1 files changed, 0 insertions, 163 deletions
diff --git a/gst/siren/gstrtpsirenpay.c b/gst/siren/gstrtpsirenpay.c
deleted file mode 100644
index dff8ae7b..00000000
--- a/gst/siren/gstrtpsirenpay.c
+++ /dev/null
@@ -1,163 +0,0 @@
-/*
- * 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);
-}