| Index: webrtc/pc/peerconnection.cc
|
| diff --git a/webrtc/pc/peerconnection.cc b/webrtc/pc/peerconnection.cc
|
| index 7111b1c53d5925cae4ddfc742a7da2a571840ee2..12891df16bedad5253c0fcc356404aba41f67ef9 100644
|
| --- a/webrtc/pc/peerconnection.cc
|
| +++ b/webrtc/pc/peerconnection.cc
|
| @@ -391,17 +391,20 @@ bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints,
|
| return mandatory_constraints_satisfied == constraints->GetMandatory().size();
|
| }
|
|
|
| -PeerConnection::PeerConnection(PeerConnectionFactory* factory)
|
| +PeerConnection::PeerConnection(PeerConnectionFactory* factory,
|
| + std::unique_ptr<RtcEventLog> event_log,
|
| + std::unique_ptr<Call> call)
|
| : factory_(factory),
|
| observer_(NULL),
|
| uma_observer_(NULL),
|
| - event_log_(RtcEventLog::Create()),
|
| + event_log_(std::move(event_log)),
|
| signaling_state_(kStable),
|
| ice_connection_state_(kIceConnectionNew),
|
| ice_gathering_state_(kIceGatheringNew),
|
| rtcp_cname_(GenerateRtcpCname()),
|
| local_streams_(StreamCollection::Create()),
|
| - remote_streams_(StreamCollection::Create()) {}
|
| + remote_streams_(StreamCollection::Create()),
|
| + call_(std::move(call)) {}
|
|
|
| PeerConnection::~PeerConnection() {
|
| TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
|
| @@ -460,10 +463,6 @@ bool PeerConnection::Initialize(
|
| return false;
|
| }
|
|
|
| - // Call must be constructed on the worker thread.
|
| - factory_->worker_thread()->Invoke<void>(
|
| - RTC_FROM_HERE, rtc::Bind(&PeerConnection::CreateCall_w,
|
| - this));
|
|
|
| session_.reset(new WebRtcSession(
|
| call_.get(), factory_->channel_manager(), configuration.media_config,
|
| @@ -2375,21 +2374,4 @@ void PeerConnection::StopRtcEventLog_w() {
|
| }
|
| }
|
|
|
| -void PeerConnection::CreateCall_w() {
|
| - RTC_DCHECK(!call_);
|
| -
|
| - const int kMinBandwidthBps = 30000;
|
| - const int kStartBandwidthBps = 300000;
|
| - const int kMaxBandwidthBps = 2000000;
|
| -
|
| - webrtc::Call::Config call_config(event_log_.get());
|
| - call_config.audio_state =
|
| - 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));
|
| -}
|
| -
|
| } // namespace webrtc
|
|
|