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jitterbuffer: add slope estimation code and debug

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Add some code to measure the sender speed vs the receiver speed. This can be
used to detect bursts.
This commit is contained in:
Wim Taymans 2009-08-31 13:02:16 +02:00
parent 4814d899c2
commit a26a2a9ff5
1 changed files with 13 additions and 4 deletions
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17
gst/rtpmanager/rtpjitterbuffer.c
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@ -203,6 +203,7 @@ calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
gint64 old;
gint pos, i;
GstClockTime gstrtptime, out_time;
guint64 slope;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
@ -258,13 +259,21 @@ calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
/* elapsed time at receiver, includes the jitter */
recv_diff = time - jbuf->base_time;
GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
GST_TIME_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (jbuf->base_time),
GST_TIME_ARGS (recv_diff));
/* measure the diff */
delta = ((gint64) recv_diff) - ((gint64) send_diff);
/* measure the slope, this gives a rought estimate between the sender speed
* and the receiver speed. This should be approximately 8, higher values
* indicate a burst (especially when the connection starts) */
if (recv_diff > 0)
slope = (send_diff * 8) / recv_diff;
else
slope = 8;
GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time),
GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope);
/* if the difference between the sender timeline and the receiver timeline
* changed too quickly we have to resync because the server likely restarted
* its timestamps. */