[asterisk-commits] oej: branch oej/chocolate-video-congestion-11 r433546 - /team/oej/chocolate-v...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Mar 27 04:40:16 CDT 2015
Author: oej
Date: Fri Mar 27 04:40:12 2015
New Revision: 433546
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=433546
Log:
Strict compilers are ... strict.
Modified:
team/oej/chocolate-video-congestion-11/res/res_rtp_asterisk.c
Modified: team/oej/chocolate-video-congestion-11/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/chocolate-video-congestion-11/res/res_rtp_asterisk.c?view=diff&rev=433546&r1=433545&r2=433546
==============================================================================
--- team/oej/chocolate-video-congestion-11/res/res_rtp_asterisk.c (original)
+++ team/oej/chocolate-video-congestion-11/res/res_rtp_asterisk.c Fri Mar 27 04:40:12 2015
@@ -4564,18 +4564,18 @@
if (rtp->f.seqno != seqno && rtp->lastrxts == timestamp) {
/* This is a new part of a larger video frame sent in multiple RTP payloads */
/* We need to count these and when the frame is over, send to the bitrate estimator */
- if (lastrxts_reuse == 0) {
- lastrxts_reuse=2;
+ if (rtp->lastrxts_reuse == 0) {
+ rtp->lastrxts_reuse=2;
} else {
- lastrxts_reuse++;
+ rtp->lastrxts_reuse++;
}
- if (multi_payload_size == 0) {
+ if (rtp->multi_payload_size == 0) {
/* Second frame */
- multi_payload_size = rtp->f.datalen + (res - hdrlen);
+ rtp->multi_payload_size = rtp->f.datalen + (res - hdrlen);
} else {
- multi_payload_size += res - hdrlen;
+ rtp->multi_payload_size += res - hdrlen;
}
- multi_payload_startts = rtp->lastrxts; /* When the first packet arrived to us */
+ rtp->multi_payload_startts = rtp->lastrxts; /* When the first packet arrived to us */
/* OEJ question: How do we measure the relative transmission time, network wise ?
Answer is propably hidden in RTCP code in this file.
*/
@@ -4589,14 +4589,11 @@
/* We have a new frame */
unsigned int transmissiontime = 0;
unsigned int bitspersec = 0;
- if (lastrxts_reuse) {
+ if (rtp->lastrxts_reuse) {
/* Calculate total transmission time for this payload */
- transmissiontime = timestamp - multi_payload_startts;
+ transmissiontime = timestamp - rtp->multi_payload_startts;
/* Can we get an idea of something here? */
- /* payload_size * 8 = bits
- transmissiontime = milliseconds */
-
- bitspersec = (unsigned int) (payload_size * 8) / (transmissiontime * 1000);
+ bitspersec = (unsigned int) (rtp->multi_payload_size * 8) / (transmissiontime * 1000);
} else {
lastrxts_reuse=1;
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