mirror of
https://gitlab.gnome.org/GNOME/calls.git
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129 lines
3.7 KiB
C
129 lines
3.7 KiB
C
/*
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* Copyright (C) 2021 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 "CallsGstRfc3551"
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#include "gst-rfc3551.h"
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#include <glib.h>
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#include <gst/gst.h>
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/* Use the following codecs in order of preference */
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static MediaCodecInfo gst_codecs[] = {
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{8, "PCMA", 8000, 1, "rtppcmapay", "rtppcmadepay", "alawenc", "alawdec", "libgstalaw.so"},
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{0, "PCMU", 8000, 1, "rtppcmupay", "rtppcmudepay", "mulawenc", "mulawdec", "libgstmulaw.so"},
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{3, "GSM", 8000, 1, "rtpgsmpay", "rtpgsmdepay", "gsmenc", "gsmdec", "libgstgsm.so"},
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{9, "G722", 8000, 1, "rtpg722pay", "rtpg722depay", "avenc_g722", "avdec_g722", "libgstlibav.so"},
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{4, "G723", 8000, 1, "rtpg723pay", "rtpg723depay", "avenc_g723_1", "avdec_g723_1", "libgstlibav.so"}, // does not seem to work
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};
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static gboolean
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media_codec_available_in_gst (MediaCodecInfo *codec) {
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gboolean available = FALSE;
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GstRegistry *registry = gst_registry_get ();
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GstPlugin *plugin = NULL;
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plugin = gst_registry_lookup (registry, codec->filename);
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available = !!plugin;
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if (plugin)
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gst_object_unref (plugin);
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g_debug ("Gstreamer plugin for %s %s available",
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codec->name, available ? "is" : "is not");
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return available;
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}
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/* media_codec_by_name:
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*
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* @name: The name of the codec
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*
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* Returns: (transfer none): A #MediaCodecInfo, if found
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*/
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MediaCodecInfo *
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media_codec_by_name (const char *name)
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{
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g_return_val_if_fail (name, NULL);
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for (guint i = 0; i < G_N_ELEMENTS (gst_codecs); i++) {
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if (g_strcmp0 (name, gst_codecs[i].name) == 0)
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return &gst_codecs[i];
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}
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return NULL;
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}
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/* media_codec_by_payload_id:
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*
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* @payload_id: The payload id (see RFC 3551, 3555, 4733, 4855)
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*
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* Returns: (transfer none): A #MediaCodecInfo, if found
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*/
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MediaCodecInfo *
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media_codec_by_payload_id (guint payload_id)
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{
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for (guint i = 0; i < G_N_ELEMENTS (gst_codecs); i++) {
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if (payload_id == gst_codecs[i].payload_id)
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return &gst_codecs[i];
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}
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return NULL;
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}
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/* media_codec_get_gst_capabilities:
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*
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* @codec: A #MediaCodecInfo
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*
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* Returns: (transfer full): The capability string describing GstCaps.
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* Used for the RTP source element.
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*/
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gchar *
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media_codec_get_gst_capabilities (MediaCodecInfo *codec)
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{
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return g_strdup_printf ("application/x-rtp,media=(string)audio,clock-rate=(int)%u"
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",encoding-name=(string)%s,payload=(int)%u",
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codec->clock_rate,
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codec->name,
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codec->payload_id);
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}
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/* media_codecs_get_candidates:
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*
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* Returns: (transfer none): A #GList of codec candidates of type #MediaCodecInfo
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*/
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GList *
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media_codecs_get_candidates (void)
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{
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GList *candidates = NULL;
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for (guint i = 0; i < G_N_ELEMENTS (gst_codecs); i++) {
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if (media_codec_available_in_gst (&gst_codecs[i])) {
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g_debug ("Adding %s to the codec candidates", gst_codecs[i].name);
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candidates = g_list_append (candidates, &gst_codecs[i]);
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}
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}
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return candidates;
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}
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