Move provider plugins into a dedicated directory

Since we will introduce another type of plugin for the policy engine
we want to have each plugin type in separate directories.

We also have to adjust:

- plugin search directories
- po file location
- update paths for calls-doc target

plugins/provider/sip/gst-rfc3551.c

@@ -0,0 +1,144 @@
+/*
+ * 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
+};
+
+
+/**
+ * media_codec_available_in_gst:
+ * @codec: A #MediaCodecInfo
+ *
+ * Returns: %TRUE if codec is available on your system, %FALSE otherwise
+ */
+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.
+ * You should check if the codec is available on your system before
+ * trying to use it with media_codec_available_in_gst()
+ */
+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.
+ * You should check if the codec is available on your system before
+ * trying to use it with media_codec_available_in_gst()
+ */
+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
+ * @use_srtp: Whether to use SRTP
+ *
+ * Returns: (transfer full): The capability string describing GstCaps.
+ * Used for the RTP source element.
+ */
+gchar *
+media_codec_get_gst_capabilities (MediaCodecInfo *codec,
+                                  gboolean        use_srtp)
+{
+  return g_strdup_printf ("application/%s,media=(string)audio,clock-rate=(int)%u"
+                          ",encoding-name=(string)%s,payload=(int)%u",
+                          use_srtp ? "x-srtp" : "x-rtp",
+                          codec->clock_rate,
+                          codec->name,
+                          codec->payload_id);
+}
+
+/* media_codecs_get_candidates:
+ *
+ * Returns: (transfer container): A #GList of codec candidates of type #MediaCodecInfo.
+ * Free the list with g_list_free when done.
+ */
+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;
+}