,
"\r\n");
}
g_string_append_printf (attribute_lines, "a=rtcp:%d\r\n", port + 1);
done:
return g_strdup_printf ("v=0\r\n"
"%s\r\n"
"%s\r\n",
media_line->str,
attribute_lines->str);
}
/* calls_sip_media_manager_static_capabilities:
*
* @self: A #CallsSipMediaManager
* @port: Should eventually come from the ICE stack
* @use_srtp: Whether to use srtp (not really handled)
*
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* Returns: (transfer full): string describing capabilities
* to be used in the session description (SDP)
*/
char *
calls_sip_media_manager_static_capabilities (CallsSipMediaManager *self,
guint port,
gboolean use_srtp)
{
g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
return calls_sip_media_manager_get_capabilities (self,
port,
use_srtp,
self->supported_codecs);
}
MediaCodecInfo*
get_best_codec (CallsSipMediaManager *self)
{
return media_codec_by_name ("PCMA");
}
/* calls_sip_media_manager_codec_candiates:
*
* @self: A #CallsSipMediaManager
*
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* Returns: (transfer none): A #GList of supported
* #MediaCodecInfo
*/
GList *
calls_sip_media_manager_codec_candidates (CallsSipMediaManager *self)
{
g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
return self->supported_codecs;
}
/* calls_sip_media_manager_get_codecs_from_sdp
*
* @self: A #CallsSipMediaManager
* @sdp: A #sdp_media_t media description
*
* Returns: (transfer container): A #GList of codecs found in the
* SDP message
*/
GList *
calls_sip_media_manager_get_codecs_from_sdp (CallsSipMediaManager *self,
sdp_media_t *sdp_media)
{
GList *codecs = NULL;
sdp_rtpmap_t *rtpmap = NULL;
g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
g_return_val_if_fail (sdp_media, NULL);
if (sdp_media->m_type != sdp_media_audio) {
g_warning ("Only the 'audio' media type is supported");
return NULL;
}
for (rtpmap = sdp_media->m_rtpmaps; rtpmap != NULL; rtpmap = rtpmap->rm_next) {
MediaCodecInfo *codec = media_codec_by_payload_id (rtpmap->rm_pt);
if (codec)
codecs = g_list_append (codecs, codec);
}
if (sdp_media->m_next != NULL)
g_warning ("Currently only a single media session is supported");
if (codecs == NULL)
g_warning ("Did not find any common codecs");
return codecs;
}