Chromium Code Reviews| Index: webrtc/pc/webrtcsession.h |
| diff --git a/webrtc/pc/webrtcsession.h b/webrtc/pc/webrtcsession.h |
| index a1339cbd61dd5e452404f6c386df59815109f652..32d2ebe9002d5dac8aa255dc6ae3f6ee90ab9e7c 100644 |
| --- a/webrtc/pc/webrtcsession.h |
| +++ b/webrtc/pc/webrtcsession.h |
| @@ -27,7 +27,6 @@ |
| #include "webrtc/p2p/base/candidate.h" |
| #include "webrtc/p2p/base/transportcontroller.h" |
| #include "webrtc/pc/datachannel.h" |
| -#include "webrtc/pc/mediacontroller.h" |
| #include "webrtc/pc/mediasession.h" |
| #ifdef HAVE_QUIC |
| @@ -55,6 +54,7 @@ namespace webrtc { |
| class IceRestartAnswerLatch; |
| class JsepIceCandidate; |
| class MediaStreamSignaling; |
| +class RtcEventLog; |
| class WebRtcSessionDescriptionFactory; |
| extern const char kBundleWithoutRtcpMux[]; |
| @@ -159,7 +159,9 @@ class WebRtcSession : |
| // |sctp_factory| may be null, in which case SCTP is treated as unsupported. |
| WebRtcSession( |
| - webrtc::MediaControllerInterface* media_controller, |
| + cricket::ChannelManager* channel_manager, |
| + const cricket::MediaConfig& media_config, |
| + RtcEventLog* event_log, |
| rtc::Thread* network_thread, |
| rtc::Thread* worker_thread, |
| rtc::Thread* signaling_thread, |
| @@ -362,6 +364,10 @@ class WebRtcSession : |
| } |
| #endif // HAVE_QUIC |
| + // To let tests override the Call object we use. |
| + protected: |
| + virtual Call* call() { return call_.get(); }; |
|
nisse-webrtc
2017/04/27 13:46:34
This is a bit ugly. It's needed *only* for the tes
Taylor Brandstetter
2017/04/30 07:21:29
I think it would make more sense to pass a Call in
|
| + |
| private: |
| // Indicates the type of SessionDescription in a call to SetLocalDescription |
| // and SetRemoteDescription. |
| @@ -546,6 +552,9 @@ class WebRtcSession : |
| void DestroyVoiceChannel(); |
| void DestroyDataChannel(); |
| + void Init_w(); |
| + void Close_w(); |
| + |
| rtc::Thread* const network_thread_; |
| rtc::Thread* const worker_thread_; |
| rtc::Thread* const signaling_thread_; |
| @@ -559,7 +568,9 @@ class WebRtcSession : |
| const std::unique_ptr<cricket::TransportController> transport_controller_; |
| const std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory_; |
| - MediaControllerInterface* media_controller_; |
| + const cricket::MediaConfig media_config_; |
| + RtcEventLog* event_log_; |
| + std::unique_ptr<Call> call_; |
| std::unique_ptr<cricket::VoiceChannel> voice_channel_; |
| std::unique_ptr<cricket::VideoChannel> video_channel_; |
| // |rtp_data_channel_| is used if in RTP data channel mode, |sctp_transport_| |