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Update libjingle to CL 53398036.

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Review URL: https://webrtc-codereview.appspot.com/2323004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4872 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mallinath@webrtc.org
2013-09-27 23:04:10 +00:00
parent 34c50c1de1
commit a27be8e4a1
41 changed files with 496 additions and 436 deletions
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16
talk/media/base/rtpdataengine.cc
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@@ -342,10 +342,6 @@ bool RtpDataMediaChannel::SendData(
<< "; 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 @@ bool RtpDataMediaChannel::SendData(
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);