mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-09 10:59:39 +00:00
1069 lines
34 KiB
C
1069 lines
34 KiB
C
/* GStreamer
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* Copyright (C) <2005> Wim Taymans <wim@fluendo.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
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*/
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/**
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* SECTION:gstrtpbasepayload
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* @short_description: Base class for RTP payloader
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*
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* Provides a base class for RTP payloaders
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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 "gstrtpbasepayload.h"
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GST_DEBUG_CATEGORY_STATIC (rtpbasepayload_debug);
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#define GST_CAT_DEFAULT (rtpbasepayload_debug)
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#define GST_RTP_BASE_PAYLOAD_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BASE_PAYLOAD, GstRTPBasePayloadPrivate))
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struct _GstRTPBasePayloadPrivate
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{
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gboolean ts_offset_random;
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gboolean seqnum_offset_random;
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gboolean ssrc_random;
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guint16 next_seqnum;
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gboolean perfect_rtptime;
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gint notified_first_timestamp;
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guint64 base_offset;
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gint64 base_rtime;
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gint64 prop_max_ptime;
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gint64 caps_max_ptime;
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};
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/* RTPBasePayload signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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/* FIXME 0.11, a better default is the Ethernet MTU of
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* 1500 - sizeof(headers) as pointed out by marcelm in IRC:
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* So an Ethernet MTU of 1500, minus 60 for the max IP, minus 8 for UDP, gives
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* 1432 bytes or so. And that should be adjusted downward further for other
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* encapsulations like PPPoE, so 1400 at most.
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*/
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#define DEFAULT_MTU 1400
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#define DEFAULT_PT 96
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#define DEFAULT_SSRC -1
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#define DEFAULT_TIMESTAMP_OFFSET -1
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#define DEFAULT_SEQNUM_OFFSET -1
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#define DEFAULT_MAX_PTIME -1
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#define DEFAULT_MIN_PTIME 0
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#define DEFAULT_PERFECT_RTPTIME TRUE
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#define DEFAULT_PTIME_MULTIPLE 0
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enum
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{
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PROP_0,
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PROP_MTU,
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PROP_PT,
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PROP_SSRC,
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PROP_TIMESTAMP_OFFSET,
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PROP_SEQNUM_OFFSET,
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PROP_MAX_PTIME,
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PROP_MIN_PTIME,
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PROP_TIMESTAMP,
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PROP_SEQNUM,
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PROP_PERFECT_RTPTIME,
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PROP_PTIME_MULTIPLE,
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PROP_LAST
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};
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static void gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass);
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static void gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload,
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gpointer g_class);
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static void gst_rtp_base_payload_finalize (GObject * object);
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static GstCaps *gst_rtp_base_payload_sink_getcaps (GstPad * pad,
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GstCaps * filter);
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static gboolean gst_rtp_base_payload_event_default (GstRTPBasePayload *
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rtpbasepayload, GstEvent * event);
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static gboolean gst_rtp_base_payload_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_rtp_base_payload_chain (GstPad * pad,
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GstBuffer * buffer);
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static GstCaps *gst_rtp_base_payload_getcaps_default (GstRTPBasePayload *
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rtpbasepayload, GstPad * pad, GstCaps * filter);
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static void gst_rtp_base_payload_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_base_payload_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_rtp_base_payload_change_state (GstElement *
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element, GstStateChange transition);
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static GstElementClass *parent_class = NULL;
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/* FIXME 0.11: API should be changed to gst_base_typ_payload_xyz */
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GType
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gst_rtp_base_payload_get_type (void)
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{
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static GType rtpbasepayload_type = 0;
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if (g_once_init_enter ((gsize *) & rtpbasepayload_type)) {
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static const GTypeInfo rtpbasepayload_info = {
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sizeof (GstRTPBasePayloadClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_rtp_base_payload_class_init,
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NULL,
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NULL,
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sizeof (GstRTPBasePayload),
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0,
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(GInstanceInitFunc) gst_rtp_base_payload_init,
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};
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g_once_init_leave ((gsize *) & rtpbasepayload_type,
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g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBasePayload",
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&rtpbasepayload_info, G_TYPE_FLAG_ABSTRACT));
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}
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return rtpbasepayload_type;
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}
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static void
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gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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g_type_class_add_private (klass, sizeof (GstRTPBasePayloadPrivate));
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_rtp_base_payload_finalize;
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gobject_class->set_property = gst_rtp_base_payload_set_property;
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gobject_class->get_property = gst_rtp_base_payload_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MTU,
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g_param_spec_uint ("mtu", "MTU",
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"Maximum size of one packet",
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28, G_MAXUINT, DEFAULT_MTU,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT,
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g_param_spec_uint ("pt", "payload type",
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"The payload type of the packets", 0, 0x80, DEFAULT_PT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC,
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g_param_spec_uint ("ssrc", "SSRC",
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"The SSRC of the packets (default == random)", 0, G_MAXUINT32,
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DEFAULT_SSRC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_TIMESTAMP_OFFSET, g_param_spec_uint ("timestamp-offset",
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"Timestamp Offset",
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"Offset to add to all outgoing timestamps (default = random)", 0,
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G_MAXUINT32, DEFAULT_TIMESTAMP_OFFSET,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET,
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g_param_spec_int ("seqnum-offset", "Sequence number Offset",
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"Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXUINT16,
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DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_PTIME,
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g_param_spec_int64 ("max-ptime", "Max packet time",
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"Maximum duration of the packet data in ns (-1 = unlimited up to MTU)",
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-1, G_MAXINT64, DEFAULT_MAX_PTIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTPBaseAudioPayload:min-ptime:
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*
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* Minimum duration of the packet data in ns (can't go above MTU)
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*
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* Since: 0.10.13
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**/
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MIN_PTIME,
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g_param_spec_int64 ("min-ptime", "Min packet time",
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"Minimum duration of the packet data in ns (can't go above MTU)",
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0, G_MAXINT64, DEFAULT_MIN_PTIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP,
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g_param_spec_uint ("timestamp", "Timestamp",
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"The RTP timestamp of the last processed packet",
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0, G_MAXUINT32, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM,
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g_param_spec_uint ("seqnum", "Sequence number",
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"The RTP sequence number of the last processed packet",
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0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTPBaseAudioPayload:perfect-rtptime:
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*
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* Try to use the offset fields to generate perfect RTP timestamps. when this
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* option is disabled, RTP timestamps are generated from the GStreamer
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* timestamps, which could result in RTP timestamps that don't increment with
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* the amount of data in the packet.
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*
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* Since: 0.10.25
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PERFECT_RTPTIME,
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g_param_spec_boolean ("perfect-rtptime", "Perfect RTP Time",
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"Generate perfect RTP timestamps when possible",
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DEFAULT_PERFECT_RTPTIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTPBaseAudioPayload:ptime-multiple:
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*
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* Force buffers to be multiples of this duration in ns (0 disables)
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*
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* Since: 0.10.29
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**/
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PTIME_MULTIPLE,
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g_param_spec_int64 ("ptime-multiple", "Packet time multiple",
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"Force buffers to be multiples of this duration in ns (0 disables)",
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0, G_MAXINT64, DEFAULT_PTIME_MULTIPLE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstelement_class->change_state = gst_rtp_base_payload_change_state;
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klass->get_caps = gst_rtp_base_payload_getcaps_default;
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klass->handle_event = gst_rtp_base_payload_event_default;
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GST_DEBUG_CATEGORY_INIT (rtpbasepayload_debug, "rtpbasepayload", 0,
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"Base class for RTP Payloaders");
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}
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static void
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gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload, gpointer g_class)
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{
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GstPadTemplate *templ;
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GstRTPBasePayloadPrivate *priv;
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rtpbasepayload->priv = priv =
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GST_RTP_BASE_PAYLOAD_GET_PRIVATE (rtpbasepayload);
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templ =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
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g_return_if_fail (templ != NULL);
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rtpbasepayload->srcpad = gst_pad_new_from_template (templ, "src");
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gst_element_add_pad (GST_ELEMENT (rtpbasepayload), rtpbasepayload->srcpad);
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templ =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink");
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g_return_if_fail (templ != NULL);
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rtpbasepayload->sinkpad = gst_pad_new_from_template (templ, "sink");
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gst_pad_set_getcaps_function (rtpbasepayload->sinkpad,
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gst_rtp_base_payload_sink_getcaps);
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gst_pad_set_event_function (rtpbasepayload->sinkpad,
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gst_rtp_base_payload_event);
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gst_pad_set_chain_function (rtpbasepayload->sinkpad,
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gst_rtp_base_payload_chain);
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gst_element_add_pad (GST_ELEMENT (rtpbasepayload), rtpbasepayload->sinkpad);
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rtpbasepayload->mtu = DEFAULT_MTU;
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rtpbasepayload->pt = DEFAULT_PT;
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rtpbasepayload->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
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rtpbasepayload->ssrc = DEFAULT_SSRC;
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rtpbasepayload->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
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priv->seqnum_offset_random = (rtpbasepayload->seqnum_offset == -1);
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priv->ts_offset_random = (rtpbasepayload->ts_offset == -1);
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priv->ssrc_random = (rtpbasepayload->ssrc == -1);
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rtpbasepayload->max_ptime = DEFAULT_MAX_PTIME;
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rtpbasepayload->min_ptime = DEFAULT_MIN_PTIME;
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rtpbasepayload->priv->perfect_rtptime = DEFAULT_PERFECT_RTPTIME;
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rtpbasepayload->abidata.ABI.ptime_multiple = DEFAULT_PTIME_MULTIPLE;
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rtpbasepayload->priv->base_offset = GST_BUFFER_OFFSET_NONE;
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rtpbasepayload->priv->base_rtime = GST_BUFFER_OFFSET_NONE;
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rtpbasepayload->media = NULL;
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rtpbasepayload->encoding_name = NULL;
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rtpbasepayload->clock_rate = 0;
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rtpbasepayload->priv->caps_max_ptime = DEFAULT_MAX_PTIME;
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rtpbasepayload->priv->prop_max_ptime = DEFAULT_MAX_PTIME;
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}
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static void
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gst_rtp_base_payload_finalize (GObject * object)
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{
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GstRTPBasePayload *rtpbasepayload;
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rtpbasepayload = GST_RTP_BASE_PAYLOAD (object);
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g_free (rtpbasepayload->media);
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rtpbasepayload->media = NULL;
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g_free (rtpbasepayload->encoding_name);
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rtpbasepayload->encoding_name = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static GstCaps *
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gst_rtp_base_payload_getcaps_default (GstRTPBasePayload * rtpbasepayload,
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GstPad * pad, GstCaps * filter)
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{
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GstCaps *caps;
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caps = GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad));
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GST_DEBUG_OBJECT (pad,
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"using pad template %p with caps %p %" GST_PTR_FORMAT,
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GST_PAD_PAD_TEMPLATE (pad), caps, caps);
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if (filter)
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caps = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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else
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caps = gst_caps_ref (caps);
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return caps;
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}
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static GstCaps *
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gst_rtp_base_payload_sink_getcaps (GstPad * pad, GstCaps * filter)
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{
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GstRTPBasePayload *rtpbasepayload;
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GstRTPBasePayloadClass *rtpbasepayload_class;
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GstCaps *caps = NULL;
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GST_DEBUG_OBJECT (pad, "getting caps");
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rtpbasepayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad));
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rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
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if (rtpbasepayload_class->get_caps)
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caps = rtpbasepayload_class->get_caps (rtpbasepayload, pad, filter);
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gst_object_unref (rtpbasepayload);
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return caps;
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}
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static gboolean
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gst_rtp_base_payload_event_default (GstRTPBasePayload * rtpbasepayload,
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GstEvent * event)
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{
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gboolean res = FALSE;
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_FLUSH_START:
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res = gst_pad_event_default (rtpbasepayload->sinkpad, event);
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break;
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case GST_EVENT_FLUSH_STOP:
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res = gst_pad_event_default (rtpbasepayload->sinkpad, event);
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gst_segment_init (&rtpbasepayload->segment, GST_FORMAT_UNDEFINED);
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break;
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case GST_EVENT_CAPS:
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{
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GstRTPBasePayloadClass *rtpbasepayload_class;
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GstCaps *caps;
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gst_event_parse_caps (event, &caps);
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GST_DEBUG_OBJECT (rtpbasepayload, "setting caps %" GST_PTR_FORMAT, caps);
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rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
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if (rtpbasepayload_class->set_caps)
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res = rtpbasepayload_class->set_caps (rtpbasepayload, caps);
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gst_event_unref (event);
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break;
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}
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case GST_EVENT_SEGMENT:
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{
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GstSegment *segment;
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segment = &rtpbasepayload->segment;
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gst_event_copy_segment (event, segment);
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rtpbasepayload->priv->base_offset = GST_BUFFER_OFFSET_NONE;
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GST_DEBUG_OBJECT (rtpbasepayload,
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"configured SEGMENT %" GST_SEGMENT_FORMAT, segment);
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res = gst_pad_event_default (rtpbasepayload->sinkpad, event);
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break;
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}
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default:
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res = gst_pad_event_default (rtpbasepayload->sinkpad, event);
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break;
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}
|
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return res;
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}
|
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|
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static gboolean
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gst_rtp_base_payload_event (GstPad * pad, GstEvent * event)
|
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{
|
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GstRTPBasePayload *rtpbasepayload;
|
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GstRTPBasePayloadClass *rtpbasepayload_class;
|
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gboolean res = FALSE;
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rtpbasepayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad));
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if (G_UNLIKELY (rtpbasepayload == NULL)) {
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gst_event_unref (event);
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return FALSE;
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}
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rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
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if (rtpbasepayload_class->handle_event)
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res = rtpbasepayload_class->handle_event (rtpbasepayload, event);
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else
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gst_event_unref (event);
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gst_object_unref (rtpbasepayload);
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return res;
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}
|
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|
|
|
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static GstFlowReturn
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gst_rtp_base_payload_chain (GstPad * pad, GstBuffer * buffer)
|
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{
|
|
GstRTPBasePayload *rtpbasepayload;
|
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GstRTPBasePayloadClass *rtpbasepayload_class;
|
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GstFlowReturn ret;
|
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|
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rtpbasepayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad));
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rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
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|
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if (!rtpbasepayload_class->handle_buffer)
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goto no_function;
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|
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ret = rtpbasepayload_class->handle_buffer (rtpbasepayload, buffer);
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|
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gst_object_unref (rtpbasepayload);
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|
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return ret;
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|
|
/* ERRORS */
|
|
no_function:
|
|
{
|
|
GST_ELEMENT_ERROR (rtpbasepayload, STREAM, NOT_IMPLEMENTED, (NULL),
|
|
("subclass did not implement handle_buffer function"));
|
|
gst_object_unref (rtpbasepayload);
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_payload_set_options:
|
|
* @payload: a #GstRTPBasePayload
|
|
* @media: the media type (typically "audio" or "video")
|
|
* @dynamic: if the payload type is dynamic
|
|
* @encoding_name: the encoding name
|
|
* @clock_rate: the clock rate of the media
|
|
*
|
|
* Set the rtp options of the payloader. These options will be set in the caps
|
|
* of the payloader. Subclasses must call this method before calling
|
|
* gst_rtp_base_payload_push() or gst_rtp_base_payload_set_outcaps().
|
|
*/
|
|
void
|
|
gst_rtp_base_payload_set_options (GstRTPBasePayload * payload,
|
|
const gchar * media, gboolean dynamic, const gchar * encoding_name,
|
|
guint32 clock_rate)
|
|
{
|
|
g_return_if_fail (payload != NULL);
|
|
g_return_if_fail (clock_rate != 0);
|
|
|
|
g_free (payload->media);
|
|
payload->media = g_strdup (media);
|
|
payload->dynamic = dynamic;
|
|
g_free (payload->encoding_name);
|
|
payload->encoding_name = g_strdup (encoding_name);
|
|
payload->clock_rate = clock_rate;
|
|
}
|
|
|
|
static gboolean
|
|
copy_fixed (GQuark field_id, const GValue * value, GstStructure * dest)
|
|
{
|
|
if (gst_value_is_fixed (value)) {
|
|
gst_structure_id_set_value (dest, field_id, value);
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
update_max_ptime (GstRTPBasePayload * rtpbasepayload)
|
|
{
|
|
if (rtpbasepayload->priv->caps_max_ptime != -1 &&
|
|
rtpbasepayload->priv->prop_max_ptime != -1)
|
|
rtpbasepayload->max_ptime = MIN (rtpbasepayload->priv->caps_max_ptime,
|
|
rtpbasepayload->priv->prop_max_ptime);
|
|
else if (rtpbasepayload->priv->caps_max_ptime != -1)
|
|
rtpbasepayload->max_ptime = rtpbasepayload->priv->caps_max_ptime;
|
|
else if (rtpbasepayload->priv->prop_max_ptime != -1)
|
|
rtpbasepayload->max_ptime = rtpbasepayload->priv->prop_max_ptime;
|
|
else
|
|
rtpbasepayload->max_ptime = DEFAULT_MAX_PTIME;
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_payload_set_outcaps:
|
|
* @payload: a #GstRTPBasePayload
|
|
* @fieldname: the first field name or %NULL
|
|
* @...: field values
|
|
*
|
|
* Configure the output caps with the optional parameters.
|
|
*
|
|
* Variable arguments should be in the form field name, field type
|
|
* (as a GType), value(s). The last variable argument should be NULL.
|
|
*
|
|
* Returns: %TRUE if the caps could be set.
|
|
*/
|
|
gboolean
|
|
gst_rtp_base_payload_set_outcaps (GstRTPBasePayload * payload,
|
|
const gchar * fieldname, ...)
|
|
{
|
|
GstCaps *srccaps, *peercaps;
|
|
gboolean res;
|
|
|
|
/* fill in the defaults, their properties cannot be negotiated. */
|
|
srccaps = gst_caps_new_simple ("application/x-rtp",
|
|
"media", G_TYPE_STRING, payload->media,
|
|
"clock-rate", G_TYPE_INT, payload->clock_rate,
|
|
"encoding-name", G_TYPE_STRING, payload->encoding_name, NULL);
|
|
|
|
GST_DEBUG_OBJECT (payload, "defaults: %" GST_PTR_FORMAT, srccaps);
|
|
|
|
if (fieldname) {
|
|
va_list varargs;
|
|
|
|
/* override with custom properties */
|
|
va_start (varargs, fieldname);
|
|
gst_caps_set_simple_valist (srccaps, fieldname, varargs);
|
|
va_end (varargs);
|
|
|
|
GST_DEBUG_OBJECT (payload, "custom added: %" GST_PTR_FORMAT, srccaps);
|
|
}
|
|
|
|
payload->priv->caps_max_ptime = DEFAULT_MAX_PTIME;
|
|
payload->abidata.ABI.ptime = 0;
|
|
|
|
/* the peer caps can override some of the defaults */
|
|
peercaps = gst_pad_peer_get_caps (payload->srcpad, srccaps);
|
|
if (peercaps == NULL) {
|
|
/* no peer caps, just add the other properties */
|
|
gst_caps_set_simple (srccaps,
|
|
"payload", G_TYPE_INT, GST_RTP_BASE_PAYLOAD_PT (payload),
|
|
"ssrc", G_TYPE_UINT, payload->current_ssrc,
|
|
"timestamp-offset", G_TYPE_UINT, payload->ts_base,
|
|
"seqnum-offset", G_TYPE_UINT, payload->seqnum_base, NULL);
|
|
|
|
GST_DEBUG_OBJECT (payload, "no peer caps: %" GST_PTR_FORMAT, srccaps);
|
|
} else {
|
|
GstCaps *temp;
|
|
GstStructure *s, *d;
|
|
const GValue *value;
|
|
gint pt;
|
|
guint max_ptime, ptime;
|
|
|
|
/* peer provides caps we can use to fixate. They are already intersected
|
|
* with our srccaps, just make them writable */
|
|
temp = gst_caps_make_writable (peercaps);
|
|
gst_caps_unref (srccaps);
|
|
|
|
if (gst_caps_is_empty (temp)) {
|
|
gst_caps_unref (temp);
|
|
return FALSE;
|
|
}
|
|
|
|
/* now fixate, start by taking the first caps */
|
|
gst_caps_truncate (temp);
|
|
|
|
/* get first structure */
|
|
s = gst_caps_get_structure (temp, 0);
|
|
|
|
if (gst_structure_get_uint (s, "maxptime", &max_ptime))
|
|
payload->priv->caps_max_ptime = max_ptime * GST_MSECOND;
|
|
|
|
if (gst_structure_get_uint (s, "ptime", &ptime))
|
|
payload->abidata.ABI.ptime = ptime * GST_MSECOND;
|
|
|
|
if (gst_structure_get_int (s, "payload", &pt)) {
|
|
/* use peer pt */
|
|
GST_RTP_BASE_PAYLOAD_PT (payload) = pt;
|
|
GST_LOG_OBJECT (payload, "using peer pt %d", pt);
|
|
} else {
|
|
if (gst_structure_has_field (s, "payload")) {
|
|
/* can only fixate if there is a field */
|
|
gst_structure_fixate_field_nearest_int (s, "payload",
|
|
GST_RTP_BASE_PAYLOAD_PT (payload));
|
|
gst_structure_get_int (s, "payload", &pt);
|
|
GST_LOG_OBJECT (payload, "using peer pt %d", pt);
|
|
} else {
|
|
/* no pt field, use the internal pt */
|
|
pt = GST_RTP_BASE_PAYLOAD_PT (payload);
|
|
gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL);
|
|
GST_LOG_OBJECT (payload, "using internal pt %d", pt);
|
|
}
|
|
}
|
|
|
|
if (gst_structure_has_field_typed (s, "ssrc", G_TYPE_UINT)) {
|
|
value = gst_structure_get_value (s, "ssrc");
|
|
payload->current_ssrc = g_value_get_uint (value);
|
|
GST_LOG_OBJECT (payload, "using peer ssrc %08x", payload->current_ssrc);
|
|
} else {
|
|
/* FIXME, fixate_nearest_uint would be even better */
|
|
gst_structure_set (s, "ssrc", G_TYPE_UINT, payload->current_ssrc, NULL);
|
|
GST_LOG_OBJECT (payload, "using internal ssrc %08x",
|
|
payload->current_ssrc);
|
|
}
|
|
|
|
if (gst_structure_has_field_typed (s, "timestamp-offset", G_TYPE_UINT)) {
|
|
value = gst_structure_get_value (s, "timestamp-offset");
|
|
payload->ts_base = g_value_get_uint (value);
|
|
GST_LOG_OBJECT (payload, "using peer timestamp-offset %u",
|
|
payload->ts_base);
|
|
} else {
|
|
/* FIXME, fixate_nearest_uint would be even better */
|
|
gst_structure_set (s, "timestamp-offset", G_TYPE_UINT, payload->ts_base,
|
|
NULL);
|
|
GST_LOG_OBJECT (payload, "using internal timestamp-offset %u",
|
|
payload->ts_base);
|
|
}
|
|
if (gst_structure_has_field_typed (s, "seqnum-offset", G_TYPE_UINT)) {
|
|
value = gst_structure_get_value (s, "seqnum-offset");
|
|
payload->seqnum_base = g_value_get_uint (value);
|
|
GST_LOG_OBJECT (payload, "using peer seqnum-offset %u",
|
|
payload->seqnum_base);
|
|
} else {
|
|
/* FIXME, fixate_nearest_uint would be even better */
|
|
gst_structure_set (s, "seqnum-offset", G_TYPE_UINT, payload->seqnum_base,
|
|
NULL);
|
|
GST_LOG_OBJECT (payload, "using internal seqnum-offset %u",
|
|
payload->seqnum_base);
|
|
}
|
|
|
|
/* make the target caps by copying over all the fixed caps, removing the
|
|
* unfixed caps. */
|
|
srccaps = gst_caps_new_empty_simple (gst_structure_get_name (s));
|
|
d = gst_caps_get_structure (srccaps, 0);
|
|
|
|
gst_structure_foreach (s, (GstStructureForeachFunc) copy_fixed, d);
|
|
|
|
gst_caps_unref (temp);
|
|
|
|
GST_DEBUG_OBJECT (payload, "with peer caps: %" GST_PTR_FORMAT, srccaps);
|
|
}
|
|
|
|
update_max_ptime (payload);
|
|
|
|
res = gst_pad_set_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), srccaps);
|
|
gst_caps_unref (srccaps);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_payload_is_filled:
|
|
* @payload: a #GstRTPBasePayload
|
|
* @size: the size of the packet
|
|
* @duration: the duration of the packet
|
|
*
|
|
* Check if the packet with @size and @duration would exceed the configured
|
|
* maximum size.
|
|
*
|
|
* Returns: %TRUE if the packet of @size and @duration would exceed the
|
|
* configured MTU or max_ptime.
|
|
*/
|
|
gboolean
|
|
gst_rtp_base_payload_is_filled (GstRTPBasePayload * payload,
|
|
guint size, GstClockTime duration)
|
|
{
|
|
if (size > payload->mtu)
|
|
return TRUE;
|
|
|
|
if (payload->max_ptime != -1 && duration >= payload->max_ptime)
|
|
return TRUE;
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
typedef struct
|
|
{
|
|
GstRTPBasePayload *payload;
|
|
guint32 ssrc;
|
|
guint16 seqnum;
|
|
guint8 pt;
|
|
GstClockTime timestamp;
|
|
guint64 offset;
|
|
guint32 rtptime;
|
|
} HeaderData;
|
|
|
|
static gboolean
|
|
find_timestamp (GstBuffer ** buffer, guint idx, HeaderData * data)
|
|
{
|
|
data->timestamp = GST_BUFFER_TIMESTAMP (*buffer);
|
|
data->offset = GST_BUFFER_OFFSET (*buffer);
|
|
|
|
/* stop when we find a timestamp. We take whatever offset is associated with
|
|
* the timestamp (if any) to do perfect timestamps when we need to. */
|
|
if (data->timestamp != -1)
|
|
return FALSE;
|
|
else
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
set_headers (GstBuffer ** buffer, guint group, guint idx, HeaderData * data)
|
|
{
|
|
GstRTPBuffer rtp;
|
|
|
|
gst_rtp_buffer_map (*buffer, GST_MAP_WRITE, &rtp);
|
|
gst_rtp_buffer_set_ssrc (&rtp, data->ssrc);
|
|
gst_rtp_buffer_set_payload_type (&rtp, data->pt);
|
|
gst_rtp_buffer_set_seq (&rtp, data->seqnum);
|
|
gst_rtp_buffer_set_timestamp (&rtp, data->rtptime);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
/* increment the seqnum for each buffer */
|
|
data->seqnum++;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* Updates the SSRC, payload type, seqnum and timestamp of the RTP buffer
|
|
* before the buffer is pushed. */
|
|
static GstFlowReturn
|
|
gst_rtp_base_payload_prepare_push (GstRTPBasePayload * payload,
|
|
gpointer obj, gboolean is_list)
|
|
{
|
|
GstRTPBasePayloadPrivate *priv;
|
|
HeaderData data;
|
|
|
|
if (payload->clock_rate == 0)
|
|
goto no_rate;
|
|
|
|
priv = payload->priv;
|
|
|
|
/* update first, so that the property is set to the last
|
|
* seqnum pushed */
|
|
payload->seqnum = priv->next_seqnum;
|
|
|
|
/* fill in the fields we want to set on all headers */
|
|
data.payload = payload;
|
|
data.seqnum = payload->seqnum;
|
|
data.ssrc = payload->current_ssrc;
|
|
data.pt = payload->pt;
|
|
|
|
/* find the first buffer with a timestamp */
|
|
if (is_list) {
|
|
data.timestamp = -1;
|
|
data.offset = GST_BUFFER_OFFSET_NONE;
|
|
gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj),
|
|
(GstBufferListFunc) find_timestamp, &data);
|
|
} else {
|
|
data.timestamp = GST_BUFFER_TIMESTAMP (GST_BUFFER_CAST (obj));
|
|
data.offset = GST_BUFFER_OFFSET (GST_BUFFER_CAST (obj));
|
|
}
|
|
|
|
/* convert to RTP time */
|
|
if (priv->perfect_rtptime && data.offset != GST_BUFFER_OFFSET_NONE &&
|
|
priv->base_offset != GST_BUFFER_OFFSET_NONE) {
|
|
/* if we have an offset, use that for making an RTP timestamp */
|
|
data.rtptime = payload->ts_base + priv->base_rtime +
|
|
data.offset - priv->base_offset;
|
|
GST_LOG_OBJECT (payload,
|
|
"Using offset %" G_GUINT64_FORMAT " for RTP timestamp", data.offset);
|
|
} else if (GST_CLOCK_TIME_IS_VALID (data.timestamp)) {
|
|
gint64 rtime;
|
|
|
|
/* no offset, use the gstreamer timestamp */
|
|
rtime = gst_segment_to_running_time (&payload->segment, GST_FORMAT_TIME,
|
|
data.timestamp);
|
|
|
|
if (rtime == -1) {
|
|
GST_LOG_OBJECT (payload, "Clipped timestamp, using base RTP timestamp");
|
|
rtime = 0;
|
|
} else {
|
|
GST_LOG_OBJECT (payload,
|
|
"Using running_time %" GST_TIME_FORMAT " for RTP timestamp",
|
|
GST_TIME_ARGS (rtime));
|
|
rtime =
|
|
gst_util_uint64_scale_int (rtime, payload->clock_rate, GST_SECOND);
|
|
priv->base_offset = data.offset;
|
|
priv->base_rtime = rtime;
|
|
}
|
|
/* add running_time in clock-rate units to the base timestamp */
|
|
data.rtptime = payload->ts_base + rtime;
|
|
} else {
|
|
GST_LOG_OBJECT (payload,
|
|
"Using previous RTP timestamp %" G_GUINT32_FORMAT, payload->timestamp);
|
|
/* no timestamp to convert, take previous timestamp */
|
|
data.rtptime = payload->timestamp;
|
|
}
|
|
|
|
/* set ssrc, payload type, seq number, caps and rtptime */
|
|
if (is_list) {
|
|
gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj),
|
|
(GstBufferListFunc) set_headers, &data);
|
|
} else {
|
|
GstBuffer *buf = GST_BUFFER_CAST (obj);
|
|
set_headers (&buf, 0, 0, &data);
|
|
}
|
|
|
|
priv->next_seqnum = data.seqnum;
|
|
payload->timestamp = data.rtptime;
|
|
|
|
GST_LOG_OBJECT (payload, "Preparing to push packet with size %"
|
|
G_GSIZE_FORMAT ", seq=%d, rtptime=%u, timestamp %" GST_TIME_FORMAT,
|
|
(is_list) ? -1 : gst_buffer_get_size (GST_BUFFER (obj)),
|
|
payload->seqnum, data.rtptime, GST_TIME_ARGS (data.timestamp));
|
|
|
|
if (g_atomic_int_compare_and_exchange (&payload->priv->
|
|
notified_first_timestamp, 1, 0)) {
|
|
g_object_notify (G_OBJECT (payload), "timestamp");
|
|
g_object_notify (G_OBJECT (payload), "seqnum");
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
no_rate:
|
|
{
|
|
GST_ELEMENT_ERROR (payload, STREAM, NOT_IMPLEMENTED, (NULL),
|
|
("subclass did not specify clock-rate"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_payload_push_list:
|
|
* @payload: a #GstRTPBasePayload
|
|
* @list: a #GstBufferList
|
|
*
|
|
* Push @list to the peer element of the payloader. The SSRC, payload type,
|
|
* seqnum and timestamp of the RTP buffer will be updated first.
|
|
*
|
|
* This function takes ownership of @list.
|
|
*
|
|
* Returns: a #GstFlowReturn.
|
|
*
|
|
* Since: 0.10.24
|
|
*/
|
|
GstFlowReturn
|
|
gst_rtp_base_payload_push_list (GstRTPBasePayload * payload,
|
|
GstBufferList * list)
|
|
{
|
|
GstFlowReturn res;
|
|
|
|
res = gst_rtp_base_payload_prepare_push (payload, list, TRUE);
|
|
|
|
if (G_LIKELY (res == GST_FLOW_OK))
|
|
res = gst_pad_push_list (payload->srcpad, list);
|
|
else
|
|
gst_buffer_list_unref (list);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_payload_push:
|
|
* @payload: a #GstRTPBasePayload
|
|
* @buffer: a #GstBuffer
|
|
*
|
|
* Push @buffer to the peer element of the payloader. The SSRC, payload type,
|
|
* seqnum and timestamp of the RTP buffer will be updated first.
|
|
*
|
|
* This function takes ownership of @buffer.
|
|
*
|
|
* Returns: a #GstFlowReturn.
|
|
*/
|
|
GstFlowReturn
|
|
gst_rtp_base_payload_push (GstRTPBasePayload * payload, GstBuffer * buffer)
|
|
{
|
|
GstFlowReturn res;
|
|
|
|
res = gst_rtp_base_payload_prepare_push (payload, buffer, FALSE);
|
|
|
|
if (G_LIKELY (res == GST_FLOW_OK))
|
|
res = gst_pad_push (payload->srcpad, buffer);
|
|
else
|
|
gst_buffer_unref (buffer);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_base_payload_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTPBasePayload *rtpbasepayload;
|
|
GstRTPBasePayloadPrivate *priv;
|
|
gint64 val;
|
|
|
|
rtpbasepayload = GST_RTP_BASE_PAYLOAD (object);
|
|
priv = rtpbasepayload->priv;
|
|
|
|
switch (prop_id) {
|
|
case PROP_MTU:
|
|
rtpbasepayload->mtu = g_value_get_uint (value);
|
|
break;
|
|
case PROP_PT:
|
|
rtpbasepayload->pt = g_value_get_uint (value);
|
|
break;
|
|
case PROP_SSRC:
|
|
val = g_value_get_uint (value);
|
|
rtpbasepayload->ssrc = val;
|
|
priv->ssrc_random = FALSE;
|
|
break;
|
|
case PROP_TIMESTAMP_OFFSET:
|
|
val = g_value_get_uint (value);
|
|
rtpbasepayload->ts_offset = val;
|
|
priv->ts_offset_random = FALSE;
|
|
break;
|
|
case PROP_SEQNUM_OFFSET:
|
|
val = g_value_get_int (value);
|
|
rtpbasepayload->seqnum_offset = val;
|
|
priv->seqnum_offset_random = (val == -1);
|
|
GST_DEBUG_OBJECT (rtpbasepayload, "seqnum offset 0x%04x, random %d",
|
|
rtpbasepayload->seqnum_offset, priv->seqnum_offset_random);
|
|
break;
|
|
case PROP_MAX_PTIME:
|
|
rtpbasepayload->priv->prop_max_ptime = g_value_get_int64 (value);
|
|
update_max_ptime (rtpbasepayload);
|
|
break;
|
|
case PROP_MIN_PTIME:
|
|
rtpbasepayload->min_ptime = g_value_get_int64 (value);
|
|
break;
|
|
case PROP_PERFECT_RTPTIME:
|
|
priv->perfect_rtptime = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_PTIME_MULTIPLE:
|
|
rtpbasepayload->abidata.ABI.ptime_multiple = g_value_get_int64 (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_base_payload_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTPBasePayload *rtpbasepayload;
|
|
GstRTPBasePayloadPrivate *priv;
|
|
|
|
rtpbasepayload = GST_RTP_BASE_PAYLOAD (object);
|
|
priv = rtpbasepayload->priv;
|
|
|
|
switch (prop_id) {
|
|
case PROP_MTU:
|
|
g_value_set_uint (value, rtpbasepayload->mtu);
|
|
break;
|
|
case PROP_PT:
|
|
g_value_set_uint (value, rtpbasepayload->pt);
|
|
break;
|
|
case PROP_SSRC:
|
|
if (priv->ssrc_random)
|
|
g_value_set_uint (value, -1);
|
|
else
|
|
g_value_set_uint (value, rtpbasepayload->ssrc);
|
|
break;
|
|
case PROP_TIMESTAMP_OFFSET:
|
|
if (priv->ts_offset_random)
|
|
g_value_set_uint (value, -1);
|
|
else
|
|
g_value_set_uint (value, (guint32) rtpbasepayload->ts_offset);
|
|
break;
|
|
case PROP_SEQNUM_OFFSET:
|
|
if (priv->seqnum_offset_random)
|
|
g_value_set_int (value, -1);
|
|
else
|
|
g_value_set_int (value, (guint16) rtpbasepayload->seqnum_offset);
|
|
break;
|
|
case PROP_MAX_PTIME:
|
|
g_value_set_int64 (value, rtpbasepayload->max_ptime);
|
|
break;
|
|
case PROP_MIN_PTIME:
|
|
g_value_set_int64 (value, rtpbasepayload->min_ptime);
|
|
break;
|
|
case PROP_TIMESTAMP:
|
|
g_value_set_uint (value, rtpbasepayload->timestamp);
|
|
break;
|
|
case PROP_SEQNUM:
|
|
g_value_set_uint (value, rtpbasepayload->seqnum);
|
|
break;
|
|
case PROP_PERFECT_RTPTIME:
|
|
g_value_set_boolean (value, priv->perfect_rtptime);
|
|
break;
|
|
case PROP_PTIME_MULTIPLE:
|
|
g_value_set_int64 (value, rtpbasepayload->abidata.ABI.ptime_multiple);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_base_payload_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstRTPBasePayload *rtpbasepayload;
|
|
GstRTPBasePayloadPrivate *priv;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtpbasepayload = GST_RTP_BASE_PAYLOAD (element);
|
|
priv = rtpbasepayload->priv;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
gst_segment_init (&rtpbasepayload->segment, GST_FORMAT_UNDEFINED);
|
|
|
|
if (priv->seqnum_offset_random)
|
|
rtpbasepayload->seqnum_base = g_random_int_range (0, G_MAXUINT16);
|
|
else
|
|
rtpbasepayload->seqnum_base = rtpbasepayload->seqnum_offset;
|
|
priv->next_seqnum = rtpbasepayload->seqnum_base;
|
|
rtpbasepayload->seqnum = rtpbasepayload->seqnum_base;
|
|
|
|
if (priv->ssrc_random)
|
|
rtpbasepayload->current_ssrc = g_random_int ();
|
|
else
|
|
rtpbasepayload->current_ssrc = rtpbasepayload->ssrc;
|
|
|
|
if (priv->ts_offset_random)
|
|
rtpbasepayload->ts_base = g_random_int ();
|
|
else
|
|
rtpbasepayload->ts_base = rtpbasepayload->ts_offset;
|
|
rtpbasepayload->timestamp = rtpbasepayload->ts_base;
|
|
g_atomic_int_set (&rtpbasepayload->priv->notified_first_timestamp, 1);
|
|
priv->base_offset = GST_BUFFER_OFFSET_NONE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
g_atomic_int_set (&rtpbasepayload->priv->notified_first_timestamp, 1);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|