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gerrit-public.fairphone.software / platform / external / webrtc / b2d5577caf90346d03b010822653720fcdfd9596
  1. b2d5577 Move tools/mb -> tools-webrtc/mb by Henrik Kjellander · 8 years ago
  2. b4ad603 Put iOS H264 High profile under a field trial by magjed · 8 years ago
  3. afd5494 Move tools/valgrind-webrtc -> tools-webrtc/valgrind by kjellander · 8 years ago
  4. 8d8816c CQ config: remove unused hide_ref_in_committed_msg. by tandrii · 8 years ago
  5. ffb865f Revert of Delete unused code from systeminfo. (patchset #3 id:40001 of https://codereview.webrtc.org/2578323005/ ) by skvlad · 8 years ago
  6. 5214a0a Add support for content hints to VideoTrack. by pbos · 8 years ago
  7. fb6ad3b Add full stack tests: by asapersson · 8 years ago
  8. 5bf9bed Remove redundant local variable from OveruseDetector::Detect. by terelius · 8 years ago
  9. b3df385 Remove device HW id -> marketing name mapping table for iOS devices. by kthelgason · 8 years ago
  10. 617ca31 Delete unused code from systeminfo. by kthelgason · 8 years ago
  11. ffecbbf Fix for integer overflow in NetEq. by ivoc · 8 years ago
  12. 70f39a3 In RtpPacket do not keep pointer to RtpHeaderExtensionMap by danilchap · 8 years ago
  13. 7e70fe2 Fix wrong log message. by kthelgason · 8 years ago
  14. 51813b3 Use NtpTime in RTCPSender::RtcpContext instead of pair of uint32_t by danilchap · 8 years ago
  15. eb538fd Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss. by terelius · 8 years ago
  16. beafee3 Move ios_helpers to sdk folder by kthelgason · 8 years ago
  17. 8d66245 Adding Åsa and Erik as video owners. by mflodman · 8 years ago
  18. 528ec3d Don't report packets with id -1 to the transport feedback adapter as they provide no value. by stefan · 8 years ago
  19. 05d6e26 Initialize packetization mode in VideoToolbox by kthelgason · 8 years ago
  20. 9a27205 whitespace by manual git cl land. by Andrii Shyshkalov · 8 years ago
  21. 2d897bd CQ whitespace. by tandrii · 8 years ago
  22. 4cd6221 iOS: Add trendline filter to field trials. by tkchin · 8 years ago
  23. 5bc3945 Fix integer overflow in ProbeController. by sergeyu · 8 years ago
  24. b3564ad Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double. by terelius · 8 years ago
  25. a97c5d2 Add ossu@ to OWNERS of audio/ and modules/audio_coding/ by ossu · 8 years ago
  26. d79f97b Fixing loopback video test by reconfiguring the encoder to correct size. by mflodman · 8 years ago
  27. 721d402 Create VideoReceiver with external VCMTiming object. by philipel · 8 years ago
  28. ac8d516 Improves release of allocated audio resources on Android. by henrika · 8 years ago
  29. 43c3821 Revert of Avoid precision loss in TrendlineEstimator from int64_t -> double conversion (patchset #7 id:120001 of https://codereview.webrtc.org/2577463002/ ) by terelius · 8 years ago
  30. 0bac07a Revert of Avoid precision loss in MedianSlopeEstimator from int64_t -> double conversion (patchset #3 id:40001 of https://codereview.webrtc.org/2578543002/ ) by terelius · 8 years ago
  31. ebcbcc3 Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss. by terelius · 8 years ago
  32. df2ceb8 Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ ) by nisse · 8 years ago
  33. bf65be5 Wire-up audio BWE with overhead. by michaelt · 8 years ago
  34. c12cbaf Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double. by terelius · 8 years ago
  35. 3168c7a Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time. by hbos · 8 years ago
  36. 6a58f33 Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2581663002/ ) by nisse · 8 years ago
  37. 7bf5369 RTCStatsIntegrationTest: TestMemberIsIDReference on all defined IDs. by hbos · 8 years ago
  38. 06035cf Reland of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2576673002/ ) by nisse · 8 years ago
  39. 0b571a1 MB: Enable memcheck for the linux_memcheck trybot. by Henrik Kjellander · 8 years ago
  40. e10e6d1 RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values. by hbos · 8 years ago
  41. 24db179 Move tools/autoroller to tools-webrtc/ + rename script by Henrik Kjellander · 8 years ago
  42. 4128649 Move all codec specific definitions from modules_include by hta · 8 years ago
  43. b5ffc14 Create top-level dir tools-webrtc and start moving things into it. by Henrik Kjellander · 8 years ago
  44. ef753e2 Remove unnecessary third_party links: nss, llvm-build, syzygy by kjellander · 8 years ago
  45. 3bc031b Remove unused items in tools/ by Henrik Kjellander · 8 years ago
  46. c3765f9 Roll chromium_revision b935b59277..5e0dca78b3 (438725:438769) by buildbot · 8 years ago
  47. f2832a0 Roll chromium_revision 9807edde11..b935b59277 (438688:438725) by buildbot · 8 years ago
  48. 2c8d94b Roll chromium_revision 699f628e13..9807edde11 (438637:438688) by buildbot · 8 years ago
  49. 2769ec6 Add WriteIsolatedOutput() functions by zijiehe · 8 years ago
  50. fba7900 Roll chromium_revision eae1bf6b1e..699f628e13 (438554:438637) by buildbot · 8 years ago
  51. 88cf05c Guard against uninitialized packetization modes. by hta · 8 years ago
  52. d30bd18 Roll chromium_revision a20ca9fe57..eae1bf6b1e (438523:438554) by buildbot · 8 years ago
  53. 9ce8cbf Roll chromium_revision 135e29eed5..a20ca9fe57 (438491:438523) by buildbot · 8 years ago
  54. 9a394f0 Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value. by hbos · 8 years ago
  55. c3c2f31 Adds basic Bluetooth support to AppRTCMobile by henrika · 8 years ago
  56. db8af2a Run 'git cl format --full' on Base64. by johan · 8 years ago
  57. 9006987 Remove deprecated RTPSender::SendPadData by danilchap · 8 years ago
  58. e2ec7c2 Remove static cast from H264SpropParameterSets. by johan · 8 years ago
  59. 930959d Improvements to the reliability of the echo detector perf test. by ivoc · 8 years ago
  60. a701469 Roll chromium_revision a8e17a3031..135e29eed5 (438476:438491) by buildbot · 8 years ago
  61. 8fc0c4c Add vector<uint8_t> to Base64 decoded data types. by johan · 8 years ago
  62. 0878f94 Delete accidental drmemory symlink by kjellander@webrtc.org · 8 years ago
  63. 665da28 Autoroller: Add --ignore-unclean-workdir flag by kjellander · 8 years ago
  64. a5bb562 Delete webrtc/transport.h. by aleloi · 8 years ago
  65. 9e1e6c5 Corrected access of null pointer in audioproc_f: by peah · 8 years ago
  66. 63e6a38 Removes verification of audio parameters on Android by henrika · 8 years ago
  67. fded4cc Roll chromium_revision 5fb8c41aea..a8e17a3031 (438448:438476) by buildbot · 8 years ago
  68. 0989fbc Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ ) by nisse · 8 years ago
  69. 7b25166 Fix for left shift of negative value in NetEq. by ivoc · 8 years ago
  70. bd6c6fa Delete method Pathname::url and base/urlencode* by nisse · 8 years ago
  71. bb66ec3 Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9 by skvlad · 8 years ago
  72. e0eae3c This CL adds the basic framework for AEC3 in the audio processing module. by peah · 8 years ago
  73. db39742 Delete unused class rtc::RegKey. by nisse · 8 years ago
  74. e5dc62a PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. by kjellander · 8 years ago
  75. 43c5a97 Delete stl_util.h. Unused since cl https://codereview.webrtc.org/2447103002 by nisse · 8 years ago
  76. 8afbc8c Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2567143003/ ) by nisse · 8 years ago
  77. 36f74e5 Delete rtc::linked_ptr. Only use, in statstypes.h, replaced with std::unique_ptr. by nisse · 8 years ago
  78. dd3c811 Roll chromium_revision cfd026f99e..5fb8c41aea (438418:438448) by buildbot · 8 years ago
  79. a5073c0 Disable AudioDeviceTest.StartPlayoutOnTwoInstances on iOS by Henrik Kjellander · 8 years ago
  80. 80df795 Roll chromium_revision e234d53ddf..cfd026f99e (438369:438418) by buildbot · 8 years ago
  81. e26b89c Roll chromium_revision b571577c64..e234d53ddf (438292:438369) by buildbot · 8 years ago
  82. 62802a1 Fixing possible crash due to RefCountedChannel assignment operator. by deadbeef · 8 years ago
  83. b236257 Fixing integer overflow when parsing bandwidth attribute. by deadbeef · 8 years ago
  84. 9396a08 Roll chromium_revision 79b1930444..b571577c64 (438242:438292) by buildbot · 8 years ago
  85. 95aa964 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago
  86. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  87. 00fd520 Roll chromium_revision 047b36f906..79b1930444 (438176:438242) by buildbot · 8 years ago
  88. b68cc75 ParseCandidate(): Refactor to fix memcheck false positive. by hnsl · 8 years ago
  89. f8b262e Roll chromium_revision e882052d97..047b36f906 (438143:438176) by buildbot · 8 years ago
  90. 301fc4a Update common_audio/smoothing_filter. by minyue · 8 years ago
  91. bfcf561 Delete VideoFrame default constructor, and IsZeroSize method. by nisse · 8 years ago
  92. 46711db Disable flaky QualityScaler tests for now. by kthelgason · 8 years ago
  93. 277b250 Refactor "secure bool" into explicit PROTO_TLS. by hnsl · 8 years ago
  94. 1c4b5bc Roll chromium_revision 632410c83c..e882052d97 (438112:438143) by buildbot · 8 years ago
  95. 38b6dbc Autoroller: Support for rolling individual DEPS entries. by kjellander · 8 years ago
  96. ef16e99 Add a gtk3 port of peerconnection_client on Linux by thomasanderson · 8 years ago
  97. 349092b Logging basic bad call detection by palmkvist · 8 years ago
  98. e381015 Revert of New PeerConnectionInterface::GetStats: No bogus default implementation. (patchset #1 id:1 of https://codereview.webrtc.org/2566143002/ ) by hbos · 8 years ago
  99. 4145989 Roll chromium_revision 2d6dcff9ac..632410c83c (438085:438112) by buildbot · 8 years ago
  100. 07e276c Rename RtpStreamReceiver::SetCodec() to ::AddCodec(). by johan · 8 years ago