mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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ca7f66f9b5
Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header extension timestamp is updated with the actual NTP time before sending the packet. However, there are some circumstances where we would like to preserve the original timestamp obtained from reference timestamp buffer metadata. This commit provides the ability to configure whether or not to update the ntp-64 header extension timestamp with the actual NTP time via the update-ntp64-header-ext boolean property. The property is also exposed via rtpbin. Default property value of TRUE will preserve existing behavior (update ntp-64 header ext with actual NTP time). Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>
449 lines
17 KiB
C
449 lines
17 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __RTP_SESSION_H__
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#define __RTP_SESSION_H__
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#include <gst/gst.h>
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#include "rtpsource.h"
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#include "rtptwcc.h"
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typedef struct _RTPSession RTPSession;
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typedef struct _RTPSessionClass RTPSessionClass;
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#define RTP_TYPE_SESSION (rtp_session_get_type())
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#define RTP_SESSION(sess) (G_TYPE_CHECK_INSTANCE_CAST((sess),RTP_TYPE_SESSION,RTPSession))
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#define RTP_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SESSION,RTPSessionClass))
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#define RTP_IS_SESSION(sess) (G_TYPE_CHECK_INSTANCE_TYPE((sess),RTP_TYPE_SESSION))
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#define RTP_IS_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SESSION))
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#define RTP_SESSION_CAST(sess) ((RTPSession *)(sess))
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#define RTP_SESSION_LOCK(sess) (g_mutex_lock (&(sess)->lock))
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#define RTP_SESSION_UNLOCK(sess) (g_mutex_unlock (&(sess)->lock))
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/**
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* RTPSessionProcessRTP:
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* @sess: an #RTPSession
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* @src: the #RTPSource
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* @buffer: the RTP buffer 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 @sess has @buffer ready for further
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* processing. Processing the buffer typically includes decoding and displaying
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* the buffer.
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*
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* Returns: a #GstFlowReturn.
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*/
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typedef GstFlowReturn (*RTPSessionProcessRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
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/**
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* RTPSessionSendRTP:
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* @sess: an #RTPSession
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* @src: the #RTPSource
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* @data: the RTP buffer or buffer list ready for sending
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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 @sess has @data (a buffer or buffer list)
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* ready for sending to all listening participants in this session.
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*
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* Returns: a #GstFlowReturn.
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*/
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typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, gpointer data, gpointer user_data);
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/**
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* RTPSessionSendRTCP:
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* @sess: an #RTPSession
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* @src: the #RTPSource
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* @buffer: the RTCP buffer ready for sending
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* @eos: if an EOS event should be pushed
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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 @sess has @buffer ready for sending to
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* all listening participants in this session.
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*
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* Returns: a #GstFlowReturn.
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*/
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typedef GstFlowReturn (*RTPSessionSendRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer,
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gboolean eos, gpointer user_data);
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/**
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* RTPSessionSyncRTCP:
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* @sess: an #RTPSession
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* @buffer: the RTCP buffer ready for synchronisation
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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 @sess has an SR @buffer ready for doing
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* synchronisation between streams.
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*
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* Returns: a #GstFlowReturn.
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*/
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typedef GstFlowReturn (*RTPSessionSyncRTCP) (RTPSession *sess, GstBuffer *buffer, gpointer user_data);
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/**
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* RTPSessionCaps:
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* @sess: an #RTPSession
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* @payload: the payload
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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 @sess needs the caps of @payload.
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*
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* Returns: the caps of @pt.
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*/
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typedef GstCaps * (*RTPSessionCaps) (RTPSession *sess, guint8 payload, gpointer user_data);
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/**
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* RTPSessionReconsider:
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* @sess: an #RTPSession
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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 @sess needs to cancel the current timeout.
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* The currently running timeout should be canceled and a new reporting interval
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* should be requested from @sess.
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*/
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typedef void (*RTPSessionReconsider) (RTPSession *sess, gpointer user_data);
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/**
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* RTPSessionRequestKeyUnit:
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* @sess: an #RTPSession
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* @ssrc: SSRC of the source related to the key unit request
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* @all_headers: %TRUE if "all-headers" property should be set on the key unit
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* request
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* @user_data: user data specified when registering
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*
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* Asks the encoder to produce a key unit as soon as possibly within the
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* bandwidth constraints
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*/
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typedef void (*RTPSessionRequestKeyUnit) (RTPSession *sess, guint32 ssrc,
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gboolean all_headers, gpointer user_data);
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/**
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* RTPSessionRequestTime:
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* @sess: an #RTPSession
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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 @sess needs the current time. The time
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* should be returned as a #GstClockTime
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*/
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typedef GstClockTime (*RTPSessionRequestTime) (RTPSession *sess,
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gpointer user_data);
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/**
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* RTPSessionNotifyNACK:
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* @sess: an #RTPSession
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* @seqnum: the missing seqnum
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* @blp: other missing seqnums
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* @ssrc: SSRC of requested stream
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* @user_data: user data specified when registering
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*
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* Notifies of NACKed frames.
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*/
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typedef void (*RTPSessionNotifyNACK) (RTPSession *sess,
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guint16 seqnum, guint16 blp, guint32 ssrc, gpointer user_data);
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/**
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* RTPSessionNotifyTWCC:
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* @user_data: user data specified when registering
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*
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* Notifies of Transport-wide congestion control packets and stats.
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*/
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typedef void (*RTPSessionNotifyTWCC) (RTPSession *sess,
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GstStructure * twcc_packets, GstStructure * twcc_stats, gpointer user_data);
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/**
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* RTPSessionReconfigure:
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* @sess: an #RTPSession
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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 @sess wants to reconfigure the
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* negotiated parameters.
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*/
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typedef void (*RTPSessionReconfigure) (RTPSession *sess, gpointer user_data);
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/**
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* RTPSessionNotifyEarlyRTCP:
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* @sess: an #RTPSession
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* @user_data: user data specified when registering
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*
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* Notifies of early RTCP being requested
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*/
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typedef void (*RTPSessionNotifyEarlyRTCP) (RTPSession *sess,
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gpointer user_data);
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/**
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* RTPSessionCallbacks:
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* @RTPSessionProcessRTP: callback to process RTP packets
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* @RTPSessionSendRTP: callback for sending RTP packets
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* @RTPSessionSendRTCP: callback for sending RTCP packets
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* @RTPSessionSyncRTCP: callback for handling SR packets
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* @RTPSessionReconsider: callback for reconsidering the timeout
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* @RTPSessionRequestKeyUnit: callback for requesting a new key unit
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* @RTPSessionRequestTime: callback for requesting the current time
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* @RTPSessionNotifyNACK: callback for notifying NACK
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* @RTPSessionNotifyTWCC: callback for notifying TWCC
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* @RTPSessionReconfigure: callback for requesting reconfiguration
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* @RTPSessionNotifyEarlyRTCP: callback for notifying early RTCP
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*
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* These callbacks can be installed on the session manager to get notification
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* when RTP and RTCP packets are ready for further processing. These callbacks
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* are not implemented with signals for performance reasons.
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*/
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typedef struct {
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RTPSessionProcessRTP process_rtp;
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RTPSessionSendRTP send_rtp;
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RTPSessionSyncRTCP sync_rtcp;
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RTPSessionSendRTCP send_rtcp;
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RTPSessionCaps caps;
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RTPSessionReconsider reconsider;
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RTPSessionRequestKeyUnit request_key_unit;
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RTPSessionRequestTime request_time;
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RTPSessionNotifyNACK notify_nack;
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RTPSessionNotifyTWCC notify_twcc;
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RTPSessionReconfigure reconfigure;
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RTPSessionNotifyEarlyRTCP notify_early_rtcp;
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} RTPSessionCallbacks;
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/**
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* RTPSession:
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* @lock: lock to protect the session
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* @source: the source of this session
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* @ssrcs: Hashtable of sources indexed by SSRC
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* @num_sources: the number of sources
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* @activecount: the number of active sources
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* @callbacks: callbacks
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* @user_data: user data passed in callbacks
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* @stats: session statistics
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* @conflicting_addresses: GList of conflicting addresses
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*
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* The RTP session manager object
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*/
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struct _RTPSession {
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GObject object;
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GMutex lock;
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guint header_len;
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guint mtu;
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GstStructure *sdes;
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guint probation;
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guint32 max_dropout_time;
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guint32 max_misorder_time;
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GstRTPProfile rtp_profile;
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gboolean reduced_size_rtcp;
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/* bandwidths */
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gboolean recalc_bandwidth;
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guint bandwidth;
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gdouble rtcp_bandwidth;
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guint rtcp_rr_bandwidth;
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guint rtcp_rs_bandwidth;
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guint32 suggested_ssrc;
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gboolean internal_ssrc_set;
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gboolean internal_ssrc_from_caps_or_property;
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/* for sender/receiver counting */
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guint32 key;
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guint32 mask_idx;
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guint32 mask;
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GHashTable *ssrcs[32];
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guint total_sources;
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guint16 generation;
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GstClockTime next_rtcp_check_time; /* tn */
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GstClockTime last_rtcp_check_time; /* tp */
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GstClockTime last_rtcp_send_time; /* t_rr_last */
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GstClockTime last_rtcp_interval; /* T_rr */
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GstClockTime start_time;
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gboolean first_rtcp;
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gboolean allow_early;
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GstClockTime next_early_rtcp_time;
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gboolean sr_req_pending;
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gboolean scheduled_bye;
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RTPSessionCallbacks callbacks;
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gpointer process_rtp_user_data;
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gpointer send_rtp_user_data;
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gpointer send_rtcp_user_data;
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gpointer sync_rtcp_user_data;
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gpointer caps_user_data;
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gpointer reconsider_user_data;
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gpointer request_key_unit_user_data;
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gpointer request_time_user_data;
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gpointer notify_nack_user_data;
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gpointer notify_twcc_user_data;
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gpointer reconfigure_user_data;
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gpointer notify_early_rtcp_user_data;
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RTPSessionStats stats;
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RTPSessionStats bye_stats;
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gboolean favor_new;
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GstClockTime rtcp_feedback_retention_window;
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guint rtcp_immediate_feedback_threshold;
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gboolean is_doing_ptp;
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GList *conflicting_addresses;
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gboolean timestamp_sender_reports;
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/* RFC6051 64-bit NTP header extension */
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guint8 send_ntp64_ext_id;
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gboolean update_ntp64_header_ext;
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/* Transport-wide cc-extension */
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RTPTWCCManager *twcc;
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RTPTWCCStats *twcc_stats;
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};
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/**
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* RTPSessionClass:
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* @on_new_ssrc: emitted when a new source is found
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* @on_bye_ssrc: emitted when a source is gone
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*
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* The session class.
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*/
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struct _RTPSessionClass {
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GObjectClass parent_class;
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/* action signals */
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RTPSource* (*get_source_by_ssrc) (RTPSession *sess, guint32 ssrc);
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/* signals */
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void (*on_new_ssrc) (RTPSession *sess, RTPSource *source);
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void (*on_ssrc_collision) (RTPSession *sess, RTPSource *source);
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void (*on_ssrc_validated) (RTPSession *sess, RTPSource *source);
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void (*on_ssrc_active) (RTPSession *sess, RTPSource *source);
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void (*on_ssrc_sdes) (RTPSession *sess, RTPSource *source);
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void (*on_bye_ssrc) (RTPSession *sess, RTPSource *source);
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void (*on_bye_timeout) (RTPSession *sess, RTPSource *source);
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void (*on_timeout) (RTPSession *sess, RTPSource *source);
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void (*on_sender_timeout) (RTPSession *sess, RTPSource *source);
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gboolean (*on_sending_rtcp) (RTPSession *sess, GstBuffer *buffer,
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gboolean early);
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void (*on_app_rtcp) (RTPSession *sess, guint subtype, guint ssrc,
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const gchar *name, GstBuffer *data);
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void (*on_feedback_rtcp) (RTPSession *sess, guint type, guint fbtype,
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guint sender_ssrc, guint media_ssrc, GstBuffer *fci);
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gboolean (*send_rtcp) (RTPSession *sess, GstClockTime max_delay);
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void (*on_receiving_rtcp) (RTPSession *sess, GstBuffer *buffer);
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void (*on_new_sender_ssrc) (RTPSession *sess, RTPSource *source);
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void (*on_sender_ssrc_active) (RTPSession *sess, RTPSource *source);
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guint (*on_sending_nacks) (RTPSession *sess, guint sender_ssrc,
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guint media_ssrc, GArray *nacks, GstBuffer *buffer);
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};
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GType rtp_session_get_type (void);
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/* create and configure */
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RTPSession* rtp_session_new (void);
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void rtp_session_reset (RTPSession *sess);
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void rtp_session_set_callbacks (RTPSession *sess,
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RTPSessionCallbacks *callbacks,
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gpointer user_data);
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void rtp_session_set_process_rtp_callback (RTPSession * sess,
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RTPSessionProcessRTP callback,
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gpointer user_data);
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void rtp_session_set_send_rtp_callback (RTPSession * sess,
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RTPSessionSendRTP callback,
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gpointer user_data);
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void rtp_session_set_send_rtcp_callback (RTPSession * sess,
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RTPSessionSendRTCP callback,
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gpointer user_data);
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void rtp_session_set_sync_rtcp_callback (RTPSession * sess,
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RTPSessionSyncRTCP callback,
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gpointer user_data);
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void rtp_session_set_caps_callback (RTPSession * sess,
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RTPSessionCaps callback,
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gpointer user_data);
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void rtp_session_set_reconsider_callback (RTPSession * sess,
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RTPSessionReconsider callback,
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gpointer user_data);
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void rtp_session_set_request_time_callback (RTPSession * sess,
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RTPSessionRequestTime callback,
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gpointer user_data);
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void rtp_session_set_bandwidth (RTPSession *sess, gdouble bandwidth);
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gdouble rtp_session_get_bandwidth (RTPSession *sess);
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void rtp_session_set_rtcp_fraction (RTPSession *sess, gdouble fraction);
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gdouble rtp_session_get_rtcp_fraction (RTPSession *sess);
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GstStructure * rtp_session_get_sdes_struct (RTPSession *sess);
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void rtp_session_set_sdes_struct (RTPSession *sess, const GstStructure *sdes);
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/* handling sources */
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guint32 rtp_session_suggest_ssrc (RTPSession *sess, gboolean *is_random);
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gboolean rtp_session_add_source (RTPSession *sess, RTPSource *src);
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guint rtp_session_get_num_sources (RTPSession *sess);
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guint rtp_session_get_num_active_sources (RTPSession *sess);
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RTPSource* rtp_session_get_source_by_ssrc (RTPSession *sess, guint32 ssrc);
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/* processing packets from receivers */
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GstFlowReturn rtp_session_process_rtp (RTPSession *sess, GstBuffer *buffer,
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GstClockTime current_time,
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GstClockTime running_time,
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guint64 ntpnstime);
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GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer *buffer,
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GstClockTime current_time,
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GstClockTime running_time,
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guint64 ntpnstime);
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/* processing packets for sending */
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void rtp_session_update_send_caps (RTPSession *sess, GstCaps *caps);
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GstFlowReturn rtp_session_send_rtp (RTPSession *sess, gpointer data, gboolean is_list,
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GstClockTime current_time, GstClockTime running_time,
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guint64 ntpnstime);
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/* scheduling bye */
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void rtp_session_mark_all_bye (RTPSession *sess, const gchar *reason);
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GstFlowReturn rtp_session_schedule_bye (RTPSession *sess, GstClockTime current_time);
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/* get interval for next RTCP interval */
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GstClockTime rtp_session_next_timeout (RTPSession *sess, GstClockTime current_time);
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GstFlowReturn rtp_session_on_timeout (RTPSession *sess, GstClockTime current_time,
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guint64 ntpnstime, GstClockTime running_time);
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/* request the transmission of an early RTCP packet */
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gboolean rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
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GstClockTime max_delay);
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/* Notify session of a request for a new key unit */
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gboolean rtp_session_request_key_unit (RTPSession * sess,
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guint32 ssrc,
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gboolean fir,
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gint count);
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gboolean rtp_session_request_nack (RTPSession * sess,
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guint32 ssrc,
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guint16 seqnum,
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GstClockTime max_delay);
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void rtp_session_update_recv_caps_structure (RTPSession * sess, const GstStructure * s);
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#endif /* __RTP_SESSION_H__ */
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