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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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f636fb8b34
Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_calculate_##TYPE), (gst_level_transform_ip): Use guint8 * instead of gpointer then vs6 know the size of data pointed when moving the pointer. * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer): Move instructions after variables declaration. * win32/vs6/autogen.dsp: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: Update vs6 project files.
337 lines
9.2 KiB
C
337 lines
9.2 KiB
C
/* GStreamer
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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 "gstrtph264pay.h"
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GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug);
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#define GST_CAT_DEFAULT (rtph264pay_debug)
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/* references:
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*
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* RFC 3984
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*/
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/* elementfactory information */
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static const GstElementDetails gst_rtp_h264pay_details =
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GST_ELEMENT_DETAILS ("RTP packet payloader",
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"Codec/Payloader/Network",
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"Payload-encode H264 video into RTP packets (RFC 3984)",
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"Laurent Glayal <spglegle@yahoo.fr>");
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static GstStaticPadTemplate gst_rtp_h264_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 ("video/x-h264")
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);
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static GstStaticPadTemplate gst_rtp_h264_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) \"video\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 90000, " "encoding-name = (string) \"H264\"")
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);
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static void gst_rtp_h264_pay_finalize (GObject * object);
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static void gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * pad,
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GstBuffer * buffer);
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GST_BOILERPLATE (GstRtpH264Pay, gst_rtp_h264_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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static void
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gst_rtp_h264_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_h264_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_h264_pay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_h264pay_details);
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}
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static void
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gst_rtp_h264_pay_class_init (GstRtpH264PayClass * 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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gobject_class->finalize = gst_rtp_h264_pay_finalize;
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gobject_class->set_property = gst_rtp_h264_pay_set_property;
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gobject_class->get_property = gst_rtp_h264_pay_get_property;
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gstelement_class->change_state = gst_rtp_h264_pay_change_state;
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gstbasertppayload_class->set_caps = gst_rtp_h264_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtph264pay_debug, "rtph264pay", 0,
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"H264 RTP Payloader");
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}
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static void
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gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay, GstRtpH264PayClass * klass)
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{
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}
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static void
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gst_rtp_h264_pay_finalize (GObject * object)
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{
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GstRtpH264Pay *rtph264pay;
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rtph264pay = GST_RTP_H264_PAY (object);
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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_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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{
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GstRtpH264Pay *rtph264pay;
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rtph264pay = GST_RTP_H264_PAY (basepayload);
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gst_basertppayload_set_options (basepayload, "video", TRUE, "H264", 90000);
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gst_basertppayload_set_outcaps (basepayload, NULL);
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return TRUE;
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}
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static GstFlowReturn
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gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpH264Pay *rtph264pay;
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GstFlowReturn ret;
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guint size, idxdata;
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GstBuffer *outbuf;
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guint8 *payload, *data, *pdata;
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guint8 nalType;
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GstClockTime timestamp;
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guint packet_len, payload_len, mtu;
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rtph264pay = GST_RTP_H264_PAY (basepayload);
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mtu = GST_BASE_RTP_PAYLOAD_MTU (rtph264pay);
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size = GST_BUFFER_SIZE (buffer);
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data = GST_BUFFER_DATA (buffer);
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timestamp = GST_BUFFER_TIMESTAMP (buffer);
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GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
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/* H264 stream analysis */
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pdata = data;
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idxdata = size;
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while (idxdata > 5 &&
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(pdata[0] != 0x00 || pdata[1] != 0x00 || pdata[2] != 0x1 ||
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(pdata[3] & 0x1f) < 1 || (pdata[3] & 0x1f) > 23)
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) {
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pdata++;
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idxdata--;
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GST_DEBUG_OBJECT (basepayload, "idxdata=%d", idxdata);
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}
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if (idxdata < 5) {
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GST_DEBUG_OBJECT (basepayload,
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"Returning GST_FLOW_OK without creating RTP packet");
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return GST_FLOW_OK;
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}
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pdata += 3;
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idxdata -= 3;
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nalType = pdata[0] & 0x1f;
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GST_DEBUG_OBJECT (basepayload, "Processing Buffer with NAL TYPE=%d", nalType);
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packet_len = gst_rtp_buffer_calc_packet_len (idxdata, 0, 0);
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if (packet_len < mtu) {
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GST_DEBUG_OBJECT (basepayload,
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"NAL Unit fit in one packet datasize=%d mtu=%d", idxdata, mtu);
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/* will fit in one packet */
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outbuf = gst_rtp_buffer_new_allocate (idxdata, 0, 0);
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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gst_rtp_buffer_set_marker (outbuf, 1);
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payload = gst_rtp_buffer_get_payload (outbuf);
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GST_DEBUG_OBJECT (basepayload, "Copying %d bytes to outbuf", idxdata);
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memcpy (payload, pdata, idxdata);
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gst_buffer_unref (buffer);
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ret = gst_basertppayload_push (basepayload, outbuf);
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return ret;
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} else {
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/* Fragmentation Units FU-A */
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guint8 nalHeader;
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guint limitedSize;
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int ii = 0, start = 1, end = 0, first = 0;
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GST_DEBUG_OBJECT (basepayload,
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"NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", idxdata, mtu);
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nalHeader = *pdata;
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pdata++;
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idxdata--;
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ret = GST_FLOW_OK;
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GST_DEBUG_OBJECT (basepayload, "Using FU-A fragmentation for data size=%d",
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idxdata);
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payload_len = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0); /* We keep 2 bytes for FU indicator and FU Header */
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while (end == 0) {
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limitedSize = idxdata < payload_len ? idxdata : payload_len;
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GST_DEBUG_OBJECT (basepayload,
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"Inside FU-A fragmentation limitedSize=%d iteration=%d", limitedSize,
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ii);
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outbuf = gst_rtp_buffer_new_allocate (limitedSize + 2, 0, 0);
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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gst_rtp_buffer_set_marker (outbuf, end);
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payload = gst_rtp_buffer_get_payload (outbuf);
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if (limitedSize == idxdata) {
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GST_DEBUG_OBJECT (basepayload, "end idxdata=%d iteration=%d", idxdata,
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ii);
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end = 1;
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}
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/* FU indicator */
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payload[0] = (nalHeader & 0x60) | 28;
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/* FU Header */
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payload[1] = (start << 7) | (end << 6) | (nalHeader & 0x1f);
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memcpy (&payload[2], pdata + first, limitedSize);
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GST_DEBUG_OBJECT (basepayload,
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"recorded %d payload bytes into packet iteration=%d", limitedSize + 2,
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ii);
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ret = gst_basertppayload_push (basepayload, outbuf);
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if (ret != GST_FLOW_OK)
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break;
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idxdata -= limitedSize;
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first += limitedSize;
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ii++;
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start = 0;
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}
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gst_buffer_unref (buffer);
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return ret;
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}
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GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
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(NULL), ("Should not be there !!"));
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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static void
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gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRtpH264Pay *rtph264pay;
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rtph264pay = GST_RTP_H264_PAY (object);
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRtpH264Pay *rtph264pay;
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rtph264pay = GST_RTP_H264_PAY (object);
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstStateChangeReturn
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gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition)
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{
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GstRtpH264Pay *rtph264pay;
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GstStateChangeReturn ret;
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rtph264pay = GST_RTP_H264_PAY (element);
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switch (transition) {
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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default:
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break;
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}
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return ret;
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}
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gboolean
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gst_rtp_h264_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtph264pay",
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GST_RANK_NONE, GST_TYPE_RTP_H264_PAY);
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}
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