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-rw-r--r--ChangeLog11
-rw-r--r--gst/rtpmanager/gstrtpjitterbuffer.c2
-rw-r--r--gst/rtpmanager/rtpsource.c4
3 files changed, 12 insertions, 5 deletions
diff --git a/ChangeLog b/ChangeLog
index f463c12d..9fc54b27 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,12 @@
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),
(gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp),
(gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_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>
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index a4ec5b48..9b5e5236 100644
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ b/gst/rtpmanager/gstrtpjitterbuffer.c
@@ -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. */
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index 50170d10..4c01a7ad 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -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 */