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jitterbuffer: make sure time does not go backwards

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When we construct a timestamp that would result in a timestamp that is earlier
than when the packet was received, reset the skew calculation as this is
probably a sign that the sender restarted or paused.

Fixes #593354
This commit is contained in:
Wim Taymans 2009-09-01 12:41:36 +02:00
parent bfb1260af4
commit 8d924611e7
3 changed files with 36 additions and 22 deletions
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24
gst/rtpmanager/gstrtpjitterbuffer.c
View file

@ -1,7 +1,7 @@
/*
* Farsight Voice+Video library
*
* Copyright 2007 Collabora Ltd,
* Copyright 2007 Collabora Ltd,
* Copyright 2007 Nokia Corporation
* @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
* Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
@ -30,17 +30,17 @@
* from a network source. It will also wait for missing packets up to a
* configurable time limit using the #GstRtpJitterBuffer:latency property.
* Packets arriving too late are considered to be lost packets.
*
*
* This element acts as a live element and so adds #GstRtpJitterBuffer:latency
* to the pipeline.
*
*
* The element needs the clock-rate of the RTP payload in order to estimate the
* delay. This information is obtained either from the caps on the sink pad or,
* when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
* To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
*
*
* This element will automatically be used inside gstrtpbin.
*
*
* <refsect2>
* <title>Example pipelines</title>
* |[
@ -300,7 +300,7 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
/**
* GstRtpJitterBuffer::latency:
*
*
* The maximum latency of the jitterbuffer. Packets will be kept in the buffer
* for at most this time.
*/
@ -310,8 +310,8 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
G_PARAM_READWRITE));
/**
* GstRtpJitterBuffer::drop-on-latency:
*
* Drop oldest buffers when the queue is completely filled.
*
* Drop oldest buffers when the queue is completely filled.
*/
g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
g_param_spec_boolean ("drop-on-latency",
@ -320,7 +320,7 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE));
/**
* GstRtpJitterBuffer::ts-offset:
*
*
* Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
* This is mainly used to ensure interstream synchronisation.
*/
@ -332,7 +332,7 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
/**
* GstRtpJitterBuffer::do-lost:
*
*
* Send out a GstRTPPacketLost event downstream when a packet is considered
* lost.
*/
@ -761,7 +761,7 @@ gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
/* this unblocks any waiting pops on the src pad task */
JBUF_SIGNAL (priv);
/* unlock clock, we just unschedule, the entry will be released by the
/* unlock clock, we just unschedule, the entry will be released by the
* locking streaming thread. */
if (priv->clock_id) {
gst_clock_id_unschedule (priv->clock_id);
@ -1211,7 +1211,7 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
* FALSE if a packet with the same seqnum was already in the queue, meaning we
* have a duplicate. */
if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
priv->clock_rate, &tail)))
priv->clock_rate, (priv->latency_ms * GST_MSECOND), &tail)))
goto duplicate;
/* signal addition of new buffer when the _loop is waiting. */

24
gst/rtpmanager/rtpjitterbuffer.c
View file

@ -183,7 +183,7 @@ rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time,
*
* Both the window and the weighting used for averaging influence the accuracy
* of the drift estimation. Finding the correct parameters turns out to be a
* compromise between accuracy and inertia.
* compromise between accuracy and inertia.
*
* We use a 2 second window or up to 512 data points, which is statistically big
* enough to catch spikes (FIXME, detect spikes).
@ -195,7 +195,7 @@ rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time,
*/
static GstClockTime
calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
guint32 clock_rate)
guint32 clock_rate, GstClockTime max_delay)
{
guint64 ext_rtptime;
guint64 send_diff, recv_diff;
@ -278,7 +278,7 @@ calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
* changed too quickly we have to resync because the server likely restarted
* its timestamps. */
if (ABS (delta - jbuf->skew) > GST_SECOND) {
GST_WARNING ("delta %" GST_TIME_FORMAT " too big, reset skew",
GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
GST_TIME_ARGS (delta - jbuf->skew));
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
send_diff = 0;
@ -386,6 +386,18 @@ no_skew:
out_time = jbuf->prev_out_time;
}
}
if (out_time + max_delay < time) {
/* if we are going to produce a timestamp that is later than the input
* timestamp, we need to reset the jitterbuffer. Likely the server paused
* temporarily */
GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (out_time),
max_delay, GST_TIME_ARGS (time));
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
out_time = time;
send_diff = 0;
}
} else
out_time = -1;
@ -404,6 +416,7 @@ no_skew:
* @buf: a buffer
* @time: a running_time when this buffer was received in nanoseconds
* @clock_rate: the clock-rate of the payload of @buf
* @max_delay: the maximum lateness of @buf
* @tail: TRUE when the tail element changed.
*
* Inserts @buf into the packet queue of @jbuf. The sequence number of the
@ -415,7 +428,8 @@ no_skew:
*/
gboolean
rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf,
GstClockTime time, guint32 clock_rate, gboolean * tail)
GstClockTime time, guint32 clock_rate, GstClockTime max_delay,
gboolean * tail)
{
GList *list;
guint32 rtptime;
@ -449,7 +463,7 @@ rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf,
* receive time, this function will retimestamp @buf with the skew corrected
* running time. */
rtptime = gst_rtp_buffer_get_timestamp (buf);
time = calculate_skew (jbuf, rtptime, time, clock_rate);
time = calculate_skew (jbuf, rtptime, time, clock_rate, max_delay);
GST_BUFFER_TIMESTAMP (buf) = time;
/* It's more likely that the packet was inserted in the front of the buffer */

10
gst/rtpmanager/rtpjitterbuffer.h
View file

@ -82,9 +82,10 @@ RTPJitterBuffer* rtp_jitter_buffer_new (void);
void rtp_jitter_buffer_reset_skew (RTPJitterBuffer *jbuf);
gboolean rtp_jitter_buffer_insert (RTPJitterBuffer *jbuf, GstBuffer *buf,
GstClockTime time,
guint32 clock_rate,
gboolean *tail);
GstClockTime time,
guint32 clock_rate,
GstClockTime max_delay,
gboolean *tail);
GstBuffer * rtp_jitter_buffer_peek (RTPJitterBuffer *jbuf);
GstBuffer * rtp_jitter_buffer_pop (RTPJitterBuffer *jbuf);
@ -95,7 +96,6 @@ 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);
guint64 *last_rtptime);
#endif /* __RTP_JITTER_BUFFER_H__ */