/* GStreamer * Copyright (C) <2005> Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more */ /** * SECTION:gstrtpbasepayload * @title: GstRTPBasePayload * @short_description: Base class for RTP payloader * * Provides a base class for RTP payloaders */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include "gstrtpbasepayload.h" #include "gstrtpmeta.h" #include "gstrtphdrext.h" GST_DEBUG_CATEGORY_STATIC (rtpbasepayload_debug); #define GST_CAT_DEFAULT (rtpbasepayload_debug) struct _GstRTPBasePayloadPrivate { gboolean ts_offset_random; gboolean seqnum_offset_random; gboolean ssrc_random; guint16 next_seqnum; gboolean perfect_rtptime; gint notified_first_timestamp; gboolean pt_set; gboolean source_info; GstBuffer *input_meta_buffer; guint64 base_offset; gint64 base_rtime; guint64 base_rtime_hz; guint64 running_time; gboolean scale_rtptime; gboolean auto_hdr_ext; gint64 prop_max_ptime; gint64 caps_max_ptime; gboolean onvif_no_rate_control; gboolean negotiated; /* We need to know whether negotiate was called in order to decide * whether we should store the input buffer as input meta in case * negotiate() gets called from the subclass' handle_buffer() implementation, * as negotiate() is where we instantiate header extensions. */ gboolean negotiate_called; gboolean delay_segment; GstEvent *pending_segment; GstCaps *subclass_srccaps; GstCaps *sinkcaps; /* array of GstRTPHeaderExtension's * */ GPtrArray *header_exts; }; /* RTPBasePayload signals and args */ enum { SIGNAL_0, SIGNAL_REQUEST_EXTENSION, SIGNAL_ADD_EXTENSION, SIGNAL_CLEAR_EXTENSIONS, LAST_SIGNAL }; static guint gst_rtp_base_payload_signals[LAST_SIGNAL] = { 0 }; /* FIXME 0.11, a better default is the Ethernet MTU of * 1500 - sizeof(headers) as pointed out by marcelm in IRC: * So an Ethernet MTU of 1500, minus 60 for the max IP, minus 8 for UDP, gives * 1432 bytes or so. And that should be adjusted downward further for other * encapsulations like PPPoE, so 1400 at most. */ #define DEFAULT_MTU 1400 #define DEFAULT_PT 96 #define DEFAULT_SSRC -1 #define DEFAULT_TIMESTAMP_OFFSET -1 #define DEFAULT_SEQNUM_OFFSET -1 #define DEFAULT_MAX_PTIME -1 #define DEFAULT_MIN_PTIME 0 #define DEFAULT_PERFECT_RTPTIME TRUE #define DEFAULT_PTIME_MULTIPLE 0 #define DEFAULT_RUNNING_TIME GST_CLOCK_TIME_NONE #define DEFAULT_SOURCE_INFO FALSE #define DEFAULT_ONVIF_NO_RATE_CONTROL FALSE #define DEFAULT_SCALE_RTPTIME TRUE #define DEFAULT_AUTO_HEADER_EXTENSION TRUE #define RTP_HEADER_EXT_ONE_BYTE_MAX_SIZE 16 #define RTP_HEADER_EXT_TWO_BYTE_MAX_SIZE 255 #define RTP_HEADER_EXT_ONE_BYTE_MAX_ID 14 #define RTP_HEADER_EXT_TWO_BYTE_MAX_ID 255 enum { PROP_0, PROP_MTU, PROP_PT, PROP_SSRC, PROP_TIMESTAMP_OFFSET, PROP_SEQNUM_OFFSET, PROP_MAX_PTIME, PROP_MIN_PTIME, PROP_TIMESTAMP, PROP_SEQNUM, PROP_PERFECT_RTPTIME, PROP_PTIME_MULTIPLE, PROP_STATS, PROP_SOURCE_INFO, PROP_ONVIF_NO_RATE_CONTROL, PROP_SCALE_RTPTIME, PROP_AUTO_HEADER_EXTENSION, PROP_EXTENSIONS, PROP_LAST }; static GParamSpec *gst_rtp_base_payload_extensions_pspec; static void gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass); static void gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload, gpointer g_class); static void gst_rtp_base_payload_finalize (GObject * object); static GstCaps *gst_rtp_base_payload_getcaps_default (GstRTPBasePayload * rtpbasepayload, GstPad * pad, GstCaps * filter); static gboolean gst_rtp_base_payload_sink_event_default (GstRTPBasePayload * rtpbasepayload, GstEvent * event); static gboolean gst_rtp_base_payload_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_rtp_base_payload_src_event_default (GstRTPBasePayload * rtpbasepayload, GstEvent * event); static gboolean gst_rtp_base_payload_src_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_rtp_base_payload_query_default (GstRTPBasePayload * rtpbasepayload, GstPad * pad, GstQuery * query); static gboolean gst_rtp_base_payload_query (GstPad * pad, GstObject * parent, GstQuery * query); static GstFlowReturn gst_rtp_base_payload_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer); static void gst_rtp_base_payload_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_base_payload_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstStateChangeReturn gst_rtp_base_payload_change_state (GstElement * element, GstStateChange transition); static gboolean gst_rtp_base_payload_negotiate (GstRTPBasePayload * payload); static void gst_rtp_base_payload_add_extension (GstRTPBasePayload * payload, GstRTPHeaderExtension * ext); static void gst_rtp_base_payload_clear_extensions (GstRTPBasePayload * payload); static void gst_rtp_base_payload_get_extensions (GstRTPBasePayload * payload, GValue * out_value); static GstElementClass *parent_class = NULL; static gint private_offset = 0; GType gst_rtp_base_payload_get_type (void) { static GType rtpbasepayload_type = 0; if (g_once_init_enter ((gsize *) & rtpbasepayload_type)) { static const GTypeInfo rtpbasepayload_info = { sizeof (GstRTPBasePayloadClass), NULL, NULL, (GClassInitFunc) gst_rtp_base_payload_class_init, NULL, NULL, sizeof (GstRTPBasePayload), 0, (GInstanceInitFunc) gst_rtp_base_payload_init, }; GType _type; _type = g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBasePayload", &rtpbasepayload_info, G_TYPE_FLAG_ABSTRACT); private_offset = g_type_add_instance_private (_type, sizeof (GstRTPBasePayloadPrivate)); g_once_init_leave ((gsize *) & rtpbasepayload_type, _type); } return rtpbasepayload_type; } static inline GstRTPBasePayloadPrivate * gst_rtp_base_payload_get_instance_private (GstRTPBasePayload * self) { return (G_STRUCT_MEMBER_P (self, private_offset)); } static GstRTPHeaderExtension * gst_rtp_base_payload_request_extension_default (GstRTPBasePayload * payload, guint ext_id, const gchar * uri) { GstRTPHeaderExtension *ext = NULL; if (!payload->priv->auto_hdr_ext) return NULL; ext = gst_rtp_header_extension_create_from_uri (uri); if (ext) { GST_DEBUG_OBJECT (payload, "Automatically enabled extension %s for uri \'%s\'", GST_ELEMENT_NAME (ext), uri); gst_rtp_header_extension_set_id (ext, ext_id); } else { GST_DEBUG_OBJECT (payload, "Didn't find any extension implementing uri \'%s\'", uri); } return ext; } static gboolean extension_accumulator (GSignalInvocationHint * ihint, GValue * return_accu, const GValue * handler_return, gpointer data) { gpointer ext; /* Call default handler if user callback didn't create the extension */ ext = g_value_get_object (handler_return); if (!ext) return TRUE; g_value_set_object (return_accu, ext); return FALSE; } static void gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; if (private_offset != 0) g_type_class_adjust_private_offset (klass, &private_offset); parent_class = g_type_class_peek_parent (klass); gobject_class->finalize = gst_rtp_base_payload_finalize; gobject_class->set_property = gst_rtp_base_payload_set_property; gobject_class->get_property = gst_rtp_base_payload_get_property; g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MTU, g_param_spec_uint ("mtu", "MTU", "Maximum size of one packet", 28, G_MAXUINT, DEFAULT_MTU, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT, g_param_spec_uint ("pt", "payload type", "The payload type of the packets", 0, 0x7f, DEFAULT_PT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC, g_param_spec_uint ("ssrc", "SSRC", "The SSRC of the packets (default == random)", 0, G_MAXUINT32, DEFAULT_SSRC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP_OFFSET, g_param_spec_uint ("timestamp-offset", "Timestamp Offset", "Offset to add to all outgoing timestamps (default = random)", 0, G_MAXUINT32, DEFAULT_TIMESTAMP_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET, g_param_spec_int ("seqnum-offset", "Sequence number Offset", "Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXUINT16, DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_PTIME, g_param_spec_int64 ("max-ptime", "Max packet time", "Maximum duration of the packet data in ns (-1 = unlimited up to MTU)", -1, G_MAXINT64, DEFAULT_MAX_PTIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTPBasePayload:min-ptime: * * Minimum duration of the packet data in ns (can't go above MTU) **/ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MIN_PTIME, g_param_spec_int64 ("min-ptime", "Min packet time", "Minimum duration of the packet data in ns (can't go above MTU)", 0, G_MAXINT64, DEFAULT_MIN_PTIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP, g_param_spec_uint ("timestamp", "Timestamp", "The RTP timestamp of the last processed packet", 0, G_MAXUINT32, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM, g_param_spec_uint ("seqnum", "Sequence number", "The RTP sequence number of the last processed packet", 0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); /** * GstRTPBasePayload:perfect-rtptime: * * Try to use the offset fields to generate perfect RTP timestamps. When this * option is disabled, RTP timestamps are generated from GST_BUFFER_PTS of * each payloaded buffer. The PTSes of buffers may not necessarily increment * with the amount of data in each input buffer, consider e.g. the case where * the buffer arrives from a network which means that the PTS is unrelated to * the amount of data. Because the RTP timestamps are generated from * GST_BUFFER_PTS this can result in RTP timestamps that also don't increment * with the amount of data in the payloaded packet. To circumvent this it is * possible to set the perfect rtptime option enabled. When this option is * enabled the payloader will increment the RTP timestamps based on * GST_BUFFER_OFFSET which relates to the amount of data in each packet * rather than the GST_BUFFER_PTS of each buffer and therefore the RTP * timestamps will more closely correlate with the amount of data in each * buffer. Currently GstRTPBasePayload is limited to handling perfect RTP * timestamps for audio streams. */ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PERFECT_RTPTIME, g_param_spec_boolean ("perfect-rtptime", "Perfect RTP Time", "Generate perfect RTP timestamps when possible", DEFAULT_PERFECT_RTPTIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTPBasePayload:ptime-multiple: * * Force buffers to be multiples of this duration in ns (0 disables) **/ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PTIME_MULTIPLE, g_param_spec_int64 ("ptime-multiple", "Packet time multiple", "Force buffers to be multiples of this duration in ns (0 disables)", 0, G_MAXINT64, DEFAULT_PTIME_MULTIPLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTPBasePayload:stats: * * Various payloader statistics retrieved atomically (and are therefore * synchroized with each other), these can be used e.g. to generate an * RTP-Info header. This property return a GstStructure named * application/x-rtp-payload-stats containing the following fields relating to * the last processed buffer and current state of the stream being payloaded: * * * `clock-rate` :#G_TYPE_UINT, clock-rate of the stream * * `running-time` :#G_TYPE_UINT64, running time * * `seqnum` :#G_TYPE_UINT, sequence number, same as #GstRTPBasePayload:seqnum * * `timestamp` :#G_TYPE_UINT, RTP timestamp, same as #GstRTPBasePayload:timestamp * * `ssrc` :#G_TYPE_UINT, The SSRC in use * * `pt` :#G_TYPE_UINT, The Payload type in use, same as #GstRTPBasePayload:pt * * `seqnum-offset` :#G_TYPE_UINT, The current offset added to the seqnum * * `timestamp-offset` :#G_TYPE_UINT, The current offset added to the timestamp **/ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_STATS, g_param_spec_boxed ("stats", "Statistics", "Various statistics", GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); /** * GstRTPBasePayload:source-info: * * Enable writing the CSRC field in allocated RTP header based on RTP source * information found in the input buffer's #GstRTPSourceMeta. * * Since: 1.16 **/ g_object_class_install_property (gobject_class, PROP_SOURCE_INFO, g_param_spec_boolean ("source-info", "RTP source information", "Write CSRC based on buffer meta RTP source information", DEFAULT_SOURCE_INFO, G_PARAM_READWRITE)); /** * GstRTPBasePayload:onvif-no-rate-control: * * Make the payloader timestamp packets according to the Rate-Control=no * behaviour specified in the ONVIF replay spec. * * Since: 1.16 */ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ONVIF_NO_RATE_CONTROL, g_param_spec_boolean ("onvif-no-rate-control", "ONVIF no rate control", "Enable ONVIF Rate-Control=no timestamping mode", DEFAULT_ONVIF_NO_RATE_CONTROL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTPBasePayload:scale-rtptime: * * Make the RTP packets' timestamps be scaled with the segment's rate * (corresponding to RTSP speed parameter). Disabling this property means * the timestamps will not be affected by the set delivery speed (RTSP speed). * * Example: A server wants to allow streaming a recorded video in double * speed but still have the timestamps correspond to the position in the * video. This is achieved by the client setting RTSP Speed to 2 while the * server has this property disabled. * * Since: 1.18 */ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SCALE_RTPTIME, g_param_spec_boolean ("scale-rtptime", "Scale RTP time", "Whether the RTP timestamp should be scaled with the rate (speed)", DEFAULT_SCALE_RTPTIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTPBasePayload:auto-header-extension: * * If enabled, the payloader will automatically try to enable all the * RTP header extensions provided in the src caps, saving the application * the need to handle these extensions manually using the * GstRTPBasePayload::request-extension: signal. * * Since: 1.20 */ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_AUTO_HEADER_EXTENSION, g_param_spec_boolean ("auto-header-extension", "Automatic RTP header extension", "Whether RTP header extensions should be automatically enabled, if an implementation is available", DEFAULT_AUTO_HEADER_EXTENSION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_rtp_base_payload_extensions_pspec = gst_param_spec_array ("extensions", "RTP header extensions", "A list of already enabled RTP header extensions", g_param_spec_object ("extension", "RTP header extension", "An already enabled RTP extension", GST_TYPE_RTP_HEADER_EXTENSION, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS), G_PARAM_READABLE | G_PARAM_STATIC_STRINGS); /** * GstRTPBasePayload:extensions: * * A list of already enabled RTP header extensions. This may be useful for finding * out which extensions are already enabled (with add-extension signal) and picking a non-conflicting * ID for a new extension that needs to be added on top of the existing ones. * * Note that the value returned by reading this property is not dynamically updated when the set of * enabled extensions changes by any of existing action signals. Rather, it represents the current state * at the time the property is read. * * Dynamic updates of this property can be received by subscribing to its corresponding "notify" signal, i.e. * "notify::extensions". * * Since: 1.24 */ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_EXTENSIONS, gst_rtp_base_payload_extensions_pspec); /** * GstRTPBasePayload::add-extension: * @object: the #GstRTPBasePayload * @ext: (transfer full): the #GstRTPHeaderExtension * * Add @ext as an extension for writing part of an RTP header extension onto * outgoing RTP packets. * * Since: 1.20 */ gst_rtp_base_payload_signals[SIGNAL_ADD_EXTENSION] = g_signal_new_class_handler ("add-extension", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_rtp_base_payload_add_extension), NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_RTP_HEADER_EXTENSION); /** * GstRTPBasePayload::request-extension: * @object: the #GstRTPBasePayload * @ext_id: the extension id being requested * @ext_uri: the extension URI being requested * * The returned @ext must be configured with the correct @ext_id and with the * necessary attributes as required by the extension implementation. * * Returns: (transfer full) (nullable): the #GstRTPHeaderExtension for @ext_id, or %NULL * * Since: 1.20 */ gst_rtp_base_payload_signals[SIGNAL_REQUEST_EXTENSION] = g_signal_new_class_handler ("request-extension", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_CALLBACK (gst_rtp_base_payload_request_extension_default), extension_accumulator, NULL, NULL, GST_TYPE_RTP_HEADER_EXTENSION, 2, G_TYPE_UINT, G_TYPE_STRING); /** * GstRTPBasePayload::clear-extensions: * @object: the #GstRTPBasePayload * * Clear all RTP header extensions used by this payloader. * * Since: 1.20 */ gst_rtp_base_payload_signals[SIGNAL_CLEAR_EXTENSIONS] = g_signal_new_class_handler ("clear-extensions", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_rtp_base_payload_clear_extensions), NULL, NULL, NULL, G_TYPE_NONE, 0); gstelement_class->change_state = gst_rtp_base_payload_change_state; klass->get_caps = gst_rtp_base_payload_getcaps_default; klass->sink_event = gst_rtp_base_payload_sink_event_default; klass->src_event = gst_rtp_base_payload_src_event_default; klass->query = gst_rtp_base_payload_query_default; GST_DEBUG_CATEGORY_INIT (rtpbasepayload_debug, "rtpbasepayload", 0, "Base class for RTP Payloaders"); } static void gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload, gpointer g_class) { GstPadTemplate *templ; GstRTPBasePayloadPrivate *priv; rtpbasepayload->priv = priv = gst_rtp_base_payload_get_instance_private (rtpbasepayload); templ = gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src"); g_return_if_fail (templ != NULL); rtpbasepayload->srcpad = gst_pad_new_from_template (templ, "src"); gst_pad_set_event_function (rtpbasepayload->srcpad, gst_rtp_base_payload_src_event); gst_element_add_pad (GST_ELEMENT (rtpbasepayload), rtpbasepayload->srcpad); templ = gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink"); g_return_if_fail (templ != NULL); rtpbasepayload->sinkpad = gst_pad_new_from_template (templ, "sink"); gst_pad_set_chain_function (rtpbasepayload->sinkpad, gst_rtp_base_payload_chain); gst_pad_set_event_function (rtpbasepayload->sinkpad, gst_rtp_base_payload_sink_event); gst_pad_set_query_function (rtpbasepayload->sinkpad, gst_rtp_base_payload_query); gst_element_add_pad (GST_ELEMENT (rtpbasepayload), rtpbasepayload->sinkpad); 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->priv->auto_hdr_ext = DEFAULT_AUTO_HEADER_EXTENSION; 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; rtpbasepayload->priv->header_exts = g_ptr_array_new_with_free_func ((GDestroyNotify) gst_object_unref); } 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_ptr_array_unref (rtpbasepayload->priv->header_exts); rtpbasepayload->priv->header_exts = 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; } case GST_EVENT_GAP: { if (G_UNLIKELY (rtpbasepayload->priv->pending_segment)) { gst_pad_push_event (rtpbasepayload->srcpad, rtpbasepayload->priv->pending_segment); rtpbasepayload->priv->pending_segment = FALSE; rtpbasepayload->priv->delay_segment = FALSE; } 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 || rtpbasepayload->priv->header_exts->len > 0 || !rtpbasepayload->priv->negotiate_called) { /* 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_METADATA, 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 (const GstIdStr * fieldname, const GValue * value, GstStructure * dest) { if (gst_value_is_fixed (value)) { gst_structure_id_str_set_value (dest, fieldname, 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; } static gboolean _set_caps (const GstIdStr * fieldname, const GValue * value, GstCaps * caps) { gst_caps_set_value (caps, gst_id_str_as_str (fieldname), value); return TRUE; } /** * gst_rtp_base_payload_set_outcaps_structure: * @payload: a #GstRTPBasePayload * @s: (nullable): a #GstStructure with the caps fields * * Configure the output caps with the optional fields. * * Returns: %TRUE if the caps could be set. * * Since: 1.20 */ gboolean gst_rtp_base_payload_set_outcaps_structure (GstRTPBasePayload * payload, GstStructure * s) { 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 (s && gst_structure_n_fields (s) > 0) { gst_structure_foreach_id_str (s, (GstStructureForeachIdStrFunc) _set_caps, srccaps); 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); } /** * 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, ...) { gboolean result; GstStructure *s = NULL; if (fieldname) { va_list varargs; s = gst_structure_new_empty ("unused"); /* override with custom properties */ va_start (varargs, fieldname); gst_structure_set_valist (s, fieldname, varargs); va_end (varargs); } result = gst_rtp_base_payload_set_outcaps_structure (payload, s); gst_clear_structure (&s); return result; } static void add_and_ref_item (GstRTPHeaderExtension * ext, GPtrArray * ret) { g_ptr_array_add (ret, gst_object_ref (ext)); } static void remove_item_from (GstRTPHeaderExtension * ext, GPtrArray * ret) { g_ptr_array_remove_fast (ret, ext); } static void add_item_to (GstRTPHeaderExtension * ext, GPtrArray * ret) { g_ptr_array_add (ret, ext); } static void add_header_ext_to_caps (GstRTPHeaderExtension * ext, GstCaps * caps) { if (!gst_rtp_header_extension_set_caps_from_attributes (ext, caps)) { GST_WARNING ("Failed to set caps from rtp header extension"); } } static gboolean gst_rtp_base_payload_negotiate (GstRTPBasePayload * payload) { GstCaps *templ, *peercaps, *srccaps; GstStructure *s, *d; gboolean res = TRUE; 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_id_str (s, (GstStructureForeachIdStrFunc) 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); { /* try to find header extension implementations for the list in the * caps */ GstStructure *s = gst_caps_get_structure (srccaps, 0); guint i, j, n_fields = gst_structure_n_fields (s); GPtrArray *header_exts = g_ptr_array_new_with_free_func (gst_object_unref); GPtrArray *to_add = g_ptr_array_new (); GPtrArray *to_remove = g_ptr_array_new (); GST_OBJECT_LOCK (payload); g_ptr_array_foreach (payload->priv->header_exts, (GFunc) add_and_ref_item, header_exts); GST_OBJECT_UNLOCK (payload); for (i = 0; i < n_fields; i++) { const gchar *field_name = gst_structure_nth_field_name (s, i); if (g_str_has_prefix (field_name, "extmap-")) { const GValue *val; const gchar *uri = NULL; gchar *nptr; guint ext_id; GstRTPHeaderExtension *ext = NULL; errno = 0; ext_id = g_ascii_strtoull (&field_name[strlen ("extmap-")], &nptr, 10); if (errno != 0 || (ext_id == 0 && field_name == nptr)) { GST_WARNING_OBJECT (payload, "could not parse id from %s", field_name); res = FALSE; goto ext_out; } val = gst_structure_get_value (s, field_name); if (G_VALUE_HOLDS_STRING (val)) { uri = g_value_get_string (val); } else if (GST_VALUE_HOLDS_ARRAY (val)) { /* the uri is the second value in the array */ const GValue *str = gst_value_array_get_value (val, 1); if (G_VALUE_HOLDS_STRING (str)) { uri = g_value_get_string (str); } } if (!uri) { GST_WARNING_OBJECT (payload, "could not get extmap uri for " "field %s", field_name); res = FALSE; goto ext_out; } /* try to find if this extension mapping already exists */ for (j = 0; j < header_exts->len; j++) { ext = g_ptr_array_index (header_exts, j); if (gst_rtp_header_extension_get_id (ext) == ext_id) { if (g_strcmp0 (uri, gst_rtp_header_extension_get_uri (ext)) == 0) { /* still matching, we're good, set attributes from caps in case * the caps have been updated */ if (!gst_rtp_header_extension_set_attributes_from_caps (ext, srccaps)) { GST_WARNING_OBJECT (payload, "Failed to configure rtp header " "extension %" GST_PTR_FORMAT " attributes from caps %" GST_PTR_FORMAT, ext, srccaps); res = FALSE; goto ext_out; } break; } else { GST_DEBUG_OBJECT (payload, "extension id %u" "was replaced with a different extension uri " "original:\'%s' vs \'%s\'", ext_id, gst_rtp_header_extension_get_uri (ext), uri); g_ptr_array_add (to_remove, ext); ext = NULL; break; } } else { ext = NULL; } } /* if no extension, attempt to request one */ if (!ext) { GST_DEBUG_OBJECT (payload, "requesting extension for id %u" " and uri %s", ext_id, uri); g_signal_emit (payload, gst_rtp_base_payload_signals[SIGNAL_REQUEST_EXTENSION], 0, ext_id, uri, &ext); GST_DEBUG_OBJECT (payload, "request returned extension %p \'%s\' " "for id %u and uri %s", ext, ext ? GST_OBJECT_NAME (ext) : "", ext_id, uri); /* We require caller to set the appropriate extension if it's required */ if (ext && gst_rtp_header_extension_get_id (ext) != ext_id) { g_warning ("\'request-extension\' signal provided an rtp header " "extension for uri \'%s\' that does not match the requested " "extension id %u", uri, ext_id); gst_clear_object (&ext); } if (ext && !gst_rtp_header_extension_set_attributes_from_caps (ext, srccaps)) { GST_WARNING_OBJECT (payload, "Failed to configure rtp header " "extension %" GST_PTR_FORMAT " attributes from caps %" GST_PTR_FORMAT, ext, srccaps); res = FALSE; g_clear_object (&ext); goto ext_out; } if (ext) { g_ptr_array_add (to_add, ext); } } } } GST_OBJECT_LOCK (payload); g_ptr_array_foreach (to_remove, (GFunc) remove_item_from, payload->priv->header_exts); g_ptr_array_foreach (to_add, (GFunc) add_item_to, payload->priv->header_exts); /* let extensions update their internal state from sinkcaps */ if (payload->priv->sinkcaps) { gint i; for (i = 0; i < payload->priv->header_exts->len; i++) { GstRTPHeaderExtension *ext; ext = g_ptr_array_index (payload->priv->header_exts, i); if (!gst_rtp_header_extension_set_non_rtp_sink_caps (ext, payload->priv->sinkcaps)) { GST_WARNING_OBJECT (payload, "Failed to update rtp header extension (%s) from sink caps", GST_OBJECT_NAME (ext)); res = FALSE; GST_OBJECT_UNLOCK (payload); goto ext_out; } } } /* add extension information to srccaps */ g_ptr_array_foreach (payload->priv->header_exts, (GFunc) add_header_ext_to_caps, srccaps); GST_OBJECT_UNLOCK (payload); g_object_notify_by_pspec (G_OBJECT (payload), gst_rtp_base_payload_extensions_pspec); ext_out: g_ptr_array_unref (to_add); g_ptr_array_unref (to_remove); g_ptr_array_unref (header_exts); } GST_DEBUG_OBJECT (payload, "configuring caps %" GST_PTR_FORMAT, srccaps); if (res) res = gst_pad_set_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), srccaps); gst_caps_unref (srccaps); gst_caps_unref (templ); out: payload->priv->negotiate_called = TRUE; 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; } 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 gst_rtp_base_payload_add_extension (GstRTPBasePayload * payload, GstRTPHeaderExtension * ext) { g_return_if_fail (GST_IS_RTP_HEADER_EXTENSION (ext)); g_return_if_fail (gst_rtp_header_extension_get_id (ext) > 0); /* XXX: check for duplicate ids? */ GST_OBJECT_LOCK (payload); g_ptr_array_add (payload->priv->header_exts, gst_object_ref (ext)); gst_pad_mark_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (payload)); GST_OBJECT_UNLOCK (payload); g_object_notify_by_pspec (G_OBJECT (payload), gst_rtp_base_payload_extensions_pspec); } static void gst_rtp_base_payload_clear_extensions (GstRTPBasePayload * payload) { GST_OBJECT_LOCK (payload); g_ptr_array_set_size (payload->priv->header_exts, 0); GST_OBJECT_UNLOCK (payload); g_object_notify_by_pspec (G_OBJECT (payload), gst_rtp_base_payload_extensions_pspec); } static void gst_rtp_base_payload_get_extensions (GstRTPBasePayload * payload, GValue * out_value) { GPtrArray *extensions; guint i; GST_OBJECT_LOCK (payload); extensions = payload->priv->header_exts; for (i = 0; i < extensions->len; ++i) { GValue value = G_VALUE_INIT; g_value_init (&value, GST_TYPE_RTP_HEADER_EXTENSION); g_value_set_object (&value, g_ptr_array_index (extensions, i)); gst_value_array_append_and_take_value (out_value, &value); } GST_OBJECT_UNLOCK (payload); } typedef struct { GstRTPBasePayload *payload; GstRTPHeaderExtensionFlags flags; GstBuffer *output; guint8 *data; gsize allocated_size; gsize written_size; gsize hdr_unit_size; gboolean abort; } HeaderExt; static void determine_header_extension_flags_size (GstRTPHeaderExtension * ext, gpointer user_data) { HeaderExt *hdr = user_data; guint ext_id; gsize max_size; hdr->flags &= gst_rtp_header_extension_get_supported_flags (ext); max_size = gst_rtp_header_extension_get_max_size (ext, hdr->payload->priv->input_meta_buffer); if (max_size > RTP_HEADER_EXT_ONE_BYTE_MAX_SIZE) hdr->flags &= ~GST_RTP_HEADER_EXTENSION_ONE_BYTE; if (max_size > RTP_HEADER_EXT_TWO_BYTE_MAX_SIZE) hdr->flags &= ~GST_RTP_HEADER_EXTENSION_TWO_BYTE; ext_id = gst_rtp_header_extension_get_id (ext); if (ext_id > RTP_HEADER_EXT_ONE_BYTE_MAX_ID) hdr->flags &= ~GST_RTP_HEADER_EXTENSION_ONE_BYTE; if (ext_id > RTP_HEADER_EXT_TWO_BYTE_MAX_ID) hdr->flags &= ~GST_RTP_HEADER_EXTENSION_TWO_BYTE; hdr->allocated_size += max_size; } static void write_header_extension (GstRTPHeaderExtension * ext, gpointer user_data) { HeaderExt *hdr = user_data; gsize remaining = hdr->allocated_size - hdr->written_size - hdr->hdr_unit_size; gsize offset = hdr->written_size + hdr->hdr_unit_size; gssize written; guint ext_id; if (hdr->abort) return; written = gst_rtp_header_extension_write (ext, hdr->payload->priv->input_meta_buffer, hdr->flags, hdr->output, &hdr->data[offset], remaining); GST_TRACE_OBJECT (hdr->payload, "extension %" GST_PTR_FORMAT " wrote %" G_GSIZE_FORMAT, ext, written); if (written == 0) { /* extension wrote no data */ return; } else if (written < 0) { GST_WARNING_OBJECT (hdr->payload, "%s failed to write extension data", GST_OBJECT_NAME (ext)); goto error; } else if (written > remaining) { /* wrote too much! */ g_error ("Overflow detected writing rtp header extensions. One of the " "instances likely did not report a large enough maximum size. " "Memory corruption has occured. Aborting"); goto error; } ext_id = gst_rtp_header_extension_get_id (ext); /* move to the beginning of the extension header */ offset -= hdr->hdr_unit_size; /* write extension header */ if (hdr->flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE) { if (written > RTP_HEADER_EXT_ONE_BYTE_MAX_SIZE) { g_critical ("Amount of data written by %s is larger than allowed with " "a one byte header.", GST_OBJECT_NAME (ext)); goto error; } hdr->data[offset] = ((ext_id & 0x0F) << 4) | ((written - 1) & 0x0F); } else if (hdr->flags & GST_RTP_HEADER_EXTENSION_TWO_BYTE) { if (written > RTP_HEADER_EXT_TWO_BYTE_MAX_SIZE) { g_critical ("Amount of data written by %s is larger than allowed with " "a two byte header.", GST_OBJECT_NAME (ext)); goto error; } hdr->data[offset] = ext_id & 0xFF; hdr->data[offset + 1] = written & 0xFF; } else { g_critical ("Don't know how to write extension data with flags 0x%x!", hdr->flags); goto error; } hdr->written_size += written + hdr->hdr_unit_size; return; error: hdr->abort = TRUE; return; } static gboolean set_headers (GstBuffer ** buffer, guint idx, gpointer user_data) { HeaderData *data = user_data; HeaderExt hdrext = { NULL, }; GstRTPBuffer rtp = { NULL, }; if (!gst_rtp_buffer_map (*buffer, GST_MAP_READWRITE, &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); GST_OBJECT_LOCK (data->payload); if (data->payload->priv->header_exts->len > 0 && data->payload->priv->input_meta_buffer) { guint wordlen; gsize extlen; guint16 bit_pattern; /* write header extensions */ hdrext.payload = data->payload; hdrext.output = *buffer; /* XXX: pre-calculate these flags and sizes? */ hdrext.flags = GST_RTP_HEADER_EXTENSION_ONE_BYTE | GST_RTP_HEADER_EXTENSION_TWO_BYTE; g_ptr_array_foreach (data->payload->priv->header_exts, (GFunc) determine_header_extension_flags_size, &hdrext); hdrext.hdr_unit_size = 0; if (hdrext.flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE) { /* prefer the one byte header */ hdrext.hdr_unit_size = 1; /* TODO: support mixed size writing modes, i.e. RFC8285 */ hdrext.flags &= ~GST_RTP_HEADER_EXTENSION_TWO_BYTE; bit_pattern = 0xBEDE; } else if (hdrext.flags & GST_RTP_HEADER_EXTENSION_TWO_BYTE) { hdrext.hdr_unit_size = 2; bit_pattern = 0x1000; } else { goto unsupported_flags; } extlen = hdrext.hdr_unit_size * data->payload->priv->header_exts->len + hdrext.allocated_size; wordlen = extlen / 4 + ((extlen % 4) ? 1 : 0); /* XXX: do we need to add to any existing extension data instead of * overwriting everything? */ gst_rtp_buffer_set_extension_data (&rtp, bit_pattern, wordlen); gst_rtp_buffer_get_extension_data (&rtp, NULL, (gpointer) & hdrext.data, &wordlen); /* from 32-bit words to bytes */ hdrext.allocated_size = wordlen * 4; g_ptr_array_foreach (data->payload->priv->header_exts, (GFunc) write_header_extension, &hdrext); if (hdrext.written_size > 0) { wordlen = hdrext.written_size / 4 + ((hdrext.written_size % 4) ? 1 : 0); /* zero-fill the hdrext padding bytes */ memset (&hdrext.data[hdrext.written_size], 0, wordlen * 4 - hdrext.written_size); gst_rtp_buffer_set_extension_data (&rtp, bit_pattern, wordlen); } else { gst_rtp_buffer_remove_extension_data (&rtp); } } GST_OBJECT_UNLOCK (data->payload); 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; } unsupported_flags: { GST_OBJECT_UNLOCK (data->payload); gst_rtp_buffer_unmap (&rtp); GST_ERROR ("Cannot add rtp header extensions with mixed header types"); 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; /* 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: (transfer full): 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: (transfer full): 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_SCALE_RTPTIME: priv->scale_rtptime = g_value_get_boolean (value); break; case PROP_AUTO_HEADER_EXTENSION: priv->auto_hdr_ext = 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_SCALE_RTPTIME: g_value_set_boolean (value, priv->scale_rtptime); break; case PROP_AUTO_HEADER_EXTENSION: g_value_set_boolean (value, priv->auto_hdr_ext); break; case PROP_EXTENSIONS: gst_rtp_base_payload_get_extensions (rtpbasepayload, value); 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; priv->negotiate_called = 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; }