Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(728)

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

Issue 2257413002: Replace interface VideoCapturerInput with VideoSinkInterface. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix EXPECT_GT order. Created 4 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/media/engine/webrtcvideoengine2.h
diff --git a/webrtc/media/engine/webrtcvideoengine2.h b/webrtc/media/engine/webrtcvideoengine2.h
index 4f8fcd83372c65c8d3894b50abb383c9f065fb40..c38d997a31964f02ed5b388f901942e833d6a413 100644
--- a/webrtc/media/engine/webrtcvideoengine2.h
+++ b/webrtc/media/engine/webrtcvideoengine2.h
@@ -243,6 +243,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
// frames are then converted from cricket frames to webrtc frames.
class WebRtcVideoSendStream
: public rtc::VideoSinkInterface<cricket::VideoFrame>,
+ public rtc::VideoSourceInterface<webrtc::VideoFrame>,
public webrtc::LoadObserver {
public:
WebRtcVideoSendStream(
@@ -262,6 +263,11 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
bool SetRtpParameters(const webrtc::RtpParameters& parameters);
webrtc::RtpParameters GetRtpParameters() const;
+ // Implements rtc::VideoSourceInterface<webrtc::VideoFrame>
stefan-webrtc 2016/09/06 12:24:38 I think we should have a comment saying what the p
perkj_webrtc 2016/09/07 15:10:37 Done.
+ void AddOrUpdateSink(VideoSinkInterface<webrtc::VideoFrame>* sink,
+ const rtc::VideoSinkWants& wants) override;
+ void RemoveSink(VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+
void OnFrame(const cricket::VideoFrame& frame) override;
bool SetVideoSend(bool mute,
const VideoOptions* options,
@@ -389,6 +395,8 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
rtc::CriticalSection lock_;
webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
+ rtc::VideoSinkInterface<webrtc::VideoFrame>* encoder_sink_
+ GUARDED_BY(lock_);
// Contains settings that are the same for all streams in the MediaChannel,
// such as codecs, header extensions, and the global bitrate limit for the
// entire channel.

Powered by Google App Engine
This is Rietveld 408576698