mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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f17db5c4ed
Inform the source when caps changed. This was removed in the port to 1.0 leaving the source unaware of the clock-rate and unable to interpollate rtp timestamps for SR packets.
360 lines
14 KiB
C
360 lines
14 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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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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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* @buffer: the RTP buffer 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 @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 (*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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* @src: the #RTPSource
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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, RTPSource *src, GstBuffer *buffer, gpointer user_data);
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/**
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* RTPSessionClockRate:
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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 clock-rate of @payload.
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*
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* Returns: the clock-rate of @pt.
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*/
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typedef gint (*RTPSessionClockRate) (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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* @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,
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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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* 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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*
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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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RTPSessionClockRate clock_rate;
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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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} 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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* @cnames: Hashtable of sources indexed by CNAME
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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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*
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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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guint probation;
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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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RTPSource *source;
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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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GHashTable *cnames;
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guint total_sources;
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GstClockTime next_rtcp_check_time;
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GstClockTime last_rtcp_send_time;
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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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gchar *bye_reason;
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gboolean sent_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 clock_rate_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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RTPSessionStats stats;
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gboolean change_ssrc;
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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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GstClockTime last_keyframe_request;
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gboolean last_keyframe_all_headers;
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};
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/**
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* RTPSessionClass:
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* @on_new_ssrc: emited when a new source is found
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* @on_bye_ssrc: emited 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_feedback_rtcp) (RTPSession *sess, guint type, guint fbtype,
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guint sender_ssrc, guint media_ssrc, GstBuffer *fci);
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void (*send_rtcp) (RTPSession *sess, GstClockTimeDiff max_delay);
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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_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_clock_rate_callback (RTPSession * sess,
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RTPSessionClockRate 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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gboolean rtp_session_set_sdes_string (RTPSession *sess, GstRTCPSDESType type,
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const gchar *cname);
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gchar* rtp_session_get_sdes_string (RTPSession *sess, GstRTCPSDESType type);
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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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RTPSource* rtp_session_get_internal_source (RTPSession *sess);
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void rtp_session_set_internal_ssrc (RTPSession *sess, guint32 ssrc);
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guint32 rtp_session_get_internal_ssrc (RTPSession *sess);
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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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RTPSource* rtp_session_get_source_by_cname (RTPSession *sess, const gchar *cname);
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RTPSource* rtp_session_create_source (RTPSession *sess);
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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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GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer *buffer,
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GstClockTime current_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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/* stopping the session */
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GstFlowReturn rtp_session_schedule_bye (RTPSession *sess, const gchar *reason,
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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 transmittion of an early RTCP packet */
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void rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
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GstClockTimeDiff 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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GstClockTime now,
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gboolean fir,
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gint count);
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#endif /* __RTP_SESSION_H__ */
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