mirror of
https://gitlab.gnome.org/GNOME/calls.git
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380 lines
11 KiB
C
380 lines
11 KiB
C
/*
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* Copyright (C) 2021-2022 Purism SPC
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*
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* This file is part of Calls.
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*
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* Calls is free software: you can redistribute it and/or modify it
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* under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 3 of the License, or
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* (at your option) any later version.
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*
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* Calls is distributed in the hope that it will be useful, but
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* 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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* General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with Calls. If not, see <http://www.gnu.org/licenses/>.
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*
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* Author: Evangelos Ribeiro Tzaras <evangelos.tzaras@puri.sm>
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*
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* SPDX-License-Identifier: GPL-3.0-or-later
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*
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*/
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#define G_LOG_DOMAIN "CallsSipMediaManager"
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#include "calls-settings.h"
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#include "calls-sip-media-manager.h"
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#include "calls-sip-media-pipeline.h"
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#include "calls-srtp-utils.h"
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#include "gst-rfc3551.h"
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#include "util.h"
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#include <gio/gio.h>
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#include <gst/gst.h>
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#include <sys/socket.h>
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/**
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* SECTION:sip-media-manager
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* @short_description: The media manager singleton
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* @Title: CallsSipMediaManager
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*
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* #CallsSipMediaManager is mainly responsible for generating appropriate
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* SDP messages for the set of supported codecs. It also holds a list of
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* #CallsSipMediaPipeline objects that are ready to be used.
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*/
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typedef struct _CallsSipMediaManager {
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GObject parent;
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CallsSettings *settings;
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GList *preferred_codecs;
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GListStore *pipelines;
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} CallsSipMediaManager;
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G_DEFINE_TYPE (CallsSipMediaManager, calls_sip_media_manager, G_TYPE_OBJECT);
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static void
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on_notify_preferred_audio_codecs (CallsSipMediaManager *self)
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{
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GList *supported_codecs;
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g_auto (GStrv) settings_codec_preference = NULL;
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g_assert (CALLS_IS_SIP_MEDIA_MANAGER (self));
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g_clear_list (&self->preferred_codecs, NULL);
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supported_codecs = media_codecs_get_candidates ();
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if (!supported_codecs) {
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g_warning ("There aren't any supported codecs installed on your system");
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return;
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}
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settings_codec_preference = calls_settings_get_preferred_audio_codecs (self->settings);
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if (!settings_codec_preference) {
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g_debug ("No audio codec preference set. Using all supported codecs");
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self->preferred_codecs = supported_codecs;
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return;
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}
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for (guint i = 0; settings_codec_preference[i] != NULL; i++) {
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MediaCodecInfo *codec = media_codec_by_name (settings_codec_preference[i]);
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if (!codec) {
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g_debug ("Did not find audio codec %s", settings_codec_preference[i]);
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continue;
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}
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if (media_codec_available_in_gst (codec))
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self->preferred_codecs = g_list_append (self->preferred_codecs, codec);
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}
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if (!self->preferred_codecs) {
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g_warning ("Cannot satisfy audio codec preference, "
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"falling back to all supported codecs");
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self->preferred_codecs = supported_codecs;
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} else {
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g_list_free (supported_codecs);
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}
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}
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static void
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add_new_pipeline (CallsSipMediaManager *self)
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{
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CallsSipMediaPipeline *pipeline;
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g_assert (CALLS_IS_SIP_MEDIA_MANAGER (self));
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pipeline = calls_sip_media_pipeline_new (NULL);
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g_list_store_append (self->pipelines, pipeline);
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}
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static void
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calls_sip_media_manager_finalize (GObject *object)
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{
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CallsSipMediaManager *self = CALLS_SIP_MEDIA_MANAGER (object);
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g_list_free (self->preferred_codecs);
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g_object_unref (self->pipelines);
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G_OBJECT_CLASS (calls_sip_media_manager_parent_class)->finalize (object);
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}
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static void
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calls_sip_media_manager_class_init (CallsSipMediaManagerClass *klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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object_class->finalize = calls_sip_media_manager_finalize;
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}
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static void
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calls_sip_media_manager_init (CallsSipMediaManager *self)
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{
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if (!gst_is_initialized ())
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gst_init (NULL, NULL);
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self->settings = calls_settings_get_default ();
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g_signal_connect_swapped (self->settings,
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"notify::preferred-audio-codecs",
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G_CALLBACK (on_notify_preferred_audio_codecs),
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self);
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on_notify_preferred_audio_codecs (self);
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self->pipelines = g_list_store_new (CALLS_TYPE_SIP_MEDIA_PIPELINE);
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add_new_pipeline (self);
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}
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/* Public functions */
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CallsSipMediaManager *
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calls_sip_media_manager_default (void)
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{
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static CallsSipMediaManager *instance = NULL;
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if (instance == NULL) {
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g_debug ("Creating CallsSipMediaManager");
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instance = g_object_new (CALLS_TYPE_SIP_MEDIA_MANAGER, NULL);
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g_object_add_weak_pointer (G_OBJECT (instance), (gpointer *) &instance);
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}
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return instance;
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}
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/* helper for calls_sip_media_manager_get_capabilities() */
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static char *
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get_connection_line (const char *ip)
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{
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int af_family;
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if (STR_IS_NULL_OR_EMPTY(ip))
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goto invalid;
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af_family = get_address_family_for_ip (ip, TRUE);
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if (af_family == AF_UNSPEC)
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goto invalid;
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return g_strdup_printf ("c=IN %s %s\r\n",
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af_family == AF_INET ? "IP4" : "IP6",
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ip);
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invalid:
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return NULL;
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}
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/* calls_sip_media_manager_get_capabilities:
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*
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* @self: A #CallsSipMediaManager
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* @port: Should eventually come from the ICE stack
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* @crypto_attributes: A #GList of #calls_srtp_crypto_attribute
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* @supported_codecs: A #GList of #MediaCodecInfo
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*
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* Returns: (transfer full): string describing capabilities
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* to be used in the session description (SDP)
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*/
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char *
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calls_sip_media_manager_get_capabilities (CallsSipMediaManager *self,
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const char *own_ip,
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gint rtp_port,
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gint rtcp_port,
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GList *crypto_attributes,
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GList *supported_codecs)
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{
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char *payload_type = crypto_attributes ? "SAVP" : "AVP";
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g_autoptr (GString) media_line = NULL;
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g_autoptr (GString) attribute_lines = NULL;
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g_autofree char *connection_line = NULL;
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GList *node;
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g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
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media_line = g_string_new (NULL);
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attribute_lines = g_string_new (NULL);
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if (supported_codecs == NULL) {
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g_warning ("No supported codecs found. Can't build meaningful SDP message");
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g_string_append_printf (media_line, "m=audio 0 RTP/AVP");
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goto done;
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}
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/* media lines look f.e like "audio 31337 RTP/AVP 9 8 0" */
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g_string_append_printf (media_line,
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"m=audio %d RTP/%s", rtp_port, payload_type);
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for (node = supported_codecs; node != NULL; node = node->next) {
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MediaCodecInfo *codec = node->data;
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g_string_append_printf (media_line, " %u", codec->payload_id);
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g_string_append_printf (attribute_lines,
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"a=rtpmap:%u %s/%u%s",
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codec->payload_id,
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codec->name,
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codec->clock_rate,
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"\r\n");
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}
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for (node = crypto_attributes; node != NULL; node = node->next) {
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calls_srtp_crypto_attribute *attr = node->data;
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g_autoptr (GError) error = NULL;
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g_autofree char *crypto_line =
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calls_srtp_print_sdp_crypto_attribute (attr, &error);
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if (!crypto_line) {
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g_warning ("Could not print SDP crypto line for tag %d: %s", attr->tag, error->message);
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continue;
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}
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g_string_append_printf (attribute_lines, "%s\r\n", crypto_line);
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}
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g_string_append_printf (attribute_lines, "a=rtcp:%d\r\n", rtcp_port);
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done:
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connection_line = get_connection_line (own_ip);
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return g_strdup_printf ("v=0\r\n"
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"%s"
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"%s\r\n"
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"%s\r\n",
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connection_line ?: "",
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media_line->str,
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attribute_lines->str);
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}
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/* calls_sip_media_manager_static_capabilities:
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*
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* @self: A #CallsSipMediaManager
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* @rtp_port: Port to use for RTP. Should eventually come from the ICE stack
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* @rtcp_port: Port to use for RTCP.Should eventually come from the ICE stack
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* @crypto_attributes: A #GList of #calls_srtp_crypto_attribute
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*
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* Returns: (transfer full): string describing capabilities
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* to be used in the session description (SDP)
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*/
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char *
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calls_sip_media_manager_static_capabilities (CallsSipMediaManager *self,
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const char *own_ip,
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gint rtp_port,
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gint rtcp_port,
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GList *crypto_attributes)
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{
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g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
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return calls_sip_media_manager_get_capabilities (self,
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own_ip,
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rtp_port,
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rtcp_port,
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crypto_attributes,
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self->preferred_codecs);
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}
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/* calls_sip_media_manager_codec_candiates:
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*
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* @self: A #CallsSipMediaManager
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*
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* Returns: (transfer none): A #GList of supported
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* #MediaCodecInfo
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*/
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GList *
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calls_sip_media_manager_codec_candidates (CallsSipMediaManager *self)
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{
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g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
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return self->preferred_codecs;
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}
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/* calls_sip_media_manager_get_codecs_from_sdp
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*
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* @self: A #CallsSipMediaManager
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* @sdp: A #sdp_media_t media description
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*
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* Returns: (transfer container): A #GList of codecs found in the
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* SDP message
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*/
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GList *
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calls_sip_media_manager_get_codecs_from_sdp (CallsSipMediaManager *self,
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sdp_media_t *sdp_media)
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{
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GList *codecs = NULL;
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sdp_rtpmap_t *rtpmap = NULL;
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g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
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g_return_val_if_fail (sdp_media, NULL);
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if (sdp_media->m_type != sdp_media_audio) {
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g_warning ("Only the 'audio' media type is supported");
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return NULL;
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}
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for (rtpmap = sdp_media->m_rtpmaps; rtpmap != NULL; rtpmap = rtpmap->rm_next) {
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MediaCodecInfo *codec = media_codec_by_payload_id (rtpmap->rm_pt);
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if (codec)
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codecs = g_list_append (codecs, codec);
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}
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if (sdp_media->m_next != NULL)
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g_warning ("Currently only a single media session is supported");
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if (codecs == NULL)
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g_warning ("Did not find any common codecs");
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return codecs;
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}
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/**
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* calls_sip_media_manager_get_pipeline:
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* @self: A #CallsSipMediaManager
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*
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* Returns: (transfer full): A #CallsSipMediaPipeline
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*/
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CallsSipMediaPipeline *
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calls_sip_media_manager_get_pipeline (CallsSipMediaManager *self)
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{
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g_autoptr (CallsSipMediaPipeline) pipeline = NULL;
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g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
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pipeline = g_list_model_get_item (G_LIST_MODEL (self->pipelines), 0);
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g_list_store_remove (self->pipelines, 0);
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/* add a pipeline for the one we just removed */
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add_new_pipeline (self);
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return pipeline;
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}
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