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Purism-Calls/plugins/provider/sip/calls-sip-media-manager.c
Evangelos Ribeiro Tzaras bb6b76107c build: Rename util.{c,h} to calls-util.{c,h}
"util" is a very generic name. This guards against accidentally
including similarly named headers from elsewhere.
2023-01-22 07:33:02 +00:00

380 lines
11 KiB
C

/*
* Copyright (C) 2021-2022 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
*
*/
#define G_LOG_DOMAIN "CallsSipMediaManager"
#include "calls-settings.h"
#include "calls-sip-media-manager.h"
#include "calls-sip-media-pipeline.h"
#include "calls-srtp-utils.h"
#include "gst-rfc3551.h"
#include "calls-util.h"
#include <gio/gio.h>
#include <gst/gst.h>
#include <sys/socket.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. It also holds a list of
* #CallsSipMediaPipeline objects that are ready to be used.
*/
typedef struct _CallsSipMediaManager {
GObject parent;
CallsSettings *settings;
GList *preferred_codecs;
GListStore *pipelines;
} CallsSipMediaManager;
G_DEFINE_TYPE (CallsSipMediaManager, calls_sip_media_manager, G_TYPE_OBJECT);
static void
on_notify_preferred_audio_codecs (CallsSipMediaManager *self)
{
GList *supported_codecs;
g_auto (GStrv) settings_codec_preference = NULL;
g_assert (CALLS_IS_SIP_MEDIA_MANAGER (self));
g_clear_list (&self->preferred_codecs, NULL);
supported_codecs = media_codecs_get_candidates ();
if (!supported_codecs) {
g_warning ("There aren't any supported codecs installed on your system");
return;
}
settings_codec_preference = calls_settings_get_preferred_audio_codecs (self->settings);
if (!settings_codec_preference) {
g_debug ("No audio codec preference set. Using all supported codecs");
self->preferred_codecs = supported_codecs;
return;
}
for (guint i = 0; settings_codec_preference[i] != NULL; i++) {
MediaCodecInfo *codec = media_codec_by_name (settings_codec_preference[i]);
if (!codec) {
g_debug ("Did not find audio codec %s", settings_codec_preference[i]);
continue;
}
if (media_codec_available_in_gst (codec))
self->preferred_codecs = g_list_append (self->preferred_codecs, codec);
}
if (!self->preferred_codecs) {
g_warning ("Cannot satisfy audio codec preference, "
"falling back to all supported codecs");
self->preferred_codecs = supported_codecs;
} else {
g_list_free (supported_codecs);
}
}
static void
add_new_pipeline (CallsSipMediaManager *self)
{
CallsSipMediaPipeline *pipeline;
g_assert (CALLS_IS_SIP_MEDIA_MANAGER (self));
pipeline = calls_sip_media_pipeline_new (NULL);
g_list_store_append (self->pipelines, pipeline);
}
static void
calls_sip_media_manager_finalize (GObject *object)
{
CallsSipMediaManager *self = CALLS_SIP_MEDIA_MANAGER (object);
g_list_free (self->preferred_codecs);
g_object_unref (self->pipelines);
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)
{
if (!gst_is_initialized ())
gst_init (NULL, NULL);
self->settings = calls_settings_get_default ();
g_signal_connect_swapped (self->settings,
"notify::preferred-audio-codecs",
G_CALLBACK (on_notify_preferred_audio_codecs),
self);
on_notify_preferred_audio_codecs (self);
self->pipelines = g_list_store_new (CALLS_TYPE_SIP_MEDIA_PIPELINE);
add_new_pipeline (self);
}
/* 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;
}
/* helper for calls_sip_media_manager_get_capabilities() */
static char *
get_connection_line (const char *ip)
{
int af_family;
if (STR_IS_NULL_OR_EMPTY(ip))
goto invalid;
af_family = get_address_family_for_ip (ip, TRUE);
if (af_family == AF_UNSPEC)
goto invalid;
return g_strdup_printf ("c=IN %s %s\r\n",
af_family == AF_INET ? "IP4" : "IP6",
ip);
invalid:
return NULL;
}
/* calls_sip_media_manager_get_capabilities:
*
* @self: A #CallsSipMediaManager
* @port: Should eventually come from the ICE stack
* @crypto_attributes: A #GList of #calls_srtp_crypto_attribute
* @supported_codecs: A #GList of #MediaCodecInfo
*
* Returns: (transfer full): string describing capabilities
* to be used in the session description (SDP)
*/
char *
calls_sip_media_manager_get_capabilities (CallsSipMediaManager *self,
const char *own_ip,
gint rtp_port,
gint rtcp_port,
GList *crypto_attributes,
GList *supported_codecs)
{
char *payload_type = crypto_attributes ? "SAVP" : "AVP";
g_autoptr (GString) media_line = NULL;
g_autoptr (GString) attribute_lines = NULL;
g_autofree char *connection_line = 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", rtp_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");
}
for (node = crypto_attributes; node != NULL; node = node->next) {
calls_srtp_crypto_attribute *attr = node->data;
g_autoptr (GError) error = NULL;
g_autofree char *crypto_line =
calls_srtp_print_sdp_crypto_attribute (attr, &error);
if (!crypto_line) {
g_warning ("Could not print SDP crypto line for tag %d: %s", attr->tag, error->message);
continue;
}
g_string_append_printf (attribute_lines, "%s\r\n", crypto_line);
}
g_string_append_printf (attribute_lines, "a=rtcp:%d\r\n", rtcp_port);
done:
connection_line = get_connection_line (own_ip);
return g_strdup_printf ("v=0\r\n"
"%s"
"%s\r\n"
"%s\r\n",
connection_line ?: "",
media_line->str,
attribute_lines->str);
}
/* calls_sip_media_manager_static_capabilities:
*
* @self: A #CallsSipMediaManager
* @rtp_port: Port to use for RTP. Should eventually come from the ICE stack
* @rtcp_port: Port to use for RTCP.Should eventually come from the ICE stack
* @crypto_attributes: A #GList of #calls_srtp_crypto_attribute
*
* Returns: (transfer full): string describing capabilities
* to be used in the session description (SDP)
*/
char *
calls_sip_media_manager_static_capabilities (CallsSipMediaManager *self,
const char *own_ip,
gint rtp_port,
gint rtcp_port,
GList *crypto_attributes)
{
g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
return calls_sip_media_manager_get_capabilities (self,
own_ip,
rtp_port,
rtcp_port,
crypto_attributes,
self->preferred_codecs);
}
/* calls_sip_media_manager_codec_candiates:
*
* @self: A #CallsSipMediaManager
*
* 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->preferred_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;
}
/**
* calls_sip_media_manager_get_pipeline:
* @self: A #CallsSipMediaManager
*
* Returns: (transfer full): A #CallsSipMediaPipeline
*/
CallsSipMediaPipeline *
calls_sip_media_manager_get_pipeline (CallsSipMediaManager *self)
{
g_autoptr (CallsSipMediaPipeline) pipeline = NULL;
g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
pipeline = g_list_model_get_item (G_LIST_MODEL (self->pipelines), 0);
g_list_store_remove (self->pipelines, 0);
/* add a pipeline for the one we just removed */
add_new_pipeline (self);
return pipeline;
}