rtp: add G722 pay and depayloader

gst/rtp/Makefile.am

@@ -23,6 +23,8 @@ libgstrtp_la_SOURCES = \
 	gstrtppcmudepay.c \
 	gstrtppcmupay.c \
 	gstrtppcmapay.c \
+	gstrtpg722depay.c \
+	gstrtpg722pay.c \
 	gstrtpg723depay.c \
 	gstrtpg723pay.c \
 	gstrtpg726pay.c \
@@ -106,6 +108,8 @@ noinst_HEADERS =			\
 		 gstrtppcmudepay.h	\
 		 gstrtppcmupay.h 	\
 		 gstrtppcmapay.h 	\
+	         gstrtpg722depay.h      \
+	         gstrtpg722pay.h        \
 	         gstrtpg723depay.h      \
 	         gstrtpg723pay.h        \
 	         gstrtpg726depay.h      \

gst/rtp/gstrtp.c

@@ -35,6 +35,8 @@
 #include "gstrtppcmapay.h"
 #include "gstrtppcmadepay.h"
 #include "gstrtppcmudepay.h"
+#include "gstrtpg722depay.h"
+#include "gstrtpg722pay.h"
 #include "gstrtpg723depay.h"
 #include "gstrtpg723pay.h"
 #include "gstrtpg726depay.h"
@@ -119,6 +121,12 @@ plugin_init (GstPlugin * plugin)
   if (!gst_rtp_ilbc_depay_plugin_init (plugin))
     return FALSE;
 
+  if (!gst_rtp_g722_depay_plugin_init (plugin))
+    return FALSE;
+
+  if (!gst_rtp_g722_pay_plugin_init (plugin))
+    return FALSE;
+
   if (!gst_rtp_g723_depay_plugin_init (plugin))
     return FALSE;
 

gst/rtp/gstrtpg722depay.c

@@ -0,0 +1,260 @@
+/* GStreamer
+ * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * 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 <string.h>
+#include <stdlib.h>
+
+#include <gst/audio/audio.h>
+#include <gst/audio/multichannel.h>
+
+#include "gstrtpg722depay.h"
+#include "gstrtpchannels.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpg722depay_debug);
+#define GST_CAT_DEFAULT (rtpg722depay_debug)
+
+static GstStaticPadTemplate gst_rtp_g722_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/G722, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
+    );
+
+static GstStaticPadTemplate gst_rtp_g722_depay_sink_template =
+    GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "clock-rate = (int) 8000, "
+        /* "channels = (int) [1, MAX]"  */
+        /* "channel-order = (string) ANY" */
+        "encoding-name = (string) \"G722\";"
+        "application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
+        "clock-rate = (int) [ 1, MAX ]"
+        /* "channels = (int) [1, MAX]" */
+        /* "emphasis = (string) ANY" */
+        /* "channel-order = (string) ANY" */
+    )
+    );
+
+GST_BOILERPLATE (GstRtpG722Depay, gst_rtp_g722_depay, GstBaseRTPDepayload,
+    GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static gboolean gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload,
+    GstCaps * caps);
+static GstBuffer *gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload,
+    GstBuffer * buf);
+
+static void
+gst_rtp_g722_depay_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_rtp_g722_depay_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_g722_depay_sink_template));
+
+  gst_element_class_set_details_simple (element_class, "RTP audio depayloader",
+      "Codec/Depayloader/Network",
+      "Extracts G722 audio from RTP packets",
+      "Wim Taymans <wim.taymans@gmail.com>");
+}
+
+static void
+gst_rtp_g722_depay_class_init (GstRtpG722DepayClass * klass)
+{
+  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+  gstbasertpdepayload_class->set_caps = gst_rtp_g722_depay_setcaps;
+  gstbasertpdepayload_class->process = gst_rtp_g722_depay_process;
+
+  GST_DEBUG_CATEGORY_INIT (rtpg722depay_debug, "rtpg722depay", 0,
+      "G722 RTP Depayloader");
+}
+
+static void
+gst_rtp_g722_depay_init (GstRtpG722Depay * rtpg722depay,
+    GstRtpG722DepayClass * klass)
+{
+  /* needed because of GST_BOILERPLATE */
+}
+
+static gint
+gst_rtp_g722_depay_parse_int (GstStructure * structure, const gchar * field,
+    gint def)
+{
+  const gchar *str;
+  gint res;
+
+  if ((str = gst_structure_get_string (structure, field)))
+    return atoi (str);
+
+  if (gst_structure_get_int (structure, field, &res))
+    return res;
+
+  return def;
+}
+
+static gboolean
+gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+{
+  GstStructure *structure;
+  GstRtpG722Depay *rtpg722depay;
+  gint clock_rate, payload, samplerate;
+  gint channels;
+  GstCaps *srccaps;
+  gboolean res;
+  const gchar *channel_order;
+  const GstRTPChannelOrder *order;
+
+  rtpg722depay = GST_RTP_G722_DEPAY (depayload);
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  payload = 96;
+  gst_structure_get_int (structure, "payload", &payload);
+  switch (payload) {
+    case GST_RTP_PAYLOAD_G722:
+      channels = 1;
+      clock_rate = 8000;
+      samplerate = 16000;
+      break;
+    default:
+      /* no fixed mapping, we need clock-rate */
+      channels = 0;
+      clock_rate = 0;
+      samplerate = 0;
+      break;
+  }
+
+  /* caps can overwrite defaults */
+  clock_rate =
+      gst_rtp_g722_depay_parse_int (structure, "clock-rate", clock_rate);
+  if (clock_rate == 0)
+    goto no_clockrate;
+
+  if (clock_rate == 8000)
+    samplerate = 16000;
+
+  if (samplerate == 0)
+    samplerate = clock_rate;
+
+  channels =
+      gst_rtp_g722_depay_parse_int (structure, "encoding-params", channels);
+  if (channels == 0) {
+    channels = gst_rtp_g722_depay_parse_int (structure, "channels", channels);
+    if (channels == 0) {
+      /* channels defaults to 1 otherwise */
+      channels = 1;
+    }
+  }
+
+  depayload->clock_rate = clock_rate;
+  rtpg722depay->rate = samplerate;
+  rtpg722depay->channels = channels;
+
+  srccaps = gst_caps_new_simple ("audio/G722",
+      "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
+
+  /* add channel positions */
+  channel_order = gst_structure_get_string (structure, "channel-order");
+
+  order = gst_rtp_channels_get_by_order (channels, channel_order);
+  if (order) {
+    gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
+        order->pos);
+  } else {
+    GstAudioChannelPosition *pos;
+
+    GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
+        (NULL), ("Unknown channel order '%s' for %d channels",
+            GST_STR_NULL (channel_order), channels));
+    /* create default NONE layout */
+    pos = gst_rtp_channels_create_default (channels);
+    gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
+    g_free (pos);
+  }
+
+  res = gst_pad_set_caps (depayload->srcpad, srccaps);
+  gst_caps_unref (srccaps);
+
+  return res;
+
+  /* ERRORS */
+no_clockrate:
+  {
+    GST_ERROR_OBJECT (depayload, "no clock-rate specified");
+    return FALSE;
+  }
+}
+
+static GstBuffer *
+gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+  GstRtpG722Depay *rtpg722depay;
+  GstBuffer *outbuf;
+  gint payload_len;
+  gboolean marker;
+
+  rtpg722depay = GST_RTP_G722_DEPAY (depayload);
+
+  payload_len = gst_rtp_buffer_get_payload_len (buf);
+
+  if (payload_len <= 0)
+    goto empty_packet;
+
+  GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len);
+
+  outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+  marker = gst_rtp_buffer_get_marker (buf);
+
+  if (marker) {
+    /* mark talk spurt with DISCONT */
+    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+  }
+
+  return outbuf;
+
+  /* ERRORS */
+empty_packet:
+  {
+    GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
+        ("Empty Payload."), (NULL));
+    return NULL;
+  }
+}
+
+gboolean
+gst_rtp_g722_depay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpg722depay",
+      GST_RANK_MARGINAL, GST_TYPE_RTP_G722_DEPAY);
+}

gst/rtp/gstrtpg722depay.h

@@ -0,0 +1,64 @@
+/* GStreamer
+ * Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * 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.
+ */
+
+#ifndef __GST_RTP_G722_DEPAY_H__
+#define __GST_RTP_G722_DEPAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpdepayload.h>
+
+G_BEGIN_DECLS
+
+/* Standard macros for defining types for this element.  */
+#define GST_TYPE_RTP_G722_DEPAY \
+  (gst_rtp_g722_depay_get_type())
+#define GST_RTP_G722_DEPAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G722_DEPAY,GstRtpG722Depay))
+#define GST_RTP_G722_DEPAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G722_DEPAY,GstRtpG722DepayClass))
+#define GST_IS_RTP_G722_DEPAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G722_DEPAY))
+#define GST_IS_RTP_G722_DEPAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G722_DEPAY))
+
+typedef struct _GstRtpG722Depay GstRtpG722Depay;
+typedef struct _GstRtpG722DepayClass GstRtpG722DepayClass;
+
+/* Definition of structure storing data for this element. */
+struct _GstRtpG722Depay
+{
+  GstBaseRTPDepayload depayload;
+
+  guint rate;
+  guint channels;
+};
+
+/* Standard definition defining a class for this element. */
+struct _GstRtpG722DepayClass
+{
+  GstBaseRTPDepayloadClass parent_class;
+};
+
+GType gst_rtp_g722_depay_get_type (void);
+
+gboolean gst_rtp_g722_depay_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_G722_DEPAY_H__ */

gst/rtp/gstrtpg722pay.c

@@ -0,0 +1,209 @@
+/* GStreamer
+ * Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * 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 <string.h>
+
+#include <gst/audio/audio.h>
+#include <gst/audio/multichannel.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpg722pay.h"
+#include "gstrtpchannels.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpg722pay_debug);
+#define GST_CAT_DEFAULT (rtpg722pay_debug)
+
+static GstStaticPadTemplate gst_rtp_g722_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/G722, " "rate = (int) 16000, " "channels = (int) 1")
+    );
+
+static GstStaticPadTemplate gst_rtp_g722_pay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "encoding-name = (string) \"G722\", "
+        "payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
+        "clock-rate = (int) 8000")
+    );
+
+static gboolean gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload,
+    GstCaps * caps);
+static GstCaps *gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload,
+    GstPad * pad);
+
+GST_BOILERPLATE (GstRtpG722Pay, gst_rtp_g722_pay, GstBaseRTPAudioPayload,
+    GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+
+static void
+gst_rtp_g722_pay_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_rtp_g722_pay_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_g722_pay_sink_template));
+
+  gst_element_class_set_details_simple (element_class, "RTP audio payloader",
+      "Codec/Payloader/Network",
+      "Payload-encode Raw audio into RTP packets (RFC 3551)",
+      "Wim Taymans <wim.taymans@gmail.com>");
+}
+
+static void
+gst_rtp_g722_pay_class_init (GstRtpG722PayClass * klass)
+{
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  gstbasertppayload_class->set_caps = gst_rtp_g722_pay_setcaps;
+  gstbasertppayload_class->get_caps = gst_rtp_g722_pay_getcaps;
+
+  GST_DEBUG_CATEGORY_INIT (rtpg722pay_debug, "rtpg722pay", 0,
+      "G722 RTP Payloader");
+}
+
+static void
+gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay, GstRtpG722PayClass * klass)
+{
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg722pay);
+
+  /* tell basertpaudiopayload that this is a sample based codec */
+  gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
+}
+
+static gboolean
+gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
+{
+  GstRtpG722Pay *rtpg722pay;
+  GstStructure *structure;
+  gint rate, channels, clock_rate;
+  gboolean res;
+  gchar *params;
+  GstAudioChannelPosition *pos;
+  const GstRTPChannelOrder *order;
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
+  rtpg722pay = GST_RTP_G722_PAY (basepayload);
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  /* first parse input caps */
+  if (!gst_structure_get_int (structure, "rate", &rate))
+    goto no_rate;
+
+  if (!gst_structure_get_int (structure, "channels", &channels))
+    goto no_channels;
+
+  /* get the channel order */
+  pos = gst_audio_get_channel_positions (structure);
+  if (pos)
+    order = gst_rtp_channels_get_by_pos (channels, pos);
+  else
+    order = NULL;
+
+  if (rate == 16000)
+    clock_rate = 8000;
+
+  gst_basertppayload_set_options (basepayload, "audio", TRUE, "G722",
+      clock_rate);
+  params = g_strdup_printf ("%d", channels);
+
+  if (!order && channels > 2) {
+    GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
+        (NULL), ("Unknown channel order for %d channels", channels));
+  }
+
+  if (order && order->name) {
+    res = gst_basertppayload_set_outcaps (basepayload,
+        "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
+        channels, "channel-order", G_TYPE_STRING, order->name, NULL);
+  } else {
+    res = gst_basertppayload_set_outcaps (basepayload,
+        "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
+        channels, NULL);
+  }
+
+  g_free (params);
+  g_free (pos);
+
+  rtpg722pay->rate = rate;
+  rtpg722pay->channels = channels;
+
+  /* octet-per-sample is 1 * channels for G722 */
+  gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+      4 * rtpg722pay->channels);
+
+  return res;
+
+  /* ERRORS */
+no_rate:
+  {
+    GST_DEBUG_OBJECT (rtpg722pay, "no rate given");
+    return FALSE;
+  }
+no_channels:
+  {
+    GST_DEBUG_OBJECT (rtpg722pay, "no channels given");
+    return FALSE;
+  }
+}
+
+static GstCaps *
+gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
+{
+  GstCaps *otherpadcaps;
+  GstCaps *caps;
+
+  otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
+  caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
+
+  if (otherpadcaps) {
+    if (!gst_caps_is_empty (otherpadcaps)) {
+      GstStructure *structure;
+
+      structure = gst_caps_get_structure (otherpadcaps, 0);
+
+      gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
+      gst_caps_set_simple (caps, "rate", G_TYPE_INT, 16000, NULL);
+    }
+    gst_caps_unref (otherpadcaps);
+  }
+  return caps;
+}
+
+gboolean
+gst_rtp_g722_pay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpg722pay",
+      GST_RANK_NONE, GST_TYPE_RTP_G722_PAY);
+}

gst/rtp/gstrtpg722pay.h

@@ -0,0 +1,61 @@
+/* GStreamer
+ * Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * 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.
+ */
+
+#ifndef __GST_RTP_G722_PAY_H__
+#define __GST_RTP_G722_PAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpaudiopayload.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_G722_PAY \
+  (gst_rtp_g722_pay_get_type())
+#define GST_RTP_G722_PAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G722_PAY,GstRtpG722Pay))
+#define GST_RTP_G722_PAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G722_PAY,GstRtpG722PayClass))
+#define GST_IS_RTP_G722_PAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G722_PAY))
+#define GST_IS_RTP_G722_PAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G722_PAY))
+
+typedef struct _GstRtpG722Pay GstRtpG722Pay;
+typedef struct _GstRtpG722PayClass GstRtpG722PayClass;
+
+struct _GstRtpG722Pay
+{
+  GstBaseRTPAudioPayload payload;
+
+  gint rate;
+  gint channels;
+};
+
+struct _GstRtpG722PayClass
+{
+  GstBaseRTPAudioPayloadClass parent_class;
+};
+
+GType gst_rtp_g722_pay_get_type (void);
+
+gboolean gst_rtp_g722_pay_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_G722_PAY_H__ */