| 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)
|
| 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)
|
| 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
|
|
|