| commit | 381da4be9c70dd90bddc4ac14593f8ed1eb81a3a | [log] [tgz] |
|---|---|---|
| author | mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Thu Apr 25 21:45:29 2013 +0000 |
| committer | mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Thu Apr 25 21:45:29 2013 +0000 |
| tree | 1ceaec3ae2f6fdde77c2c0712f1db2b9a2e8a38d | |
| parent | 8392cd9edd859b835f566a7eb212b075dca01deb [diff] [blame] |
VCM: Adding API for the size(duration) of the jitter buffer. Refers to the duration in time of the frames which are ready to be sent to the decoder. Review URL: https://webrtc-codereview.appspot.com/1319004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3903 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/video_coding/main/source/receiver.cc b/webrtc/modules/video_coding/main/source/receiver.cc index fffe609..65d7a68 100644 --- a/webrtc/modules/video_coding/main/source/receiver.cc +++ b/webrtc/modules/video_coding/main/source/receiver.cc
@@ -432,6 +432,10 @@ return 0; } +int VCMReceiver::RenderBufferSizeMs() { + return jitter_buffer_.RenderBufferSizeMs(); +} + void VCMReceiver::UpdateState(VCMReceiverState new_state) { CriticalSectionScoped cs(crit_sect_); assert(!(state_ == kPassive && new_state == kWaitForPrimaryDecode));