Index: webrtc/pc/peerconnection.cc |
diff --git a/webrtc/pc/peerconnection.cc b/webrtc/pc/peerconnection.cc |
index 65203fdc731f1ef8e861e3971800d811fa2c61cc..c461b79757d905d6a408b2f9355d582ba687eca7 100644 |
--- a/webrtc/pc/peerconnection.cc |
+++ b/webrtc/pc/peerconnection.cc |
@@ -398,10 +398,15 @@ PeerConnection::PeerConnection(PeerConnectionFactory* factory) |
signaling_state_(kStable), |
ice_connection_state_(kIceConnectionNew), |
ice_gathering_state_(kIceGatheringNew), |
+#if defined(HAVE_WEBRTC_VOICE) && defined(HAVE_WEBRTC_VIDEO) |
Taylor Brandstetter
2017/05/11 04:43:11
||?
Zhi Huang
2017/05/12 20:05:33
Done.
|
event_log_(RtcEventLog::Create()), |
+#else |
+ event_log_(std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl())), |
+#endif |
rtcp_cname_(GenerateRtcpCname()), |
local_streams_(StreamCollection::Create()), |
- remote_streams_(StreamCollection::Create()) {} |
+ remote_streams_(StreamCollection::Create()) { |
+} |
PeerConnection::~PeerConnection() { |
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection"); |
@@ -2326,6 +2331,7 @@ void PeerConnection::StopRtcEventLog_w() { |
} |
void PeerConnection::CreateCall_w() { |
+#if defined(HAVE_WEBRTC_VOICE) && defined(HAVE_WEBRTC_VIDEO) |
Taylor Brandstetter
2017/05/11 04:43:11
||?
Zhi Huang
2017/05/12 20:05:33
Done.
|
RTC_DCHECK(!call_); |
const int kMinBandwidthBps = 30000; |
@@ -2340,6 +2346,7 @@ void PeerConnection::CreateCall_w() { |
call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps; |
call_.reset(webrtc::Call::Create(call_config)); |
+#endif |
} |
} // namespace webrtc |