Log - 40220194bb03637cbc5b21ef36b6abd4e20cbd75 - fp2-dev/platform/external/chromium_org/third_party/webrtc

d1d198b Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics(). by stefan@webrtc.org · 10 years ago
55b0f2e Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 10 years ago
09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
8c95e83 Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
f8ec08e Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
2fd91bd Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
88b558f Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
6845de7 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
7f54561 Fix deadlock in RegisterPreDecodeImageCallback. by pbos@webrtc.org · 10 years ago
4d61a36 Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. by stefan@webrtc.org · 10 years ago
2d3624c Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
4a15560 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
800136d Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
d1e7fac Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
7d99cd4 Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 10 years ago
49812e6 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
b70db6d Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
ffea4ce Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
1430bc3 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
b113981 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
ee234be Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
9b30fd3 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
5fdd10a Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
47f0c41 Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
2e98d45 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
3dc7ff3 Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
c4af4cf Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). by asapersson@webrtc.org · 11 years ago
ecfef19 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
63301bd Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
31a8ce7 Removing FrameForStorage by mikhal@webrtc.org · 11 years ago
9b7bdee Revert r4562 by elham@webrtc.org · 11 years ago
a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
ece3d35 Added choice of decode error mode to loopback test. by agalusza@google.com · 11 years ago
7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
d893b3f Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
f43029b Revert "Avoid acquiring VCM::_receiveCritSect during decode callback." by wuchengli@chromium.org · 11 years ago
b0af417 Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
b89eed3 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
46088d2 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
0291c80 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
20cfda6 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
f40e9b6 - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
453f9c0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
ac6d919 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 11 years ago
7bc7e02 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
67879bc WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
208a648 Always set render delay in ViEChannel::RegisterExternalDecoder. by pbos@webrtc.org · 11 years ago
065b64d Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
2daec4c Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
55e6f58 Stop and restart fix. by mflodman@webrtc.org · 11 years ago
0329e59 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 11 years ago
9d6fcb3 Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
cd1ac8b Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 11 years ago
1619664 Adding a send side API for streaming by mikhal@webrtc.org · 11 years ago
f314c80 Added API to get receive side video delay. by mflodman@webrtc.org · 12 years ago
78696d3 Wire up CallStats to provide modules with correct RTT. by mflodman@webrtc.org · 12 years ago
5e87b5f Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016 by pwestin@webrtc.org · 12 years ago
b6d9cfc Revert the revert in r2988 since that wasn't the issue. by mflodman@webrtc.org · 12 years ago
9f269d2 Reverse Merged r2884 & r2888 from trunk. by vikasmarwaha@webrtc.org · 12 years ago
3bbed74 Switching to I420VideoFrame by mikhal@webrtc.org · 12 years ago
b015cbe Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago