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webrtc / src.git / 6f5707e184f798db335527d3d7757347cdce3be3 / webrtc / modules / rtp_rtcp / source / rtcp_format_remb_unittest.cc
  1. 717d147 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 12 years ago
  2. 66b2e5c Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 12 years ago
  3. d900e8b Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 12 years ago
  4. 6c35e0b Reorganize test targets in WebRTC by kjellander@webrtc.org · 12 years ago
  5. a048d7c Include files from webrtc/.. paths in rtp_rtcp/ by pbos@webrtc.org · 12 years ago
  6. 29b2219 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 12 years ago
  7. 3004c79 Fix clang errors in non-GYP_DEFINES=clang=1 build by pbos@webrtc.org · 12 years ago
  8. 2f44673 WebRtc_Word32 => int32_t for rtp_rtcp/ by pbos@webrtc.org · 12 years ago
  9. 0cb48a0 Set SingleStream BWE in unittests. by stefan@webrtc.org · 12 years ago
  10. b586507 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 12 years ago
  11. a678a3b Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests. by stefan@webrtc.org · 12 years ago
  12. 20ed36d Break out RtpClock to system_wrappers and make it more generic. by stefan@webrtc.org · 12 years ago
  13. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc]
  14. 976a7e6 Adding support for jointly estimating bandwidth using all streams from the same sending client. by stefan@webrtc.org · 13 years ago
  15. bd7aeba Expose a set of options to the OveruseDetector supporting experiments by astor@webrtc.org · 13 years ago
  16. 9354cc9 Refactoring the receive-side bandwidth estimation into its own module. by stefan@webrtc.org · 13 years ago
  17. cb89c6f Revert 2363 - Refactoring the receive-side bandwidth estimation into its own module. by bjornv@webrtc.org · 13 years ago
  18. f728814 Refactoring the receive-side bandwidth estimation into its own module. by stefan@webrtc.org · 13 years ago
  19. 2853dde Refactor the internal API to the rtp/rtcp module. by pwestin@webrtc.org · 13 years ago
  20. 3c383ab Revert 2211 - Refactor the internal API to the rtp/rtcp module. by turaj@webrtc.org · 13 years ago
  21. 0774838 Refactor the internal API to the rtp/rtcp module. by pwestin@webrtc.org · 13 years ago
  22. d6d014f Fixes memory leaks introduced in 1698. by henrike@webrtc.org · 13 years ago
  23. 78088c2 Removed warnings on Windows and enabled warnings-as-errors on Windows. by phoglund@webrtc.org · 13 years ago
  24. d1a860b Fixed GCC 4.6 errors (mostly 'unused variable' errors and incorrect usage of EXPECT_EQ with booleans. by phoglund@webrtc.org · 13 years ago
  25. 0644b1d Introduce a mockable RtpRtcpClock interface replacing ModuleRTPUtility time functions by pwestin@webrtc.org · 13 years ago
  26. 741da94 Added support for new RTCP message REMB (remote estimated max bitrate) by pwestin@webrtc.org · 14 years ago