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Replace flooding logs in rtp_sender.cc with a comment.

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Started occurring after:
https://webrtc-codereview.appspot.com/11129004

BUG=3153
R=andresp@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5916 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2014-04-15 21:26:34 +00:00
parent 36eda7cf0e
commit 2c3f1abb69
1 changed files with 2 additions and 2 deletions
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4
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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@ -1295,7 +1295,7 @@ bool RTPSender::UpdateAudioLevel(uint8_t *rtp_packet,
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionAudioLevel);
if (extension_block_pos < 0) {
LOG(LS_WARNING) << "Failed to update audio level, not registered.";
// The feature is not enabled.
return false;
}
int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
@ -1337,7 +1337,7 @@ bool RTPSender::UpdateAbsoluteSendTime(
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionAbsoluteSendTime);
if (extension_block_pos < 0) {
LOG(LS_WARNING) << "Failed to update absolute send time, not registered.";
// The feature is not enabled.
return false;
}
int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;