1 files changed, 121 insertions, 0 deletions
diff --git a/subprojects/gst-plugins-good/gst/rtp/gstrtpsirendepay.c b/subprojects/gst-plugins-good/gst/rtp/gstrtpsirendepay.c
new file mode 100644
index 0000000000..86a9dfffe1
--- /dev/null
+++ b/subprojects/gst-plugins-good/gst/rtp/gstrtpsirendepay.c
@@ -0,0 +1,121 @@
+/*
+ * Siren Depayloader Gst Element
+ *
+ *   @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include <string.h>
+#include <stdlib.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/audio/audio.h>
+#include "gstrtpelements.h"
+#include "gstrtpsirendepay.h"
+#include "gstrtputils.h"
+
+static GstStaticPadTemplate gst_rtp_siren_depay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "clock-rate = (int) 16000, " "encoding-name = (string) \"SIREN\"")
+    /* This is the default, so the peer doesn't have to specify it */
+    /*  " "dct-length = (int) 320") */
+    );
+
+     static GstStaticPadTemplate gst_rtp_siren_depay_src_template =
+         GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")
+    );
+
+     static GstBuffer *gst_rtp_siren_depay_process (GstRTPBaseDepayload *
+    depayload, GstRTPBuffer * rtp);
+     static gboolean gst_rtp_siren_depay_setcaps (GstRTPBaseDepayload *
+    depayload, GstCaps * caps);
+
+G_DEFINE_TYPE (GstRTPSirenDepay, gst_rtp_siren_depay,
+    GST_TYPE_RTP_BASE_DEPAYLOAD);
+GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpsirendepay, "rtpsirendepay",
+    GST_RANK_SECONDARY, GST_TYPE_RTP_SIREN_DEPAY, rtp_element_init (plugin));
+
+     static void gst_rtp_siren_depay_class_init (GstRTPSirenDepayClass * klass)
+{
+  GstElementClass *gstelement_class;
+  GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
+
+  gstelement_class = (GstElementClass *) klass;
+  gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
+
+  gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_siren_depay_process;
+  gstrtpbasedepayload_class->set_caps = gst_rtp_siren_depay_setcaps;
+
+  gst_element_class_add_static_pad_template (gstelement_class,
+      &gst_rtp_siren_depay_src_template);
+  gst_element_class_add_static_pad_template (gstelement_class,
+      &gst_rtp_siren_depay_sink_template);
+  gst_element_class_set_static_metadata (gstelement_class,
+      "RTP Siren packet depayloader", "Codec/Depayloader/Network/RTP",
+      "Extracts Siren audio from RTP packets",
+      "Philippe Kalaf <philippe.kalaf@collabora.co.uk>");
+}
+
+static void
+gst_rtp_siren_depay_init (GstRTPSirenDepay * rtpsirendepay)
+{
+
+}
+
+static gboolean
+gst_rtp_siren_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
+{
+  GstCaps *srccaps;
+  gboolean ret;
+
+  srccaps = gst_caps_new_simple ("audio/x-siren",
+      "dct-length", G_TYPE_INT, 320, NULL);
+  ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
+
+  GST_DEBUG ("set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
+  gst_caps_unref (srccaps);
+
+  /* always fixed clock rate of 16000 */
+  depayload->clock_rate = 16000;
+
+  return ret;
+}
+
+static GstBuffer *
+gst_rtp_siren_depay_process (GstRTPBaseDepayload * depayload,
+    GstRTPBuffer * rtp)
+{
+  GstBuffer *outbuf;
+
+  outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
+
+  if (outbuf) {
+    gst_rtp_drop_non_audio_meta (depayload, outbuf);
+  }
+
+  return outbuf;
+}