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208 lines
6.4 KiB
C
208 lines
6.4 KiB
C
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/* GStreamer
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* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) 2005 Edgard Lima <edgard.lima@indt.org.br>
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* Copyright (C) 2005 Zeeshan Ali <zeenix@gmail.com>
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* Copyright (C) 2008 Axis Communications <dev-gstreamer@axis.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpg726depay.h"
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#define DEFAULT_BIT_RATE 32000
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#define SAMPLE_RATE 8000
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#define LAYOUT_G726 "g726"
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/* elementfactory information */
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static const GstElementDetails gst_rtp_g726depay_details =
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GST_ELEMENT_DETAILS ("RTP packet depayloader",
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"Codec/Depayloader/Network",
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"Extracts G.726 audio from RTP packets",
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"Axis Communications <dev-gstreamer@axis.com>");
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/* RtpG726Depay signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0
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};
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static GstStaticPadTemplate gst_rtp_g726_depay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"encoding-name = (string) { \"G726\", \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\"}, "
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"clock-rate = (int) 8000;")
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);
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static GstStaticPadTemplate gst_rtp_g726_depay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-adpcm, "
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"channels = (int) 1, "
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"rate = (int) 8000, "
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"bitrate = (int) { 16000, 24000, 32000, 40000 }, "
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"layout = (string) \"g726\"")
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);
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static GstBuffer *gst_rtp_g726_depay_process (GstBaseRTPDepayload * depayload,
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GstBuffer * buf);
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static gboolean gst_rtp_g726_depay_setcaps (GstBaseRTPDepayload * depayload,
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GstCaps * caps);
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GST_BOILERPLATE (GstRtpG726Depay, gst_rtp_g726_depay, GstBaseRTPDepayload,
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GST_TYPE_BASE_RTP_DEPAYLOAD);
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static void
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gst_rtp_g726_depay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_g726_depay_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_g726_depay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_g726depay_details);
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}
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static void
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gst_rtp_g726_depay_class_init (GstRtpG726DepayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gstbasertpdepayload_class->process = gst_rtp_g726_depay_process;
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gstbasertpdepayload_class->set_caps = gst_rtp_g726_depay_setcaps;
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}
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static void
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gst_rtp_g726_depay_init (GstRtpG726Depay * rtpG726depay,
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GstRtpG726DepayClass * klass)
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{
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GstBaseRTPDepayload *depayload;
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depayload = GST_BASE_RTP_DEPAYLOAD (rtpG726depay);
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gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
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}
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static gboolean
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gst_rtp_g726_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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{
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GstCaps *srccaps;
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GstStructure *structure;
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gboolean ret;
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gint clock_rate = 8000; /* default */
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const gchar *encoding_name;
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gint bitrate;
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "clock-rate", &clock_rate);
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depayload->clock_rate = clock_rate;
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encoding_name = gst_structure_get_string (structure, "encoding-name");
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if (encoding_name == NULL || g_ascii_strcasecmp (encoding_name, "G726") == 0) {
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bitrate = DEFAULT_BIT_RATE;
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} else if (g_ascii_strcasecmp (encoding_name, "G726-16") == 0) {
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bitrate = 16000;
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} else if (g_ascii_strcasecmp (encoding_name, "G726-24") == 0) {
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bitrate = 24000;
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} else if (g_ascii_strcasecmp (encoding_name, "G726-32") == 0) {
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bitrate = 32000;
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} else if (g_ascii_strcasecmp (encoding_name, "G726-40") == 0) {
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bitrate = 40000;
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} else {
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GST_WARNING ("Could not determine bitrate from encoding-name (%s)",
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encoding_name);
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ret = FALSE;
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goto done;
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}
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GST_DEBUG ("RTP G.726 depayloader, bitrate set to %d\n", bitrate);
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srccaps = gst_caps_new_simple ("audio/x-adpcm",
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"channels", G_TYPE_INT, 1,
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"rate", G_TYPE_INT, clock_rate,
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"bitrate", G_TYPE_INT, bitrate,
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"layout", G_TYPE_STRING, LAYOUT_G726, NULL);
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ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
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done:
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return ret;
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}
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static GstBuffer *
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gst_rtp_g726_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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{
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GstCaps *srccaps;
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GstBuffer *outbuf = NULL;
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GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
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GST_BUFFER_SIZE (buf),
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gst_rtp_buffer_get_marker (buf),
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gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
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srccaps = GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
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if (!srccaps) {
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/* Set the default caps */
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srccaps = gst_caps_new_simple ("audio/x-adpcm",
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"channels", G_TYPE_INT, 1,
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"rate", G_TYPE_INT, SAMPLE_RATE,
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"bitrate", G_TYPE_INT, DEFAULT_BIT_RATE,
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"layout", G_TYPE_STRING, LAYOUT_G726, NULL);
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gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
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}
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outbuf = gst_rtp_buffer_get_payload_buffer (buf);
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return outbuf;
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}
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gboolean
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gst_rtp_g726_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpg726depay",
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GST_RANK_MARGINAL, GST_TYPE_RTP_G726_DEPAY);
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}
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