Chromium Code Reviews| 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 |