gstreamer/gst/rtpmanager/rtpjitterbuffer.h
Andrew 76792a5c20 rtpjitterbuffer: Don't always reset PTS to 0 after a gap
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
2017-02-26 12:41:19 +02:00

195 lines
7.6 KiB
C

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __RTP_JITTER_BUFFER_H__
#define __RTP_JITTER_BUFFER_H__
#include <gst/gst.h>
#include <gst/rtp/gstrtcpbuffer.h>
typedef struct _RTPJitterBuffer RTPJitterBuffer;
typedef struct _RTPJitterBufferClass RTPJitterBufferClass;
typedef struct _RTPJitterBufferItem RTPJitterBufferItem;
#define RTP_TYPE_JITTER_BUFFER (rtp_jitter_buffer_get_type())
#define RTP_JITTER_BUFFER(src) (G_TYPE_CHECK_INSTANCE_CAST((src),RTP_TYPE_JITTER_BUFFER,RTPJitterBuffer))
#define RTP_JITTER_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_JITTER_BUFFER,RTPJitterBufferClass))
#define RTP_IS_JITTER_BUFFER(src) (G_TYPE_CHECK_INSTANCE_TYPE((src),RTP_TYPE_JITTER_BUFFER))
#define RTP_IS_JITTER_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_JITTER_BUFFER))
#define RTP_JITTER_BUFFER_CAST(src) ((RTPJitterBuffer *)(src))
/**
* RTPJitterBufferMode:
* @RTP_JITTER_BUFFER_MODE_NONE: don't do any skew correction, outgoing
* timestamps are calculated directly from the RTP timestamps. This mode is
* good for recording but not for real-time applications.
* @RTP_JITTER_BUFFER_MODE_SLAVE: calculate the skew between sender and receiver
* and produce smoothed adjusted outgoing timestamps. This mode is good for
* low latency communications.
* @RTP_JITTER_BUFFER_MODE_BUFFER: buffer packets between low/high watermarks.
* This mode is good for streaming communication.
* @RTP_JITTER_BUFFER_MODE_SYNCED: sender and receiver clocks are synchronized,
* like #RTP_JITTER_BUFFER_MODE_SLAVE but skew is assumed to be 0. Good for
* low latency communication when sender and receiver clocks are
* synchronized and there is thus no clock skew.
* @RTP_JITTER_BUFFER_MODE_LAST: last buffer mode.
*
* The different buffer modes for a jitterbuffer.
*/
typedef enum {
RTP_JITTER_BUFFER_MODE_NONE = 0,
RTP_JITTER_BUFFER_MODE_SLAVE = 1,
RTP_JITTER_BUFFER_MODE_BUFFER = 2,
/* FIXME 3 is missing because it was used for 'auto' in jitterbuffer */
RTP_JITTER_BUFFER_MODE_SYNCED = 4,
RTP_JITTER_BUFFER_MODE_LAST
} RTPJitterBufferMode;
#define RTP_TYPE_JITTER_BUFFER_MODE (rtp_jitter_buffer_mode_get_type())
GType rtp_jitter_buffer_mode_get_type (void);
#define RTP_JITTER_BUFFER_MAX_WINDOW 512
/**
* RTPJitterBuffer:
*
* A JitterBuffer in the #RTPSession
*/
struct _RTPJitterBuffer {
GObject object;
GQueue *packets;
RTPJitterBufferMode mode;
GstClockTime delay;
/* for buffering */
gboolean buffering;
guint64 low_level;
guint64 high_level;
/* for calculating skew */
gboolean need_resync;
GstClockTime base_time;
GstClockTime base_rtptime;
GstClockTime media_clock_base_time;
guint32 clock_rate;
GstClockTime base_extrtp;
GstClockTime prev_out_time;
guint64 ext_rtptime;
guint64 last_rtptime;
gint64 window[RTP_JITTER_BUFFER_MAX_WINDOW];
guint window_pos;
guint window_size;
gboolean window_filling;
gint64 window_min;
gint64 skew;
gint64 prev_send_diff;
gboolean buffering_disabled;
GMutex clock_lock;
GstClock *pipeline_clock;
GstClock *media_clock;
gulong media_clock_synced_id;
guint64 media_clock_offset;
gboolean rfc7273_sync;
};
struct _RTPJitterBufferClass {
GObjectClass parent_class;
};
/**
* RTPJitterBufferItem:
* @data: the data of the item
* @next: pointer to next item
* @prev: pointer to previous item
* @type: the type of @data, used freely by caller
* @dts: input DTS
* @pts: output PTS
* @seqnum: seqnum, the seqnum is used to insert the item in the
* right position in the jitterbuffer and detect duplicates. Use -1 to
* append.
* @count: amount of seqnum in this item
* @rtptime: rtp timestamp
*
* An object containing an RTP packet or event.
*/
struct _RTPJitterBufferItem {
gpointer data;
GList *next;
GList *prev;
guint type;
GstClockTime dts;
GstClockTime pts;
guint seqnum;
guint count;
guint rtptime;
};
GType rtp_jitter_buffer_get_type (void);
/* managing lifetime */
RTPJitterBuffer* rtp_jitter_buffer_new (void);
RTPJitterBufferMode rtp_jitter_buffer_get_mode (RTPJitterBuffer *jbuf);
void rtp_jitter_buffer_set_mode (RTPJitterBuffer *jbuf, RTPJitterBufferMode mode);
GstClockTime rtp_jitter_buffer_get_delay (RTPJitterBuffer *jbuf);
void rtp_jitter_buffer_set_delay (RTPJitterBuffer *jbuf, GstClockTime delay);
void rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer *jbuf, guint32 clock_rate);
guint32 rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer *jbuf);
void rtp_jitter_buffer_set_media_clock (RTPJitterBuffer *jbuf, GstClock * clock, guint64 clock_offset);
void rtp_jitter_buffer_set_pipeline_clock (RTPJitterBuffer *jbuf, GstClock * clock);
gboolean rtp_jitter_buffer_get_rfc7273_sync (RTPJitterBuffer *jbuf);
void rtp_jitter_buffer_set_rfc7273_sync (RTPJitterBuffer *jbuf, gboolean rfc7273_sync);
void rtp_jitter_buffer_reset_skew (RTPJitterBuffer *jbuf);
gboolean rtp_jitter_buffer_insert (RTPJitterBuffer *jbuf,
RTPJitterBufferItem *item,
gboolean *head, gint *percent);
void rtp_jitter_buffer_disable_buffering (RTPJitterBuffer *jbuf, gboolean disabled);
RTPJitterBufferItem * rtp_jitter_buffer_peek (RTPJitterBuffer *jbuf);
RTPJitterBufferItem * rtp_jitter_buffer_pop (RTPJitterBuffer *jbuf, gint *percent);
void rtp_jitter_buffer_flush (RTPJitterBuffer *jbuf,
GFunc free_func, gpointer user_data);
gboolean rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf);
void rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering);
gint rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf);
guint rtp_jitter_buffer_num_packets (RTPJitterBuffer *jbuf);
guint32 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer *jbuf);
void rtp_jitter_buffer_get_sync (RTPJitterBuffer *jbuf, guint64 *rtptime,
guint64 *timestamp, guint32 *clock_rate,
guint64 *last_rtptime);
GstClockTime rtp_jitter_buffer_calculate_pts (RTPJitterBuffer * jbuf, GstClockTime dts, gboolean estimated_dts,
guint32 rtptime, GstClockTime base_time);
#endif /* __RTP_JITTER_BUFFER_H__ */