Update libjingle to CL 53398036.
Review URL: https://webrtc-codereview.appspot.com/2323004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4872 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/base/rtpdataengine.cc b/talk/media/base/rtpdataengine.cc
index 2a23c10..1254b29 100644
--- a/talk/media/base/rtpdataengine.cc
+++ b/talk/media/base/rtpdataengine.cc
@@ -342,10 +342,6 @@
<< "; already sent " << send_limiter_->used_in_period()
<< "/" << send_limiter_->max_per_period();
return false;
- } else {
- LOG(LS_VERBOSE) << "Sending data packet of len=" << packet_len
- << "; already sent " << send_limiter_->used_in_period()
- << "/" << send_limiter_->max_per_period();
}
RtpHeader header;
@@ -363,12 +359,12 @@
packet.AppendData(&kReservedSpace, sizeof(kReservedSpace));
packet.AppendData(payload.data(), payload.length());
- // Uncomment this for easy debugging.
- // LOG(LS_INFO) << "Sent packet: "
- // << " stream=" << found_stream.id
- // << ", seqnum=" << header.seq_num
- // << ", timestamp=" << header.timestamp
- // << ", len=" << data_len;
+ LOG(LS_VERBOSE) << "Sent RTP data packet: "
+ << " stream=" << found_stream.id
+ << " ssrc=" << header.ssrc
+ << ", seqnum=" << header.seq_num
+ << ", timestamp=" << header.timestamp
+ << ", len=" << payload.length();
MediaChannel::SendPacket(&packet);
send_limiter_->Use(packet_len, now);