Update talk to 59410372.
R=jiayl@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5367 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 2aa6b8c..95e8c85 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1769,7 +1769,7 @@
}
if (dscp_option_changed) {
talk_base::DiffServCodePoint dscp = talk_base::DSCP_DEFAULT;
- if (options.dscp.GetWithDefaultIfUnset(false))
+ if (options_.dscp.GetWithDefaultIfUnset(false))
dscp = kAudioDscpValue;
if (MediaChannel::SetDscp(dscp) != 0) {
LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
@@ -1879,7 +1879,7 @@
// this, but double-check to be sure.
webrtc::CodecInst voe_codec;
if (!engine()->FindWebRtcCodec(*it, &voe_codec)) {
- LOG(LS_WARNING) << "Unknown codec " << ToString(voe_codec);
+ LOG(LS_WARNING) << "Unknown codec " << ToString(*it);
continue;
}
@@ -2431,14 +2431,14 @@
}
// Use the same SSRC as our default channel (so the RTCP reports are correct).
- unsigned int send_ssrc;
+ unsigned int send_ssrc = 0;
webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp();
if (rtp->GetLocalSSRC(voe_channel(), send_ssrc) == -1) {
- LOG_RTCERR2(GetSendSSRC, channel, send_ssrc);
+ LOG_RTCERR1(GetSendSSRC, channel);
return false;
}
if (rtp->SetLocalSSRC(channel, send_ssrc) == -1) {
- LOG_RTCERR2(SetSendSSRC, channel, send_ssrc);
+ LOG_RTCERR1(SetSendSSRC, channel);
return false;
}