gst/rtp/gstrtpamrdepay.c - imx-gst-plugins-good - Git at Google

 /* GStreamer
  * Copyright (C) <2005> 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., 51 Franklin St, Fifth Floor,
  * Boston, MA 02110-1301, USA.
  */

 /**
  * SECTION:element-rtpamrdepay
  * @see_also: rtpamrpay
  *
  * Extract AMR audio from RTP packets according to RFC 3267.
  * For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
  *
  * <refsect2>
  * <title>Example pipeline</title>
  * |[
  * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)AMR, encoding-params=(string)1, octet-align=(string)1, payload=(int)96' ! rtpamrdepay ! amrnbdec ! pulsesink
  * ]| This example pipeline will depayload and decode an RTP AMR stream. Refer to
  * the rtpamrpay example to create the RTP stream.
  * </refsect2>
  */

 /*
  * RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
  * Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
  * Wideband (AMR-WB) Audio Codecs.
  *
  */
 #ifdef HAVE_CONFIG_H
 #  include "config.h"
 #endif

 #include <gst/rtp/gstrtpbuffer.h>
 #include <gst/audio/audio.h>

 #include <stdlib.h>
 #include <string.h>
 #include "gstrtpamrdepay.h"
 #include "gstrtputils.h"

 GST_DEBUG_CATEGORY_STATIC (rtpamrdepay_debug);
 #define GST_CAT_DEFAULT (rtpamrdepay_debug)

 /* RtpAMRDepay signals and args */
 enum
 {
   /* FILL ME */
   LAST_SIGNAL
 };

 enum
 {
   PROP_0
 };

 /* input is an RTP packet
  *
  * params see RFC 3267, section 8.1
  */
 static GstStaticPadTemplate gst_rtp_amr_depay_sink_template =
     GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "media = (string) \"audio\", "
         "clock-rate = (int) 8000, " "encoding-name = (string) \"AMR\", "
         /* This is the default, so the peer doesn't have to specify it
          * "encoding-params = (string) \"1\", " */
         /* NOTE that all values must be strings in orde to be able to do SDP <->
          * GstCaps mapping. */
         "octet-align = (string) \"1\";"
         /* following options are not needed for a decoder
          *
          "crc = (string) { \"0\", \"1\" }, "
          "robust-sorting = (string) \"0\", "
          "interleaving = (string) \"0\";"
          "mode-set = (int) [ 0, 7 ], "
          "mode-change-period = (int) [ 1, MAX ], "
          "mode-change-neighbor = (boolean) { TRUE, FALSE }, "
          "maxptime = (int) [ 20, MAX ], "
          "ptime = (int) [ 20, MAX ]"
          */
         "application/x-rtp, "
         "media = (string) \"audio\", "
         "clock-rate = (int) 16000, " "encoding-name = (string) \"AMR-WB\", "
         /* This is the default, so the peer doesn't have to specify it
          * "encoding-params = (string) \"1\", " */
         /* NOTE that all values must be strings in orde to be able to do SDP <->
          * GstCaps mapping. */
         "octet-align = (string) \"1\";"
         /* following options are not needed for a decoder
          *
          "crc = (string) { \"0\", \"1\" }, "
          "robust-sorting = (string) \"0\", "
          "interleaving = (string) \"0\""
          "mode-set = (int) [ 0, 7 ], "
          "mode-change-period = (int) [ 1, MAX ], "
          "mode-change-neighbor = (boolean) { TRUE, FALSE }, "
          "maxptime = (int) [ 20, MAX ], "
          "ptime = (int) [ 20, MAX ]"
          */
     )
     );

 static GstStaticPadTemplate gst_rtp_amr_depay_src_template =
     GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000;"
         "audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000")
     );

 static gboolean gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload,
     GstCaps * caps);
 static GstBuffer *gst_rtp_amr_depay_process (GstRTPBaseDepayload * depayload,
     GstRTPBuffer * rtp);

 #define gst_rtp_amr_depay_parent_class parent_class
 G_DEFINE_TYPE (GstRtpAMRDepay, gst_rtp_amr_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);

 static void
 gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass)
 {
   GstElementClass *gstelement_class;
   GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;

   gstelement_class = (GstElementClass *) klass;
   gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;

   gst_element_class_add_static_pad_template (gstelement_class,
       &gst_rtp_amr_depay_src_template);
   gst_element_class_add_static_pad_template (gstelement_class,
       &gst_rtp_amr_depay_sink_template);

   gst_element_class_set_static_metadata (gstelement_class,
       "RTP AMR depayloader", "Codec/Depayloader/Network/RTP",
       "Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
       "Wim Taymans <wim.taymans@gmail.com>");

   gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_amr_depay_process;
   gstrtpbasedepayload_class->set_caps = gst_rtp_amr_depay_setcaps;

   GST_DEBUG_CATEGORY_INIT (rtpamrdepay_debug, "rtpamrdepay", 0,
       "AMR/AMR-WB RTP Depayloader");
 }

 static void
 gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay)
 {
   GstRTPBaseDepayload *depayload;

   depayload = GST_RTP_BASE_DEPAYLOAD (rtpamrdepay);

   gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
 }

 static gboolean
 gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
 {
   GstStructure *structure;
   GstCaps *srccaps;
   GstRtpAMRDepay *rtpamrdepay;
   const gchar *params;
   const gchar *str, *type;
   gint clock_rate, need_clock_rate;
   gboolean res;

   rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);

   structure = gst_caps_get_structure (caps, 0);

   /* figure out the mode first and set the clock rates */
   if ((str = gst_structure_get_string (structure, "encoding-name"))) {
     if (strcmp (str, "AMR") == 0) {
       rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB;
       need_clock_rate = 8000;
       type = "audio/AMR";
     } else if (strcmp (str, "AMR-WB") == 0) {
       rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
       need_clock_rate = 16000;
       type = "audio/AMR-WB";
     } else
       goto invalid_mode;
   } else
     goto invalid_mode;

   if (!(str = gst_structure_get_string (structure, "octet-align")))
     rtpamrdepay->octet_align = FALSE;
   else
     rtpamrdepay->octet_align = (atoi (str) == 1);

   if (!(str = gst_structure_get_string (structure, "crc")))
     rtpamrdepay->crc = FALSE;
   else
     rtpamrdepay->crc = (atoi (str) == 1);

   if (rtpamrdepay->crc) {
     /* crc mode implies octet aligned mode */
     rtpamrdepay->octet_align = TRUE;
   }

   if (!(str = gst_structure_get_string (structure, "robust-sorting")))
     rtpamrdepay->robust_sorting = FALSE;
   else
     rtpamrdepay->robust_sorting = (atoi (str) == 1);

   if (rtpamrdepay->robust_sorting) {
     /* robust_sorting mode implies octet aligned mode */
     rtpamrdepay->octet_align = TRUE;
   }

   if (!(str = gst_structure_get_string (structure, "interleaving")))
     rtpamrdepay->interleaving = FALSE;
   else
     rtpamrdepay->interleaving = (atoi (str) == 1);

   if (rtpamrdepay->interleaving) {
     /* interleaving mode implies octet aligned mode */
     rtpamrdepay->octet_align = TRUE;
   }

   if (!(params = gst_structure_get_string (structure, "encoding-params")))
     rtpamrdepay->channels = 1;
   else {
     rtpamrdepay->channels = atoi (params);
   }

   if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
     clock_rate = need_clock_rate;
   depayload->clock_rate = clock_rate;

   /* we require 1 channel, 8000 Hz, octet aligned, no CRC,
    * no robust sorting, no interleaving for now */
   if (rtpamrdepay->channels != 1)
     return FALSE;
   if (clock_rate != need_clock_rate)
     return FALSE;
   if (rtpamrdepay->octet_align != TRUE)
     return FALSE;
   if (rtpamrdepay->robust_sorting != FALSE)
     return FALSE;
   if (rtpamrdepay->interleaving != FALSE)
     return FALSE;

   srccaps = gst_caps_new_simple (type,
       "channels", G_TYPE_INT, rtpamrdepay->channels,
       "rate", G_TYPE_INT, clock_rate, NULL);
   res = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
   gst_caps_unref (srccaps);

   return res;

   /* ERRORS */
 invalid_mode:
   {
     GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
     return FALSE;
   }
 }

 /* -1 is invalid */
 static const gint nb_frame_size[16] = {
   12, 13, 15, 17, 19, 20, 26, 31,
   5, -1, -1, -1, -1, -1, -1, 0
 };

 static const gint wb_frame_size[16] = {
   17, 23, 32, 36, 40, 46, 50, 58,
   60, 5, -1, -1, -1, -1, -1, 0
 };

 static GstBuffer *
 gst_rtp_amr_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
 {
   GstRtpAMRDepay *rtpamrdepay;
   const gint *frame_size;
   GstBuffer *outbuf = NULL;
   gint payload_len;
   GstMapInfo map;

   rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);

   /* setup frame size pointer */
   if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
     frame_size = nb_frame_size;
   else
     frame_size = wb_frame_size;

   /* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
    * no robust sorting, no interleaving data is to be depayloaded */
   {
     guint8 *payload, *p, *dp;
     gint i, num_packets, num_nonempty_packets;
     gint amr_len;
     gint ILL, ILP;

     payload_len = gst_rtp_buffer_get_payload_len (rtp);

     /* need at least 2 bytes for the header */
     if (payload_len < 2)
       goto too_small;

     payload = gst_rtp_buffer_get_payload (rtp);

     /* depay CMR. The CMR is used by the sender to request
      * a new encoding mode.
      *
      *  0 1 2 3 4 5 6 7
      * +-+-+-+-+-+-+-+-+
      * | CMR   |R|R|R|R|
      * +-+-+-+-+-+-+-+-+
      */
     /* CMR = (payload[0] & 0xf0) >> 4; */

     /* strip CMR header now, pack FT and the data for the decoder */
     payload_len -= 1;
     payload += 1;

     GST_DEBUG_OBJECT (rtpamrdepay, "payload len %d", payload_len);

     if (rtpamrdepay->interleaving) {
       ILL = (payload[0] & 0xf0) >> 4;
       ILP = (payload[0] & 0x0f);

       payload_len -= 1;
       payload += 1;

       if (ILP > ILL)
         goto wrong_interleaving;
     }

     /*
      *  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
      * +-+-+-+-+-+-+-+-+..
      * |F|  FT   |Q|P|P| more FT..
      * +-+-+-+-+-+-+-+-+..
      */
     /* count number of packets by counting the FTs. Also
      * count number of amr data bytes and number of non-empty
      * packets (this is also the number of CRCs if present). */
     amr_len = 0;
     num_nonempty_packets = 0;
     num_packets = 0;
     for (i = 0; i < payload_len; i++) {
       gint fr_size;
       guint8 FT;

       FT = (payload[i] & 0x78) >> 3;

       fr_size = frame_size[FT];
       GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
       if (fr_size == -1)
         goto wrong_framesize;

       if (fr_size > 0) {
         amr_len += fr_size;
         num_nonempty_packets++;
       }
       num_packets++;

       if ((payload[i] & 0x80) == 0)
         break;
     }

     if (rtpamrdepay->crc) {
       /* data len + CRC len + header bytes should be smaller than payload_len */
       if (num_packets + num_nonempty_packets + amr_len > payload_len)
         goto wrong_length_1;
     } else {
       /* data len + header bytes should be smaller than payload_len */
       if (num_packets + amr_len > payload_len)
         goto wrong_length_2;
     }

     outbuf = gst_buffer_new_and_alloc (payload_len);

     /* point to destination */
     gst_buffer_map (outbuf, &map, GST_MAP_WRITE);

     /* point to first data packet */
     p = map.data;
     dp = payload + num_packets;
     if (rtpamrdepay->crc) {
       /* skip CRC if present */
       dp += num_nonempty_packets;
     }

     for (i = 0; i < num_packets; i++) {
       gint fr_size;

       /* copy FT, clear F bit */
       *p++ = payload[i] & 0x7f;

       fr_size = frame_size[(payload[i] & 0x78) >> 3];
       if (fr_size > 0) {
         /* copy data packet, FIXME, calc CRC here. */
         memcpy (p, dp, fr_size);

         p += fr_size;
         dp += fr_size;
       }
     }
     gst_buffer_unmap (outbuf, &map);

     /* we can set the duration because each packet is 20 milliseconds */
     GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;

     if (gst_rtp_buffer_get_marker (rtp)) {
       /* marker bit marks a buffer after a talkspurt. */
       GST_DEBUG_OBJECT (depayload, "marker bit was set");
       GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
     }

     GST_DEBUG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
         gst_buffer_get_size (outbuf));

     gst_rtp_copy_audio_meta (rtpamrdepay, outbuf, rtp->buffer);
   }

   return outbuf;

   /* ERRORS */
 too_small:
   {
     GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
         (NULL), ("AMR RTP payload too small (%d)", payload_len));
     goto bad_packet;
   }
 wrong_interleaving:
   {
     GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
         (NULL), ("AMR RTP wrong interleaving"));
     goto bad_packet;
   }
 wrong_framesize:
   {
     GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
         (NULL), ("AMR RTP frame size == -1"));
     goto bad_packet;
   }
 wrong_length_1:
   {
     GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
         (NULL), ("AMR RTP wrong length 1"));
     goto bad_packet;
   }
 wrong_length_2:
   {
     GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
         (NULL), ("AMR RTP wrong length 2"));
     goto bad_packet;
   }
 bad_packet:
   {
     /* no fatal error */
     return NULL;
   }
 }

 gboolean
 gst_rtp_amr_depay_plugin_init (GstPlugin * plugin)
 {
   return gst_element_register (plugin, "rtpamrdepay",
       GST_RANK_SECONDARY, GST_TYPE_RTP_AMR_DEPAY);
 }
	/* GStreamer
	* Copyright (C) <2005> 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., 51 Franklin St, Fifth Floor,
	* Boston, MA 02110-1301, USA.
	*/

	/**
	* SECTION:element-rtpamrdepay
	* @see_also: rtpamrpay
	*
	* Extract AMR audio from RTP packets according to RFC 3267.
	* For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
	*
	* <refsect2>
	* <title>Example pipeline</title>
	* \|[
	* gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)AMR, encoding-params=(string)1, octet-align=(string)1, payload=(int)96' ! rtpamrdepay ! amrnbdec ! pulsesink
	* ]\| This example pipeline will depayload and decode an RTP AMR stream. Refer to
	* the rtpamrpay example to create the RTP stream.
	* </refsect2>
	*/

	/*
	* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
	* Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
	* Wideband (AMR-WB) Audio Codecs.
	*
	*/
	#ifdef HAVE_CONFIG_H
	# include "config.h"
	#endif

	#include <gst/rtp/gstrtpbuffer.h>
	#include <gst/audio/audio.h>

	#include <stdlib.h>
	#include <string.h>
	#include "gstrtpamrdepay.h"
	#include "gstrtputils.h"

	GST_DEBUG_CATEGORY_STATIC (rtpamrdepay_debug);
	#define GST_CAT_DEFAULT (rtpamrdepay_debug)

	/* RtpAMRDepay signals and args */
	enum
	{
	/* FILL ME */
	LAST_SIGNAL
	};

	enum
	{
	PROP_0
	};

	/* input is an RTP packet
	*
	* params see RFC 3267, section 8.1
	*/
	static GstStaticPadTemplate gst_rtp_amr_depay_sink_template =
	GST_STATIC_PAD_TEMPLATE ("sink",
	GST_PAD_SINK,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("application/x-rtp, "
	"media = (string) \"audio\", "
	"clock-rate = (int) 8000, " "encoding-name = (string) \"AMR\", "
	/* This is the default, so the peer doesn't have to specify it
	* "encoding-params = (string) \"1\", " */
	/* NOTE that all values must be strings in orde to be able to do SDP <->
	* GstCaps mapping. */
	"octet-align = (string) \"1\";"
	/* following options are not needed for a decoder
	*
	"crc = (string) { \"0\", \"1\" }, "
	"robust-sorting = (string) \"0\", "
	"interleaving = (string) \"0\";"
	"mode-set = (int) [ 0, 7 ], "
	"mode-change-period = (int) [ 1, MAX ], "
	"mode-change-neighbor = (boolean) { TRUE, FALSE }, "
	"maxptime = (int) [ 20, MAX ], "
	"ptime = (int) [ 20, MAX ]"
	*/
	"application/x-rtp, "
	"media = (string) \"audio\", "
	"clock-rate = (int) 16000, " "encoding-name = (string) \"AMR-WB\", "
	/* This is the default, so the peer doesn't have to specify it
	* "encoding-params = (string) \"1\", " */
	/* NOTE that all values must be strings in orde to be able to do SDP <->
	* GstCaps mapping. */
	"octet-align = (string) \"1\";"
	/* following options are not needed for a decoder
	*
	"crc = (string) { \"0\", \"1\" }, "
	"robust-sorting = (string) \"0\", "
	"interleaving = (string) \"0\""
	"mode-set = (int) [ 0, 7 ], "
	"mode-change-period = (int) [ 1, MAX ], "
	"mode-change-neighbor = (boolean) { TRUE, FALSE }, "
	"maxptime = (int) [ 20, MAX ], "
	"ptime = (int) [ 20, MAX ]"
	*/
	)
	);

	static GstStaticPadTemplate gst_rtp_amr_depay_src_template =
	GST_STATIC_PAD_TEMPLATE ("src",
	GST_PAD_SRC,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000;"
	"audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000")
	);

	static gboolean gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload,
	GstCaps * caps);
	static GstBuffer gst_rtp_amr_depay_process (GstRTPBaseDepayload depayload,
	GstRTPBuffer * rtp);

	#define gst_rtp_amr_depay_parent_class parent_class
	G_DEFINE_TYPE (GstRtpAMRDepay, gst_rtp_amr_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);

	static void
	gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass)
	{
	GstElementClass *gstelement_class;
	GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;

	gstelement_class = (GstElementClass *) klass;
	gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;

	gst_element_class_add_static_pad_template (gstelement_class,
	&gst_rtp_amr_depay_src_template);
	gst_element_class_add_static_pad_template (gstelement_class,
	&gst_rtp_amr_depay_sink_template);

	gst_element_class_set_static_metadata (gstelement_class,
	"RTP AMR depayloader", "Codec/Depayloader/Network/RTP",
	"Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
	"Wim Taymans <wim.taymans@gmail.com>");

	gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_amr_depay_process;
	gstrtpbasedepayload_class->set_caps = gst_rtp_amr_depay_setcaps;

	GST_DEBUG_CATEGORY_INIT (rtpamrdepay_debug, "rtpamrdepay", 0,
	"AMR/AMR-WB RTP Depayloader");
	}

	static void
	gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay)
	{
	GstRTPBaseDepayload *depayload;

	depayload = GST_RTP_BASE_DEPAYLOAD (rtpamrdepay);

	gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
	}

	static gboolean
	gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
	{
	GstStructure *structure;
	GstCaps *srccaps;
	GstRtpAMRDepay *rtpamrdepay;
	const gchar *params;
	const gchar str, type;
	gint clock_rate, need_clock_rate;
	gboolean res;

	rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);

	structure = gst_caps_get_structure (caps, 0);

	/* figure out the mode first and set the clock rates */
	if ((str = gst_structure_get_string (structure, "encoding-name"))) {
	if (strcmp (str, "AMR") == 0) {
	rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB;
	need_clock_rate = 8000;
	type = "audio/AMR";
	} else if (strcmp (str, "AMR-WB") == 0) {
	rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
	need_clock_rate = 16000;
	type = "audio/AMR-WB";
	} else
	goto invalid_mode;
	} else
	goto invalid_mode;

	if (!(str = gst_structure_get_string (structure, "octet-align")))
	rtpamrdepay->octet_align = FALSE;
	else
	rtpamrdepay->octet_align = (atoi (str) == 1);

	if (!(str = gst_structure_get_string (structure, "crc")))
	rtpamrdepay->crc = FALSE;
	else
	rtpamrdepay->crc = (atoi (str) == 1);

	if (rtpamrdepay->crc) {
	/* crc mode implies octet aligned mode */
	rtpamrdepay->octet_align = TRUE;
	}

	if (!(str = gst_structure_get_string (structure, "robust-sorting")))
	rtpamrdepay->robust_sorting = FALSE;
	else
	rtpamrdepay->robust_sorting = (atoi (str) == 1);

	if (rtpamrdepay->robust_sorting) {
	/* robust_sorting mode implies octet aligned mode */
	rtpamrdepay->octet_align = TRUE;
	}

	if (!(str = gst_structure_get_string (structure, "interleaving")))
	rtpamrdepay->interleaving = FALSE;
	else
	rtpamrdepay->interleaving = (atoi (str) == 1);

	if (rtpamrdepay->interleaving) {
	/* interleaving mode implies octet aligned mode */
	rtpamrdepay->octet_align = TRUE;
	}

	if (!(params = gst_structure_get_string (structure, "encoding-params")))
	rtpamrdepay->channels = 1;
	else {
	rtpamrdepay->channels = atoi (params);
	}

	if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
	clock_rate = need_clock_rate;
	depayload->clock_rate = clock_rate;

	/* we require 1 channel, 8000 Hz, octet aligned, no CRC,
	* no robust sorting, no interleaving for now */
	if (rtpamrdepay->channels != 1)
	return FALSE;
	if (clock_rate != need_clock_rate)
	return FALSE;
	if (rtpamrdepay->octet_align != TRUE)
	return FALSE;
	if (rtpamrdepay->robust_sorting != FALSE)
	return FALSE;
	if (rtpamrdepay->interleaving != FALSE)
	return FALSE;

	srccaps = gst_caps_new_simple (type,
	"channels", G_TYPE_INT, rtpamrdepay->channels,
	"rate", G_TYPE_INT, clock_rate, NULL);
	res = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
	gst_caps_unref (srccaps);

	return res;

	/* ERRORS */
	invalid_mode:
	{
	GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
	return FALSE;
	}
	}

	/* -1 is invalid */
	static const gint nb_frame_size[16] = {
	12, 13, 15, 17, 19, 20, 26, 31,
	5, -1, -1, -1, -1, -1, -1, 0
	};

	static const gint wb_frame_size[16] = {
	17, 23, 32, 36, 40, 46, 50, 58,
	60, 5, -1, -1, -1, -1, -1, 0
	};

	static GstBuffer *
	gst_rtp_amr_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
	{
	GstRtpAMRDepay *rtpamrdepay;
	const gint *frame_size;
	GstBuffer *outbuf = NULL;
	gint payload_len;
	GstMapInfo map;

	rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);

	/* setup frame size pointer */
	if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
	frame_size = nb_frame_size;
	else
	frame_size = wb_frame_size;

	/* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
	* no robust sorting, no interleaving data is to be depayloaded */
	{
	guint8 payload, p, *dp;
	gint i, num_packets, num_nonempty_packets;
	gint amr_len;
	gint ILL, ILP;

	payload_len = gst_rtp_buffer_get_payload_len (rtp);

	/* need at least 2 bytes for the header */
	if (payload_len < 2)
	goto too_small;

	payload = gst_rtp_buffer_get_payload (rtp);

	/* depay CMR. The CMR is used by the sender to request
	* a new encoding mode.
	*
	* 0 1 2 3 4 5 6 7
	* +-+-+-+-+-+-+-+-+
	* \| CMR \|R\|R\|R\|R\|
	* +-+-+-+-+-+-+-+-+
	*/
	/* CMR = (payload[0] & 0xf0) >> 4; */

	/* strip CMR header now, pack FT and the data for the decoder */
	payload_len -= 1;
	payload += 1;

	GST_DEBUG_OBJECT (rtpamrdepay, "payload len %d", payload_len);

	if (rtpamrdepay->interleaving) {
	ILL = (payload[0] & 0xf0) >> 4;
	ILP = (payload[0] & 0x0f);

	payload_len -= 1;
	payload += 1;

	if (ILP > ILL)
	goto wrong_interleaving;
	}

	/*
	* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
	* +-+-+-+-+-+-+-+-+..
	* \|F\| FT \|Q\|P\|P\| more FT..
	* +-+-+-+-+-+-+-+-+..
	*/
	/* count number of packets by counting the FTs. Also
	* count number of amr data bytes and number of non-empty
	* packets (this is also the number of CRCs if present). */
	amr_len = 0;
	num_nonempty_packets = 0;
	num_packets = 0;
	for (i = 0; i < payload_len; i++) {
	gint fr_size;
	guint8 FT;

	FT = (payload[i] & 0x78) >> 3;

	fr_size = frame_size[FT];
	GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
	if (fr_size == -1)
	goto wrong_framesize;

	if (fr_size > 0) {
	amr_len += fr_size;
	num_nonempty_packets++;
	}
	num_packets++;

	if ((payload[i] & 0x80) == 0)
	break;
	}

	if (rtpamrdepay->crc) {
	/* data len + CRC len + header bytes should be smaller than payload_len */
	if (num_packets + num_nonempty_packets + amr_len > payload_len)
	goto wrong_length_1;
	} else {
	/* data len + header bytes should be smaller than payload_len */
	if (num_packets + amr_len > payload_len)
	goto wrong_length_2;
	}

	outbuf = gst_buffer_new_and_alloc (payload_len);

	/* point to destination */
	gst_buffer_map (outbuf, &map, GST_MAP_WRITE);

	/* point to first data packet */
	p = map.data;
	dp = payload + num_packets;
	if (rtpamrdepay->crc) {
	/* skip CRC if present */
	dp += num_nonempty_packets;
	}

	for (i = 0; i < num_packets; i++) {
	gint fr_size;

	/* copy FT, clear F bit */
	*p++ = payload[i] & 0x7f;

	fr_size = frame_size[(payload[i] & 0x78) >> 3];
	if (fr_size > 0) {
	/* copy data packet, FIXME, calc CRC here. */
	memcpy (p, dp, fr_size);

	p += fr_size;
	dp += fr_size;
	}
	}
	gst_buffer_unmap (outbuf, &map);

	/* we can set the duration because each packet is 20 milliseconds */
	GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;

	if (gst_rtp_buffer_get_marker (rtp)) {
	/* marker bit marks a buffer after a talkspurt. */
	GST_DEBUG_OBJECT (depayload, "marker bit was set");
	GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
	}

	GST_DEBUG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
	gst_buffer_get_size (outbuf));

	gst_rtp_copy_audio_meta (rtpamrdepay, outbuf, rtp->buffer);
	}

	return outbuf;

	/* ERRORS */
	too_small:
	{
	GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
	(NULL), ("AMR RTP payload too small (%d)", payload_len));
	goto bad_packet;
	}
	wrong_interleaving:
	{
	GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
	(NULL), ("AMR RTP wrong interleaving"));
	goto bad_packet;
	}
	wrong_framesize:
	{
	GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
	(NULL), ("AMR RTP frame size == -1"));
	goto bad_packet;
	}
	wrong_length_1:
	{
	GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
	(NULL), ("AMR RTP wrong length 1"));
	goto bad_packet;
	}
	wrong_length_2:
	{
	GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
	(NULL), ("AMR RTP wrong length 2"));
	goto bad_packet;
	}
	bad_packet:
	{
	/* no fatal error */
	return NULL;
	}
	}

	gboolean
	gst_rtp_amr_depay_plugin_init (GstPlugin * plugin)
	{
	return gst_element_register (plugin, "rtpamrdepay",
	GST_RANK_SECONDARY, GST_TYPE_RTP_AMR_DEPAY);
	}