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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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712 lines
21 KiB
C
712 lines
21 KiB
C
/* GStreamer
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* Copyright (C) <2010> 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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 <gst/video/video.h>
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#include "gstrtpelements.h"
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#include "gstrtpgstpay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_pay_debug);
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#define GST_CAT_DEFAULT gst_rtp_pay_debug
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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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* |C| CV |D|0|0|0| ETYPE | MBZ |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Frag_offset |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*
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* C: caps inlined flag
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* When C set, first part of payload contains caps definition. Caps definition
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* starts with variable-length length prefix and then a string of that length.
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* the length is encoded in big endian 7 bit chunks, the top 1 bit of a byte
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* is the continuation marker and the 7 next bits the data. A continuation
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* marker of 1 means that the next byte contains more data.
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*
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* CV: caps version, 0 = caps from SDP, 1 - 7 inlined caps
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* D: delta unit buffer
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* ETYPE: type of event. Payload contains the event, prefixed with a
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* variable length field.
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* 0 = NO event
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* 1 = GST_EVENT_TAG
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* 2 = GST_EVENT_CUSTOM_DOWNSTREAM
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* 3 = GST_EVENT_CUSTOM_BOTH
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* 4 = GST_EVENT_STREAM_START
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*/
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static GstStaticPadTemplate gst_rtp_gst_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_ANY);
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static GstStaticPadTemplate gst_rtp_gst_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) \"application\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 90000, " "encoding-name = (string) \"X-GST\"")
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);
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enum
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{
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PROP_0,
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PROP_CONFIG_INTERVAL
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};
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#define DEFAULT_CONFIG_INTERVAL 0
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static void gst_rtp_gst_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_gst_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_rtp_gst_pay_finalize (GObject * obj);
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static GstStateChangeReturn gst_rtp_gst_pay_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_rtp_gst_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_gst_pay_handle_buffer (GstRTPBasePayload * payload,
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GstBuffer * buffer);
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static gboolean gst_rtp_gst_pay_sink_event (GstRTPBasePayload * payload,
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GstEvent * event);
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static gboolean gst_rtp_gst_pay_src_event (GstRTPBasePayload * payload,
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GstEvent * event);
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#define gst_rtp_gst_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpGSTPay, gst_rtp_gst_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpgstpay, "rtpgstpay", GST_RANK_NONE,
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GST_TYPE_RTP_GST_PAY, rtp_element_init (plugin));
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static void
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gst_rtp_gst_pay_class_init (GstRtpGSTPayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gobject_class->set_property = gst_rtp_gst_pay_set_property;
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gobject_class->get_property = gst_rtp_gst_pay_get_property;
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gobject_class->finalize = gst_rtp_gst_pay_finalize;
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_CONFIG_INTERVAL,
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g_param_spec_uint ("config-interval",
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"Caps/Tags Send Interval",
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"Interval for sending caps and TAG events in seconds (0 = disabled)",
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0, 3600, DEFAULT_CONFIG_INTERVAL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
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);
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gstelement_class->change_state = gst_rtp_gst_pay_change_state;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_gst_pay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_gst_pay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP GStreamer payloader", "Codec/Payloader/Network/RTP",
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"Payload GStreamer buffers as RTP packets",
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"Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasepayload_class->set_caps = gst_rtp_gst_pay_setcaps;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_gst_pay_handle_buffer;
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gstrtpbasepayload_class->sink_event = gst_rtp_gst_pay_sink_event;
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gstrtpbasepayload_class->src_event = gst_rtp_gst_pay_src_event;
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GST_DEBUG_CATEGORY_INIT (gst_rtp_pay_debug, "rtpgstpay", 0,
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"rtpgstpay element");
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}
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static void
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gst_rtp_gst_pay_init (GstRtpGSTPay * rtpgstpay)
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{
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rtpgstpay->adapter = gst_adapter_new ();
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rtpgstpay->pending_buffers = NULL;
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gst_rtp_base_payload_set_options (GST_RTP_BASE_PAYLOAD (rtpgstpay),
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"application", TRUE, "X-GST", 90000);
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rtpgstpay->last_config = GST_CLOCK_TIME_NONE;
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rtpgstpay->taglist = NULL;
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rtpgstpay->config_interval = DEFAULT_CONFIG_INTERVAL;
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}
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static void
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gst_rtp_gst_pay_reset (GstRtpGSTPay * rtpgstpay, gboolean full)
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{
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rtpgstpay->last_config = GST_CLOCK_TIME_NONE;
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gst_adapter_clear (rtpgstpay->adapter);
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rtpgstpay->flags &= 0x70;
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rtpgstpay->etype = 0;
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if (rtpgstpay->pending_buffers)
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gst_buffer_list_unref (rtpgstpay->pending_buffers);
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rtpgstpay->pending_buffers = NULL;
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if (full) {
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if (rtpgstpay->taglist)
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gst_tag_list_unref (rtpgstpay->taglist);
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rtpgstpay->taglist = NULL;
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g_free (rtpgstpay->stream_id);
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rtpgstpay->stream_id = NULL;
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rtpgstpay->current_CV = 0;
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rtpgstpay->next_CV = 0;
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}
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}
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static void
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gst_rtp_gst_pay_finalize (GObject * obj)
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{
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GstRtpGSTPay *rtpgstpay;
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rtpgstpay = GST_RTP_GST_PAY (obj);
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gst_rtp_gst_pay_reset (rtpgstpay, TRUE);
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g_object_unref (rtpgstpay->adapter);
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G_OBJECT_CLASS (parent_class)->finalize (obj);
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}
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static void
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gst_rtp_gst_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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GstRtpGSTPay *rtpgstpay;
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rtpgstpay = GST_RTP_GST_PAY (object);
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switch (prop_id) {
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case PROP_CONFIG_INTERVAL:
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rtpgstpay->config_interval = g_value_get_uint (value);
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break;
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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_gst_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRtpGSTPay *rtpgstpay;
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rtpgstpay = GST_RTP_GST_PAY (object);
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switch (prop_id) {
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case PROP_CONFIG_INTERVAL:
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g_value_set_uint (value, rtpgstpay->config_interval);
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break;
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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_gst_pay_change_state (GstElement * element, GstStateChange transition)
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{
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GstRtpGSTPay *rtpgstpay;
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GstStateChangeReturn ret;
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rtpgstpay = GST_RTP_GST_PAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_rtp_gst_pay_reset (rtpgstpay, TRUE);
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break;
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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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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_rtp_gst_pay_reset (rtpgstpay, TRUE);
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break;
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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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#define RTP_HEADER_LEN 12
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static gboolean
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gst_rtp_gst_pay_create_from_adapter (GstRtpGSTPay * rtpgstpay,
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GstClockTime timestamp)
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{
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guint avail, mtu;
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guint frag_offset;
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avail = gst_adapter_available (rtpgstpay->adapter);
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if (avail == 0)
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return FALSE;
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mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpgstpay);
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if (!rtpgstpay->pending_buffers)
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rtpgstpay->pending_buffers =
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gst_buffer_list_new_sized ((avail / (mtu - (RTP_HEADER_LEN + 8))) + 1);
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frag_offset = 0;
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while (avail) {
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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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GstBuffer *outbuf;
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GstRTPBuffer rtp = { NULL };
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GstBuffer *paybuf;
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/* this will be the total length of the packet */
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packet_len = gst_rtp_buffer_calc_packet_len (8 + avail, 0, 0);
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/* fill one MTU or all available bytes */
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towrite = MIN (packet_len, mtu);
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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 header */
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outbuf =
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gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
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(rtpgstpay), 8, 0, 0);
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gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
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payload = gst_rtp_buffer_get_payload (&rtp);
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GST_DEBUG_OBJECT (rtpgstpay, "new packet len %u, frag %u", packet_len,
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frag_offset);
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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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* |C| CV |D|0|0|0| ETYPE | MBZ |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Frag_offset |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*/
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payload[0] = rtpgstpay->flags;
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payload[1] = rtpgstpay->etype;
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payload[2] = payload[3] = 0;
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payload[4] = frag_offset >> 24;
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payload[5] = frag_offset >> 16;
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payload[6] = frag_offset >> 8;
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payload[7] = frag_offset & 0xff;
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payload += 8;
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payload_len -= 8;
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frag_offset += payload_len;
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avail -= payload_len;
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if (avail == 0) {
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gst_rtp_buffer_set_marker (&rtp, TRUE);
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MARKER);
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}
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gst_rtp_buffer_unmap (&rtp);
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/* create a new buf to hold the payload */
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GST_DEBUG_OBJECT (rtpgstpay, "take %u bytes from adapter", payload_len);
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paybuf = gst_adapter_take_buffer_fast (rtpgstpay->adapter, payload_len);
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if (GST_BUFFER_FLAG_IS_SET (paybuf, GST_BUFFER_FLAG_DELTA_UNIT))
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DELTA_UNIT);
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/* create a new group to hold the rtp header and the payload */
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gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpgstpay), outbuf, paybuf, 0);
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outbuf = gst_buffer_append (outbuf, paybuf);
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GST_BUFFER_PTS (outbuf) = timestamp;
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/* and add to list */
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gst_buffer_list_insert (rtpgstpay->pending_buffers, -1, outbuf);
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}
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rtpgstpay->flags &= 0x70;
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rtpgstpay->etype = 0;
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return TRUE;
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}
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static gboolean
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retimestamp_buffer (GstBuffer ** buffer, guint idx, gpointer user_data)
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{
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GstClockTime *timestamp = user_data;
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GST_BUFFER_PTS (*buffer) = *timestamp;
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return TRUE;
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}
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static GstFlowReturn
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gst_rtp_gst_pay_flush (GstRtpGSTPay * rtpgstpay, GstClockTime timestamp)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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gst_rtp_gst_pay_create_from_adapter (rtpgstpay, timestamp);
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if (rtpgstpay->pending_buffers) {
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// make sure all buffers in the buffer list have the correct timestamp.
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// If we created packets based on an event they would have
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// GST_CLOCK_TIME_NONE as PTS.
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gst_buffer_list_foreach (rtpgstpay->pending_buffers, retimestamp_buffer,
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×tamp);
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/* push the whole buffer list at once */
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ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpgstpay),
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rtpgstpay->pending_buffers);
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rtpgstpay->pending_buffers = NULL;
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}
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return ret;
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}
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static GstBuffer *
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make_data_buffer (GstRtpGSTPay * rtpgstpay, gchar * data, guint size)
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{
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guint plen;
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guint8 *ptr;
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GstBuffer *outbuf;
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GstMapInfo map;
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/* calculate length */
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plen = 1;
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while (size >> (7 * plen))
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plen++;
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outbuf = gst_buffer_new_allocate (NULL, plen + size, NULL);
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gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
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ptr = map.data;
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/* write length */
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while (plen) {
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plen--;
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*ptr++ = ((plen > 0) ? 0x80 : 0) | ((size >> (7 * plen)) & 0x7f);
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}
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/* copy data */
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memcpy (ptr, data, size);
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gst_buffer_unmap (outbuf, &map);
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return outbuf;
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}
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static void
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gst_rtp_gst_pay_send_caps (GstRtpGSTPay * rtpgstpay, guint8 cv, GstCaps * caps)
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{
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gchar *capsstr;
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guint capslen;
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GstBuffer *outbuf;
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if (rtpgstpay->flags == ((1 << 7) | (cv << 4))) {
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/* If caps for the current CV are pending in the adapter already, do
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* nothing at all here
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*/
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return;
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} else if (rtpgstpay->flags & (1 << 7)) {
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/* Create a new standalone caps packet if caps were already pending.
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* The next caps are going to be merged with the following buffer or
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* sent standalone if another event is sent first */
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gst_rtp_gst_pay_create_from_adapter (rtpgstpay, GST_CLOCK_TIME_NONE);
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}
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capsstr = gst_caps_to_string (caps);
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capslen = strlen (capsstr);
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/* for 0 byte */
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capslen++;
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GST_DEBUG_OBJECT (rtpgstpay, "sending caps=%s", capsstr);
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/* make a data buffer of it */
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outbuf = make_data_buffer (rtpgstpay, capsstr, capslen);
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g_free (capsstr);
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/* store in adapter, we don't flush yet, buffer might follow */
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rtpgstpay->flags = (1 << 7) | (cv << 4);
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gst_adapter_push (rtpgstpay->adapter, outbuf);
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}
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static gboolean
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gst_rtp_gst_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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|
{
|
|
GstRtpGSTPay *rtpgstpay;
|
|
gboolean res;
|
|
gchar *capsstr, *capsenc, *capsver;
|
|
guint capslen;
|
|
|
|
rtpgstpay = GST_RTP_GST_PAY (payload);
|
|
|
|
capsstr = gst_caps_to_string (caps);
|
|
capslen = strlen (capsstr);
|
|
|
|
/* encode without 0 byte */
|
|
capsenc = g_base64_encode ((guchar *) capsstr, capslen);
|
|
GST_DEBUG_OBJECT (payload, "caps=%s, caps(base64)=%s", capsstr, capsenc);
|
|
g_free (capsstr);
|
|
|
|
/* Send the new caps */
|
|
rtpgstpay->current_CV = rtpgstpay->next_CV;
|
|
rtpgstpay->next_CV = (rtpgstpay->next_CV + 1) & 0x7;
|
|
gst_rtp_gst_pay_send_caps (rtpgstpay, rtpgstpay->current_CV, caps);
|
|
|
|
/* make caps for SDP */
|
|
capsver = g_strdup_printf ("%d", rtpgstpay->current_CV);
|
|
res =
|
|
gst_rtp_base_payload_set_outcaps (payload, "caps", G_TYPE_STRING, capsenc,
|
|
"capsversion", G_TYPE_STRING, capsver, NULL);
|
|
g_free (capsenc);
|
|
g_free (capsver);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_gst_pay_send_event (GstRtpGSTPay * rtpgstpay, guint etype,
|
|
GstEvent * event)
|
|
{
|
|
const GstStructure *s;
|
|
gchar *estr;
|
|
guint elen;
|
|
GstBuffer *outbuf;
|
|
|
|
/* Create the standalone caps packet if an inlined caps was pending */
|
|
gst_rtp_gst_pay_create_from_adapter (rtpgstpay, GST_CLOCK_TIME_NONE);
|
|
|
|
s = gst_event_get_structure (event);
|
|
|
|
estr = gst_structure_to_string (s);
|
|
elen = strlen (estr);
|
|
/* for 0 byte */
|
|
elen++;
|
|
outbuf = make_data_buffer (rtpgstpay, estr, elen);
|
|
GST_DEBUG_OBJECT (rtpgstpay, "sending event=%s", estr);
|
|
g_free (estr);
|
|
|
|
rtpgstpay->etype = etype;
|
|
gst_adapter_push (rtpgstpay->adapter, outbuf);
|
|
/* Create the event packet now to avoid conflict with data/caps packets */
|
|
gst_rtp_gst_pay_create_from_adapter (rtpgstpay, GST_CLOCK_TIME_NONE);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_gst_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
|
|
{
|
|
gboolean ret;
|
|
GstRtpGSTPay *rtpgstpay;
|
|
guint etype = 0;
|
|
|
|
rtpgstpay = GST_RTP_GST_PAY (payload);
|
|
|
|
if (gst_video_event_is_force_key_unit (event)) {
|
|
g_atomic_int_set (&rtpgstpay->force_config, TRUE);
|
|
}
|
|
|
|
if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
|
|
// We must flush at this point, as no next input frame is expected
|
|
gst_rtp_gst_pay_flush (rtpgstpay, GST_CLOCK_TIME_NONE);
|
|
}
|
|
|
|
ret =
|
|
GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload,
|
|
gst_event_ref (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_rtp_gst_pay_reset (rtpgstpay, FALSE);
|
|
break;
|
|
case GST_EVENT_TAG:{
|
|
GstTagList *tags;
|
|
|
|
gst_event_parse_tag (event, &tags);
|
|
|
|
if (gst_tag_list_get_scope (tags) == GST_TAG_SCOPE_STREAM) {
|
|
GstTagList *old;
|
|
|
|
GST_DEBUG_OBJECT (rtpgstpay, "storing stream tags %" GST_PTR_FORMAT,
|
|
tags);
|
|
if ((old = rtpgstpay->taglist))
|
|
gst_tag_list_unref (old);
|
|
rtpgstpay->taglist = gst_tag_list_ref (tags);
|
|
}
|
|
etype = 1;
|
|
break;
|
|
}
|
|
case GST_EVENT_CUSTOM_DOWNSTREAM:
|
|
etype = 2;
|
|
break;
|
|
case GST_EVENT_CUSTOM_BOTH:
|
|
etype = 3;
|
|
break;
|
|
case GST_EVENT_STREAM_START:{
|
|
const gchar *stream_id = NULL;
|
|
|
|
if (rtpgstpay->taglist)
|
|
gst_tag_list_unref (rtpgstpay->taglist);
|
|
rtpgstpay->taglist = NULL;
|
|
|
|
gst_event_parse_stream_start (event, &stream_id);
|
|
if (stream_id) {
|
|
g_free (rtpgstpay->stream_id);
|
|
rtpgstpay->stream_id = g_strdup (stream_id);
|
|
}
|
|
etype = 4;
|
|
break;
|
|
}
|
|
default:
|
|
GST_LOG_OBJECT (rtpgstpay, "no event for %s",
|
|
GST_EVENT_TYPE_NAME (event));
|
|
break;
|
|
}
|
|
if (etype) {
|
|
GST_DEBUG_OBJECT (rtpgstpay, "make event type %d for %s",
|
|
etype, GST_EVENT_TYPE_NAME (event));
|
|
gst_rtp_gst_pay_send_event (rtpgstpay, etype, event);
|
|
// do not flush events here yet as they would get no timestamp at all or
|
|
// the timestamp of the previous buffer, both of which are bogus. We need
|
|
// to wait until the next actual input frame to know the timestamp that
|
|
// applies to the event.
|
|
}
|
|
|
|
gst_event_unref (event);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_gst_pay_src_event (GstRTPBasePayload * payload, GstEvent * event)
|
|
{
|
|
GstRtpGSTPay *rtpgstpay;
|
|
|
|
rtpgstpay = GST_RTP_GST_PAY (payload);
|
|
|
|
if (gst_video_event_is_force_key_unit (event)) {
|
|
g_atomic_int_set (&rtpgstpay->force_config, TRUE);
|
|
}
|
|
|
|
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->src_event (payload, event);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_gst_pay_send_config (GstRtpGSTPay * rtpgstpay,
|
|
GstClockTime running_time)
|
|
{
|
|
GstPad *pad = GST_RTP_BASE_PAYLOAD_SINKPAD (rtpgstpay);
|
|
GstCaps *caps = NULL;
|
|
GstEvent *tag = NULL;
|
|
GstEvent *stream_start = NULL;
|
|
|
|
GST_DEBUG_OBJECT (rtpgstpay, "time to send config");
|
|
/* Send tags */
|
|
if (rtpgstpay->taglist && !gst_tag_list_is_empty (rtpgstpay->taglist))
|
|
tag = gst_event_new_tag (gst_tag_list_ref (rtpgstpay->taglist));
|
|
if (tag) {
|
|
/* Send start-stream to clear tags */
|
|
if (rtpgstpay->stream_id)
|
|
stream_start = gst_event_new_stream_start (rtpgstpay->stream_id);
|
|
if (stream_start) {
|
|
gst_rtp_gst_pay_send_event (rtpgstpay, 4, stream_start);
|
|
gst_event_unref (stream_start);
|
|
}
|
|
gst_rtp_gst_pay_send_event (rtpgstpay, 1, tag);
|
|
gst_event_unref (tag);
|
|
}
|
|
/* send caps */
|
|
caps = gst_pad_get_current_caps (pad);
|
|
if (caps) {
|
|
gst_rtp_gst_pay_send_caps (rtpgstpay, rtpgstpay->current_CV, caps);
|
|
gst_caps_unref (caps);
|
|
}
|
|
rtpgstpay->last_config = running_time;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_gst_pay_handle_buffer (GstRTPBasePayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstRtpGSTPay *rtpgstpay;
|
|
GstClockTime timestamp, running_time;
|
|
|
|
rtpgstpay = GST_RTP_GST_PAY (basepayload);
|
|
|
|
timestamp = GST_BUFFER_PTS (buffer);
|
|
running_time =
|
|
gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME,
|
|
timestamp);
|
|
|
|
/* check if we need to send the caps and taglist now */
|
|
if (rtpgstpay->config_interval > 0
|
|
|| g_atomic_int_compare_and_exchange (&rtpgstpay->force_config, TRUE,
|
|
FALSE)) {
|
|
GST_DEBUG_OBJECT (rtpgstpay,
|
|
"running time %" GST_TIME_FORMAT ", last config %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (running_time), GST_TIME_ARGS (rtpgstpay->last_config));
|
|
|
|
if (running_time != GST_CLOCK_TIME_NONE &&
|
|
rtpgstpay->last_config != GST_CLOCK_TIME_NONE) {
|
|
guint64 diff;
|
|
|
|
/* calculate diff between last SPS/PPS in milliseconds */
|
|
if (running_time > rtpgstpay->last_config)
|
|
diff = running_time - rtpgstpay->last_config;
|
|
else
|
|
diff = 0;
|
|
|
|
GST_DEBUG_OBJECT (rtpgstpay,
|
|
"interval since last config %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
|
|
|
|
/* bigger than interval, queue SPS/PPS */
|
|
if (GST_TIME_AS_SECONDS (diff) >= rtpgstpay->config_interval)
|
|
gst_rtp_gst_pay_send_config (rtpgstpay, running_time);
|
|
} else {
|
|
gst_rtp_gst_pay_send_config (rtpgstpay, running_time);
|
|
}
|
|
}
|
|
|
|
/* caps always from SDP for now */
|
|
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT))
|
|
rtpgstpay->flags |= (1 << 3);
|
|
|
|
gst_adapter_push (rtpgstpay->adapter, buffer);
|
|
ret = gst_rtp_gst_pay_flush (rtpgstpay, timestamp);
|
|
|
|
return ret;
|
|
}
|