1 files changed, 0 insertions, 149 deletions
diff --git a/gst/rtp/gstrtpspeexenc.c b/gst/rtp/gstrtpspeexenc.c
deleted file mode 100644
index 97e3bf33..00000000
--- a/gst/rtp/gstrtpspeexenc.c
+++ /dev/null
@@ -1,149 +0,0 @@
-/* GStreamer
- * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
- *
- * 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 
- */
-
-#ifdef HAVE_CONFIG_H
-#  include "config.h"
-#endif
-
-#include <stdlib.h>
-#include <string.h>
-#include <gst/rtp/gstrtpbuffer.h>
-
-#include "gstrtpspeexenc.h"
-
-/* elementfactory information */
-static GstElementDetails gst_rtpspeexenc_details = {
-  "RTP packet parser",
-  "Codec/Encoder/Network",
-  "Encodes Speex audio into a RTP packet",
-  "Edgard Lima <edgard.lima@indt.org.br>"
-};
-
-static GstStaticPadTemplate gst_rtpspeexenc_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-speex")
-    );
-
-static GstStaticPadTemplate gst_rtpspeexenc_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) 110, "       /* guaranties compatibility with Linphone
-                                                                                                           Could be [96,127] See page 34 at http://www.ietf.org/rfc/rfc3551.txt */
-        "clock-rate =  (int) [6000, 48000], "
-        "encoding-name = (string) \"speex\", "
-        "encoding-params = (string) \"1\"")
-    );
-
-static gboolean gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload,
-    GstCaps * caps);
-static GstFlowReturn gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * payload,
-    GstBuffer * buffer);
-
-GST_BOILERPLATE (GstRtpSPEEXEnc, gst_rtpspeexenc, GstBaseRTPPayload,
-    GST_TYPE_BASE_RTP_PAYLOAD);
-
-static void
-gst_rtpspeexenc_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_rtpspeexenc_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtpspeexenc_src_template));
-  gst_element_class_set_details (element_class, &gst_rtpspeexenc_details);
-}
-
-static void
-gst_rtpspeexenc_class_init (GstRtpSPEEXEncClass * 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_rtpspeexenc_setcaps;
-  gstbasertppayload_class->handle_buffer = gst_rtpspeexenc_handle_buffer;
-}
-
-static void
-gst_rtpspeexenc_init (GstRtpSPEEXEnc * rtpspeexenc, GstRtpSPEEXEncClass * klass)
-{
-  GST_BASE_RTP_PAYLOAD (rtpspeexenc)->clock_rate = 8000;
-  GST_BASE_RTP_PAYLOAD_PT (rtpspeexenc) = 110;  /* Create String */
-}
-
-static gboolean
-gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
-{
-  gst_basertppayload_set_options (payload, "audio", FALSE, "speex", 8000);
-  gst_basertppayload_set_outcaps (payload, NULL);
-
-  return TRUE;
-}
-
-static GstFlowReturn
-gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * basepayload,
-    GstBuffer * buffer)
-{
-  GstRtpSPEEXEnc *rtpspeexenc;
-  guint size, payload_len;
-  GstBuffer *outbuf;
-  guint8 *payload, *data;
-  GstClockTime timestamp;
-  GstFlowReturn ret;
-
-  rtpspeexenc = GST_RTP_SPEEX_ENC (basepayload);
-
-  size = GST_BUFFER_SIZE (buffer);
-  timestamp = GST_BUFFER_TIMESTAMP (buffer);
-
-  /* FIXME, only one SPEEX frame per RTP packet for now */
-  payload_len = size;
-
-  outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
-  /* FIXME, assert for now */
-  g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpspeexenc));
-
-  /* copy timestamp */
-  GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
-  /* get payload */
-  payload = gst_rtpbuffer_get_payload (outbuf);
-
-  data = GST_BUFFER_DATA (buffer);
-
-  /* copy data in payload */
-  memcpy (&payload[0], data, size);
-
-  gst_buffer_unref (buffer);
-
-  ret = gst_basertppayload_push (basepayload, outbuf);
-
-  return ret;
-}
-
-gboolean
-gst_rtpspeexenc_plugin_init (GstPlugin * plugin)
-{
-  return gst_element_register (plugin, "rtpspeexenc",
-      GST_RANK_NONE, GST_TYPE_RTP_SPEEX_ENC);
-}