1
0
Fork 0
mirror of https://gitlab.gnome.org/GNOME/calls.git synced 2024-12-26 14:27:35 +00:00
Purism-Calls/plugins/sip/gst-rfc3551.c
2021-06-18 11:12:13 +02:00

129 lines
3.7 KiB
C

/*
* 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
*
*/
#define G_LOG_DOMAIN "CallsGstRfc3551"
#include "gst-rfc3551.h"
#include <glib.h>
#include <gst/gst.h>
/* Use the following codecs in order of preference */
static MediaCodecInfo gst_codecs[] = {
{8, "PCMA", 8000, 1, "rtppcmapay", "rtppcmadepay", "alawenc", "alawdec", "libgstalaw.so"},
{0, "PCMU", 8000, 1, "rtppcmupay", "rtppcmudepay", "mulawenc", "mulawdec", "libgstmulaw.so"},
{3, "GSM", 8000, 1, "rtpgsmpay", "rtpgsmdepay", "gsmenc", "gsmdec", "libgstgsm.so"},
{9, "G722", 8000, 1, "rtpg722pay", "rtpg722depay", "avenc_g722", "avdec_g722", "libgstlibav.so"},
{4, "G723", 8000, 1, "rtpg723pay", "rtpg723depay", "avenc_g723_1", "avdec_g723_1", "libgstlibav.so"}, // does not seem to work
};
static gboolean
media_codec_available_in_gst (MediaCodecInfo *codec) {
gboolean available = FALSE;
GstRegistry *registry = gst_registry_get ();
GstPlugin *plugin = NULL;
plugin = gst_registry_lookup (registry, codec->filename);
available = !!plugin;
if (plugin)
gst_object_unref (plugin);
g_debug ("Gstreamer plugin for %s %s available",
codec->name, available ? "is" : "is not");
return available;
}
/* media_codec_by_name:
*
* @name: The name of the codec
*
* Returns: (transfer none): A #MediaCodecInfo, if found
*/
MediaCodecInfo *
media_codec_by_name (const char *name)
{
g_return_val_if_fail (name, NULL);
for (guint i = 0; i < G_N_ELEMENTS (gst_codecs); i++) {
if (g_strcmp0 (name, gst_codecs[i].name) == 0)
return &gst_codecs[i];
}
return NULL;
}
/* media_codec_by_payload_id:
*
* @payload_id: The payload id (see RFC 3551, 3555, 4733, 4855)
*
* Returns: (transfer none): A #MediaCodecInfo, if found
*/
MediaCodecInfo *
media_codec_by_payload_id (guint payload_id)
{
for (guint i = 0; i < G_N_ELEMENTS (gst_codecs); i++) {
if (payload_id == gst_codecs[i].payload_id)
return &gst_codecs[i];
}
return NULL;
}
/* media_codec_get_gst_capabilities:
*
* @codec: A #MediaCodecInfo
*
* Returns: (transfer full): The capability string describing GstCaps.
* Used for the RTP source element.
*/
gchar *
media_codec_get_gst_capabilities (MediaCodecInfo *codec)
{
return g_strdup_printf ("application/x-rtp,media=(string)audio,clock-rate=(int)%u"
",encoding-name=(string)%s,payload=(int)%u",
codec->clock_rate,
codec->name,
codec->payload_id);
}
/* media_codecs_get_candidates:
*
* Returns: (transfer none): A #GList of codec candidates of type #MediaCodecInfo
*/
GList *
media_codecs_get_candidates (void)
{
GList *candidates = NULL;
for (guint i = 0; i < G_N_ELEMENTS (gst_codecs); i++) {
if (media_codec_available_in_gst (&gst_codecs[i])) {
g_debug ("Adding %s to the codec candidates", gst_codecs[i].name);
candidates = g_list_append (candidates, &gst_codecs[i]);
}
}
return candidates;
}