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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).
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3 changed files with 12 additions and 5 deletions
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ChangeLog
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ChangeLog
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@ -1,3 +1,10 @@
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2008-07-03 Peter Kjellerstedt <pkj@axis.com>
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* ChangeLog:
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* gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop):
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* gst/rtpmanager/rtpsource.c: (rtp_source_get_new_sr):
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Corrected a typo (interpollate -> interpolate).
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2008-07-03 Peter Kjellerstedt <pkj@axis.com>
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2008-07-03 Peter Kjellerstedt <pkj@axis.com>
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* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
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* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
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@ -7110,7 +7117,7 @@
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* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
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* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
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Handle cases where input buffers have no timestamps so that no clock
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Handle cases where input buffers have no timestamps so that no clock
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skew can be calculated, in this case interpollate timestamps based on
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skew can be calculated, in this case interpolate timestamps based on
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rtp timestamp and assume a 0 clock skew.
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rtp timestamp and assume a 0 clock skew.
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2007-09-28 Wim Taymans <wim.taymans@gmail.com>
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2007-09-28 Wim Taymans <wim.taymans@gmail.com>
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@ -7647,7 +7654,7 @@
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Make the clock skew window a little bigger.
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Make the clock skew window a little bigger.
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Apply the clock skew to all buffers, not just one with a new timestamp.
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Apply the clock skew to all buffers, not just one with a new timestamp.
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Calculate and debug sender clock drift.
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Calculate and debug sender clock drift.
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Use extended last timestamp to interpollate for SR reports.
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Use extended last timestamp to interpolate for SR reports.
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2007-09-12 Tim-Philipp Müller <tim at centricular dot net>
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2007-09-12 Tim-Philipp Müller <tim at centricular dot net>
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@ -1128,7 +1128,7 @@ again:
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GST_DEBUG_OBJECT (jitterbuffer,
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GST_DEBUG_OBJECT (jitterbuffer,
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"out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
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"out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
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GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
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GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
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/* interpollate between the current time and the last time based on
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/* interpolate between the current time and the last time based on
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* number of packets we are missing, this is the estimated duration
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* number of packets we are missing, this is the estimated duration
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* for the missing packet based on equidistant packet spacing. Also make
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* for the missing packet based on equidistant packet spacing. Also make
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* sure we never go negative. */
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* sure we never go negative. */
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@ -1184,7 +1184,7 @@ rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
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g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
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g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
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/* use the sync params to interpollate the date->time member to rtptime. We
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/* use the sync params to interpolate the date->time member to rtptime. We
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* use the last sent timestamp and rtptime as reference points. We assume
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* use the last sent timestamp and rtptime as reference points. We assume
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* that the slope of the rtptime vs timestamp curve is 1, which is certainly
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* that the slope of the rtptime vs timestamp curve is 1, which is certainly
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* sufficient for the frequency at which we report SR and the rate we send
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* sufficient for the frequency at which we report SR and the rate we send
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@ -1211,7 +1211,7 @@ rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
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t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
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t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
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}
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}
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} else {
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} else {
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GST_WARNING ("no clock-rate, cannot interpollate rtp time");
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GST_WARNING ("no clock-rate, cannot interpolate rtp time");
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}
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}
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/* convert the NTP time in nanoseconds to 32.32 fixed point */
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/* convert the NTP time in nanoseconds to 32.32 fixed point */
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