Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 2d897bd31bb3a3daa7e9fd1ab7560d45968b4fce
  1. bd44bb0 Fix out of bound reads in ParseIceServerUrl() for various input. by hnsl · 8 years ago
  2. b010b8f Roll chromium_revision 4537fa801e..88e7649411 (437826:437837) by buildbot · 8 years ago
  3. 65a1e77 Try to deflake VideoSendStream tests with ULPFEC. by brandtr · 8 years ago
  4. e448dd5 RTCIceCandidatePairStats.consentRequestsSent set by RTCStatsCollector by hbos · 8 years ago
  5. b29b9c8 Replace VideoCaptureDataCallback by VideoSinkInterface. by nisse · 8 years ago
  6. 99f7bfd Change MANUAL -> DISABLED for ScreenCapturerIntegrationTest tests by Henrik Kjellander · 8 years ago
  7. 9e3e0da Roll chromium_revision d33aa11bc5..4537fa801e (437814:437826) by buildbot · 8 years ago
  8. 951fe73 Roll chromium_revision 5f112f29f4..d33aa11bc5 (437807:437814) by buildbot · 8 years ago
  9. bbfed52 Set OPENSSL_EC_NAMED_CURVE explicitly on EC key so that certificate has ASN1 OID and NIST curve info. Without this openSSL handshake negotiation fails throwing NO_SHARED_CIPHER error. the change made is along the lines of openssl behavior documented here: https://wiki.openssl.org/index.php/Elliptic_Curve_Diffie_Hellman#ECDH_and_Named_Curves by ssaroha · 8 years ago
  10. ae875f2 Roll chromium_revision 05b6f4be7e..5f112f29f4 (437804:437807) by buildbot · 8 years ago
  11. 2cdec07 Roll chromium_revision 73ac3ff0ec..05b6f4be7e (437797:437804) by buildbot · 8 years ago
  12. 95bfe2d Roll chromium_revision f10b6162b6..73ac3ff0ec (437786:437797) by buildbot · 8 years ago
  13. ba41428 Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface. by minyue · 8 years ago
  14. 50bac86 Roll chromium_revision cf064891c2..f10b6162b6 (437785:437786) by buildbot · 8 years ago
  15. 5c8b757 Roll chromium_revision 211f7ed09d..cf064891c2 (437781:437785) by buildbot · 8 years ago
  16. 18d592c Roll chromium_revision 3c957a583f..211f7ed09d (437777:437781) by buildbot · 8 years ago
  17. 0acf8fa Roll chromium_revision 8ca234a6e4..3c957a583f (437773:437777) by buildbot · 8 years ago
  18. d1a38b5 Implement the "needs-ice-restart" logic for SetConfiguration. by deadbeef · 8 years ago
  19. 3edec7c Adding error enum to be used by PeerConnectionInterface methods. by deadbeef · 8 years ago
  20. d00ff0b Roll chromium_revision ba3c4f9491..8ca234a6e4 (437769:437773) by buildbot · 8 years ago
  21. 78b6dc8 Roll chromium_revision a229e828a6..ba3c4f9491 (437767:437769) by buildbot · 8 years ago
  22. fe719ac Roll chromium_revision 39e086ae3e..a229e828a6 (437763:437767) by buildbot · 8 years ago
  23. a1c0f9b Roll chromium_revision 7b5206fba7..39e086ae3e (437761:437763) by buildbot · 8 years ago
  24. f71dbf9 Roll chromium_revision bd86a87396..7b5206fba7 (437742:437761) by buildbot · 8 years ago
  25. 3910792 Roll chromium_revision b671a5234f..bd86a87396 (437704:437742) by buildbot · 8 years ago
  26. a6cc7f1 Roll chromium_revision 71d41b68d1..b671a5234f (437664:437704) by buildbot · 8 years ago
  27. 8d1649d MANUAL tests of GDI capturers by zijiehe · 8 years ago
  28. c5eefaf Roll chromium_revision 55c13aa961..71d41b68d1 (437596:437664) by buildbot · 8 years ago
  29. 82dbd33 Roll chromium_revision 3387c1777b..55c13aa961 (437554:437596) by buildbot · 8 years ago
  30. 7800cbb Roll chromium_revision 1289bbe752..3387c1777b (437522:437554) by buildbot · 8 years ago
  31. 1eac19f Update version of third_party/gtest-parallel. by ehmaldonado · 8 years ago
  32. 5493b8a Remove extra uses of basictypes.h. by pbos · 8 years ago
  33. 3536463 Only store sequence numbers for media stream in FlexFEC end-to-end test. by brandtr · 8 years ago
  34. 08ab2ee Roll chromium_revision b558081912..1289bbe752 (437509:437522) by buildbot · 8 years ago
  35. ca87b62 Disable failing perf test on Android. by kthelgason · 8 years ago
  36. 5025463 Modify JavaToStdString to allow ISO-8859-1 encoded strings. by minyue · 8 years ago
  37. 5a38836 Implement Theil-Sen's method for fitting a line to noisy data (used in bandwidth estimation). by terelius · 8 years ago
  38. 02cd4d6 RTCInboundRTPStreamStats.packetsLost set by RTCStatsCollector. by hbos · 8 years ago
  39. d82f512 RTCIceCandidatePairStats.requestsReceived defined by RTCStatsCollector. by hbos · 8 years ago
  40. 33ce889 Reland of Bump up scaling limit for MediaCodec. (patchset #1 id:1 of https://codereview.webrtc.org/2562963002/ ) by kthelgason · 8 years ago
  41. 0e738d4 Convert seconds_since_epoch to integer. by ehmaldonado · 8 years ago
  42. 5ad5de3 During AEC development, it is handy to be able to simulate different by peah · 8 years ago
  43. df80fd1 When recreating a call based on an aecdump recording the nearend used by peah · 8 years ago
  44. a90799d Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ ) by asapersson · 8 years ago
  45. 1cd0a0a Revert of Bump up scaling limit for MediaCodec. (patchset #3 id:40001 of https://codereview.webrtc.org/2566533002/ ) by kthelgason · 8 years ago
  46. 3312e18 Roll chromium_revision 2387860409..b558081912 (437485:437509) by buildbot · 8 years ago
  47. 3e9b133 Bump up scaling limit for MediaCodec. by kthelgason · 8 years ago
  48. 3b11d00 Disable TestWithNewVideoJitterBuffer/EndToEndTest.SendsAndReceivesH264 on Memcheck. by asapersson · 8 years ago
  49. 17a32c4 whitespace CL to trigger builds by Henrik Kjellander · 8 years ago
  50. 55512c2 Roll chromium_revision 247b2e4c45..2387860409 (437457:437485) by buildbot · 8 years ago
  51. 61aeaf1 Roll chromium_revision 5c49d183f4..247b2e4c45 (437395:437457) by buildbot · 8 years ago
  52. 87af0ac Roll chromium_revision c1ab029096..5c49d183f4 (437359:437395) by buildbot · 8 years ago
  53. 2200de3 Roll chromium_revision 51fb98171a..c1ab029096 (437294:437359) by buildbot · 8 years ago
  54. 7aa9b91 Fix issue with deprecated CongestionController interface not working. by stefan · 8 years ago
  55. c553910 Roll chromium_revision aa2290d47f..51fb98171a (437235:437294) by buildbot · 8 years ago
  56. e83f4b3 Enable screen capturer tests for Linux / DirectX capturer / magnifier capturer by zijiehe · 8 years ago
  57. 36df2d7 Refactor webrtc/modules/video_{capture,coding} for GN check by ehmaldonado · 8 years ago
  58. f7c6d72 Rename RtpStreamReceiver::IsFecEnabled to RtpStreamReceiver::IsUlpfecEnabled. by brandtr · 8 years ago
  59. 8d193a7 Do not update OnReceivedRtcpReceiverReport if report block list is empty (and rtt zero). by asapersson · 8 years ago
  60. c847417 Reland of Add ability to scale to arbitrary factors (patchset #1 id:1 of https://codereview.webrtc.org/2557323002/ ) by kthelgason · 8 years ago
  61. 7dada5e Delete deprecated CongestionController constructor and packet_router method. by nisse · 8 years ago
  62. 01653b1 Add a tool to fix (some) errors reported by gn gen --check. by ehmaldonado · 8 years ago
  63. 0287db0 Re-enable disabled VideoProcessorIntegrationTest tests by kjellander · 8 years ago
  64. 0582e6c Add FlexFEC settings toggle in Android AppRTCMobile. by brandtr · 8 years ago
  65. 2a85f9c Roll chromium_revision bd4fdcd8d3..aa2290d47f (437221:437235) by buildbot · 8 years ago
  66. 10daf86 Simplify an always true condition. by nisse · 8 years ago
  67. d003557 Fix error in VideoFileRenderer_nativeI420Scale. by sakal · 8 years ago
  68. 1cfbd60 Generalize FlexfecReceiveStream::Config. by brandtr · 8 years ago
  69. 446fcb6 Clean up FlexfecReceiveStream ctor signatures. by brandtr · 8 years ago
  70. 64c4a7e Refactor webrtc/modules/audio_processing for GN check by ehmaldonado · 8 years ago
  71. b0a1111 Decode h264 fmtp sprop-parameter-sets to binary. by johan · 8 years ago
  72. c95d810 Roll chromium_revision b4a2bcaef4..bd4fdcd8d3 (437198:437221) by buildbot · 8 years ago
  73. 623427c Injectable output rate calculater for AudioMixer. by aleloi · 8 years ago
  74. 9abd275 Remove unused arguments and variable in MediaOptimization. by asapersson · 8 years ago
  75. 7722a4c Revert of Add ability to scale to arbitrary factors (patchset #7 id:120001 of https://codereview.webrtc.org/2555483005/ ) by kthelgason · 8 years ago
  76. 710c335 Add ability to scale to arbitrary factors by kthelgason · 8 years ago
  77. b39db84 Refactoring: Declare cricket::Codec constructors protected. by hta · 8 years ago
  78. 06a6984 Android classreferenceholder.h: Reorder function declaration keywords by magjed · 8 years ago
  79. 7e1070d Roll chromium_revision 3cd4e5505f..b4a2bcaef4 (436604:437198) by kjellander · 8 years ago
  80. 2cd872a Log BitBlt failure by zijiehe · 8 years ago
  81. c4adabf Create the Java Wrapper of RtpReceiverObserverInterface. by zhihuang · 8 years ago
  82. dd33e63 Translate isolated-script-test-output to dump_json_test_results in gtest-parallel-wrapper.py by ehmaldonado · 8 years ago
  83. 157a34d Add kwiberg@ to gtest-parallel OWNERS. by Peter Boström · 8 years ago
  84. 676e08f Refactor webrtc/{api,audio} and modules/audio_coding for GN check by kjellander · 8 years ago
  85. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  86. 7495c8c Clean up redundant include of ../webrtc_overrides by ehmaldonado · 8 years ago
  87. 5a4eb51 Roll gtest-parallel. by ehmaldonado · 8 years ago
  88. eca373f Adding OnReceivedOverhead to AudioEncoder. by minyue · 8 years ago
  89. ac382f3 Make ostream<< for enum class H264PacketizationMode by hta · 8 years ago
  90. e36c46e Use SSL_CTX_set_max_proto_version instead of SSL_CTX_set_max_version. by davidben · 8 years ago
  91. d3de4ab Remove deprecated comments by zijiehe · 8 years ago
  92. 49f34fd Relanding: Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  93. 57fd726 Revert of Refactoring that removes P2PTransport and DtlsTransport classes. (patchset #9 id:150001 of https://codereview.webrtc.org/2517883002/ ) by deadbeef · 8 years ago
  94. dd87d58 Add File::Open / Create functions to take an rtc::Pathname by zijiehe · 8 years ago
  95. bd28681 Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  96. ebbe4f2 Set the preferred DSCP value for Rtp data channel to be DSCP_AF41. by zhihuang · 8 years ago
  97. 66f99a4 Roll chromium_revision d8bf23963e..3cd4e5505f (436579:436604) by buildbot · 8 years ago
  98. 6d314c7 Reject XR TargetBitrate items with unsupported layer indices by sprang · 8 years ago
  99. 99cc10f Roll chromium_revision 092e72e03b..d8bf23963e (436556:436579) by buildbot · 8 years ago
  100. 9aa9688 Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ ) by hta · 8 years ago