1. 78f8386 Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) by nisse · 7 years ago
  2. 642a074 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  3. eac9d2a Revert of Remove typedefs.h from webrtc/ root (part 1) (patchset #3 id:40001 of https://codereview.webrtc.org/3007253002/ ) by kjellander · 7 years ago
  4. 3b7353e Remove typedefs.h from webrtc/ root (part 1) by solenberg · 7 years ago
  5. 5e02d38 Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) by nisse · 7 years ago
  6. a9e87e4 Revert of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #3 id:40001 of https://codereview.webrtc.org/3006993002/ ) by nisse · 7 years ago
  7. 4e3045f Delete Rtx-related methods from RTPPayloadRegistry. by nisse · 7 years ago
  8. 6ae8262 Add reporting of googContentType via GetStats on send side by ilnik · 7 years ago
  9. 3fa5b7b Fix FrameConfigs used for VP8 with four temporal layers. by sprang · 7 years ago
  10. 2e14b15 Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) by nisse · 7 years ago
  11. 5b18967 Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  12. 1a9dee5 Tightening visibility and removing a public_dep. by mbonadei · 7 years ago
  13. 1f11d1a Implement googContentType GetStats metric reported on receive side. by ilnik · 7 years ago
  14. 19cf6be Unwrap picture ids in the RtpFrameReferencerFinder. by philipel · 7 years ago
  15. dfa8f5e Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) by nisse · 7 years ago
  16. 5eb0c3b Change reporting of timing frames conditions in GetStats on receive side by ilnik · 7 years ago
  17. daf49be Use RtxReceiveStream. by nisse · 7 years ago
  18. 36189cd Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  19. 9c6e107 Fix alr tests config by ilnik · 7 years ago
  20. 054582b Replace remaining gflags usages with rtc_base/flags by oprypin · 7 years ago
  21. 00ed864 Recently we moved webrtc/base to webrtc/rtc_base, so these by mbonadei · 7 years ago
  22. 42e4711 Piggybacking simulcast id and ALR experiment id into video content type extension. by ilnik · 7 years ago
  23. 47db89a Add SentToInputFpsRatioPercent UMA metric on send side by ilnik · 7 years ago
  24. 878f987 Add a new frame generator that cycles through randomly generated slides. by erikvarga · 7 years ago
  25. 0a6becd Replace gflags usages with rtc_base/flags in all targets based on test_main by oprypin · 7 years ago
  26. 5bf39fc Removing dependencies on stub headers within WebRTC. by mbonadei · 7 years ago
  27. a2c4b05 Update jpeg writer to compile on iOS and document it better by ilnik · 7 years ago
  28. c205a94 Now that https://codereview.webrtc.org/3003643002 is landed we can by mbonadei · 7 years ago
  29. beb74d6 Update video_replay tool to be able to dump .jpg files. by philipel · 7 years ago
  30. 3ef80a2 Reverse |rtx_payload_types| map, and rename. by nisse · 7 years ago
  31. 7f62201 Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker by eladalon · 7 years ago
  32. 775b7c5 Fix places that trigger no-unused-lambda-capture - change to using static-constexpr. by eladalon · 7 years ago
  33. ed56bdc Make CodecType conversion functions non-optional. by kthelgason · 7 years ago
  34. 3ba7282 Delete unneeded includes of atomic32.h. by nisse · 7 years ago
  35. acbfa68 Add experiment to disable ulpfec. by stefan · 7 years ago
  36. 6a5ac8f Report max interframe delay over window insdead of interframe delay sum by ilnik · 7 years ago
  37. 7d8db23 Fix places that trigger no-unused-lambda-capture by eladalon · 7 years ago
  38. 3a071f6 Let Call::OnRecoveredPacket parse RTP header extensions. by brandtr · 7 years ago
  39. de7e264 Remove WebRTC-videocontenttypeextension field trial completely by ilnik · 7 years ago
  40. 7f56e63 Ignore inter-frame delay stats samples when stream is inactive by sprang · 7 years ago
  41. 53431d2 Move PacedSender ownership to RtpTransportControllerSend. by Stefan Holmer · 7 years ago
  42. 80d7b62 Reland of Add Jpeg frame writer for test support. by ilnik · 7 years ago
  43. f0c86c0 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
  44. 0f007ea Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  45. aff1234 Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ ) by philipel · 7 years ago
  46. fd5a68e Revert of Add Jpeg frame writer for test support. (patchset #12 id:220001 of https://codereview.webrtc.org/2990563002/ ) by charujain · 7 years ago
  47. 1dae68b Add Jpeg frame writer for test support. by ilnik · 7 years ago
  48. 21950c6 Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2995153002/ ) by philipel · 7 years ago
  49. cee6642 Change ThreadChecker to SequencedTaskChecker in VideoReceiveStream by eladalon · 7 years ago
  50. 0df2c66 Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ ) by sprang · 7 years ago
  51. 90a1756 Reland of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:1 of https://codereview.webrtc.org/2995173002/ ) by emircan · 7 years ago
  52. 7f88afe Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #2 id:20001 of https://codereview.webrtc.org/2925253002/ ) by emircan · 7 years ago
  53. cc76c33 Revert of quest keyframes more frequently on stream start/decoding error. (patchset #2 id:170001 of https://codereview.webrtc.org/2996823002/ ) by tkchin · 7 years ago
  54. 7647da7 Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings. by asapersson · 7 years ago
  55. de39669 Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled. by asapersson · 7 years ago
  56. e8a5ec5 Reland of Make the acceptable queue in the cwnd experiment configurable. (patchset #1 id:1 of https://codereview.webrtc.org/2999893002/ ) by stefan · 7 years ago
  57. 9db127b Reland of Add functionality which limits the number of bytes on the network. (patchset #1 id:1 of https://codereview.webrtc.org/3001653002/ ) by stefan · 7 years ago
  58. 0a27bbc Delete unneeded Start and Stop methods on FlexfecReceiveStream. by Niels Möller · 7 years ago
  59. 786989c Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ ) by emircan · 7 years ago
  60. de6c74c Replace absolute path with relative path for GN files. by Jianjun Zhu · 7 years ago
  61. a58c48e Add a flags field to video timing extension. by sprang · 7 years ago
  62. 62d8f35 Fix incorrect InterframeDelayMaxInMs histogram metrics by sprang · 7 years ago
  63. a562a59 Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ ) by stefan · 7 years ago
  64. e7df174 Revert of Make the acceptable queue in the cwnd experiment configurable. (patchset #1 id:1 of https://codereview.webrtc.org/2998753002/ ) by stefan · 7 years ago
  65. 4a604c8 Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ ) by philipel · 7 years ago
  66. e1dcdc7 Make the acceptable queue in the cwnd experiment configurable. by stefan · 7 years ago
  67. 1ac8f7d Fix (1) EndToEndTest.InitialProbing and (2) EndToEndTest.TriggerMidCallProbing by eladalon · 7 years ago
  68. d8a48fb Workaround for PacketBuffer bug. by philipel · 7 years ago
  69. f0920fe Revert of Request keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.webrtc.org/2993793002/ ) by deadbeef · 7 years ago
  70. b0be841 Add functionality which limits the number of bytes on the network. by stefan · 7 years ago
  71. 6ab33cf Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  72. 504c822 Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock by srte · 7 years ago
  73. e9507d1 Fix the video buffer size should take rtt into consideration by gustavogb · 7 years ago
  74. 6e1beec Request keyframes more frequently on stream start/decoding error. by philipel · 7 years ago
  75. cb712cb Add an experiment for stricter pacing and ALR probing. by stefan · 7 years ago
  76. 63b0b01 Renamed fields in common_types.h/RtcpStatistics. by srte · 7 years ago
  77. 783ce68 Rename ViEEncoder to VideoStreamEncoder by mflodman · 7 years ago
  78. 92bdb64 Tracking mock_process_thread with a GN target by mbonadei · 7 years ago
  79. 2c1cbc1 Protected streams report RTP messages directly to the FlexFec streams by eladalon · 7 years ago
  80. a207a3a Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates by eladalon · 7 years ago
  81. 2f3edcc Remove NullReceiveStatistics by danilchap · 7 years ago
  82. 5fb3e75 Only one implementation of MockRtpPacketSink once by eladalon · 7 years ago
  83. 708d0cd Simplify FakeReceiveStatistics in video send stream tests by danilchap · 7 years ago
  84. 78133f0 Report minimum PSNR in VideoQualityTest and save corresponding frame to file by ilnik · 7 years ago
  85. 0667476 Remove traces from {send,receive}_statistics_proxy.cc by ehmaldonado · 7 years ago
  86. 5410662 Remove setting configuration parameter to itself. by danilchap · 7 years ago
  87. 2bc62f3 Remove remains of webrtc/base by ehmaldonado · 7 years ago
  88. 34cf86e Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback by ilnik · 8 years ago
  89. 00a0701 Let alr detector use a budged to detect underuse. by tschumim · 8 years ago
  90. 59d5575 Use relative paths in GN files. by jianjun.zhu · 8 years ago
  91. bc05a6a Move RTP keep-alive config from VideoSendStream::Config to Call::Config by sprang · 8 years ago
  92. 2b6f358 Fix video_replay tool to respect RTX stream and fix default parameters. by ilnik · 8 years ago
  93. 440909e Fix flaky test VideoSendStreamTest.SendsKeepAlive by sprang · 8 years ago
  94. 4f870fc Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 8 years ago
  95. c6c814d Remove remains of webrtc/base by ehmaldonado · 8 years ago
  96. a5b6c52 Remove webrtc::VideoEncoderFactory by magjed · 8 years ago
  97. 4b941f0 Report interframe delay sum in old GetStats by ilnik · 8 years ago
  98. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  99. e662d12 Implement RTP keepalive in native stack. by sprang · 8 years ago
  100. 37342b9 Report timing frames info in GetStats. by ilnik · 8 years ago