mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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f1b3ed37c6
This patch introduces a property which, if set to FALSE, prevents RTP basepayloader from scaling the RTP time when a segment's rate is not equal to 1.0. The specification is ambiguous on this subject and some clients expect the timestamps not to be scaled.
1852 lines
60 KiB
C
1852 lines
60 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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* @title: 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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#include "gstrtpmeta.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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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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gboolean pt_set;
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gboolean source_info;
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GstBuffer *input_meta_buffer;
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guint8 twcc_ext_id;
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guint64 base_offset;
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gint64 base_rtime;
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guint64 base_rtime_hz;
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guint64 running_time;
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gboolean scale_rtptime;
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gint64 prop_max_ptime;
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gint64 caps_max_ptime;
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gboolean onvif_no_rate_control;
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gboolean negotiated;
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gboolean delay_segment;
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GstEvent *pending_segment;
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GstCaps *subclass_srccaps;
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GstCaps *sinkcaps;
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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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#define DEFAULT_RUNNING_TIME GST_CLOCK_TIME_NONE
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#define DEFAULT_SOURCE_INFO FALSE
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#define DEFAULT_ONVIF_NO_RATE_CONTROL FALSE
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#define DEFAULT_TWCC_EXT_ID 0
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#define DEFAULT_SCALE_RTPTIME TRUE
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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_STATS,
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PROP_SOURCE_INFO,
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PROP_ONVIF_NO_RATE_CONTROL,
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PROP_TWCC_EXT_ID,
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PROP_SCALE_RTPTIME,
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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_getcaps_default (GstRTPBasePayload *
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rtpbasepayload, GstPad * pad, GstCaps * filter);
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static gboolean gst_rtp_base_payload_sink_event_default (GstRTPBasePayload *
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rtpbasepayload, GstEvent * event);
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static gboolean gst_rtp_base_payload_sink_event (GstPad * pad,
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GstObject * parent, GstEvent * event);
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static gboolean gst_rtp_base_payload_src_event_default (GstRTPBasePayload *
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rtpbasepayload, GstEvent * event);
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static gboolean gst_rtp_base_payload_src_event (GstPad * pad,
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GstObject * parent, GstEvent * event);
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static gboolean gst_rtp_base_payload_query_default (GstRTPBasePayload *
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rtpbasepayload, GstPad * pad, GstQuery * query);
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static gboolean gst_rtp_base_payload_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static GstFlowReturn gst_rtp_base_payload_chain (GstPad * pad,
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GstObject * parent, GstBuffer * buffer);
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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 gboolean gst_rtp_base_payload_negotiate (GstRTPBasePayload * payload);
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static GstElementClass *parent_class = NULL;
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static gint private_offset = 0;
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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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GType _type;
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_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBasePayload",
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&rtpbasepayload_info, G_TYPE_FLAG_ABSTRACT);
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private_offset =
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g_type_add_instance_private (_type, sizeof (GstRTPBasePayloadPrivate));
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g_once_init_leave ((gsize *) & rtpbasepayload_type, _type);
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}
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return rtpbasepayload_type;
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}
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static inline GstRTPBasePayloadPrivate *
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gst_rtp_base_payload_get_instance_private (GstRTPBasePayload * self)
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{
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return (G_STRUCT_MEMBER_P (self, private_offset));
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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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if (private_offset != 0)
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g_type_class_adjust_private_offset (klass, &private_offset);
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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, 0x7f, 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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* GstRTPBasePayload: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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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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* GstRTPBasePayload: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 GST_BUFFER_PTS of
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* each payloaded buffer. The PTSes of buffers may not necessarily increment
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* with the amount of data in each input buffer, consider e.g. the case where
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* the buffer arrives from a network which means that the PTS is unrelated to
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* the amount of data. Because the RTP timestamps are generated from
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* GST_BUFFER_PTS this can result in RTP timestamps that also don't increment
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* with the amount of data in the payloaded packet. To circumvent this it is
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* possible to set the perfect rtptime option enabled. When this option is
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* enabled the payloader will increment the RTP timestamps based on
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* GST_BUFFER_OFFSET which relates to the amount of data in each packet
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* rather than the GST_BUFFER_PTS of each buffer and therefore the RTP
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* timestamps will more closely correlate with the amount of data in each
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* buffer. Currently GstRTPBasePayload is limited to handling perfect RTP
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* timestamps for audio streams.
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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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* GstRTPBasePayload: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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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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/**
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* GstRTPBasePayload:stats:
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*
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* Various payloader statistics retrieved atomically (and are therefore
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* synchroized with each other), these can be used e.g. to generate an
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* RTP-Info header. This property return a GstStructure named
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* application/x-rtp-payload-stats containing the following fields relating to
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* the last processed buffer and current state of the stream being payloaded:
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*
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* * `clock-rate` :#G_TYPE_UINT, clock-rate of the stream
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* * `running-time` :#G_TYPE_UINT64, running time
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* * `seqnum` :#G_TYPE_UINT, sequence number, same as #GstRTPBasePayload:seqnum
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* * `timestamp` :#G_TYPE_UINT, RTP timestamp, same as #GstRTPBasePayload:timestamp
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* * `ssrc` :#G_TYPE_UINT, The SSRC in use
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* * `pt` :#G_TYPE_UINT, The Payload type in use, same as #GstRTPBasePayload:pt
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* * `seqnum-offset` :#G_TYPE_UINT, The current offset added to the seqnum
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* * `timestamp-offset` :#G_TYPE_UINT, The current offset added to the timestamp
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**/
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_STATS,
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g_param_spec_boxed ("stats", "Statistics", "Various statistics",
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GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTPBasePayload:source-info:
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*
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* Enable writing the CSRC field in allocated RTP header based on RTP source
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* information found in the input buffer's #GstRTPSourceMeta.
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*
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* Since: 1.16
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**/
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g_object_class_install_property (gobject_class, PROP_SOURCE_INFO,
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g_param_spec_boolean ("source-info", "RTP source information",
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"Write CSRC based on buffer meta RTP source information",
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DEFAULT_SOURCE_INFO, G_PARAM_READWRITE));
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/**
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* GstRTPBasePayload:onvif-no-rate-control:
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*
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* Make the payloader timestamp packets according to the Rate-Control=no
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* behaviour specified in the ONVIF replay spec.
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*
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* Since: 1.16
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_ONVIF_NO_RATE_CONTROL, g_param_spec_boolean ("onvif-no-rate-control",
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"ONVIF no rate control",
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"Enable ONVIF Rate-Control=no timestamping mode",
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DEFAULT_ONVIF_NO_RATE_CONTROL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTPBasePayload:twcc-ext-id:
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*
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* The RTP header-extension ID used for tagging buffers with Transport-Wide
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* Congestion Control sequence-numbers.
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*
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* To use this across multiple bundled streams (transport wide), the
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* GstRTPFunnel can mux TWCC sequence-numbers together.
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*
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* This is experimental, as it is still a draft and not yet a standard.
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*
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* Since: 1.18
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*/
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g_object_class_install_property (gobject_class, PROP_TWCC_EXT_ID,
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g_param_spec_uint ("twcc-ext-id",
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"Transport-wide Congestion Control Extension ID (experimental)",
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"The RTP header-extension ID to use for tagging buffers with "
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"Transport-wide Congestion Control sequencenumbers (0 = disable)",
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0, 15, DEFAULT_TWCC_EXT_ID,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTPBasePayload:scale-rtptime:
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*
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* Make the RTP packets' timestamps be scaled with the segment's rate
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* (corresponding to RTSP speed parameter). Disabling this property means
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* the timestamps will not be affected by the set delivery speed (RTSP speed).
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*
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* Example: A server wants to allow streaming a recorded video in double
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* speed but still have the timestamps correspond to the position in the
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* video. This is achieved by the client setting RTSP Speed to 2 while the
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* server has this property disabled.
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*
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* Since: 1.18
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SCALE_RTPTIME,
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g_param_spec_boolean ("scale-rtptime", "Scale RTP time",
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"Whether the RTP timestamp should be scaled with the rate (speed)",
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DEFAULT_SCALE_RTPTIME, 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->sink_event = gst_rtp_base_payload_sink_event_default;
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klass->src_event = gst_rtp_base_payload_src_event_default;
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klass->query = gst_rtp_base_payload_query_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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|
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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_instance_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_pad_set_event_function (rtpbasepayload->srcpad,
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gst_rtp_base_payload_src_event);
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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_chain_function (rtpbasepayload->sinkpad,
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gst_rtp_base_payload_chain);
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gst_pad_set_event_function (rtpbasepayload->sinkpad,
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gst_rtp_base_payload_sink_event);
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gst_pad_set_query_function (rtpbasepayload->sinkpad,
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gst_rtp_base_payload_query);
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gst_element_add_pad (GST_ELEMENT (rtpbasepayload), rtpbasepayload->sinkpad);
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|
|
rtpbasepayload->mtu = DEFAULT_MTU;
|
|
rtpbasepayload->pt = DEFAULT_PT;
|
|
rtpbasepayload->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
|
|
rtpbasepayload->ssrc = DEFAULT_SSRC;
|
|
rtpbasepayload->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
|
|
priv->running_time = DEFAULT_RUNNING_TIME;
|
|
priv->seqnum_offset_random = (rtpbasepayload->seqnum_offset == -1);
|
|
priv->ts_offset_random = (rtpbasepayload->ts_offset == -1);
|
|
priv->ssrc_random = (rtpbasepayload->ssrc == -1);
|
|
priv->pt_set = FALSE;
|
|
priv->source_info = DEFAULT_SOURCE_INFO;
|
|
|
|
rtpbasepayload->max_ptime = DEFAULT_MAX_PTIME;
|
|
rtpbasepayload->min_ptime = DEFAULT_MIN_PTIME;
|
|
rtpbasepayload->priv->perfect_rtptime = DEFAULT_PERFECT_RTPTIME;
|
|
rtpbasepayload->ptime_multiple = DEFAULT_PTIME_MULTIPLE;
|
|
rtpbasepayload->priv->base_offset = GST_BUFFER_OFFSET_NONE;
|
|
rtpbasepayload->priv->base_rtime_hz = GST_BUFFER_OFFSET_NONE;
|
|
rtpbasepayload->priv->onvif_no_rate_control = DEFAULT_ONVIF_NO_RATE_CONTROL;
|
|
rtpbasepayload->priv->scale_rtptime = DEFAULT_SCALE_RTPTIME;
|
|
|
|
rtpbasepayload->media = NULL;
|
|
rtpbasepayload->encoding_name = NULL;
|
|
|
|
rtpbasepayload->clock_rate = 0;
|
|
|
|
rtpbasepayload->priv->caps_max_ptime = DEFAULT_MAX_PTIME;
|
|
rtpbasepayload->priv->prop_max_ptime = DEFAULT_MAX_PTIME;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_base_payload_finalize (GObject * object)
|
|
{
|
|
GstRTPBasePayload *rtpbasepayload;
|
|
|
|
rtpbasepayload = GST_RTP_BASE_PAYLOAD (object);
|
|
|
|
g_free (rtpbasepayload->media);
|
|
rtpbasepayload->media = NULL;
|
|
g_free (rtpbasepayload->encoding_name);
|
|
rtpbasepayload->encoding_name = NULL;
|
|
|
|
gst_caps_replace (&rtpbasepayload->priv->subclass_srccaps, NULL);
|
|
gst_caps_replace (&rtpbasepayload->priv->sinkcaps, NULL);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_rtp_base_payload_getcaps_default (GstRTPBasePayload * rtpbasepayload,
|
|
GstPad * pad, GstCaps * filter)
|
|
{
|
|
GstCaps *caps;
|
|
|
|
caps = GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad));
|
|
GST_DEBUG_OBJECT (pad,
|
|
"using pad template %p with caps %p %" GST_PTR_FORMAT,
|
|
GST_PAD_PAD_TEMPLATE (pad), caps, caps);
|
|
|
|
if (filter)
|
|
caps = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
|
|
else
|
|
caps = gst_caps_ref (caps);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_base_payload_sink_event_default (GstRTPBasePayload * rtpbasepayload,
|
|
GstEvent * event)
|
|
{
|
|
GstObject *parent = GST_OBJECT_CAST (rtpbasepayload);
|
|
gboolean res = FALSE;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_START:
|
|
res = gst_pad_event_default (rtpbasepayload->sinkpad, parent, event);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
res = gst_pad_event_default (rtpbasepayload->sinkpad, parent, event);
|
|
gst_segment_init (&rtpbasepayload->segment, GST_FORMAT_UNDEFINED);
|
|
gst_event_replace (&rtpbasepayload->priv->pending_segment, NULL);
|
|
break;
|
|
case GST_EVENT_CAPS:
|
|
{
|
|
GstRTPBasePayloadClass *rtpbasepayload_class;
|
|
GstCaps *caps;
|
|
|
|
gst_event_parse_caps (event, &caps);
|
|
GST_DEBUG_OBJECT (rtpbasepayload, "setting caps %" GST_PTR_FORMAT, caps);
|
|
|
|
gst_caps_replace (&rtpbasepayload->priv->sinkcaps, caps);
|
|
|
|
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
|
|
if (rtpbasepayload_class->set_caps)
|
|
res = rtpbasepayload_class->set_caps (rtpbasepayload, caps);
|
|
else
|
|
res = gst_rtp_base_payload_negotiate (rtpbasepayload);
|
|
|
|
rtpbasepayload->priv->negotiated = res;
|
|
|
|
gst_event_unref (event);
|
|
break;
|
|
}
|
|
case GST_EVENT_SEGMENT:
|
|
{
|
|
GstSegment *segment;
|
|
|
|
segment = &rtpbasepayload->segment;
|
|
gst_event_copy_segment (event, segment);
|
|
|
|
rtpbasepayload->priv->base_offset = GST_BUFFER_OFFSET_NONE;
|
|
|
|
GST_DEBUG_OBJECT (rtpbasepayload,
|
|
"configured SEGMENT %" GST_SEGMENT_FORMAT, segment);
|
|
if (rtpbasepayload->priv->delay_segment) {
|
|
gst_event_replace (&rtpbasepayload->priv->pending_segment, event);
|
|
gst_event_unref (event);
|
|
res = TRUE;
|
|
} else {
|
|
res = gst_pad_event_default (rtpbasepayload->sinkpad, parent, event);
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_event_default (rtpbasepayload->sinkpad, parent, event);
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_base_payload_sink_event (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
GstRTPBasePayload *rtpbasepayload;
|
|
GstRTPBasePayloadClass *rtpbasepayload_class;
|
|
gboolean res = FALSE;
|
|
|
|
rtpbasepayload = GST_RTP_BASE_PAYLOAD (parent);
|
|
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
|
|
|
|
if (rtpbasepayload_class->sink_event)
|
|
res = rtpbasepayload_class->sink_event (rtpbasepayload, event);
|
|
else
|
|
gst_event_unref (event);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_base_payload_src_event_default (GstRTPBasePayload * rtpbasepayload,
|
|
GstEvent * event)
|
|
{
|
|
GstObject *parent = GST_OBJECT_CAST (rtpbasepayload);
|
|
gboolean res = TRUE, forward = TRUE;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_CUSTOM_UPSTREAM:
|
|
{
|
|
const GstStructure *s = gst_event_get_structure (event);
|
|
|
|
if (gst_structure_has_name (s, "GstRTPCollision")) {
|
|
guint ssrc = 0;
|
|
|
|
if (!gst_structure_get_uint (s, "ssrc", &ssrc))
|
|
ssrc = -1;
|
|
|
|
GST_DEBUG_OBJECT (rtpbasepayload, "collided ssrc: %" G_GUINT32_FORMAT,
|
|
ssrc);
|
|
|
|
/* choose another ssrc for our stream */
|
|
if (ssrc == rtpbasepayload->current_ssrc) {
|
|
GstCaps *caps;
|
|
guint suggested_ssrc = 0;
|
|
|
|
if (gst_structure_get_uint (s, "suggested-ssrc", &suggested_ssrc))
|
|
rtpbasepayload->current_ssrc = suggested_ssrc;
|
|
|
|
while (ssrc == rtpbasepayload->current_ssrc)
|
|
rtpbasepayload->current_ssrc = g_random_int ();
|
|
|
|
caps = gst_pad_get_current_caps (rtpbasepayload->srcpad);
|
|
if (caps) {
|
|
caps = gst_caps_make_writable (caps);
|
|
gst_caps_set_simple (caps,
|
|
"ssrc", G_TYPE_UINT, rtpbasepayload->current_ssrc, NULL);
|
|
res = gst_pad_set_caps (rtpbasepayload->srcpad, caps);
|
|
gst_caps_unref (caps);
|
|
}
|
|
|
|
/* the event was for us */
|
|
forward = FALSE;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (forward)
|
|
res = gst_pad_event_default (rtpbasepayload->srcpad, parent, event);
|
|
else
|
|
gst_event_unref (event);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_base_payload_src_event (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
GstRTPBasePayload *rtpbasepayload;
|
|
GstRTPBasePayloadClass *rtpbasepayload_class;
|
|
gboolean res = FALSE;
|
|
|
|
rtpbasepayload = GST_RTP_BASE_PAYLOAD (parent);
|
|
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
|
|
|
|
if (rtpbasepayload_class->src_event)
|
|
res = rtpbasepayload_class->src_event (rtpbasepayload, event);
|
|
else
|
|
gst_event_unref (event);
|
|
|
|
return res;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_rtp_base_payload_query_default (GstRTPBasePayload * rtpbasepayload,
|
|
GstPad * pad, GstQuery * query)
|
|
{
|
|
gboolean res = FALSE;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstRTPBasePayloadClass *rtpbasepayload_class;
|
|
GstCaps *filter, *caps;
|
|
|
|
gst_query_parse_caps (query, &filter);
|
|
GST_DEBUG_OBJECT (rtpbasepayload, "getting caps with filter %"
|
|
GST_PTR_FORMAT, filter);
|
|
|
|
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
|
|
if (rtpbasepayload_class->get_caps) {
|
|
caps = rtpbasepayload_class->get_caps (rtpbasepayload, pad, filter);
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
res = TRUE;
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
res =
|
|
gst_pad_query_default (pad, GST_OBJECT_CAST (rtpbasepayload), query);
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_base_payload_query (GstPad * pad, GstObject * parent, GstQuery * query)
|
|
{
|
|
GstRTPBasePayload *rtpbasepayload;
|
|
GstRTPBasePayloadClass *rtpbasepayload_class;
|
|
gboolean res = FALSE;
|
|
|
|
rtpbasepayload = GST_RTP_BASE_PAYLOAD (parent);
|
|
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
|
|
|
|
if (rtpbasepayload_class->query)
|
|
res = rtpbasepayload_class->query (rtpbasepayload, pad, query);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_base_payload_chain (GstPad * pad, GstObject * parent,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRTPBasePayload *rtpbasepayload;
|
|
GstRTPBasePayloadClass *rtpbasepayload_class;
|
|
GstFlowReturn ret;
|
|
|
|
rtpbasepayload = GST_RTP_BASE_PAYLOAD (parent);
|
|
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
|
|
|
|
if (!rtpbasepayload_class->handle_buffer)
|
|
goto no_function;
|
|
|
|
if (!rtpbasepayload->priv->negotiated)
|
|
goto not_negotiated;
|
|
|
|
if (rtpbasepayload->priv->source_info) {
|
|
/* Save a copy of meta (instead of taking an extra reference before
|
|
* handle_buffer) to make the meta available when allocating a output
|
|
* buffer. */
|
|
rtpbasepayload->priv->input_meta_buffer = gst_buffer_new ();
|
|
gst_buffer_copy_into (rtpbasepayload->priv->input_meta_buffer, buffer,
|
|
GST_BUFFER_COPY_META, 0, -1);
|
|
}
|
|
|
|
if (gst_pad_check_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload))) {
|
|
if (!gst_rtp_base_payload_negotiate (rtpbasepayload)) {
|
|
gst_pad_mark_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload));
|
|
if (GST_PAD_IS_FLUSHING (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload))) {
|
|
goto flushing;
|
|
} else {
|
|
goto negotiate_failed;
|
|
}
|
|
}
|
|
}
|
|
|
|
ret = rtpbasepayload_class->handle_buffer (rtpbasepayload, buffer);
|
|
|
|
gst_buffer_replace (&rtpbasepayload->priv->input_meta_buffer, NULL);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
no_function:
|
|
{
|
|
GST_ELEMENT_ERROR (rtpbasepayload, STREAM, NOT_IMPLEMENTED, (NULL),
|
|
("subclass did not implement handle_buffer function"));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
not_negotiated:
|
|
{
|
|
GST_ELEMENT_ERROR (rtpbasepayload, CORE, NEGOTIATION, (NULL),
|
|
("No input format was negotiated, i.e. no caps event was received. "
|
|
"Perhaps you need a parser or typefind element before the payloader"));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
negotiate_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpbasepayload, "Not negotiated");
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpbasepayload, "we are flushing");
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* 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;
|
|
|
|
/* 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);
|
|
}
|
|
|
|
gst_caps_replace (&payload->priv->subclass_srccaps, srccaps);
|
|
gst_caps_unref (srccaps);
|
|
|
|
return gst_rtp_base_payload_negotiate (payload);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_base_payload_negotiate (GstRTPBasePayload * payload)
|
|
{
|
|
GstCaps *templ, *peercaps, *srccaps;
|
|
GstStructure *s, *d;
|
|
gboolean res;
|
|
|
|
payload->priv->caps_max_ptime = DEFAULT_MAX_PTIME;
|
|
payload->ptime = 0;
|
|
|
|
gst_pad_check_reconfigure (payload->srcpad);
|
|
|
|
templ = gst_pad_get_pad_template_caps (payload->srcpad);
|
|
|
|
if (payload->priv->subclass_srccaps) {
|
|
GstCaps *tmp = gst_caps_intersect (payload->priv->subclass_srccaps,
|
|
templ);
|
|
gst_caps_unref (templ);
|
|
templ = tmp;
|
|
}
|
|
|
|
peercaps = gst_pad_peer_query_caps (payload->srcpad, templ);
|
|
|
|
if (peercaps == NULL) {
|
|
/* no peer caps, just add the other properties */
|
|
|
|
srccaps = gst_caps_copy (templ);
|
|
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;
|
|
const GValue *value;
|
|
gboolean have_pt = FALSE;
|
|
gboolean have_ts_offset = FALSE;
|
|
gboolean have_seqnum_offset = FALSE;
|
|
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);
|
|
peercaps = NULL;
|
|
|
|
if (gst_caps_is_empty (temp)) {
|
|
gst_caps_unref (temp);
|
|
gst_caps_unref (templ);
|
|
res = FALSE;
|
|
goto out;
|
|
}
|
|
|
|
/* We prefer the pt, timestamp-offset, seqnum-offset from the
|
|
* property (if set), or any previously configured value over what
|
|
* downstream prefers. Only if downstream can't accept that, or the
|
|
* properties were not set, we fall back to choosing downstream's
|
|
* preferred value
|
|
*
|
|
* For ssrc we prefer any value downstream suggests, otherwise
|
|
* the property value or as a last resort a random value.
|
|
* This difference for ssrc is implemented for retaining backwards
|
|
* compatibility with changing rtpsession's internal-ssrc property.
|
|
*
|
|
* FIXME 2.0: All these properties should go away and be negotiated
|
|
* via caps only!
|
|
*/
|
|
|
|
/* try to use the previously set pt, or the one from the property */
|
|
if (payload->priv->pt_set || gst_pad_has_current_caps (payload->srcpad)) {
|
|
GstCaps *probe_caps = gst_caps_copy (templ);
|
|
GstCaps *intersection;
|
|
|
|
gst_caps_set_simple (probe_caps, "payload", G_TYPE_INT,
|
|
GST_RTP_BASE_PAYLOAD_PT (payload), NULL);
|
|
intersection = gst_caps_intersect (probe_caps, temp);
|
|
|
|
if (!gst_caps_is_empty (intersection)) {
|
|
GST_LOG_OBJECT (payload, "Using selected pt %d",
|
|
GST_RTP_BASE_PAYLOAD_PT (payload));
|
|
have_pt = TRUE;
|
|
gst_caps_unref (temp);
|
|
temp = intersection;
|
|
} else {
|
|
GST_WARNING_OBJECT (payload, "Can't use selected pt %d",
|
|
GST_RTP_BASE_PAYLOAD_PT (payload));
|
|
gst_caps_unref (intersection);
|
|
}
|
|
gst_caps_unref (probe_caps);
|
|
}
|
|
|
|
/* If we got no pt above, select one now */
|
|
if (!have_pt) {
|
|
gint pt;
|
|
|
|
/* get first structure */
|
|
s = gst_caps_get_structure (temp, 0);
|
|
|
|
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_RTP_BASE_PAYLOAD_PT (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);
|
|
}
|
|
}
|
|
s = NULL;
|
|
}
|
|
|
|
/* If we got no ssrc above, select one now */
|
|
/* get first structure */
|
|
s = gst_caps_get_structure (temp, 0);
|
|
|
|
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 but we
|
|
* don't support uint ranges so how likely is it that anybody
|
|
* uses a list of possible ssrcs */
|
|
gst_structure_set (s, "ssrc", G_TYPE_UINT, payload->current_ssrc, NULL);
|
|
GST_LOG_OBJECT (payload, "using internal ssrc %08x",
|
|
payload->current_ssrc);
|
|
}
|
|
s = NULL;
|
|
|
|
/* try to select the previously used timestamp-offset, or the one from the property */
|
|
if (!payload->priv->ts_offset_random
|
|
|| gst_pad_has_current_caps (payload->srcpad)) {
|
|
GstCaps *probe_caps = gst_caps_copy (templ);
|
|
GstCaps *intersection;
|
|
|
|
gst_caps_set_simple (probe_caps, "timestamp-offset", G_TYPE_UINT,
|
|
payload->ts_base, NULL);
|
|
intersection = gst_caps_intersect (probe_caps, temp);
|
|
|
|
if (!gst_caps_is_empty (intersection)) {
|
|
GST_LOG_OBJECT (payload, "Using selected timestamp-offset %u",
|
|
payload->ts_base);
|
|
gst_caps_unref (temp);
|
|
temp = intersection;
|
|
have_ts_offset = TRUE;
|
|
} else {
|
|
GST_WARNING_OBJECT (payload, "Can't use selected timestamp-offset %u",
|
|
payload->ts_base);
|
|
gst_caps_unref (intersection);
|
|
}
|
|
gst_caps_unref (probe_caps);
|
|
}
|
|
|
|
/* If we got no timestamp-offset above, select one now */
|
|
if (!have_ts_offset) {
|
|
/* get first structure */
|
|
s = gst_caps_get_structure (temp, 0);
|
|
|
|
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 but we
|
|
* don't support uint ranges so how likely is it that anybody
|
|
* uses a list of possible timestamp-offsets */
|
|
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);
|
|
}
|
|
s = NULL;
|
|
}
|
|
|
|
/* try to select the previously used seqnum-offset, or the one from the property */
|
|
if (!payload->priv->seqnum_offset_random
|
|
|| gst_pad_has_current_caps (payload->srcpad)) {
|
|
GstCaps *probe_caps = gst_caps_copy (templ);
|
|
GstCaps *intersection;
|
|
|
|
gst_caps_set_simple (probe_caps, "seqnum-offset", G_TYPE_UINT,
|
|
payload->seqnum_base, NULL);
|
|
intersection = gst_caps_intersect (probe_caps, temp);
|
|
|
|
if (!gst_caps_is_empty (intersection)) {
|
|
GST_LOG_OBJECT (payload, "Using selected seqnum-offset %u",
|
|
payload->seqnum_base);
|
|
gst_caps_unref (temp);
|
|
temp = intersection;
|
|
have_seqnum_offset = TRUE;
|
|
} else {
|
|
GST_WARNING_OBJECT (payload, "Can't use selected seqnum-offset %u",
|
|
payload->seqnum_base);
|
|
gst_caps_unref (intersection);
|
|
}
|
|
gst_caps_unref (probe_caps);
|
|
}
|
|
|
|
/* If we got no seqnum-offset above, select one now */
|
|
if (!have_seqnum_offset) {
|
|
/* get first structure */
|
|
s = gst_caps_get_structure (temp, 0);
|
|
|
|
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);
|
|
payload->priv->next_seqnum = payload->seqnum_base;
|
|
payload->seqnum = payload->seqnum_base;
|
|
payload->priv->seqnum_offset_random = FALSE;
|
|
} else {
|
|
/* FIXME, fixate_nearest_uint would be even better but we
|
|
* don't support uint ranges so how likely is it that anybody
|
|
* uses a list of possible seqnum-offsets */
|
|
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);
|
|
}
|
|
|
|
s = NULL;
|
|
}
|
|
|
|
/* now fixate, start by taking the first caps */
|
|
temp = 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->ptime = ptime * GST_MSECOND;
|
|
|
|
/* make the target caps by copying over all the fixed fields, removing the
|
|
* unfixed fields. */
|
|
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);
|
|
}
|
|
|
|
if (payload->priv->sinkcaps != NULL) {
|
|
s = gst_caps_get_structure (payload->priv->sinkcaps, 0);
|
|
if (g_str_has_prefix (gst_structure_get_name (s), "video")) {
|
|
gboolean has_framerate;
|
|
gint num, denom;
|
|
|
|
GST_DEBUG_OBJECT (payload, "video caps: %" GST_PTR_FORMAT,
|
|
payload->priv->sinkcaps);
|
|
|
|
has_framerate = gst_structure_get_fraction (s, "framerate", &num, &denom);
|
|
if (has_framerate && num == 0 && denom == 1) {
|
|
has_framerate =
|
|
gst_structure_get_fraction (s, "max-framerate", &num, &denom);
|
|
}
|
|
|
|
if (has_framerate) {
|
|
gchar str[G_ASCII_DTOSTR_BUF_SIZE];
|
|
gdouble framerate;
|
|
|
|
gst_util_fraction_to_double (num, denom, &framerate);
|
|
g_ascii_dtostr (str, G_ASCII_DTOSTR_BUF_SIZE, framerate);
|
|
d = gst_caps_get_structure (srccaps, 0);
|
|
gst_structure_set (d, "a-framerate", G_TYPE_STRING, str, NULL);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (payload, "with video caps: %" GST_PTR_FORMAT, srccaps);
|
|
}
|
|
}
|
|
|
|
update_max_ptime (payload);
|
|
|
|
|
|
if (payload->priv->twcc_ext_id > 0) {
|
|
/* TODO: put this as a separate utility-function for RTP extensions */
|
|
gchar *name = g_strdup_printf ("extmap-%u", payload->priv->twcc_ext_id);
|
|
gst_caps_set_simple (srccaps, name, G_TYPE_STRING,
|
|
"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
|
|
NULL);
|
|
g_free (name);
|
|
}
|
|
|
|
res = gst_pad_set_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), srccaps);
|
|
gst_caps_unref (srccaps);
|
|
gst_caps_unref (templ);
|
|
|
|
out:
|
|
|
|
if (!res)
|
|
gst_pad_mark_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
|
|
|
|
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 dts;
|
|
GstClockTime pts;
|
|
guint64 offset;
|
|
guint32 rtptime;
|
|
guint8 twcc_ext_id;
|
|
} HeaderData;
|
|
|
|
static gboolean
|
|
find_timestamp (GstBuffer ** buffer, guint idx, gpointer user_data)
|
|
{
|
|
HeaderData *data = user_data;
|
|
data->dts = GST_BUFFER_DTS (*buffer);
|
|
data->pts = GST_BUFFER_PTS (*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->pts != -1)
|
|
return FALSE;
|
|
else
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
_set_twcc_seq (GstRTPBuffer * rtp, guint16 seq, guint8 ext_id)
|
|
{
|
|
guint16 data;
|
|
if (ext_id == 0 || ext_id > 14)
|
|
return;
|
|
GST_WRITE_UINT16_BE (&data, seq);
|
|
gst_rtp_buffer_add_extension_onebyte_header (rtp, ext_id, &data, 2);
|
|
}
|
|
|
|
static gboolean
|
|
set_headers (GstBuffer ** buffer, guint idx, gpointer user_data)
|
|
{
|
|
HeaderData *data = user_data;
|
|
GstRTPBuffer rtp = { NULL, };
|
|
|
|
if (!gst_rtp_buffer_map (*buffer, GST_MAP_WRITE, &rtp))
|
|
goto map_failed;
|
|
|
|
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);
|
|
_set_twcc_seq (&rtp, data->seqnum, data->twcc_ext_id);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
/* increment the seqnum for each buffer */
|
|
data->seqnum++;
|
|
|
|
return TRUE;
|
|
/* ERRORS */
|
|
map_failed:
|
|
{
|
|
GST_ERROR ("failed to map buffer %p", *buffer);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
foreach_metadata_drop (GstBuffer * buffer, GstMeta ** meta, gpointer user_data)
|
|
{
|
|
GType drop_api_type = (GType) user_data;
|
|
const GstMetaInfo *info = (*meta)->info;
|
|
|
|
if (info->api == drop_api_type)
|
|
*meta = NULL;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
filter_meta (GstBuffer ** buffer, guint idx, gpointer user_data)
|
|
{
|
|
return gst_buffer_foreach_meta (*buffer, foreach_metadata_drop,
|
|
(gpointer) GST_RTP_SOURCE_META_API_TYPE);
|
|
}
|
|
|
|
/* 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;
|
|
data.twcc_ext_id = priv->twcc_ext_id;
|
|
|
|
/* find the first buffer with a timestamp */
|
|
if (is_list) {
|
|
data.dts = -1;
|
|
data.pts = -1;
|
|
data.offset = GST_BUFFER_OFFSET_NONE;
|
|
gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), find_timestamp, &data);
|
|
} else {
|
|
data.dts = GST_BUFFER_DTS (GST_BUFFER_CAST (obj));
|
|
data.pts = GST_BUFFER_PTS (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) {
|
|
/* generate perfect RTP time by adding together the base timestamp, the
|
|
* running time of the first buffer and difference between the offset of the
|
|
* first buffer and the offset of the current buffer. */
|
|
guint64 offset = data.offset - priv->base_offset;
|
|
data.rtptime = payload->ts_base + priv->base_rtime_hz + offset;
|
|
|
|
GST_LOG_OBJECT (payload,
|
|
"Using offset %" G_GUINT64_FORMAT " for RTP timestamp", data.offset);
|
|
|
|
/* store buffer's running time */
|
|
GST_LOG_OBJECT (payload,
|
|
"setting running-time to %" G_GUINT64_FORMAT,
|
|
data.offset - priv->base_offset);
|
|
priv->running_time = priv->base_rtime + data.offset - priv->base_offset;
|
|
} else if (GST_CLOCK_TIME_IS_VALID (data.pts)) {
|
|
guint64 rtime_ns;
|
|
guint64 rtime_hz;
|
|
|
|
/* no offset, use the gstreamer pts */
|
|
if (priv->onvif_no_rate_control || !priv->scale_rtptime)
|
|
rtime_ns = gst_segment_to_stream_time (&payload->segment,
|
|
GST_FORMAT_TIME, data.pts);
|
|
else
|
|
rtime_ns =
|
|
gst_segment_to_running_time (&payload->segment, GST_FORMAT_TIME,
|
|
data.pts);
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (rtime_ns)) {
|
|
GST_LOG_OBJECT (payload, "Clipped pts, using base RTP timestamp");
|
|
rtime_hz = 0;
|
|
} else {
|
|
GST_LOG_OBJECT (payload,
|
|
"Using running_time %" GST_TIME_FORMAT " for RTP timestamp",
|
|
GST_TIME_ARGS (rtime_ns));
|
|
rtime_hz =
|
|
gst_util_uint64_scale_int (rtime_ns, payload->clock_rate, GST_SECOND);
|
|
priv->base_offset = data.offset;
|
|
priv->base_rtime_hz = rtime_hz;
|
|
}
|
|
|
|
/* add running_time in clock-rate units to the base timestamp */
|
|
data.rtptime = payload->ts_base + rtime_hz;
|
|
|
|
/* store buffer's running time */
|
|
if (priv->perfect_rtptime) {
|
|
GST_LOG_OBJECT (payload,
|
|
"setting running-time to %" G_GUINT64_FORMAT, rtime_hz);
|
|
priv->running_time = rtime_hz;
|
|
} else {
|
|
GST_LOG_OBJECT (payload,
|
|
"setting running-time to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (rtime_ns));
|
|
priv->running_time = rtime_ns;
|
|
}
|
|
} 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 */
|
|
/* remove unwanted meta */
|
|
if (is_list) {
|
|
gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), set_headers, &data);
|
|
gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), filter_meta, NULL);
|
|
/* sequence number has increased more if this was a buffer list */
|
|
payload->seqnum = data.seqnum - 1;
|
|
} else {
|
|
GstBuffer *buf = GST_BUFFER_CAST (obj);
|
|
set_headers (&buf, 0, &data);
|
|
filter_meta (&buf, 0, NULL);
|
|
}
|
|
|
|
priv->next_seqnum = data.seqnum;
|
|
payload->timestamp = data.rtptime;
|
|
|
|
GST_LOG_OBJECT (payload, "Preparing to push %s with size %"
|
|
G_GSIZE_FORMAT ", seq=%d, rtptime=%u, pts %" GST_TIME_FORMAT,
|
|
(is_list) ? "list" : "packet",
|
|
(is_list) ? gst_buffer_list_length (GST_BUFFER_LIST_CAST (obj)) :
|
|
gst_buffer_get_size (GST_BUFFER (obj)),
|
|
payload->seqnum, data.rtptime, GST_TIME_ARGS (data.pts));
|
|
|
|
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.
|
|
*/
|
|
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)) {
|
|
if (G_UNLIKELY (payload->priv->pending_segment)) {
|
|
gst_pad_push_event (payload->srcpad, payload->priv->pending_segment);
|
|
payload->priv->pending_segment = FALSE;
|
|
payload->priv->delay_segment = FALSE;
|
|
}
|
|
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)) {
|
|
if (G_UNLIKELY (payload->priv->pending_segment)) {
|
|
gst_pad_push_event (payload->srcpad, payload->priv->pending_segment);
|
|
payload->priv->pending_segment = FALSE;
|
|
payload->priv->delay_segment = FALSE;
|
|
}
|
|
res = gst_pad_push (payload->srcpad, buffer);
|
|
} else {
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_payload_allocate_output_buffer:
|
|
* @payload: a #GstRTPBasePayload
|
|
* @payload_len: the length of the payload
|
|
* @pad_len: the amount of padding
|
|
* @csrc_count: the minimum number of CSRC entries
|
|
*
|
|
* Allocate a new #GstBuffer with enough data to hold an RTP packet with
|
|
* minimum @csrc_count CSRCs, a payload length of @payload_len and padding of
|
|
* @pad_len. If @payload has #GstRTPBasePayload:source-info %TRUE additional
|
|
* CSRCs may be allocated and filled with RTP source information.
|
|
*
|
|
* Returns: A newly allocated buffer that can hold an RTP packet with given
|
|
* parameters.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
GstBuffer *
|
|
gst_rtp_base_payload_allocate_output_buffer (GstRTPBasePayload * payload,
|
|
guint payload_len, guint8 pad_len, guint8 csrc_count)
|
|
{
|
|
GstBuffer *buffer = NULL;
|
|
|
|
if (payload->priv->input_meta_buffer != NULL) {
|
|
GstRTPSourceMeta *meta =
|
|
gst_buffer_get_rtp_source_meta (payload->priv->input_meta_buffer);
|
|
if (meta != NULL) {
|
|
guint total_csrc_count, idx, i;
|
|
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
|
|
|
|
total_csrc_count = csrc_count + meta->csrc_count +
|
|
(meta->ssrc_valid ? 1 : 0);
|
|
total_csrc_count = MIN (total_csrc_count, 15);
|
|
buffer = gst_rtp_buffer_new_allocate (payload_len, pad_len,
|
|
total_csrc_count);
|
|
|
|
gst_rtp_buffer_map (buffer, GST_MAP_READWRITE, &rtp);
|
|
|
|
/* Skip CSRC fields requested by derived class and fill CSRCs from meta.
|
|
* Finally append the SSRC as a new CSRC. */
|
|
idx = csrc_count;
|
|
for (i = 0; i < meta->csrc_count && idx < 15; i++, idx++)
|
|
gst_rtp_buffer_set_csrc (&rtp, idx, meta->csrc[i]);
|
|
if (meta->ssrc_valid && idx < 15)
|
|
gst_rtp_buffer_set_csrc (&rtp, idx, meta->ssrc);
|
|
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
}
|
|
}
|
|
|
|
if (buffer == NULL)
|
|
buffer = gst_rtp_buffer_new_allocate (payload_len, pad_len, csrc_count);
|
|
|
|
return buffer;
|
|
}
|
|
|
|
static GstStructure *
|
|
gst_rtp_base_payload_create_stats (GstRTPBasePayload * rtpbasepayload)
|
|
{
|
|
GstRTPBasePayloadPrivate *priv;
|
|
GstStructure *s;
|
|
|
|
priv = rtpbasepayload->priv;
|
|
|
|
s = gst_structure_new ("application/x-rtp-payload-stats",
|
|
"clock-rate", G_TYPE_UINT, (guint) rtpbasepayload->clock_rate,
|
|
"running-time", G_TYPE_UINT64, priv->running_time,
|
|
"seqnum", G_TYPE_UINT, (guint) rtpbasepayload->seqnum,
|
|
"timestamp", G_TYPE_UINT, (guint) rtpbasepayload->timestamp,
|
|
"ssrc", G_TYPE_UINT, rtpbasepayload->current_ssrc,
|
|
"pt", G_TYPE_UINT, rtpbasepayload->pt,
|
|
"seqnum-offset", G_TYPE_UINT, (guint) rtpbasepayload->seqnum_base,
|
|
"timestamp-offset", G_TYPE_UINT, (guint) rtpbasepayload->ts_base, NULL);
|
|
|
|
return s;
|
|
}
|
|
|
|
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);
|
|
priv->pt_set = TRUE;
|
|
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->ptime_multiple = g_value_get_int64 (value);
|
|
break;
|
|
case PROP_SOURCE_INFO:
|
|
gst_rtp_base_payload_set_source_info_enabled (rtpbasepayload,
|
|
g_value_get_boolean (value));
|
|
break;
|
|
case PROP_ONVIF_NO_RATE_CONTROL:
|
|
priv->onvif_no_rate_control = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_TWCC_EXT_ID:
|
|
priv->twcc_ext_id = g_value_get_uint (value);
|
|
break;
|
|
case PROP_SCALE_RTPTIME:
|
|
priv->scale_rtptime = g_value_get_boolean (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->ptime_multiple);
|
|
break;
|
|
case PROP_STATS:
|
|
g_value_take_boxed (value,
|
|
gst_rtp_base_payload_create_stats (rtpbasepayload));
|
|
break;
|
|
case PROP_SOURCE_INFO:
|
|
g_value_set_boolean (value,
|
|
gst_rtp_base_payload_is_source_info_enabled (rtpbasepayload));
|
|
break;
|
|
case PROP_ONVIF_NO_RATE_CONTROL:
|
|
g_value_set_boolean (value, priv->onvif_no_rate_control);
|
|
break;
|
|
case PROP_TWCC_EXT_ID:
|
|
g_value_set_uint (value, priv->twcc_ext_id);
|
|
break;
|
|
case PROP_SCALE_RTPTIME:
|
|
g_value_set_boolean (value, priv->scale_rtptime);
|
|
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);
|
|
rtpbasepayload->priv->delay_segment = TRUE;
|
|
gst_event_replace (&rtpbasepayload->priv->pending_segment, NULL);
|
|
|
|
if (priv->seqnum_offset_random)
|
|
rtpbasepayload->seqnum_base = g_random_int_range (0, G_MAXINT16);
|
|
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;
|
|
priv->running_time = DEFAULT_RUNNING_TIME;
|
|
g_atomic_int_set (&rtpbasepayload->priv->notified_first_timestamp, 1);
|
|
priv->base_offset = GST_BUFFER_OFFSET_NONE;
|
|
priv->negotiated = FALSE;
|
|
gst_caps_replace (&rtpbasepayload->priv->subclass_srccaps, NULL);
|
|
gst_caps_replace (&rtpbasepayload->priv->sinkcaps, NULL);
|
|
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_PAUSED_TO_READY:
|
|
gst_event_replace (&rtpbasepayload->priv->pending_segment, NULL);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_payload_set_source_info_enabled:
|
|
* @payload: a #GstRTPBasePayload
|
|
* @enable: whether to add contributing sources to RTP packets
|
|
*
|
|
* Enable or disable adding contributing sources to RTP packets from
|
|
* #GstRTPSourceMeta.
|
|
*
|
|
* Since: 1.16
|
|
**/
|
|
void
|
|
gst_rtp_base_payload_set_source_info_enabled (GstRTPBasePayload * payload,
|
|
gboolean enable)
|
|
{
|
|
payload->priv->source_info = enable;
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_payload_is_source_info_enabled:
|
|
* @payload: a #GstRTPBasePayload
|
|
*
|
|
* Queries whether the payloader will add contributing sources (CSRCs) to the
|
|
* RTP header from #GstRTPSourceMeta.
|
|
*
|
|
* Returns: %TRUE if source-info is enabled.
|
|
*
|
|
* Since: 1.16
|
|
**/
|
|
gboolean
|
|
gst_rtp_base_payload_is_source_info_enabled (GstRTPBasePayload * payload)
|
|
{
|
|
return payload->priv->source_info;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_rtp_base_payload_get_source_count:
|
|
* @payload: a #GstRTPBasePayload
|
|
* @buffer: (transfer none): a #GstBuffer, typically the buffer to payload
|
|
*
|
|
* Count the total number of RTP sources found in the meta of @buffer, which
|
|
* will be automically added by gst_rtp_base_payload_allocate_output_buffer().
|
|
* If #GstRTPBasePayload:source-info is %FALSE the count will be 0.
|
|
*
|
|
* Returns: The number of sources.
|
|
*
|
|
* Since: 1.16
|
|
**/
|
|
guint
|
|
gst_rtp_base_payload_get_source_count (GstRTPBasePayload * payload,
|
|
GstBuffer * buffer)
|
|
{
|
|
guint count = 0;
|
|
|
|
g_return_val_if_fail (buffer != NULL, 0);
|
|
|
|
if (gst_rtp_base_payload_is_source_info_enabled (payload)) {
|
|
GstRTPSourceMeta *meta = gst_buffer_get_rtp_source_meta (buffer);
|
|
if (meta != NULL)
|
|
count = gst_rtp_source_meta_get_source_count (meta);
|
|
}
|
|
|
|
return count;
|
|
}
|