Created Speex payloader and depayloader; Optimize G711 payloader to use adapter and send packets until MTU size.

Original commit message from CVS:
Created Speex payloader and depayloader; Optimize G711 payloader to use adapter and send packets until MTU size.

16 files changed, 1001 insertions, 63 deletions

diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am
index 877dbefa..d1c98e98 100644
--- a/gst/rtp/Makefile.am
+++ b/gst/rtp/Makefile.am
@@ -16,8 +16,10 @@ libgstrtp_la_SOURCES = \
 	gstrtph263enc.c \
 	gstasteriskh263.c \
 	gstrtpmp4venc.c \
-	gstrtpmp4vdec.c
-
+	gstrtpmp4vdec.c \
+	gstrtpspeexenc.c \
+	gstrtpspeexdec.c
+	
 #gstrtpL16enc.c gstrtpL16parse.c gstrtpgsmenc.c gstrtpgsmparse.c 
 
 if HAVE_WINSOCK2_H
@@ -48,4 +50,6 @@ noinst_HEADERS = gstrtpL16enc.h 	\
 		 gstrtpmp4vdec.h	\
 		 gstrtpdec.h		\
 		 gstrtph263enc.h	\
-		 gstasteriskh263.h
+		 gstasteriskh263.h  \
+		 gstrtpspeexenc.h	\
+		 gstrtpspeexdec.h
diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c
index 07a77e48..889e3748 100644
--- a/gst/rtp/gstrtp.c
+++ b/gst/rtp/gstrtp.c
@@ -36,6 +36,8 @@
 #include "gstasteriskh263.h"
 #include "gstrtpmp4venc.h"
 #include "gstrtpmp4vdec.h"
+#include "gstrtpspeexenc.h"
+#include "gstrtpspeexdec.h"
 
 static gboolean
 plugin_init (GstPlugin * plugin)
@@ -85,6 +87,12 @@ plugin_init (GstPlugin * plugin)
   if (!gst_rtpmp4vdec_plugin_init (plugin))
     return FALSE;
 
+  if (!gst_rtpspeexenc_plugin_init (plugin))
+    return FALSE;
+
+  if (!gst_rtpspeexdec_plugin_init (plugin))
+    return FALSE;
+
   return TRUE;
 }
 
diff --git a/gst/rtp/gstrtpg711dec.c b/gst/rtp/gstrtpg711dec.c
index e39f7d5b..216dc386 100644
--- a/gst/rtp/gstrtpg711dec.c
+++ b/gst/rtp/gstrtpg711dec.c
@@ -48,12 +48,12 @@ static GstStaticPadTemplate gst_rtpg711dec_sink_template =
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) 0, "
+        "payload = (int) [ 0, 255 ], "
         "clock-rate = (int) 8000, "
         "encoding-name = (string) \"PCMU\"; "
         "application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) 8, "
+        "payload = (int) [ 0, 255 ], "
         "clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
 
     );
diff --git a/gst/rtp/gstrtpg711depay.c b/gst/rtp/gstrtpg711depay.c
index e39f7d5b..216dc386 100644
--- a/gst/rtp/gstrtpg711depay.c
+++ b/gst/rtp/gstrtpg711depay.c
@@ -48,12 +48,12 @@ static GstStaticPadTemplate gst_rtpg711dec_sink_template =
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) 0, "
+        "payload = (int) [ 0, 255 ], "
         "clock-rate = (int) 8000, "
         "encoding-name = (string) \"PCMU\"; "
         "application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) 8, "
+        "payload = (int) [ 0, 255 ], "
         "clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
 
     );
diff --git a/gst/rtp/gstrtpg711enc.c b/gst/rtp/gstrtpg711enc.c
index e0c70678..e83c888b 100644
--- a/gst/rtp/gstrtpg711enc.c
+++ b/gst/rtp/gstrtpg711enc.c
@@ -45,12 +45,12 @@ static GstStaticPadTemplate gst_rtpg711enc_src_template =
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) 0, "
+        "payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
         "clock-rate = (int) 8000, "
         "encoding-name = (string) \"PCMU\"; "
         "application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) 8, "
+        "payload = (int) " GST_RTP_PAYLOAD_PCMA_STRING ", "
         "clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
     );
 
@@ -58,6 +58,7 @@ static gboolean gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload,
     GstCaps * caps);
 static GstFlowReturn gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * payload,
     GstBuffer * buffer);
+static void gst_rtpg711enc_finalize (GObject * object);
 
 GST_BOILERPLATE (GstRtpG711Enc, gst_rtpg711enc, GstBaseRTPPayload,
     GST_TYPE_BASE_RTP_PAYLOAD);
@@ -86,6 +87,7 @@ gst_rtpg711enc_class_init (GstRtpG711EncClass * klass)
   gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
 
   parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+  gobject_class->finalize = gst_rtpg711enc_finalize;
 
   gstbasertppayload_class->set_caps = gst_rtpg711enc_setcaps;
   gstbasertppayload_class->handle_buffer = gst_rtpg711enc_handle_buffer;
@@ -94,9 +96,23 @@ gst_rtpg711enc_class_init (GstRtpG711EncClass * klass)
 static void
 gst_rtpg711enc_init (GstRtpG711Enc * rtpg711enc, GstRtpG711EncClass * klass)
 {
+  rtpg711enc->adapter = gst_adapter_new ();
   GST_BASE_RTP_PAYLOAD (rtpg711enc)->clock_rate = 8000;
 }
 
+static void
+gst_rtpg711enc_finalize (GObject * object)
+{
+  GstRtpG711Enc *rtpg711enc;
+
+  rtpg711enc = GST_RTP_G711_ENC (object);
+
+  g_object_unref (rtpg711enc->adapter);
+  rtpg711enc->adapter = NULL;
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
 static gboolean
 gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
 {
@@ -109,9 +125,11 @@ gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   stname = gst_structure_get_name (structure);
 
   if (0 == strcmp ("audio/x-mulaw", stname)) {
-    gst_basertppayload_set_options (payload, "audio", TRUE, "PCMU", 8000);
+    payload->pt = GST_RTP_PAYLOAD_PCMU;
+    gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
   } else if (0 == strcmp ("audio/x-alaw", stname)) {
-    gst_basertppayload_set_options (payload, "audio", TRUE, "PCMA", 8000);
+    payload->pt = GST_RTP_PAYLOAD_PCMA;
+    gst_basertppayload_set_options (payload, "audio", FALSE, "PCMA", 8000);
   } else {
     return FALSE;
   }
@@ -122,41 +140,88 @@ gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
 }
 
 static GstFlowReturn
+gst_rtpg711enc_flush (GstRtpG711Enc * rtpg711enc)
+{
+  guint avail;
+  GstBuffer *outbuf;
+  GstFlowReturn ret;
+
+  /* the data available in the adapter is either smaller
+   * than the MTU or bigger. In the case it is smaller, the complete
+   * adapter contents can be put in one packet.  */
+  avail = gst_adapter_available (rtpg711enc->adapter);
+
+  ret = GST_FLOW_OK;
+
+  while (avail > 0) {
+    guint towrite;
+    guint8 *payload;
+    guint8 *data;
+    guint payload_len;
+    guint packet_len;
+
+    /* this will be the total lenght of the packet */
+    packet_len = gst_rtpbuffer_calc_packet_len (avail, 0, 0);
+    /* fill one MTU or all available bytes */
+    towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpg711enc));
+    /* this is the payload length */
+    payload_len = gst_rtpbuffer_calc_payload_len (towrite, 0, 0);
+    /* create buffer to hold the payload */
+    outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
+
+    /* copy payload */
+    gst_rtpbuffer_set_payload_type (outbuf,
+        GST_BASE_RTP_PAYLOAD_PT (rtpg711enc));
+    payload = gst_rtpbuffer_get_payload (outbuf);
+    data = (guint8 *) gst_adapter_peek (rtpg711enc->adapter, payload_len);
+    memcpy (payload, data, payload_len);
+    gst_adapter_flush (rtpg711enc->adapter, payload_len);
+
+    avail -= payload_len;
+
+    GST_BUFFER_TIMESTAMP (outbuf) = rtpg711enc->first_ts;
+    ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpg711enc), outbuf);
+  }
+
+  return ret;
+}
+
+static GstFlowReturn
 gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * basepayload,
     GstBuffer * buffer)
 {
   GstRtpG711Enc *rtpg711enc;
-  guint size, payload_len;
-  GstBuffer *outbuf;
-  guint8 *payload, *data;
-  GstClockTime timestamp;
+  guint size, packet_len, avail;
   GstFlowReturn ret;
+  GstClockTime duration;
 
   rtpg711enc = GST_RTP_G711_ENC (basepayload);
 
   size = GST_BUFFER_SIZE (buffer);
-  timestamp = GST_BUFFER_TIMESTAMP (buffer);
+  duration = GST_BUFFER_TIMESTAMP (buffer);
 
-  /* FIXME, only one G711 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 (rtpg711enc));
-
-  /* copy timestamp */
-  GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
-  /* get payload */
-  payload = gst_rtpbuffer_get_payload (outbuf);
-
-  data = GST_BUFFER_DATA (buffer);
+  avail = gst_adapter_available (rtpg711enc->adapter);
+  if (avail == 0) {
+    rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+    rtpg711enc->duration = 0;
+  }
 
-  /* copy data in payload */
-  memcpy (&payload[0], data, size);
+  /* get packet length of data and see if we exceeded MTU. */
+  packet_len = gst_rtpbuffer_calc_packet_len (avail + size, 0, 0);
 
-  gst_buffer_unref (buffer);
+  /* if this buffer is going to overflow the packet, flush what we
+   * have. */
+  if (gst_basertppayload_is_filled (basepayload,
+          packet_len, rtpg711enc->duration + duration)) {
+    ret = gst_rtpg711enc_flush (rtpg711enc);
+    rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+    rtpg711enc->duration = 0;
+  } else {
+    ret = GST_FLOW_OK;
+  }
 
-  ret = gst_basertppayload_push (basepayload, outbuf);
+  gst_adapter_push (rtpg711enc->adapter, buffer);
+  rtpg711enc->duration += duration;
 
   return ret;
 }
diff --git a/gst/rtp/gstrtpg711enc.h b/gst/rtp/gstrtpg711enc.h
index e3ab1fe5..f8ca6dd4 100644
--- a/gst/rtp/gstrtpg711enc.h
+++ b/gst/rtp/gstrtpg711enc.h
@@ -18,6 +18,7 @@
 
 #include <gst/gst.h>
 #include <gst/rtp/gstbasertppayload.h>
+#include <gst/base/gstadapter.h>
 
 G_BEGIN_DECLS
 
@@ -38,8 +39,10 @@ typedef struct _GstRtpG711EncClass GstRtpG711EncClass;
 struct _GstRtpG711Enc
 {
   GstBaseRTPPayload payload;
-
-  gint frequency;
+  GstAdapter *adapter;
+  
+  GstClockTime first_ts;
+  GstClockTime duration;  
 };
 
 struct _GstRtpG711EncClass
diff --git a/gst/rtp/gstrtpg711pay.c b/gst/rtp/gstrtpg711pay.c
index e0c70678..e83c888b 100644
--- a/gst/rtp/gstrtpg711pay.c
+++ b/gst/rtp/gstrtpg711pay.c
@@ -45,12 +45,12 @@ static GstStaticPadTemplate gst_rtpg711enc_src_template =
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) 0, "
+        "payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
         "clock-rate = (int) 8000, "
         "encoding-name = (string) \"PCMU\"; "
         "application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) 8, "
+        "payload = (int) " GST_RTP_PAYLOAD_PCMA_STRING ", "
         "clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
     );
 
@@ -58,6 +58,7 @@ static gboolean gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload,
     GstCaps * caps);
 static GstFlowReturn gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * payload,
     GstBuffer * buffer);
+static void gst_rtpg711enc_finalize (GObject * object);
 
 GST_BOILERPLATE (GstRtpG711Enc, gst_rtpg711enc, GstBaseRTPPayload,
     GST_TYPE_BASE_RTP_PAYLOAD);
@@ -86,6 +87,7 @@ gst_rtpg711enc_class_init (GstRtpG711EncClass * klass)
   gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
 
   parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+  gobject_class->finalize = gst_rtpg711enc_finalize;
 
   gstbasertppayload_class->set_caps = gst_rtpg711enc_setcaps;
   gstbasertppayload_class->handle_buffer = gst_rtpg711enc_handle_buffer;
@@ -94,9 +96,23 @@ gst_rtpg711enc_class_init (GstRtpG711EncClass * klass)
 static void
 gst_rtpg711enc_init (GstRtpG711Enc * rtpg711enc, GstRtpG711EncClass * klass)
 {
+  rtpg711enc->adapter = gst_adapter_new ();
   GST_BASE_RTP_PAYLOAD (rtpg711enc)->clock_rate = 8000;
 }
 
+static void
+gst_rtpg711enc_finalize (GObject * object)
+{
+  GstRtpG711Enc *rtpg711enc;
+
+  rtpg711enc = GST_RTP_G711_ENC (object);
+
+  g_object_unref (rtpg711enc->adapter);
+  rtpg711enc->adapter = NULL;
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
 static gboolean
 gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
 {
@@ -109,9 +125,11 @@ gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   stname = gst_structure_get_name (structure);
 
   if (0 == strcmp ("audio/x-mulaw", stname)) {
-    gst_basertppayload_set_options (payload, "audio", TRUE, "PCMU", 8000);
+    payload->pt = GST_RTP_PAYLOAD_PCMU;
+    gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
   } else if (0 == strcmp ("audio/x-alaw", stname)) {
-    gst_basertppayload_set_options (payload, "audio", TRUE, "PCMA", 8000);
+    payload->pt = GST_RTP_PAYLOAD_PCMA;
+    gst_basertppayload_set_options (payload, "audio", FALSE, "PCMA", 8000);
   } else {
     return FALSE;
   }
@@ -122,41 +140,88 @@ gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
 }
 
 static GstFlowReturn
+gst_rtpg711enc_flush (GstRtpG711Enc * rtpg711enc)
+{
+  guint avail;
+  GstBuffer *outbuf;
+  GstFlowReturn ret;
+
+  /* the data available in the adapter is either smaller
+   * than the MTU or bigger. In the case it is smaller, the complete
+   * adapter contents can be put in one packet.  */
+  avail = gst_adapter_available (rtpg711enc->adapter);
+
+  ret = GST_FLOW_OK;
+
+  while (avail > 0) {
+    guint towrite;
+    guint8 *payload;
+    guint8 *data;
+    guint payload_len;
+    guint packet_len;
+
+    /* this will be the total lenght of the packet */
+    packet_len = gst_rtpbuffer_calc_packet_len (avail, 0, 0);
+    /* fill one MTU or all available bytes */
+    towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpg711enc));
+    /* this is the payload length */
+    payload_len = gst_rtpbuffer_calc_payload_len (towrite, 0, 0);
+    /* create buffer to hold the payload */
+    outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
+
+    /* copy payload */
+    gst_rtpbuffer_set_payload_type (outbuf,
+        GST_BASE_RTP_PAYLOAD_PT (rtpg711enc));
+    payload = gst_rtpbuffer_get_payload (outbuf);
+    data = (guint8 *) gst_adapter_peek (rtpg711enc->adapter, payload_len);
+    memcpy (payload, data, payload_len);
+    gst_adapter_flush (rtpg711enc->adapter, payload_len);
+
+    avail -= payload_len;
+
+    GST_BUFFER_TIMESTAMP (outbuf) = rtpg711enc->first_ts;
+    ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpg711enc), outbuf);
+  }
+
+  return ret;
+}
+
+static GstFlowReturn
 gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * basepayload,
     GstBuffer * buffer)
 {
   GstRtpG711Enc *rtpg711enc;
-  guint size, payload_len;
-  GstBuffer *outbuf;
-  guint8 *payload, *data;
-  GstClockTime timestamp;
+  guint size, packet_len, avail;
   GstFlowReturn ret;
+  GstClockTime duration;
 
   rtpg711enc = GST_RTP_G711_ENC (basepayload);
 
   size = GST_BUFFER_SIZE (buffer);
-  timestamp = GST_BUFFER_TIMESTAMP (buffer);
+  duration = GST_BUFFER_TIMESTAMP (buffer);
 
-  /* FIXME, only one G711 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 (rtpg711enc));
-
-  /* copy timestamp */
-  GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
-  /* get payload */
-  payload = gst_rtpbuffer_get_payload (outbuf);
-
-  data = GST_BUFFER_DATA (buffer);
+  avail = gst_adapter_available (rtpg711enc->adapter);
+  if (avail == 0) {
+    rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+    rtpg711enc->duration = 0;
+  }
 
-  /* copy data in payload */
-  memcpy (&payload[0], data, size);
+  /* get packet length of data and see if we exceeded MTU. */
+  packet_len = gst_rtpbuffer_calc_packet_len (avail + size, 0, 0);
 
-  gst_buffer_unref (buffer);
+  /* if this buffer is going to overflow the packet, flush what we
+   * have. */
+  if (gst_basertppayload_is_filled (basepayload,
+          packet_len, rtpg711enc->duration + duration)) {
+    ret = gst_rtpg711enc_flush (rtpg711enc);
+    rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+    rtpg711enc->duration = 0;
+  } else {
+    ret = GST_FLOW_OK;
+  }
 
-  ret = gst_basertppayload_push (basepayload, outbuf);
+  gst_adapter_push (rtpg711enc->adapter, buffer);
+  rtpg711enc->duration += duration;
 
   return ret;
 }
diff --git a/gst/rtp/gstrtpg711pay.h b/gst/rtp/gstrtpg711pay.h
index e3ab1fe5..f8ca6dd4 100644
--- a/gst/rtp/gstrtpg711pay.h
+++ b/gst/rtp/gstrtpg711pay.h
@@ -18,6 +18,7 @@
 
 #include <gst/gst.h>
 #include <gst/rtp/gstbasertppayload.h>
+#include <gst/base/gstadapter.h>
 
 G_BEGIN_DECLS
 
@@ -38,8 +39,10 @@ typedef struct _GstRtpG711EncClass GstRtpG711EncClass;
 struct _GstRtpG711Enc
 {
   GstBaseRTPPayload payload;
-
-  gint frequency;
+  GstAdapter *adapter;
+  
+  GstClockTime first_ts;
+  GstClockTime duration;  
 };
 
 struct _GstRtpG711EncClass
diff --git a/gst/rtp/gstrtpspeexdec.c b/gst/rtp/gstrtpspeexdec.c
new file mode 100644
index 00000000..febcef4a
--- /dev/null
+++ b/gst/rtp/gstrtpspeexdec.c
@@ -0,0 +1,143 @@
+/* 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 <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include "gstrtpspeexdec.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtp_speexdec_details = {
+  "RTP packet parser",
+  "Codec/Parser/Network",
+  "Extracts Speex audio from RTP packets",
+  "Edgard Lima <edgard.lima@indt.org.br>"
+};
+
+/* RtpSPEEXDec signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  ARG_0
+};
+
+static GstStaticPadTemplate gst_rtpspeexdec_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) [ 96, 127 ], "
+        "clock-rate =  (int) [6000, 48000], "
+        "encoding-name = (string) \"speex\", "
+        "encoding-params = (string) \"1\"")
+    );
+
+static GstStaticPadTemplate gst_rtpspeexdec_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-speex")
+    );
+
+static GstBuffer *gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload,
+    GstBuffer * buf);
+static gboolean gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload,
+    GstCaps * caps);
+
+GST_BOILERPLATE (GstRtpSPEEXDec, gst_rtpspeexdec, GstBaseRTPDepayload,
+    GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static void
+gst_rtpspeexdec_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_rtpspeexdec_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtpspeexdec_sink_template));
+  gst_element_class_set_details (element_class, &gst_rtp_speexdec_details);
+}
+
+static void
+gst_rtpspeexdec_class_init (GstRtpSPEEXDecClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+  parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+  gstbasertpdepayload_class->process = gst_rtpspeexdec_process;
+  gstbasertpdepayload_class->set_caps = gst_rtpspeexdec_setcaps;
+}
+
+static void
+gst_rtpspeexdec_init (GstRtpSPEEXDec * rtpspeexdec, GstRtpSPEEXDecClass * klass)
+{
+  GST_BASE_RTP_DEPAYLOAD (rtpspeexdec)->clock_rate = 8000;
+}
+
+static gboolean
+gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+{
+  GstCaps *srccaps;
+  gboolean ret;
+
+  srccaps = gst_static_pad_template_get_caps (&gst_rtpspeexdec_src_template);
+  ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+
+  gst_caps_unref (srccaps);
+  return ret;
+}
+
+static GstBuffer *
+gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+  GstBuffer *outbuf = NULL;
+  gint payload_len;
+  guint8 *payload;
+
+  GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
+      GST_BUFFER_SIZE (buf),
+      gst_rtpbuffer_get_marker (buf),
+      gst_rtpbuffer_get_timestamp (buf), gst_rtpbuffer_get_seq (buf));
+
+  payload_len = gst_rtpbuffer_get_payload_len (buf);
+  payload = gst_rtpbuffer_get_payload (buf);
+
+  outbuf = gst_buffer_new_and_alloc (payload_len);
+  memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
+  return outbuf;
+}
+
+gboolean
+gst_rtpspeexdec_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpspeexdec",
+      GST_RANK_NONE, GST_TYPE_RTP_SPEEX_DEC);
+}
diff --git a/gst/rtp/gstrtpspeexdec.h b/gst/rtp/gstrtpspeexdec.h
new file mode 100644
index 00000000..603a7bf0
--- /dev/null
+++ b/gst/rtp/gstrtpspeexdec.h
@@ -0,0 +1,51 @@
+/* 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 
+ */
+ 
+#ifndef __GST_RTP_SPEEX_DEC_H__
+#define __GST_RTP_SPEEX_DEC_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpdepayload.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRtpSPEEXDec GstRtpSPEEXDec;
+typedef struct _GstRtpSPEEXDecClass GstRtpSPEEXDecClass;
+
+#define GST_TYPE_RTP_SPEEX_DEC \
+  (gst_rtpspeexdec_get_type())
+#define GST_RTP_SPEEX_DEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
+#define GST_RTP_SPEEX_DEC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
+#define GST_IS_RTP_SPEEX_DEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_DEC))
+#define GST_IS_RTP_SPEEX_DEC_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_DEC))
+
+struct _GstRtpSPEEXDec
+{
+  GstBaseRTPDepayload depayload;
+};
+
+struct _GstRtpSPEEXDecClass
+{
+  GstBaseRTPDepayloadClass parent_class;
+};
+
+gboolean gst_rtpspeexdec_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_SPEEX_DEC_H__ */
diff --git a/gst/rtp/gstrtpspeexdepay.c b/gst/rtp/gstrtpspeexdepay.c
new file mode 100644
index 00000000..febcef4a
--- /dev/null
+++ b/gst/rtp/gstrtpspeexdepay.c
@@ -0,0 +1,143 @@
+/* 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 <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include "gstrtpspeexdec.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtp_speexdec_details = {
+  "RTP packet parser",
+  "Codec/Parser/Network",
+  "Extracts Speex audio from RTP packets",
+  "Edgard Lima <edgard.lima@indt.org.br>"
+};
+
+/* RtpSPEEXDec signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  ARG_0
+};
+
+static GstStaticPadTemplate gst_rtpspeexdec_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) [ 96, 127 ], "
+        "clock-rate =  (int) [6000, 48000], "
+        "encoding-name = (string) \"speex\", "
+        "encoding-params = (string) \"1\"")
+    );
+
+static GstStaticPadTemplate gst_rtpspeexdec_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-speex")
+    );
+
+static GstBuffer *gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload,
+    GstBuffer * buf);
+static gboolean gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload,
+    GstCaps * caps);
+
+GST_BOILERPLATE (GstRtpSPEEXDec, gst_rtpspeexdec, GstBaseRTPDepayload,
+    GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static void
+gst_rtpspeexdec_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_rtpspeexdec_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtpspeexdec_sink_template));
+  gst_element_class_set_details (element_class, &gst_rtp_speexdec_details);
+}
+
+static void
+gst_rtpspeexdec_class_init (GstRtpSPEEXDecClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+  parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+  gstbasertpdepayload_class->process = gst_rtpspeexdec_process;
+  gstbasertpdepayload_class->set_caps = gst_rtpspeexdec_setcaps;
+}
+
+static void
+gst_rtpspeexdec_init (GstRtpSPEEXDec * rtpspeexdec, GstRtpSPEEXDecClass * klass)
+{
+  GST_BASE_RTP_DEPAYLOAD (rtpspeexdec)->clock_rate = 8000;
+}
+
+static gboolean
+gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+{
+  GstCaps *srccaps;
+  gboolean ret;
+
+  srccaps = gst_static_pad_template_get_caps (&gst_rtpspeexdec_src_template);
+  ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+
+  gst_caps_unref (srccaps);
+  return ret;
+}
+
+static GstBuffer *
+gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+  GstBuffer *outbuf = NULL;
+  gint payload_len;
+  guint8 *payload;
+
+  GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
+      GST_BUFFER_SIZE (buf),
+      gst_rtpbuffer_get_marker (buf),
+      gst_rtpbuffer_get_timestamp (buf), gst_rtpbuffer_get_seq (buf));
+
+  payload_len = gst_rtpbuffer_get_payload_len (buf);
+  payload = gst_rtpbuffer_get_payload (buf);
+
+  outbuf = gst_buffer_new_and_alloc (payload_len);
+  memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
+  return outbuf;
+}
+
+gboolean
+gst_rtpspeexdec_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpspeexdec",
+      GST_RANK_NONE, GST_TYPE_RTP_SPEEX_DEC);
+}
diff --git a/gst/rtp/gstrtpspeexdepay.h b/gst/rtp/gstrtpspeexdepay.h
new file mode 100644
index 00000000..603a7bf0
--- /dev/null
+++ b/gst/rtp/gstrtpspeexdepay.h
@@ -0,0 +1,51 @@
+/* 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 
+ */
+ 
+#ifndef __GST_RTP_SPEEX_DEC_H__
+#define __GST_RTP_SPEEX_DEC_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpdepayload.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRtpSPEEXDec GstRtpSPEEXDec;
+typedef struct _GstRtpSPEEXDecClass GstRtpSPEEXDecClass;
+
+#define GST_TYPE_RTP_SPEEX_DEC \
+  (gst_rtpspeexdec_get_type())
+#define GST_RTP_SPEEX_DEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
+#define GST_RTP_SPEEX_DEC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
+#define GST_IS_RTP_SPEEX_DEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_DEC))
+#define GST_IS_RTP_SPEEX_DEC_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_DEC))
+
+struct _GstRtpSPEEXDec
+{
+  GstBaseRTPDepayload depayload;
+};
+
+struct _GstRtpSPEEXDecClass
+{
+  GstBaseRTPDepayloadClass parent_class;
+};
+
+gboolean gst_rtpspeexdec_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_SPEEX_DEC_H__ */
diff --git a/gst/rtp/gstrtpspeexenc.c b/gst/rtp/gstrtpspeexenc.c
new file mode 100644
index 00000000..97e3bf33
--- /dev/null
+++ b/gst/rtp/gstrtpspeexenc.c
@@ -0,0 +1,149 @@
+/* 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);
+}
diff --git a/gst/rtp/gstrtpspeexenc.h b/gst/rtp/gstrtpspeexenc.h
new file mode 100644
index 00000000..066b272d
--- /dev/null
+++ b/gst/rtp/gstrtpspeexenc.h
@@ -0,0 +1,52 @@
+/* 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 
+ */
+
+
+#ifndef __GST_RTP_SPEEX_ENC_H__
+#define __GST_RTP_SPEEX_ENC_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertppayload.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRtpSPEEXEnc GstRtpSPEEXEnc;
+typedef struct _GstRtpSPEEXEncClass GstRtpSPEEXEncClass;
+
+#define GST_TYPE_RTP_SPEEX_ENC \
+  (gst_rtpspeexenc_get_type())
+#define GST_RTP_SPEEX_ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
+#define GST_RTP_SPEEX_ENC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
+#define GST_IS_RTP_SPEEX_ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_ENC))
+#define GST_IS_RTP_SPEEX_ENC_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_ENC))
+
+struct _GstRtpSPEEXEnc
+{
+  GstBaseRTPPayload payload;
+};
+
+struct _GstRtpSPEEXEncClass
+{
+  GstBaseRTPPayloadClass parent_class;
+};
+
+gboolean gst_rtpspeexenc_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_SPEEX_ENC_H__ */
diff --git a/gst/rtp/gstrtpspeexpay.c b/gst/rtp/gstrtpspeexpay.c
new file mode 100644
index 00000000..97e3bf33
--- /dev/null
+++ b/gst/rtp/gstrtpspeexpay.c
@@ -0,0 +1,149 @@
+/* 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);
+}
diff --git a/gst/rtp/gstrtpspeexpay.h b/gst/rtp/gstrtpspeexpay.h
new file mode 100644
index 00000000..066b272d
--- /dev/null
+++ b/gst/rtp/gstrtpspeexpay.h
@@ -0,0 +1,52 @@
+/* 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 
+ */
+
+
+#ifndef __GST_RTP_SPEEX_ENC_H__
+#define __GST_RTP_SPEEX_ENC_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertppayload.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRtpSPEEXEnc GstRtpSPEEXEnc;
+typedef struct _GstRtpSPEEXEncClass GstRtpSPEEXEncClass;
+
+#define GST_TYPE_RTP_SPEEX_ENC \
+  (gst_rtpspeexenc_get_type())
+#define GST_RTP_SPEEX_ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
+#define GST_RTP_SPEEX_ENC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
+#define GST_IS_RTP_SPEEX_ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_ENC))
+#define GST_IS_RTP_SPEEX_ENC_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_ENC))
+
+struct _GstRtpSPEEXEnc
+{
+  GstBaseRTPPayload payload;
+};
+
+struct _GstRtpSPEEXEncClass
+{
+  GstBaseRTPPayloadClass parent_class;
+};
+
+gboolean gst_rtpspeexenc_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_SPEEX_ENC_H__ */