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2af907ecab
Original commit message from CVS: fix element descriptions
234 lines
7 KiB
C
234 lines
7 KiB
C
/* 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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*
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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
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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 <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpg711pay.h"
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/* elementfactory information */
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static GstElementDetails gst_rtp_g711_pay_details = {
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"RTP packet parser",
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"Codec/Payloader/Network",
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"Payload-encodes PCMU/PCMA audio into a RTP packet",
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"Edgard Lima <edgard.lima@indt.org.br>"
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};
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static GstStaticPadTemplate gst_rtp_g711_pay_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 ("audio/x-mulaw, channels=(int)1, rate=(int)8000 ;"
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"audio/x-alaw, channels=(int)1, rate=(int)8000")
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);
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static GstStaticPadTemplate gst_rtp_g711_pay_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 ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) \"PCMU\"; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_PCMA_STRING ", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
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);
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static gboolean gst_rtp_g711_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_g711_pay_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buffer);
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static void gst_rtp_g711_pay_finalize (GObject * object);
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GST_BOILERPLATE (GstRtpG711Pay, gst_rtp_g711_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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static void
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gst_rtp_g711_pay_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_g711_pay_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_g711_pay_src_template));
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gst_element_class_set_details (element_class, &gst_rtp_g711_pay_details);
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}
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static void
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gst_rtp_g711_pay_class_init (GstRtpG711PayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
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gobject_class->finalize = gst_rtp_g711_pay_finalize;
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gstbasertppayload_class->set_caps = gst_rtp_g711_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_g711_pay_handle_buffer;
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}
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static void
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gst_rtp_g711_pay_init (GstRtpG711Pay * rtpg711pay, GstRtpG711PayClass * klass)
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{
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rtpg711pay->adapter = gst_adapter_new ();
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GST_BASE_RTP_PAYLOAD (rtpg711pay)->clock_rate = 8000;
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}
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static void
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gst_rtp_g711_pay_finalize (GObject * object)
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{
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GstRtpG711Pay *rtpg711pay;
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rtpg711pay = GST_RTP_G711_PAY (object);
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g_object_unref (rtpg711pay->adapter);
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rtpg711pay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_rtp_g711_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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const char *stname;
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GstStructure *structure;
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structure = gst_caps_get_structure (caps, 0);
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stname = gst_structure_get_name (structure);
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if (0 == strcmp ("audio/x-mulaw", stname)) {
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payload->pt = GST_RTP_PAYLOAD_PCMU;
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gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
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} else if (0 == strcmp ("audio/x-alaw", stname)) {
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payload->pt = GST_RTP_PAYLOAD_PCMA;
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gst_basertppayload_set_options (payload, "audio", FALSE, "PCMA", 8000);
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} else {
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return FALSE;
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}
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gst_basertppayload_set_outcaps (payload, NULL);
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return TRUE;
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}
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static GstFlowReturn
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gst_rtp_g711_pay_flush (GstRtpG711Pay * rtpg711pay)
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{
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guint avail;
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GstBuffer *outbuf;
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GstFlowReturn ret;
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/* the data available in the adapter is either smaller
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* than the MTU or bigger. In the case it is smaller, the complete
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* adapter contents can be put in one packet. */
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avail = gst_adapter_available (rtpg711pay->adapter);
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ret = GST_FLOW_OK;
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while (avail > 0) {
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guint towrite;
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guint8 *payload;
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guint8 *data;
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guint payload_len;
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guint packet_len;
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/* this will be the total lenght of the packet */
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packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
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/* fill one MTU or all available bytes */
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towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpg711pay));
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/* this is the payload length */
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payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
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/* create buffer to hold the payload */
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outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
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/* copy payload */
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gst_rtp_buffer_set_payload_type (outbuf,
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GST_BASE_RTP_PAYLOAD_PT (rtpg711pay));
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payload = gst_rtp_buffer_get_payload (outbuf);
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data = (guint8 *) gst_adapter_peek (rtpg711pay->adapter, payload_len);
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memcpy (payload, data, payload_len);
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gst_adapter_flush (rtpg711pay->adapter, payload_len);
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avail -= payload_len;
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GST_BUFFER_TIMESTAMP (outbuf) = rtpg711pay->first_ts;
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpg711pay), outbuf);
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}
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return ret;
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}
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static GstFlowReturn
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gst_rtp_g711_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpG711Pay *rtpg711pay;
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guint size, packet_len, avail;
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GstFlowReturn ret;
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GstClockTime duration;
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rtpg711pay = GST_RTP_G711_PAY (basepayload);
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size = GST_BUFFER_SIZE (buffer);
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duration = GST_BUFFER_TIMESTAMP (buffer);
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avail = gst_adapter_available (rtpg711pay->adapter);
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if (avail == 0) {
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rtpg711pay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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rtpg711pay->duration = 0;
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}
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/* get packet length of data and see if we exceeded MTU. */
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packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
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/* if this buffer is going to overflow the packet, flush what we
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* have. */
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if (gst_basertppayload_is_filled (basepayload,
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packet_len, rtpg711pay->duration + duration)) {
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ret = gst_rtp_g711_pay_flush (rtpg711pay);
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rtpg711pay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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rtpg711pay->duration = 0;
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} else {
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ret = GST_FLOW_OK;
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}
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gst_adapter_push (rtpg711pay->adapter, buffer);
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rtpg711pay->duration += duration;
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return ret;
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}
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gboolean
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gst_rtp_g711_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpg711pay",
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GST_RANK_NONE, GST_TYPE_RTP_G711_PAY);
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}
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