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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-20 23:36:38 +00:00
20a27a545a
Add signal to pause the jitterbuffer. This will be emitted from gstrtpbin when one of the jitterbuffers is buffering. Make rtpbin collect the buffering messages and post a new buffering message with the min value. Remove the stats callback from jitterbuffer but pass a percent integer to functions that affect the buffering state of the jitterbuffer. This allows us then to post buffering messages from outside of the jitterbuffer lock.
134 lines
5.2 KiB
C
134 lines
5.2 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_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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#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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*
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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_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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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 level;
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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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GstClockTime base_time;
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GstClockTime base_rtptime;
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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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};
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struct _RTPJitterBufferClass {
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GObjectClass parent_class;
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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_reset_skew (RTPJitterBuffer *jbuf);
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gboolean rtp_jitter_buffer_insert (RTPJitterBuffer *jbuf, GstBuffer *buf,
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GstClockTime time,
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guint32 clock_rate,
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gboolean *tail, gint *percent);
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GstBuffer * rtp_jitter_buffer_peek (RTPJitterBuffer *jbuf);
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GstBuffer * rtp_jitter_buffer_pop (RTPJitterBuffer *jbuf, gint *percent);
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void rtp_jitter_buffer_flush (RTPJitterBuffer *jbuf);
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gboolean rtp_jitter_buffer_is_buffering (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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#endif /* __RTP_JITTER_BUFFER_H__ */
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