[FFmpeg-cvslog] rtpdec: Use the right logging context

Diego Biurrun git at videolan.org
Wed Feb 24 19:25:46 CET 2016


ffmpeg | branch: master | Diego Biurrun <diego at biurrun.de> | Thu Feb 18 11:08:11 2016 +0100| [8d918a98aa24134a043d578ef45bae363dbed9db] | committer: Diego Biurrun

rtpdec: Use the right logging context

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8d918a98aa24134a043d578ef45bae363dbed9db
---

 libavformat/rtpdec.c |   18 ++++++++----------
 1 file changed, 8 insertions(+), 10 deletions(-)

diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 823e03c..dfeb529 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -143,8 +143,7 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
         switch (buf[1]) {
         case RTCP_SR:
             if (payload_len < 20) {
-                av_log(NULL, AV_LOG_ERROR,
-                       "Invalid length for RTCP SR packet\n");
+                av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n");
                 return AVERROR_INVALIDDATA;
             }
 
@@ -514,8 +513,8 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
     s->st                  = st;
     s->queue_size          = queue_size;
 
-    av_log(s->st ? s->st->codec : NULL, AV_LOG_VERBOSE,
-            "setting jitter buffer size to %d\n", s->queue_size);
+    av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
+           s->queue_size);
 
     rtp_init_statistics(&s->statistics, 0);
     if (st) {
@@ -616,7 +615,7 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
     st = s->st;
     // only do something with this if all the rtp checks pass...
     if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
-        av_log(st ? st->codec : NULL, AV_LOG_ERROR,
+        av_log(s->ic, AV_LOG_ERROR,
                "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
                payload_type, seq, ((s->seq + 1) & 0xffff));
         return -1;
@@ -730,7 +729,7 @@ static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
         return -1;
 
     if (!has_next_packet(s))
-        av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
+        av_log(s->ic, AV_LOG_WARNING,
                "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
 
     /* Parse the first packet in the queue, and dequeue it */
@@ -797,7 +796,7 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
         int16_t diff = seq - s->seq;
         if (diff < 0) {
             /* Packet older than the previously emitted one, drop */
-            av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
+            av_log(s->ic, AV_LOG_WARNING,
                    "RTP: dropping old packet received too late\n");
             return -1;
         } else if (diff <= 1) {
@@ -813,8 +812,7 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
             /* Return the first enqueued packet if the queue is full,
              * even if we're missing something */
             if (s->queue_len >= s->queue_size) {
-                av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
-                       "jitter buffer full\n");
+                av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
                 return rtp_parse_queued_packet(s, pkt);
             }
             return -1;
@@ -864,7 +862,7 @@ int ff_parse_fmtp(AVFormatContext *s,
     int value_size = strlen(p) + 1;
 
     if (!(value = av_malloc(value_size))) {
-        av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
+        av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
         return AVERROR(ENOMEM);
     }
 



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