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76792a5c20
In function rtp_jitter_buffer_calculate_pts: If gap in incoming RTP timestamps is more than (3 * jbuf->clock_rate) we call rtp_jitter_buffer_reset_skew which resets pts to 0. So components down the pipeline (playes, mixers) just skip frames/samples until pts becomes equal to pts before gap. In version 1.10.2 and before this checking was bypassed for packets with "estimated dts", and gaps were handled correctly. https://bugzilla.gnome.org/show_bug.cgi?id=778341
195 lines
7.6 KiB
C
195 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __RTP_JITTER_BUFFER_H__
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#define __RTP_JITTER_BUFFER_H__
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#include <gst/gst.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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typedef struct _RTPJitterBuffer RTPJitterBuffer;
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typedef struct _RTPJitterBufferClass RTPJitterBufferClass;
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typedef struct _RTPJitterBufferItem RTPJitterBufferItem;
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#define RTP_TYPE_JITTER_BUFFER (rtp_jitter_buffer_get_type())
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#define RTP_JITTER_BUFFER(src) (G_TYPE_CHECK_INSTANCE_CAST((src),RTP_TYPE_JITTER_BUFFER,RTPJitterBuffer))
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#define RTP_JITTER_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_JITTER_BUFFER,RTPJitterBufferClass))
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#define RTP_IS_JITTER_BUFFER(src) (G_TYPE_CHECK_INSTANCE_TYPE((src),RTP_TYPE_JITTER_BUFFER))
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#define RTP_IS_JITTER_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_JITTER_BUFFER))
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#define RTP_JITTER_BUFFER_CAST(src) ((RTPJitterBuffer *)(src))
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/**
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* RTPJitterBufferMode:
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* @RTP_JITTER_BUFFER_MODE_NONE: don't do any skew correction, outgoing
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* timestamps are calculated directly from the RTP timestamps. This mode is
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* good for recording but not for real-time applications.
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* @RTP_JITTER_BUFFER_MODE_SLAVE: calculate the skew between sender and receiver
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* and produce smoothed adjusted outgoing timestamps. This mode is good for
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* low latency communications.
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* @RTP_JITTER_BUFFER_MODE_BUFFER: buffer packets between low/high watermarks.
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* This mode is good for streaming communication.
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* @RTP_JITTER_BUFFER_MODE_SYNCED: sender and receiver clocks are synchronized,
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* like #RTP_JITTER_BUFFER_MODE_SLAVE but skew is assumed to be 0. Good for
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* low latency communication when sender and receiver clocks are
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* synchronized and there is thus no clock skew.
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* @RTP_JITTER_BUFFER_MODE_LAST: last buffer mode.
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*
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* The different buffer modes for a jitterbuffer.
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*/
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typedef enum {
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RTP_JITTER_BUFFER_MODE_NONE = 0,
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RTP_JITTER_BUFFER_MODE_SLAVE = 1,
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RTP_JITTER_BUFFER_MODE_BUFFER = 2,
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/* FIXME 3 is missing because it was used for 'auto' in jitterbuffer */
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RTP_JITTER_BUFFER_MODE_SYNCED = 4,
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RTP_JITTER_BUFFER_MODE_LAST
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} RTPJitterBufferMode;
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#define RTP_TYPE_JITTER_BUFFER_MODE (rtp_jitter_buffer_mode_get_type())
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GType rtp_jitter_buffer_mode_get_type (void);
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#define RTP_JITTER_BUFFER_MAX_WINDOW 512
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/**
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* RTPJitterBuffer:
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*
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* A JitterBuffer in the #RTPSession
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*/
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struct _RTPJitterBuffer {
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GObject object;
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GQueue *packets;
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RTPJitterBufferMode mode;
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GstClockTime delay;
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/* for buffering */
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gboolean buffering;
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guint64 low_level;
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guint64 high_level;
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/* for calculating skew */
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gboolean need_resync;
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GstClockTime base_time;
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GstClockTime base_rtptime;
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GstClockTime media_clock_base_time;
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guint32 clock_rate;
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GstClockTime base_extrtp;
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GstClockTime prev_out_time;
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guint64 ext_rtptime;
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guint64 last_rtptime;
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gint64 window[RTP_JITTER_BUFFER_MAX_WINDOW];
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guint window_pos;
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guint window_size;
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gboolean window_filling;
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gint64 window_min;
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gint64 skew;
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gint64 prev_send_diff;
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gboolean buffering_disabled;
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GMutex clock_lock;
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GstClock *pipeline_clock;
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GstClock *media_clock;
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gulong media_clock_synced_id;
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guint64 media_clock_offset;
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gboolean rfc7273_sync;
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};
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struct _RTPJitterBufferClass {
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GObjectClass parent_class;
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};
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/**
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* RTPJitterBufferItem:
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* @data: the data of the item
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* @next: pointer to next item
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* @prev: pointer to previous item
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* @type: the type of @data, used freely by caller
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* @dts: input DTS
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* @pts: output PTS
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* @seqnum: seqnum, the seqnum is used to insert the item in the
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* right position in the jitterbuffer and detect duplicates. Use -1 to
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* append.
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* @count: amount of seqnum in this item
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* @rtptime: rtp timestamp
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*
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* An object containing an RTP packet or event.
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*/
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struct _RTPJitterBufferItem {
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gpointer data;
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GList *next;
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GList *prev;
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guint type;
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GstClockTime dts;
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GstClockTime pts;
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guint seqnum;
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guint count;
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guint rtptime;
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};
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GType rtp_jitter_buffer_get_type (void);
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/* managing lifetime */
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RTPJitterBuffer* rtp_jitter_buffer_new (void);
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RTPJitterBufferMode rtp_jitter_buffer_get_mode (RTPJitterBuffer *jbuf);
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void rtp_jitter_buffer_set_mode (RTPJitterBuffer *jbuf, RTPJitterBufferMode mode);
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GstClockTime rtp_jitter_buffer_get_delay (RTPJitterBuffer *jbuf);
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void rtp_jitter_buffer_set_delay (RTPJitterBuffer *jbuf, GstClockTime delay);
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void rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer *jbuf, guint32 clock_rate);
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guint32 rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer *jbuf);
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void rtp_jitter_buffer_set_media_clock (RTPJitterBuffer *jbuf, GstClock * clock, guint64 clock_offset);
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void rtp_jitter_buffer_set_pipeline_clock (RTPJitterBuffer *jbuf, GstClock * clock);
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gboolean rtp_jitter_buffer_get_rfc7273_sync (RTPJitterBuffer *jbuf);
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void rtp_jitter_buffer_set_rfc7273_sync (RTPJitterBuffer *jbuf, gboolean rfc7273_sync);
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void rtp_jitter_buffer_reset_skew (RTPJitterBuffer *jbuf);
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gboolean rtp_jitter_buffer_insert (RTPJitterBuffer *jbuf,
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RTPJitterBufferItem *item,
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gboolean *head, gint *percent);
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void rtp_jitter_buffer_disable_buffering (RTPJitterBuffer *jbuf, gboolean disabled);
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RTPJitterBufferItem * rtp_jitter_buffer_peek (RTPJitterBuffer *jbuf);
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RTPJitterBufferItem * rtp_jitter_buffer_pop (RTPJitterBuffer *jbuf, gint *percent);
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void rtp_jitter_buffer_flush (RTPJitterBuffer *jbuf,
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GFunc free_func, gpointer user_data);
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gboolean rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf);
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void rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering);
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gint rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf);
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guint rtp_jitter_buffer_num_packets (RTPJitterBuffer *jbuf);
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guint32 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer *jbuf);
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void rtp_jitter_buffer_get_sync (RTPJitterBuffer *jbuf, guint64 *rtptime,
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guint64 *timestamp, guint32 *clock_rate,
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guint64 *last_rtptime);
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GstClockTime rtp_jitter_buffer_calculate_pts (RTPJitterBuffer * jbuf, GstClockTime dts, gboolean estimated_dts,
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guint32 rtptime, GstClockTime base_time);
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#endif /* __RTP_JITTER_BUFFER_H__ */
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