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Unified Diff: talk/media/webrtc/webrtcvideoengine2.h

Issue 1594973006: New rtc::VideoSinkInterface. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Workaround to not break chrome. Created 4 years, 11 months ago
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Index: talk/media/webrtc/webrtcvideoengine2.h
diff --git a/talk/media/webrtc/webrtcvideoengine2.h b/talk/media/webrtc/webrtcvideoengine2.h
index 6bace03c20fdbcf2db835f958ecdd81b81564d7b..b5efa9516dae76e64ea39382a49dd6ba53baa6bb 100644
--- a/talk/media/webrtc/webrtcvideoengine2.h
+++ b/talk/media/webrtc/webrtcvideoengine2.h
@@ -40,6 +40,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
+#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/call.h"
#include "webrtc/transport.h"
#include "webrtc/video_frame.h"
@@ -414,7 +415,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
bool SmoothsRenderedFrames() const override;
bool IsDefaultStream() const;
- void SetRenderer(cricket::VideoRenderer* renderer);
+ void SetSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink);
VideoReceiverInfo GetVideoReceiverInfo();
@@ -452,18 +453,18 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
const bool disable_prerenderer_smoothing_;
- rtc::CriticalSection renderer_lock_;
- cricket::VideoRenderer* renderer_ GUARDED_BY(renderer_lock_);
- int last_width_ GUARDED_BY(renderer_lock_);
- int last_height_ GUARDED_BY(renderer_lock_);
+ rtc::CriticalSection sink_lock_;
+ rtc::VideoSinkInterface<cricket::VideoFrame>* sink_ GUARDED_BY(sink_lock_);
+ int last_width_ GUARDED_BY(sink_lock_);
+ int last_height_ GUARDED_BY(sink_lock_);
// Expands remote RTP timestamps to int64_t to be able to estimate how long
// the stream has been running.
rtc::TimestampWrapAroundHandler timestamp_wraparound_handler_
- GUARDED_BY(renderer_lock_);
- int64_t first_frame_timestamp_ GUARDED_BY(renderer_lock_);
+ GUARDED_BY(sink_lock_);
+ int64_t first_frame_timestamp_ GUARDED_BY(sink_lock_);
// Start NTP time is estimated as current remote NTP time (estimated from
// RTCP) minus the elapsed time, as soon as remote NTP time is available.
- int64_t estimated_remote_start_ntp_time_ms_ GUARDED_BY(renderer_lock_);
+ int64_t estimated_remote_start_ntp_time_ms_ GUARDED_BY(sink_lock_);
};
void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine);
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