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

 /* 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., 51 Franklin St, Fifth Floor,
  * Boston, MA 02110-1301, USA.
  */

 #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 "gstrtpqcelpdepay.h"
 #include "gstrtputils.h"

 GST_DEBUG_CATEGORY_STATIC (rtpqcelpdepay_debug);
 #define GST_CAT_DEFAULT (rtpqcelpdepay_debug)

 /* references:
  *
  * RFC 2658 - RTP Payload Format for PureVoice(tm) Audio
  */
 #define FRAME_DURATION (20 * GST_MSECOND)

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

 enum
 {
   PROP_0
 };

 static GstStaticPadTemplate gst_rtp_qcelp_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) \"QCELP\"; "
         "application/x-rtp, "
         "media = (string) \"audio\", "
         "payload = (int) " GST_RTP_PAYLOAD_QCELP_STRING ", "
         "clock-rate = (int) 8000")
     );

 static GstStaticPadTemplate gst_rtp_qcelp_depay_src_template =
 GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/qcelp, " "channels = (int) 1," "rate = (int) 8000")
     );

 static void gst_rtp_qcelp_depay_finalize (GObject * object);

 static gboolean gst_rtp_qcelp_depay_setcaps (GstRTPBaseDepayload * depayload,
     GstCaps * caps);
 static GstBuffer *gst_rtp_qcelp_depay_process (GstRTPBaseDepayload * depayload,
     GstRTPBuffer * rtp);

 #define gst_rtp_qcelp_depay_parent_class parent_class
 G_DEFINE_TYPE (GstRtpQCELPDepay, gst_rtp_qcelp_depay,
     GST_TYPE_RTP_BASE_DEPAYLOAD);

 static void
 gst_rtp_qcelp_depay_class_init (GstRtpQCELPDepayClass * klass)
 {
   GObjectClass *gobject_class;
   GstElementClass *gstelement_class;
   GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;

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

   gobject_class->finalize = gst_rtp_qcelp_depay_finalize;

   gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_qcelp_depay_process;
   gstrtpbasedepayload_class->set_caps = gst_rtp_qcelp_depay_setcaps;

   gst_element_class_add_static_pad_template (gstelement_class,
       &gst_rtp_qcelp_depay_src_template);
   gst_element_class_add_static_pad_template (gstelement_class,
       &gst_rtp_qcelp_depay_sink_template);

   gst_element_class_set_static_metadata (gstelement_class,
       "RTP QCELP depayloader", "Codec/Depayloader/Network/RTP",
       "Extracts QCELP (PureVoice) audio from RTP packets (RFC 2658)",
       "Wim Taymans <wim.taymans@gmail.com>");

   GST_DEBUG_CATEGORY_INIT (rtpqcelpdepay_debug, "rtpqcelpdepay", 0,
       "QCELP RTP Depayloader");
 }

 static void
 gst_rtp_qcelp_depay_init (GstRtpQCELPDepay * rtpqcelpdepay)
 {
 }

 static void
 gst_rtp_qcelp_depay_finalize (GObject * object)
 {
   GstRtpQCELPDepay *depay;

   depay = GST_RTP_QCELP_DEPAY (object);

   if (depay->packets != NULL) {
     g_ptr_array_foreach (depay->packets, (GFunc) gst_buffer_unref, NULL);
     g_ptr_array_free (depay->packets, TRUE);
     depay->packets = NULL;
   }

   G_OBJECT_CLASS (parent_class)->finalize (object);
 }


 static gboolean
 gst_rtp_qcelp_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
 {
   GstCaps *srccaps;
   gboolean res;

   srccaps = gst_caps_new_simple ("audio/qcelp",
       "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
   res = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
   gst_caps_unref (srccaps);

   return res;
 }

 static const gint frame_size[16] = {
   1, 4, 8, 17, 35, -8, 0, 0,
   0, 0, 0, 0, 0, 0, 1, 0
 };

 /* get the frame length, 0 is invalid, negative values are invalid but can be
  * recovered from. */
 static gint
 get_frame_len (GstRtpQCELPDepay * depay, guint8 frame_type)
 {
   if (frame_type >= G_N_ELEMENTS (frame_size))
     return 0;

   return frame_size[frame_type];
 }

 static guint
 count_packets (GstRtpQCELPDepay * depay, guint8 * data, guint size)
 {
   guint count = 0;

   while (size > 0) {
     gint frame_len;

     frame_len = get_frame_len (depay, data[0]);

     /* 0 is invalid and we throw away the remainder of the frames */
     if (frame_len == 0)
       break;

     if (frame_len < 0)
       frame_len = -frame_len;

     if (frame_len > size)
       break;

     size -= frame_len;
     data += frame_len;
     count++;
   }
   return count;
 }

 static void
 flush_packets (GstRtpQCELPDepay * depay)
 {
   guint i, size;

   GST_DEBUG_OBJECT (depay, "flushing packets");

   size = depay->packets->len;

   for (i = 0; i < size; i++) {
     GstBuffer *outbuf;

     outbuf = g_ptr_array_index (depay->packets, i);
     g_ptr_array_index (depay->packets, i) = NULL;

     gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (depay), outbuf);
   }

   /* and reset interleaving state */
   depay->interleaved = FALSE;
   depay->bundling = 0;
 }

 static void
 add_packet (GstRtpQCELPDepay * depay, guint LLL, guint NNN, guint index,
     GstBuffer * outbuf)
 {
   guint idx;
   GstBuffer *old;

   /* figure out the position in the array, note that index is never 0 because we
    * push those packets immediately. */
   idx = NNN + ((LLL + 1) * (index - 1));

   GST_DEBUG_OBJECT (depay, "adding packet at index %u", idx);
   /* free old buffer (should not happen) */
   old = g_ptr_array_index (depay->packets, idx);
   if (old)
     gst_buffer_unref (old);

   /* store new buffer */
   g_ptr_array_index (depay->packets, idx) = outbuf;
 }

 static GstBuffer *
 create_erasure_buffer (GstRtpQCELPDepay * depay)
 {
   GstBuffer *outbuf;
   GstMapInfo map;

   outbuf = gst_buffer_new_and_alloc (1);
   gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
   map.data[0] = 14;
   gst_buffer_unmap (outbuf, &map);

   return outbuf;
 }

 static GstBuffer *
 gst_rtp_qcelp_depay_process (GstRTPBaseDepayload * depayload,
     GstRTPBuffer * rtp)
 {
   GstRtpQCELPDepay *depay;
   GstBuffer *outbuf;
   GstClockTime timestamp;
   guint payload_len, offset, index;
   guint8 *payload;
   guint LLL, NNN;

   depay = GST_RTP_QCELP_DEPAY (depayload);

   payload_len = gst_rtp_buffer_get_payload_len (rtp);

   if (payload_len < 2)
     goto too_small;

   timestamp = GST_BUFFER_PTS (rtp->buffer);

   payload = gst_rtp_buffer_get_payload (rtp);

   /*  0 1 2 3 4 5 6 7
    * +-+-+-+-+-+-+-+-+
    * |RR | LLL | NNN |
    * +-+-+-+-+-+-+-+-+
    */
   /* RR = payload[0] >> 6; */
   LLL = (payload[0] & 0x38) >> 3;
   NNN = (payload[0] & 0x07);

   payload_len--;
   payload++;

   GST_DEBUG_OBJECT (depay, "LLL %u, NNN %u", LLL, NNN);

   if (LLL > 5)
     goto invalid_lll;

   if (NNN > LLL)
     goto invalid_nnn;

   if (LLL != 0) {
     /* we are interleaved */
     if (!depay->interleaved) {
       guint size;

       GST_DEBUG_OBJECT (depay, "starting interleaving group");
       /* bundling is not allowed to change in one interleave group */
       depay->bundling = count_packets (depay, payload, payload_len);
       GST_DEBUG_OBJECT (depay, "got bundling of %u", depay->bundling);
       /* we have one bundle where NNN goes from 0 to L, we don't store the index
        * 0 frames, so L+1 packets. Each packet has 'bundling - 1' packets */
       size = (depay->bundling - 1) * (LLL + 1);
       /* create the array to hold the packets */
       if (depay->packets == NULL)
         depay->packets = g_ptr_array_sized_new (size);
       GST_DEBUG_OBJECT (depay, "created packet array of size %u", size);
       g_ptr_array_set_size (depay->packets, size);
       /* we were previously not interleaved, figure out how much space we
        * need to deinterleave */
       depay->interleaved = TRUE;
     }
   } else {
     /* we are not interleaved */
     if (depay->interleaved) {
       GST_DEBUG_OBJECT (depay, "stopping interleaving");
       /* flush packets if we were previously interleaved */
       flush_packets (depay);
     }
     depay->bundling = 0;
   }

   index = 0;
   offset = 1;

   while (payload_len > 0) {
     gint frame_len;
     gboolean do_erasure;

     frame_len = get_frame_len (depay, payload[0]);
     GST_DEBUG_OBJECT (depay, "got frame len %d", frame_len);

     if (frame_len == 0)
       goto invalid_frame;

     if (frame_len < 0) {
       /* need to add an erasure frame but we can recover */
       frame_len = -frame_len;
       do_erasure = TRUE;
     } else {
       do_erasure = FALSE;
     }

     if (frame_len > payload_len)
       goto invalid_frame;

     if (do_erasure) {
       /* create erasure frame */
       outbuf = create_erasure_buffer (depay);
     } else {
       /* each frame goes into its buffer */
       outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, offset, frame_len);
     }

     GST_BUFFER_PTS (outbuf) = timestamp;
     GST_BUFFER_DURATION (outbuf) = FRAME_DURATION;

     gst_rtp_drop_non_audio_meta (depayload, outbuf);

     if (!depay->interleaved || index == 0) {
       /* not interleaved or first frame in packet, just push */
       gst_rtp_base_depayload_push (depayload, outbuf);

       if (timestamp != -1)
         timestamp += FRAME_DURATION;
     } else {
       /* put in interleave buffer */
       add_packet (depay, LLL, NNN, index, outbuf);

       if (timestamp != -1)
         timestamp += (FRAME_DURATION * (LLL + 1));
     }

     payload_len -= frame_len;
     payload += frame_len;
     offset += frame_len;
     index++;

     /* discard excess packets */
     if (depay->bundling > 0 && depay->bundling <= index)
       break;
   }
   while (index < depay->bundling) {
     GST_DEBUG_OBJECT (depay, "filling with erasure buffer");
     /* fill remainder with erasure packets */
     outbuf = create_erasure_buffer (depay);
     add_packet (depay, LLL, NNN, index, outbuf);
     index++;
   }
   if (depay->interleaved && LLL == NNN) {
     GST_DEBUG_OBJECT (depay, "interleave group ended, flushing");
     /* we have the complete interleave group, flush */
     flush_packets (depay);
   }

   return NULL;

   /* ERRORS */
 too_small:
   {
     GST_ELEMENT_WARNING (depay, STREAM, DECODE,
         (NULL), ("QCELP RTP payload too small (%d)", payload_len));
     return NULL;
   }
 invalid_lll:
   {
     GST_ELEMENT_WARNING (depay, STREAM, DECODE,
         (NULL), ("QCELP RTP invalid LLL received (%d)", LLL));
     return NULL;
   }
 invalid_nnn:
   {
     GST_ELEMENT_WARNING (depay, STREAM, DECODE,
         (NULL), ("QCELP RTP invalid NNN received (%d)", NNN));
     return NULL;
   }
 invalid_frame:
   {
     GST_ELEMENT_WARNING (depay, STREAM, DECODE,
         (NULL), ("QCELP RTP invalid frame received"));
     return NULL;
   }
 }

 gboolean
 gst_rtp_qcelp_depay_plugin_init (GstPlugin * plugin)
 {
   return gst_element_register (plugin, "rtpqcelpdepay",
       GST_RANK_SECONDARY, GST_TYPE_RTP_QCELP_DEPAY);
 }
	/* 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., 51 Franklin St, Fifth Floor,
	* Boston, MA 02110-1301, USA.
	*/

	#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 "gstrtpqcelpdepay.h"
	#include "gstrtputils.h"

	GST_DEBUG_CATEGORY_STATIC (rtpqcelpdepay_debug);
	#define GST_CAT_DEFAULT (rtpqcelpdepay_debug)

	/* references:
	*
	* RFC 2658 - RTP Payload Format for PureVoice(tm) Audio
	*/
	#define FRAME_DURATION (20 * GST_MSECOND)

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

	enum
	{
	PROP_0
	};

	static GstStaticPadTemplate gst_rtp_qcelp_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) \"QCELP\"; "
	"application/x-rtp, "
	"media = (string) \"audio\", "
	"payload = (int) " GST_RTP_PAYLOAD_QCELP_STRING ", "
	"clock-rate = (int) 8000")
	);

	static GstStaticPadTemplate gst_rtp_qcelp_depay_src_template =
	GST_STATIC_PAD_TEMPLATE ("src",
	GST_PAD_SRC,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("audio/qcelp, " "channels = (int) 1," "rate = (int) 8000")
	);

	static void gst_rtp_qcelp_depay_finalize (GObject * object);

	static gboolean gst_rtp_qcelp_depay_setcaps (GstRTPBaseDepayload * depayload,
	GstCaps * caps);
	static GstBuffer gst_rtp_qcelp_depay_process (GstRTPBaseDepayload depayload,
	GstRTPBuffer * rtp);

	#define gst_rtp_qcelp_depay_parent_class parent_class
	G_DEFINE_TYPE (GstRtpQCELPDepay, gst_rtp_qcelp_depay,
	GST_TYPE_RTP_BASE_DEPAYLOAD);

	static void
	gst_rtp_qcelp_depay_class_init (GstRtpQCELPDepayClass * klass)
	{
	GObjectClass *gobject_class;
	GstElementClass *gstelement_class;
	GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;

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

	gobject_class->finalize = gst_rtp_qcelp_depay_finalize;

	gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_qcelp_depay_process;
	gstrtpbasedepayload_class->set_caps = gst_rtp_qcelp_depay_setcaps;

	gst_element_class_add_static_pad_template (gstelement_class,
	&gst_rtp_qcelp_depay_src_template);
	gst_element_class_add_static_pad_template (gstelement_class,
	&gst_rtp_qcelp_depay_sink_template);

	gst_element_class_set_static_metadata (gstelement_class,
	"RTP QCELP depayloader", "Codec/Depayloader/Network/RTP",
	"Extracts QCELP (PureVoice) audio from RTP packets (RFC 2658)",
	"Wim Taymans <wim.taymans@gmail.com>");

	GST_DEBUG_CATEGORY_INIT (rtpqcelpdepay_debug, "rtpqcelpdepay", 0,
	"QCELP RTP Depayloader");
	}

	static void
	gst_rtp_qcelp_depay_init (GstRtpQCELPDepay * rtpqcelpdepay)
	{
	}

	static void
	gst_rtp_qcelp_depay_finalize (GObject * object)
	{
	GstRtpQCELPDepay *depay;

	depay = GST_RTP_QCELP_DEPAY (object);

	if (depay->packets != NULL) {
	g_ptr_array_foreach (depay->packets, (GFunc) gst_buffer_unref, NULL);
	g_ptr_array_free (depay->packets, TRUE);
	depay->packets = NULL;
	}

	G_OBJECT_CLASS (parent_class)->finalize (object);
	}


	static gboolean
	gst_rtp_qcelp_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
	{
	GstCaps *srccaps;
	gboolean res;

	srccaps = gst_caps_new_simple ("audio/qcelp",
	"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
	res = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
	gst_caps_unref (srccaps);

	return res;
	}

	static const gint frame_size[16] = {
	1, 4, 8, 17, 35, -8, 0, 0,
	0, 0, 0, 0, 0, 0, 1, 0
	};

	/* get the frame length, 0 is invalid, negative values are invalid but can be
	* recovered from. */
	static gint
	get_frame_len (GstRtpQCELPDepay * depay, guint8 frame_type)
	{
	if (frame_type >= G_N_ELEMENTS (frame_size))
	return 0;

	return frame_size[frame_type];
	}

	static guint
	count_packets (GstRtpQCELPDepay * depay, guint8 * data, guint size)
	{
	guint count = 0;

	while (size > 0) {
	gint frame_len;

	frame_len = get_frame_len (depay, data[0]);

	/* 0 is invalid and we throw away the remainder of the frames */
	if (frame_len == 0)
	break;

	if (frame_len < 0)
	frame_len = -frame_len;

	if (frame_len > size)
	break;

	size -= frame_len;
	data += frame_len;
	count++;
	}
	return count;
	}

	static void
	flush_packets (GstRtpQCELPDepay * depay)
	{
	guint i, size;

	GST_DEBUG_OBJECT (depay, "flushing packets");

	size = depay->packets->len;

	for (i = 0; i < size; i++) {
	GstBuffer *outbuf;

	outbuf = g_ptr_array_index (depay->packets, i);
	g_ptr_array_index (depay->packets, i) = NULL;

	gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (depay), outbuf);
	}

	/* and reset interleaving state */
	depay->interleaved = FALSE;
	depay->bundling = 0;
	}

	static void
	add_packet (GstRtpQCELPDepay * depay, guint LLL, guint NNN, guint index,
	GstBuffer * outbuf)
	{
	guint idx;
	GstBuffer *old;

	/* figure out the position in the array, note that index is never 0 because we
	* push those packets immediately. */
	idx = NNN + ((LLL + 1) * (index - 1));

	GST_DEBUG_OBJECT (depay, "adding packet at index %u", idx);
	/* free old buffer (should not happen) */
	old = g_ptr_array_index (depay->packets, idx);
	if (old)
	gst_buffer_unref (old);

	/* store new buffer */
	g_ptr_array_index (depay->packets, idx) = outbuf;
	}

	static GstBuffer *
	create_erasure_buffer (GstRtpQCELPDepay * depay)
	{
	GstBuffer *outbuf;
	GstMapInfo map;

	outbuf = gst_buffer_new_and_alloc (1);
	gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
	map.data[0] = 14;
	gst_buffer_unmap (outbuf, &map);

	return outbuf;
	}

	static GstBuffer *
	gst_rtp_qcelp_depay_process (GstRTPBaseDepayload * depayload,
	GstRTPBuffer * rtp)
	{
	GstRtpQCELPDepay *depay;
	GstBuffer *outbuf;
	GstClockTime timestamp;
	guint payload_len, offset, index;
	guint8 *payload;
	guint LLL, NNN;

	depay = GST_RTP_QCELP_DEPAY (depayload);

	payload_len = gst_rtp_buffer_get_payload_len (rtp);

	if (payload_len < 2)
	goto too_small;

	timestamp = GST_BUFFER_PTS (rtp->buffer);

	payload = gst_rtp_buffer_get_payload (rtp);

	/* 0 1 2 3 4 5 6 7
	* +-+-+-+-+-+-+-+-+
	* \|RR \| LLL \| NNN \|
	* +-+-+-+-+-+-+-+-+
	*/
	/* RR = payload[0] >> 6; */
	LLL = (payload[0] & 0x38) >> 3;
	NNN = (payload[0] & 0x07);

	payload_len--;
	payload++;

	GST_DEBUG_OBJECT (depay, "LLL %u, NNN %u", LLL, NNN);

	if (LLL > 5)
	goto invalid_lll;

	if (NNN > LLL)
	goto invalid_nnn;

	if (LLL != 0) {
	/* we are interleaved */
	if (!depay->interleaved) {
	guint size;

	GST_DEBUG_OBJECT (depay, "starting interleaving group");
	/* bundling is not allowed to change in one interleave group */
	depay->bundling = count_packets (depay, payload, payload_len);
	GST_DEBUG_OBJECT (depay, "got bundling of %u", depay->bundling);
	/* we have one bundle where NNN goes from 0 to L, we don't store the index
	* 0 frames, so L+1 packets. Each packet has 'bundling - 1' packets */
	size = (depay->bundling - 1) * (LLL + 1);
	/* create the array to hold the packets */
	if (depay->packets == NULL)
	depay->packets = g_ptr_array_sized_new (size);
	GST_DEBUG_OBJECT (depay, "created packet array of size %u", size);
	g_ptr_array_set_size (depay->packets, size);
	/* we were previously not interleaved, figure out how much space we
	* need to deinterleave */
	depay->interleaved = TRUE;
	}
	} else {
	/* we are not interleaved */
	if (depay->interleaved) {
	GST_DEBUG_OBJECT (depay, "stopping interleaving");
	/* flush packets if we were previously interleaved */
	flush_packets (depay);
	}
	depay->bundling = 0;
	}

	index = 0;
	offset = 1;

	while (payload_len > 0) {
	gint frame_len;
	gboolean do_erasure;

	frame_len = get_frame_len (depay, payload[0]);
	GST_DEBUG_OBJECT (depay, "got frame len %d", frame_len);

	if (frame_len == 0)
	goto invalid_frame;

	if (frame_len < 0) {
	/* need to add an erasure frame but we can recover */
	frame_len = -frame_len;
	do_erasure = TRUE;
	} else {
	do_erasure = FALSE;
	}

	if (frame_len > payload_len)
	goto invalid_frame;

	if (do_erasure) {
	/* create erasure frame */
	outbuf = create_erasure_buffer (depay);
	} else {
	/* each frame goes into its buffer */
	outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, offset, frame_len);
	}

	GST_BUFFER_PTS (outbuf) = timestamp;
	GST_BUFFER_DURATION (outbuf) = FRAME_DURATION;

	gst_rtp_drop_non_audio_meta (depayload, outbuf);

	if (!depay->interleaved \|\| index == 0) {
	/* not interleaved or first frame in packet, just push */
	gst_rtp_base_depayload_push (depayload, outbuf);

	if (timestamp != -1)
	timestamp += FRAME_DURATION;
	} else {
	/* put in interleave buffer */
	add_packet (depay, LLL, NNN, index, outbuf);

	if (timestamp != -1)
	timestamp += (FRAME_DURATION * (LLL + 1));
	}

	payload_len -= frame_len;
	payload += frame_len;
	offset += frame_len;
	index++;

	/* discard excess packets */
	if (depay->bundling > 0 && depay->bundling <= index)
	break;
	}
	while (index < depay->bundling) {
	GST_DEBUG_OBJECT (depay, "filling with erasure buffer");
	/* fill remainder with erasure packets */
	outbuf = create_erasure_buffer (depay);
	add_packet (depay, LLL, NNN, index, outbuf);
	index++;
	}
	if (depay->interleaved && LLL == NNN) {
	GST_DEBUG_OBJECT (depay, "interleave group ended, flushing");
	/* we have the complete interleave group, flush */
	flush_packets (depay);
	}

	return NULL;

	/* ERRORS */
	too_small:
	{
	GST_ELEMENT_WARNING (depay, STREAM, DECODE,
	(NULL), ("QCELP RTP payload too small (%d)", payload_len));
	return NULL;
	}
	invalid_lll:
	{
	GST_ELEMENT_WARNING (depay, STREAM, DECODE,
	(NULL), ("QCELP RTP invalid LLL received (%d)", LLL));
	return NULL;
	}
	invalid_nnn:
	{
	GST_ELEMENT_WARNING (depay, STREAM, DECODE,
	(NULL), ("QCELP RTP invalid NNN received (%d)", NNN));
	return NULL;
	}
	invalid_frame:
	{
	GST_ELEMENT_WARNING (depay, STREAM, DECODE,
	(NULL), ("QCELP RTP invalid frame received"));
	return NULL;
	}
	}

	gboolean
	gst_rtp_qcelp_depay_plugin_init (GstPlugin * plugin)
	{
	return gst_element_register (plugin, "rtpqcelpdepay",
	GST_RANK_SECONDARY, GST_TYPE_RTP_QCELP_DEPAY);
	}