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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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ef8dfd7873
Move RB info from receiver reports into the internal source that the RR are about, and deprecate (but retain) the old mapping where each external source has only a single RB entry in the rtp statistics. The old method is broken if a remote peer uses a single ssrc to send receiver reports for more than one of our internal sources, other as multiple RB in a single packet, or alternate RB in different reports. In each case only the most recent entry was kept, overwriting data for other internal sources. In multicast scenarios each internal source may receive multiple receiver reports from different peers. To support that, all received RR's are now stored into a hash table indexed by the sender's SSRC, and all RRs are placed into an array when generating statistics, so that the information from all peers is retrievable. The current deficient behaviour (adding RB info into non-internal RTPSources) is deprecated but kept in order to be backward compatible, and retained that way in the generated statistics structure. Refs [1] https://tools.ietf.org/html/rfc3550#section-6.4.1 Based on a patch by Fede Claramonte <fclaramonte@twilio.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7424>
317 lines
11 KiB
C
317 lines
11 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
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* Copyright (C) 2015 Kurento (http://kurento.org/)
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* @author: Miguel París <mparisdiaz@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __RTP_SOURCE_H__
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#define __RTP_SOURCE_H__
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#include <gst/gst.h>
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#include <gst/rtp/rtp.h>
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#include <gst/net/gstnetaddressmeta.h>
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#include <gio/gio.h>
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#include "rtpstats.h"
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/* the default number of consecutive RTP packets we need to receive before the
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* source is considered valid */
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#define RTP_NO_PROBATION 0
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#define RTP_DEFAULT_PROBATION 2
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#define RTP_SEQ_MOD (1 << 16)
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typedef struct _RTPSource RTPSource;
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typedef struct _RTPSourceClass RTPSourceClass;
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#define RTP_TYPE_SOURCE (rtp_source_get_type())
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#define RTP_SOURCE(src) (G_TYPE_CHECK_INSTANCE_CAST((src),RTP_TYPE_SOURCE,RTPSource))
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#define RTP_SOURCE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SOURCE,RTPSourceClass))
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#define RTP_IS_SOURCE(src) (G_TYPE_CHECK_INSTANCE_TYPE((src),RTP_TYPE_SOURCE))
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#define RTP_IS_SOURCE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SOURCE))
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#define RTP_SOURCE_CAST(src) ((RTPSource *)(src))
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/**
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* RTP_SOURCE_IS_ACTIVE:
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* @src: an #RTPSource
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*
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* Check if @src is active. A source is active when it has been validated
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* and has not yet received a BYE packet.
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*/
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#define RTP_SOURCE_IS_ACTIVE(src) (src->validated && !src->marked_bye)
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/**
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* RTP_SOURCE_IS_SENDER:
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* @src: an #RTPSource
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*
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* Check if @src is a sender.
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*/
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#define RTP_SOURCE_IS_SENDER(src) (src->is_sender)
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/**
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* RTP_SOURCE_IS_MARKED_BYE:
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* @src: an #RTPSource
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*
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* Check if @src is a marked as BYE.
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*/
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#define RTP_SOURCE_IS_MARKED_BYE(src) (src->marked_bye)
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/**
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* RTPSourcePushRTP:
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* @src: an #RTPSource
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* @data: the RTP buffer or buffer list ready for processing
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* @user_data: user data specified when registering
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*
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* This callback will be called when @src has @buffer ready for further
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* processing.
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*
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* Returns: a #GstFlowReturn.
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*/
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typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, gpointer data,
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gpointer user_data);
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/**
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* RTPSourceCaps:
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* @src: an #RTPSource
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* @payload: a payload type
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* @user_data: user data specified when registering
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*
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* This callback will be called when @src needs the caps of the
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* @payload.
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*
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* Returns: a caps for @payload.
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*/
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typedef GstCaps * (*RTPSourceCaps) (RTPSource *src, guint8 payload, gpointer user_data);
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/**
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* RTPSourceCallbacks:
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* @push_rtp: a packet becomes available for handling
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* @caps: a caps is requested
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* @get_time: the current clock time is requested
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*
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* Callbacks performed by #RTPSource when actions need to be performed.
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*/
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typedef struct {
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RTPSourcePushRTP push_rtp;
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RTPSourceCaps caps;
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} RTPSourceCallbacks;
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/**
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* RTPConflictingAddress:
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* @address: #GSocketAddress which conflicted
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* @last_conflict_time: time when the last conflict was seen
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*
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* This structure is used to account for addresses that have conflicted to find
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* loops.
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*/
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typedef struct {
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GSocketAddress *address;
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GstClockTime time;
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} RTPConflictingAddress;
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/**
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* RTPSource:
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*
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* A source in the #RTPSession
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*
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* @conflicting_addresses: GList of conflicting addresses
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*/
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struct _RTPSource {
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GObject object;
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/*< private >*/
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guint32 ssrc;
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/* If not -1 then this is the SSRC of the corresponding media RTPSource */
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guint32 media_ssrc;
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guint16 generation;
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GHashTable *reported_in_sr_of; /* set of SSRCs */
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guint probation;
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guint curr_probation;
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gboolean validated;
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gboolean internal;
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gboolean is_csrc;
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gboolean is_sender;
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gboolean closing;
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GstStructure *sdes;
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gboolean marked_bye;
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gchar *bye_reason;
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gboolean sent_bye;
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GSocketAddress *rtp_from;
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GSocketAddress *rtcp_from;
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gint payload;
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GstCaps *caps;
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gint clock_rate;
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gint32 seqnum_offset;
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GstClockTime bye_time;
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GstClockTime last_activity;
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GstClockTime last_rtp_activity;
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GstClockTime last_rtime;
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GstClockTime last_rtptime;
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/* for bitrate estimation */
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guint64 bitrate;
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GstClockTime prev_rtime;
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guint64 bytes_sent;
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guint64 bytes_received;
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GQueue *packets;
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RTPPacketRateCtx packet_rate_ctx;
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guint32 max_dropout_time;
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guint32 max_misorder_time;
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RTPSourceCallbacks callbacks;
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gpointer user_data;
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RTPSourceStats stats;
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RTPReceiverReport last_rr; /* last_rr sent for this source */
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GHashTable *received_rr; /* set of sender SSRC -> (RTPReceiveReport *) */
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GList *conflicting_addresses;
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GQueue *retained_feedback;
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gboolean send_pli;
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gboolean send_fir;
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guint8 current_send_fir_seqnum;
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gint last_fir_count;
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GstClockTime last_keyframe_request;
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gboolean send_nack;
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GArray *nacks;
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GArray *nack_deadlines;
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gboolean pt_set;
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guint8 pt;
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gboolean disable_rtcp;
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};
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struct _RTPSourceClass {
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GObjectClass parent_class;
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};
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GType rtp_source_get_type (void);
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/* managing lifetime of sources */
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RTPSource* rtp_source_new (guint32 ssrc);
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void rtp_source_set_callbacks (RTPSource *src, RTPSourceCallbacks *cb, gpointer data);
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/* properties */
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guint32 rtp_source_get_ssrc (RTPSource *src);
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void rtp_source_set_as_csrc (RTPSource *src);
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gboolean rtp_source_is_as_csrc (RTPSource *src);
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gboolean rtp_source_is_active (RTPSource *src);
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gboolean rtp_source_is_validated (RTPSource *src);
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gboolean rtp_source_is_sender (RTPSource *src);
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void rtp_source_mark_bye (RTPSource *src, const gchar *reason);
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gboolean rtp_source_is_marked_bye (RTPSource *src);
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gchar * rtp_source_get_bye_reason (RTPSource *src);
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void rtp_source_update_send_caps (RTPSource *src, GstCaps *caps);
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/* SDES info */
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const GstStructure *
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rtp_source_get_sdes_struct (RTPSource * src);
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gboolean rtp_source_set_sdes_struct (RTPSource * src, GstStructure *sdes);
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/* handling network address */
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void rtp_source_set_rtp_from (RTPSource *src, GSocketAddress *address);
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void rtp_source_set_rtcp_from (RTPSource *src, GSocketAddress *address);
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/* handling RTP */
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GstFlowReturn rtp_source_process_rtp (RTPSource *src, RTPPacketInfo *pinfo);
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GstFlowReturn rtp_source_send_rtp (RTPSource *src, RTPPacketInfo *pinfo);
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/* RTCP messages */
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void rtp_source_process_sr (RTPSource *src, GstClockTime time, guint64 ntptime,
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guint32 rtptime, guint32 packet_count, guint32 octet_count);
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void rtp_source_process_rb (RTPSource *src, guint32 ssrc, guint32 sender_ssrc,
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guint64 ntpnstime, guint8 fractionlost,
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gint32 packetslost, guint32 exthighestseq, guint32 jitter,
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guint32 lsr, guint32 dlsr);
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gboolean rtp_source_get_new_sr (RTPSource *src, guint64 ntpnstime, GstClockTime running_time,
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guint64 *ntptime, guint32 *rtptime, guint32 *packet_count,
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guint32 *octet_count);
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gboolean rtp_source_get_new_rb (RTPSource *src, GstClockTime time, guint8 *fractionlost,
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gint32 *packetslost, guint32 *exthighestseq, guint32 *jitter,
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guint32 *lsr, guint32 *dlsr);
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gboolean rtp_source_get_last_sr (RTPSource *src, GstClockTime *time, guint64 *ntptime,
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guint32 *rtptime, guint32 *packet_count,
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guint32 *octet_count);
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gboolean rtp_source_get_last_rb (RTPSource *src, guint32 * ssrc, guint8 *fractionlost, gint32 *packetslost,
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guint32 *exthighestseq, guint32 *jitter,
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guint32 *lsr, guint32 *dlsr, guint32 *round_trip);
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void rtp_source_reset (RTPSource * src);
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gboolean rtp_source_find_conflicting_address (RTPSource * src,
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GSocketAddress *address,
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GstClockTime time);
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void rtp_source_add_conflicting_address (RTPSource * src,
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GSocketAddress *address,
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GstClockTime time);
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gboolean find_conflicting_address (GList * conflicting_address,
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GSocketAddress * address,
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GstClockTime time);
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GList * add_conflicting_address (GList * conflicting_addresses,
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GSocketAddress * address,
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GstClockTime time);
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GList * timeout_conflicting_addresses (GList * conflicting_addresses,
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GstClockTime current_time);
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void rtp_conflicting_address_free (RTPConflictingAddress * addr);
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void rtp_source_timeout (RTPSource * src,
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GstClockTime current_time,
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GstClockTime running_time,
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GstClockTime feedback_retention_window);
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void rtp_source_retain_rtcp_packet (RTPSource * src,
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GstRTCPPacket *pkt,
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GstClockTime running_time);
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gboolean rtp_source_has_retained (RTPSource * src,
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GCompareFunc func,
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gconstpointer data);
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void rtp_source_register_nack (RTPSource * src,
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guint16 seqnum,
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GstClockTime deadline);
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guint16 * rtp_source_get_nacks (RTPSource * src, guint *n_nacks);
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GstClockTime * rtp_source_get_nack_deadlines (RTPSource * src, guint *n_nacks);
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void rtp_source_clear_nacks (RTPSource * src, guint n_nacks);
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#endif /* __RTP_SOURCE_H__ */
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