Chromium Code Reviews| Index: webrtc/video/vie_sync_module.h |
| diff --git a/webrtc/video/vie_sync_module.h b/webrtc/video/vie_sync_module.h |
| index 5724ce799a7307caba2c0c0162b14f1b72affefe..b08dd8a79965b252d0f746dfda67db76eb6ab268 100644 |
| --- a/webrtc/video/vie_sync_module.h |
| +++ b/webrtc/video/vie_sync_module.h |
| @@ -23,8 +23,10 @@ |
| namespace webrtc { |
| +class Clock; |
| class RtpRtcp; |
| class VideoCodingModule; |
| +class VideoFrame; |
| class ViEChannel; |
| class VoEVideoSync; |
| @@ -42,9 +44,14 @@ class ViESyncModule : public Module { |
| int64_t TimeUntilNextProcess() override; |
| void Process() override; |
| + // Returns the absolute value of the sync offset between the video |frame| and |
| + // the current played out audio frame (or -1 on error). |
|
pbos-webrtc
2016/03/04 14:34:27
-1 if not configured/in use?
åsapersson
2016/03/09 15:44:35
Done.
|
| + int64_t GetStreamSyncOffsetInMs(const VideoFrame& frame); |
|
stefan-webrtc
2016/03/04 14:32:17
Would it be reasonable to request a unittest here
åsapersson
2016/03/09 15:44:35
I will look into adding unittest for this class. M
stefan-webrtc
2016/03/09 15:59:30
Sure no problem.
|
| + |
| private: |
| rtc::CriticalSection data_cs_; |
| VideoCodingModule* const vcm_; |
| + Clock* const clock_; |
| RtpReceiver* video_receiver_; |
| RtpRtcp* video_rtp_rtcp_; |
| int voe_channel_id_; |