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).

diff --git a/ChangeLog b/ChangeLog
index f463c12da070da7b37a442f76536681872cf8885..9fc54b27e3bbdaa567f5b2ddc326b18b6a7a128f 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -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>
 

diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index a4ec5b487b9ab5bba1c15f8d990a90ab9a4c5ac0..9b5e523636dd9f01a2cb8928bb48361c4ad60d6e 100644 (file)
--- 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 50170d10374a35ed49c1d9f48e7b440e6cfe1f3f..4c01a7ad5884451341672f1f196ee8f82177b881 100644 (file)
--- 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 */