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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b7e5b2741f9cbd68e7be54e3445d28d266477c92 / video_engine / vie_receiver.cc
  1. 903e746 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 10 years ago
  2. 99153ba First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class. by asapersson@webrtc.org · 10 years ago
  3. dd0f8b2 Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log. by wu@webrtc.org · 10 years ago
  4. 2fae0d1 Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe. by wu@webrtc.org · 10 years ago
  5. 3468f20 Remove WEBRTC_TRACE uses in video_engine/ by pbos@webrtc.org · 10 years ago
  6. d2fb259 Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. by wu@webrtc.org · 10 years ago
  7. 39d9fa5 Remove timestamp_extrapolator's dependency to Clock and vcm defines. by wu@webrtc.org · 10 years ago
  8. 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
  9. 093fc0b Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  10. 9d10769 Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 10 years ago
  11. 9968131 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
  12. 2d3624c Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
  13. fbd6f47 Fixes RTX related bugs. by stefan@webrtc.org · 10 years ago
  14. 800136d Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
  15. d1e7fac Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  16. 7d99cd4 Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  17. efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  18. 12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  19. 984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  20. 7e97e4c Fix for making sure that the packet in order checks are done prior to updating the last received packet state. by stefan@webrtc.org · 11 years ago
  21. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  22. a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  23. c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  24. efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
  25. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  26. d69e2f4 Access receiving_ under receive_cs critical section by braveyao@webrtc.org · 11 years ago
  27. 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
  28. b89eed3 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
  29. 46088d2 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
  30. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  31. 20cfda6 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
  32. d8ecee5 Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
  33. 4e5f983 Fix a return value mismatch introduced in r4129. by stefan@webrtc.org · 11 years ago
  34. 6696fba Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
  35. e3b52e6 Don't return an estimated receive BW for channels not receiving video. by mflodman@webrtc.org · 11 years ago
  36. b7716d8 - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. by solenberg@webrtc.org · 11 years ago
  37. 281cff8 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  38. 67879bc WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
  39. 065b64d Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
  40. 912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  41. 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
  42. cd1ac8b Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 11 years ago
  43. 2a5dbce Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 11 years ago
  44. 1755c25 Made TickTime immutable, rewrote tick utils to be fakeable. by phoglund@webrtc.org · 12 years ago
  45. e75e29e Fixes a bitrate mismatch between sender and receiver. by stefan@webrtc.org · 12 years ago
  46. b015cbe Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago