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274 lines
8 KiB
C
274 lines
8 KiB
C
/* GStreamer
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* Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.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 <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpmpapay.h"
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/* elementfactory information */
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static const GstElementDetails gst_rtp_mpapay_details =
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GST_ELEMENT_DETAILS ("RTP MPEG audio payloader",
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"Codec/Payloader/Network",
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"Payload MPEG audio as RTP packets (RFC 2038)",
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"Wim Taymans <wim.taymans@gmail.com>");
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static GstStaticPadTemplate gst_rtp_mpa_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/mpeg, " "mpegversion = (int) 1")
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);
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static GstStaticPadTemplate gst_rtp_mpa_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_MPA_STRING ", "
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"clock-rate = (int) 90000; "
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"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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"clock-rate = (int) 90000, " "encoding-name = (string) \"MPA\"")
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);
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static void gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass);
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static void gst_rtp_mpa_pay_base_init (GstRtpMPAPayClass * klass);
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static void gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay);
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static void gst_rtp_mpa_pay_finalize (GObject * object);
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static gboolean gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * payload,
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GstBuffer * buffer);
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static GstBaseRTPPayloadClass *parent_class = NULL;
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static GType
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gst_rtp_mpa_pay_get_type (void)
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{
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static GType rtpmpapay_type = 0;
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if (!rtpmpapay_type) {
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static const GTypeInfo rtpmpapay_info = {
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sizeof (GstRtpMPAPayClass),
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(GBaseInitFunc) gst_rtp_mpa_pay_base_init,
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NULL,
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(GClassInitFunc) gst_rtp_mpa_pay_class_init,
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NULL,
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NULL,
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sizeof (GstRtpMPAPay),
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0,
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(GInstanceInitFunc) gst_rtp_mpa_pay_init,
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};
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rtpmpapay_type =
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g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMPAPay",
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&rtpmpapay_info, 0);
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}
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return rtpmpapay_type;
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}
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static void
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gst_rtp_mpa_pay_base_init (GstRtpMPAPayClass * 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_mpa_pay_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_mpa_pay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_mpapay_details);
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}
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static void
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gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_rtp_mpa_pay_finalize;
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gstbasertppayload_class->set_caps = gst_rtp_mpa_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_mpa_pay_handle_buffer;
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}
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static void
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gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay)
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{
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rtpmpapay->adapter = gst_adapter_new ();
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}
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static void
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gst_rtp_mpa_pay_finalize (GObject * object)
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{
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GstRtpMPAPay *rtpmpapay;
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rtpmpapay = GST_RTP_MPA_PAY (object);
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g_object_unref (rtpmpapay->adapter);
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rtpmpapay->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_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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gboolean res;
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gst_basertppayload_set_options (payload, "audio", TRUE, "MPA", 90000);
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res = gst_basertppayload_set_outcaps (payload, NULL);
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return res;
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}
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static GstFlowReturn
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gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
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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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guint16 frag_offset;
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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. In the case the
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* adapter has more than one MTU, we need to split the MPA data
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* over multiple packets. The frag_offset in each packet header
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* needs to be updated with the position in the MPA frame. */
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avail = gst_adapter_available (rtpmpapay->adapter);
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ret = GST_FLOW_OK;
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frag_offset = 0;
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while (avail > 0) {
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guint towrite;
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guint8 *payload;
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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 (4 + 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 (rtpmpapay));
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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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payload_len -= 4;
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gst_rtp_buffer_set_payload_type (outbuf, GST_RTP_PAYLOAD_MPA);
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/*
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* 0 1 2 3
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | MBZ | Frag_offset |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*/
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payload = gst_rtp_buffer_get_payload (outbuf);
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payload[0] = 0;
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payload[1] = 0;
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payload[2] = frag_offset >> 8;
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payload[3] = frag_offset & 0xff;
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gst_adapter_copy (rtpmpapay->adapter, &payload[4], 0, payload_len);
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gst_adapter_flush (rtpmpapay->adapter, payload_len);
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avail -= payload_len;
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frag_offset += payload_len;
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if (avail == 0)
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gst_rtp_buffer_set_marker (outbuf, TRUE);
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GST_BUFFER_TIMESTAMP (outbuf) = rtpmpapay->first_ts;
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GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpapay), 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_mpa_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpMPAPay *rtpmpapay;
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GstFlowReturn ret;
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guint size, avail;
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guint packet_len;
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GstClockTime duration;
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rtpmpapay = GST_RTP_MPA_PAY (basepayload);
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size = GST_BUFFER_SIZE (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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avail = gst_adapter_available (rtpmpapay->adapter);
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if (avail == 0) {
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rtpmpapay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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rtpmpapay->duration = 0;
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}
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/* get packet length of previous data and this new data,
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* payload length includes a 4 byte header */
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packet_len = gst_rtp_buffer_calc_packet_len (4 + 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, rtpmpapay->duration + duration)) {
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ret = gst_rtp_mpa_pay_flush (rtpmpapay);
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rtpmpapay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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rtpmpapay->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 (rtpmpapay->adapter, buffer);
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rtpmpapay->duration += duration;
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return ret;
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}
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gboolean
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gst_rtp_mpa_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpmpapay",
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GST_RANK_NONE, GST_TYPE_RTP_MPA_PAY);
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}
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