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

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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 committed by Tim-Philipp Müller
parent e2f49d9ccf
commit fd44690d4f
2 changed files with 3 additions and 3 deletions
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2
gst/rtpmanager/gstrtpjitterbuffer.c
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@ -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. */

4
gst/rtpmanager/rtpsource.c
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 */