1 files changed, 163 insertions, 0 deletions
diff --git a/gst/siren/gstrtpsirenpay.c b/gst/siren/gstrtpsirenpay.c
new file mode 100644
index 00000000..dff8ae7b
--- /dev/null
+++ b/gst/siren/gstrtpsirenpay.c
@@ -0,0 +1,163 @@
+/*
+ * Siren Payloader 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstrtpsirenpay.h"
+#include <gst/rtp/gstrtpbuffer.h>
+
+/* elementfactory information */
+static GstElementDetails gst_rtpsirenpay_details = {
+  "RTP Payloader for Siren Audio",
+  "Codec/Payloader/Network",
+  "Packetize Siren audio streams into RTP packets",
+  "Youness Alaoui <kakaroto@kakaroto.homelinux.net>"
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtpsirenpay_debug);
+#define GST_CAT_DEFAULT (rtpsirenpay_debug)
+
+static GstStaticPadTemplate gst_rtpsirenpay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")
+    );
+
+static GstStaticPadTemplate gst_rtpsirenpay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "clock-rate = (int) 16000, "
+        "encoding-name = (string) \"SIREN\", "
+        "dct-length = (int) 320")
+    );
+
+static gboolean gst_rtpsirenpay_setcaps (GstBaseRTPPayload * payload,
+    GstCaps * caps);
+
+GST_BOILERPLATE (GstRTPSirenPay, gst_rtpsirenpay, GstBaseRTPAudioPayload,
+    GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+
+static void
+gst_rtpsirenpay_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtpsirenpay_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtpsirenpay_src_template));
+  gst_element_class_set_details (element_class, &gst_rtpsirenpay_details);
+}
+
+static void
+gst_rtpsirenpay_class_init (GstRTPSirenPayClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+
+  gstbasertppayload_class->set_caps = gst_rtpsirenpay_setcaps;
+
+  GST_DEBUG_CATEGORY_INIT (rtpsirenpay_debug, "rtpsirenpay", 0,
+      "siren audio RTP payloader");
+}
+
+static void
+gst_rtpsirenpay_init (GstRTPSirenPay * rtpsirenpay, GstRTPSirenPayClass * klass)
+{
+  GstBaseRTPPayload *basertppayload;
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+
+  basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay);
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay);
+
+  /* we don't set the payload type, it should be set by the application using
+   * the pt property or the default 96 will be used */
+  basertppayload->clock_rate = 16000;
+
+  /* tell basertpaudiopayload that this is a frame based codec */
+  gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
+}
+
+static gboolean
+gst_rtpsirenpay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
+{
+  GstRTPSirenPay *rtpsirenpay;
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  gboolean ret;
+  gint dct_length;
+  GstStructure *structure;
+  const char *payload_name;
+
+  rtpsirenpay = GST_RTP_SIREN_PAY (basertppayload);
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  gst_structure_get_int (structure, "dct-length", &dct_length);
+  if (dct_length != 320)
+    goto wrong_dct;
+
+  payload_name = gst_structure_get_name (structure);
+  if (g_strcasecmp ("audio/x-siren", payload_name))
+    goto wrong_caps;
+
+  gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN", 16000);
+  /* set options for this frame based audio codec */
+  gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40);
+
+  ret = gst_basertppayload_set_outcaps (basertppayload, NULL);
+
+  return TRUE;
+
+  /* ERRORS */
+wrong_dct:
+  {
+    GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d", dct_length);
+    return FALSE;
+  }
+wrong_caps:
+  {
+    GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s",
+        payload_name);
+    return FALSE;
+  }
+}
+
+gboolean
+gst_rtp_siren_pay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpsirenpay",
+      GST_RANK_NONE, GST_TYPE_RTP_SIREN_PAY);
+}