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a35d1dde42
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (on_sender_timeout), (create_session), (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpbin.h: Add signal to notify listeners when a sender becomes a receiver. Tweak lip-sync code, don't store our own copy of the ts-offset of the jitterbuffer, don't adjust sync if the change is less than 4msec. Get the RTP timestamp <-> GStreamer timestamp relation directly from the jitterbuffer instead of our inaccurate version from the source. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_get_sync): * gst/rtpmanager/gstrtpjitterbuffer.h: Add G_LIKELY macros, use global defines for max packet reorder and dropouts. Reset the jitterbuffer clock skew detection when packets seqnums are changed unexpectedly. * gst/rtpmanager/gstrtpsession.c: (on_sender_timeout), (gst_rtp_session_class_init), (gst_rtp_session_init): * gst/rtpmanager/gstrtpsession.h: Add sender timeout signal. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew), (rtp_jitter_buffer_insert), (rtp_jitter_buffer_get_sync): * gst/rtpmanager/rtpjitterbuffer.h: Add some G_LIKELY macros. Keep track of the extended RTP timestamp so that we can report the RTP timestamp <-> GStreamer timestamp relation for lip-sync. Remove server timestamp gap detection code, the server can sometimes make a huge gap in timestamps (talk spurts,...) see #549774. Detect timetamp weirdness instead by observing the sender/receiver timestamp relation and resync if it changes more than 1 second. Add method to report about the current rtp <-> gst timestamp relation which is needed for lip-sync. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (on_sender_timeout), (check_collision), (rtp_session_process_sr), (session_cleanup): * gst/rtpmanager/rtpsession.h: Add sender timeout signal. Remove inaccurate rtp <-> gst timestamp relation code, the jitterbuffer can now do an accurate reporting about this. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (rtp_source_update_caps), (calculate_jitter), (rtp_source_process_rtp): * gst/rtpmanager/rtpsource.h: Remove inaccurate rtp <-> gst timestamp relation code. * gst/rtpmanager/rtpstats.h: Define global max-reorder and max-dropout constants for use in various subsystems.
217 lines
7.6 KiB
C
217 lines
7.6 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_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/gstrtcpbuffer.h>
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#include <gst/netbuffer/gstnetbuffer.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->received_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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* RTPSourcePushRTP:
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* @src: an #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 @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, GstBuffer *buffer,
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gpointer user_data);
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/**
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* RTPSourceClockRate:
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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 clock-rate of the
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* @payload.
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*
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* Returns: a clock-rate for @payload.
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*/
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typedef gint (*RTPSourceClockRate) (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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* @clock_rate: a clock-rate 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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RTPSourceClockRate clock_rate;
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} RTPSourceCallbacks;
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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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struct _RTPSource {
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GObject object;
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/*< private >*/
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guint32 ssrc;
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gint probation;
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gboolean validated;
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gboolean is_csrc;
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gboolean is_sender;
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guint8 *sdes[9];
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guint sdes_len[9];
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gboolean received_bye;
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gchar *bye_reason;
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gboolean have_rtp_from;
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GstNetAddress rtp_from;
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gboolean have_rtcp_from;
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GstNetAddress rtcp_from;
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guint8 payload;
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GstCaps *caps;
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gint clock_rate;
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gint32 seqnum_base;
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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_rtptime;
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GstClockTime last_ntpnstime;
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GQueue *packets;
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RTPSourceCallbacks callbacks;
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gpointer user_data;
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RTPSourceStats stats;
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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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gboolean rtp_source_received_bye (RTPSource *src);
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gchar * rtp_source_get_bye_reason (RTPSource *src);
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void rtp_source_update_caps (RTPSource *src, GstCaps *caps);
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/* SDES info */
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gboolean rtp_source_set_sdes (RTPSource *src, GstRTCPSDESType type,
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const guint8 *data, guint len);
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gboolean rtp_source_set_sdes_string (RTPSource *src, GstRTCPSDESType type,
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const gchar *data);
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gboolean rtp_source_get_sdes (RTPSource *src, GstRTCPSDESType type,
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guint8 **data, guint *len);
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gchar* rtp_source_get_sdes_string (RTPSource *src, GstRTCPSDESType type);
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/* handling network address */
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void rtp_source_set_rtp_from (RTPSource *src, GstNetAddress *address);
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void rtp_source_set_rtcp_from (RTPSource *src, GstNetAddress *address);
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/* handling RTP */
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GstFlowReturn rtp_source_process_rtp (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival);
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GstFlowReturn rtp_source_send_rtp (RTPSource *src, GstBuffer *buffer, guint64 ntpnstime);
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/* RTCP messages */
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void rtp_source_process_bye (RTPSource *src, const gchar *reason);
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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, 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_new_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_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, guint8 *fractionlost, gint32 *packetslost,
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guint32 *exthighestseq, guint32 *jitter,
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guint32 *lsr, guint32 *dlsr);
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void rtp_source_reset (RTPSource * src);
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#endif /* __RTP_SOURCE_H__ */
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