| Index: webrtc/media/engine/webrtcvideoengine2.h
|
| diff --git a/webrtc/media/engine/webrtcvideoengine2.h b/webrtc/media/engine/webrtcvideoengine2.h
|
| index 2c7d36ccb5cc820b5d20fa6317d1dbc950ca0f49..c40014a39e95c6b606587893916825b88568b0ad 100644
|
| --- a/webrtc/media/engine/webrtcvideoengine2.h
|
| +++ b/webrtc/media/engine/webrtcvideoengine2.h
|
| @@ -22,13 +22,14 @@
|
| #include "webrtc/base/asyncinvoker.h"
|
| #include "webrtc/base/criticalsection.h"
|
| #include "webrtc/base/networkroute.h"
|
| +#include "webrtc/base/optional.h"
|
| #include "webrtc/base/thread_annotations.h"
|
| #include "webrtc/base/thread_checker.h"
|
| -#include "webrtc/media/base/videosinkinterface.h"
|
| -#include "webrtc/media/base/videosourceinterface.h"
|
| #include "webrtc/call/call.h"
|
| #include "webrtc/call/flexfec_receive_stream.h"
|
| #include "webrtc/media/base/mediaengine.h"
|
| +#include "webrtc/media/base/videosinkinterface.h"
|
| +#include "webrtc/media/base/videosourceinterface.h"
|
| #include "webrtc/media/engine/webrtcvideodecoderfactory.h"
|
| #include "webrtc/media/engine/webrtcvideoencoderfactory.h"
|
| #include "webrtc/video_receive_stream.h"
|
| @@ -81,15 +82,12 @@ class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
|
| uint32_t ssrc) override;
|
|
|
| rtc::VideoSinkInterface<webrtc::VideoFrame>* GetDefaultSink() const;
|
| - void SetDefaultSink(VideoMediaChannel* channel,
|
| + void SetDefaultSink(WebRtcVideoChannel2* channel,
|
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink);
|
|
|
| - uint32_t default_recv_ssrc() const { return default_recv_ssrc_; }
|
| -
|
| virtual ~DefaultUnsignalledSsrcHandler() = default;
|
|
|
| private:
|
| - uint32_t default_recv_ssrc_;
|
| rtc::VideoSinkInterface<webrtc::VideoFrame>* default_sink_;
|
| };
|
|
|
| @@ -177,6 +175,8 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
|
| // Implemented for VideoMediaChannelTest.
|
| bool sending() const { return sending_; }
|
|
|
| + rtc::Optional<uint32_t> GetDefaultReceiveStreamSsrc();
|
| +
|
| // AdaptReason is used for expressing why a WebRtcVideoSendStream request
|
| // a lower input frame size than the currently configured camera input frame
|
| // size. There can be more than one reason OR:ed together.
|
|
|