| Index: webrtc/pc/peerconnection.cc
|
| diff --git a/webrtc/pc/peerconnection.cc b/webrtc/pc/peerconnection.cc
|
| index 852dd39de5bc21eed1461340f796d4ef0132e55f..eed4c0d974bf03ed9c3a7af249b6b36502cd099f 100644
|
| --- a/webrtc/pc/peerconnection.cc
|
| +++ b/webrtc/pc/peerconnection.cc
|
| @@ -220,6 +220,10 @@ bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
|
|
|
| namespace webrtc {
|
|
|
| +std::unique_ptr<RtcEventLog> CreateRtcEventLog();
|
| +
|
| +Call* CreateCall(const Call::Config& config);
|
| +
|
| bool PeerConnectionInterface::RTCConfiguration::operator==(
|
| const PeerConnectionInterface::RTCConfiguration& o) const {
|
| // This static_assert prevents us from accidentally breaking operator==.
|
| @@ -395,7 +399,7 @@ PeerConnection::PeerConnection(PeerConnectionFactory* factory)
|
| : factory_(factory),
|
| observer_(NULL),
|
| uma_observer_(NULL),
|
| - event_log_(RtcEventLog::Create()),
|
| + event_log_(CreateRtcEventLog()),
|
| signaling_state_(kStable),
|
| ice_connection_state_(kIceConnectionNew),
|
| ice_gathering_state_(kIceGatheringNew),
|
| @@ -2335,13 +2339,17 @@ void PeerConnection::CreateCall_w() {
|
| const int kMaxBandwidthBps = 2000000;
|
|
|
| webrtc::Call::Config call_config(event_log_.get());
|
| + if (!factory_->channel_manager()->media_engine()) {
|
| + LOG(LS_WARNING)
|
| + << "Failed to create the Call because the media engine is unset.";
|
| + return;
|
| + }
|
| call_config.audio_state =
|
| - factory_->channel_manager() ->media_engine()->GetAudioState();
|
| + factory_->channel_manager()->media_engine()->GetAudioState();
|
| call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
|
| call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
|
| call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
|
| -
|
| - call_.reset(webrtc::Call::Create(call_config));
|
| + call_.reset(CreateCall(call_config));
|
| }
|
|
|
| } // namespace webrtc
|
|
|