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c88bfc0b3e
When syncing to an RFC7273 clock this will add the original reconstructed reference clock timestamp to buffers in form of a GstReferenceTimestampMeta. This is useful when we want to process or analyse data based on the original timestamps untainted by any local adjustments, for example reconstruct AES67 audio streams with sample accuracy. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1964>
222 lines
9 KiB
C
222 lines
9 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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#define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
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#define ITEM_TYPE_BUFFER 0
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#define ITEM_TYPE_LOST 1
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#define ITEM_TYPE_EVENT 2
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#define ITEM_TYPE_QUERY 3
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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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* @data_free: Function to free @data (optional)
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*
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* An object containing an RTP packet or event. First members of this structure
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* copied from GList so they can be inserted into lists without doing more
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* allocations.
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*/
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struct _RTPJitterBufferItem {
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/* a GList */
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gpointer data;
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GList *next;
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GList *prev;
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/* item metadata */
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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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GDestroyNotify free_data;
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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_append_event (RTPJitterBuffer * jbuf, GstEvent * event);
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gboolean rtp_jitter_buffer_append_query (RTPJitterBuffer * jbuf, GstQuery * query);
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gboolean rtp_jitter_buffer_append_lost_event (RTPJitterBuffer * jbuf, GstEvent * event,
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guint16 seqnum, guint lost_packets);
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gboolean rtp_jitter_buffer_append_buffer (RTPJitterBuffer * jbuf, GstBuffer * buf,
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GstClockTime dts, GstClockTime pts,
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guint16 seqnum, guint rtptime,
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gboolean * duplicate, 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, gint gap,
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gboolean is_rtx, GstClockTime * p_ntp_time);
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gboolean rtp_jitter_buffer_can_fast_start (RTPJitterBuffer * jbuf, gint num_packet);
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gboolean rtp_jitter_buffer_is_full (RTPJitterBuffer * jbuf);
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void rtp_jitter_buffer_free_item (RTPJitterBufferItem * item);
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#endif /* __RTP_JITTER_BUFFER_H__ */
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