gst/rtpmanager/: Corrected a typo (interpollate -> interpolate).

Original commit message from CVS:
* ChangeLog:
* gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/rtpsource.c: (rtp_source_get_new_sr):
Corrected a typo (interpollate -> interpolate).
This commit is contained in:
Peter Kjellerstedt 2008-07-03 14:44:51 +00:00
parent e6d85e6a1e
commit 4e2dcf4015
3 changed files with 12 additions and 5 deletions

View file

@ -1,3 +1,10 @@
2008-07-03 Peter Kjellerstedt <pkj@axis.com>
* ChangeLog:
* gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/rtpsource.c: (rtp_source_get_new_sr):
Corrected a typo (interpollate -> interpolate).
2008-07-03 Peter Kjellerstedt <pkj@axis.com>
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
@ -7110,7 +7117,7 @@
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Handle cases where input buffers have no timestamps so that no clock
skew can be calculated, in this case interpollate timestamps based on
skew can be calculated, in this case interpolate timestamps based on
rtp timestamp and assume a 0 clock skew.
2007-09-28 Wim Taymans <wim.taymans@gmail.com>
@ -7647,7 +7654,7 @@
Make the clock skew window a little bigger.
Apply the clock skew to all buffers, not just one with a new timestamp.
Calculate and debug sender clock drift.
Use extended last timestamp to interpollate for SR reports.
Use extended last timestamp to interpolate for SR reports.
2007-09-12 Tim-Philipp Müller <tim at centricular dot net>

View file

@ -1128,7 +1128,7 @@ again:
GST_DEBUG_OBJECT (jitterbuffer,
"out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
/* interpollate between the current time and the last time based on
/* interpolate between the current time and the last time based on
* number of packets we are missing, this is the estimated duration
* for the missing packet based on equidistant packet spacing. Also make
* sure we never go negative. */

View file

@ -1184,7 +1184,7 @@ rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
/* use the sync params to interpollate the date->time member to rtptime. We
/* use the sync params to interpolate the date->time member to rtptime. We
* use the last sent timestamp and rtptime as reference points. We assume
* that the slope of the rtptime vs timestamp curve is 1, which is certainly
* sufficient for the frequency at which we report SR and the rate we send
@ -1211,7 +1211,7 @@ rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
}
} else {
GST_WARNING ("no clock-rate, cannot interpollate rtp time");
GST_WARNING ("no clock-rate, cannot interpolate rtp time");
}
/* convert the NTP time in nanoseconds to 32.32 fixed point */