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gerrit-public.fairphone.software / platform / external / webrtc / 9c0914f93801eb0c081ae733bd744290e17e5bef
  1. 9c0914f Do not crop DesktopFrame if the size won't change by Zijie He · 7 years ago
  2. 2059bb3 Adding Obj-C binding for RTCConfiguration::max_ipv6_networks. by deadbeef · 7 years ago
  3. ecf3d53 Add histogram for FallbackDesktopCapturerWrapper and BlankDetectorDesktopCapturerWrapper by Zijie He · 7 years ago
  4. d21eab3 Add "max_ipv6_networks" field to RTCConfiguration. by deadbeef · 7 years ago
  5. 3427f53 Relanding: Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting. by deadbeef · 7 years ago
  6. 58f1725 Add gn dependency between ana_debug_dump_proto and ana_config_proto. by jbudorick · 7 years ago
  7. 74544f9 Return translated position in MouseCursorMonitor by Zijie He · 7 years ago
  8. 54c7215 Fix issues with NetworkMonitor singleton when used by multiple clients. by deadbeef · 7 years ago
  9. 8e24556 Disable SeqNumUnwrapper death tests to avoid breaking downstream builds. by philipel · 7 years ago
  10. 7956c0f Implemented a new sequence number unwrapper in sequence_number_util.h. by philipel · 7 years ago
  11. 8de1826 Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by minyue-webrtc · 7 years ago
  12. edd6cec Roll chromium_revision 4dc4bd62ba..1236993289 (489546:489596) by buildbot · 7 years ago
  13. 7df370b Revert "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by Minyue Li · 7 years ago
  14. 4a88120 Allow AudioSendStream to reconfig AudioNetworkAdaptor by minyue-webrtc · 7 years ago
  15. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
  16. 22d162d Roll chromium_revision 226fdf1148..4dc4bd62ba (489501:489546) by buildbot · 7 years ago
  17. bc88531 Roll chromium_revision c92ec9bd3c..226fdf1148 (489455:489501) by buildbot · 7 years ago
  18. b38f386 Update native plugin dll for turn servers and video. by gyzhou · 7 years ago
  19. 3b673c6 Removed file RTCCameraVideoCapturer.mm that isn't needed by jtteh · 7 years ago
  20. b1c9d1d Avoid that previous settings in APM are overwritten by WebRtcVoiceEngine by peah · 7 years ago
  21. 0748277 Roll chromium_revision f470dd6dfd..c92ec9bd3c (489395:489455) by buildbot · 7 years ago
  22. d1d6c5a Add jamiewalch to OWNERS. by jamiewalch · 7 years ago
  23. dba4f94 Roll chromium_revision 88beb225b9..f470dd6dfd (489316:489395) by buildbot · 7 years ago
  24. 2f0803c Roll chromium_revision fd8f995919..88beb225b9 (488572:489316) by buildbot · 7 years ago
  25. 61b0ed0 [iOS] Fix incorrectly oriented frames when rapidly switching between cameras. by jtteh · 7 years ago
  26. 96b69bd Refactor composing report blocks for rtcp Sender/Receiver reports. by danilchap · 7 years ago
  27. 7fb11d7 Shrink critical-section scope in ReceiveStatisticsImpl::GetActiveStatisticians() by eladalon · 7 years ago
  28. 6209dcd Add SetReportBlocks to rtcp Sender/Receive Report classes. by danilchap · 7 years ago
  29. fb14312 Reland of Injectable Obj-C video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2980173002/ ) by kthelgason · 7 years ago
  30. 8337727 Remove deprecated RtpRtcp::SetAudioPacketSize by danilchap · 7 years ago
  31. e264a9e Rename isolated_output to test_output and add a method to get the test_output directory. by ilnik · 7 years ago
  32. d5e3d0f Creating a more generic memcheck suppression by mbonadei · 7 years ago
  33. 42f44f9 Get rid of unnecessary cast of FlexfecReceiveStreamImpl to FlexfecReceiveStream by eladalon · 7 years ago
  34. 59cac99 Report minimum PSNR in VideoQualityTest and save corresponding frame to file by ilnik · 7 years ago
  35. d3f3c34 Remove NullObjectReceiveStatistics() in rtp_rtcp module by danilchap · 7 years ago
  36. a04d9c3 Remove RtpRtcp::RemoteRTCPStat(RTCPSenderInfo*) as unused by danilchap · 7 years ago
  37. d0727bf Fix NSInteger formatting warning from clang 6 by oprypin · 7 years ago
  38. 1e64cfa Fix autoroller in accordance to upstream change by oprypin · 7 years ago
  39. ec390b5 When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded by korniltsev.anatoly · 7 years ago
  40. d083e85 Remove traces from {send,receive}_statistics_proxy.cc by ehmaldonado · 7 years ago
  41. 65e1f94 Throttle log message in FrameBuffer. by philipel · 7 years ago
  42. c43d565 Remove setting configuration parameter to itself. by danilchap · 7 years ago
  43. cc8b906 iOS AppRTCMobile: Close peerconnection when disconnecting by magjed · 7 years ago
  44. e029d99 Integer overflow bug in low_cut_filter. by Alex Loiko · 7 years ago
  45. fe43df1 Ignore NewApi Android Lint warning + Roll chromium_revision by ehmaldonado · 7 years ago
  46. bf82021 Disable some Opus tests pending an update by flim · 7 years ago
  47. f3a48ab Delete unused field from AndroidVideoTrackSource by korniltsev.anatoly · 7 years ago
  48. 48e4d6d Add zijiehe@chromium.org as OWNERS in WebRTC DesktopCapturer related logic by Zijie He · 7 years ago
  49. cd66a77 Create new constructors and fields to support a better mouse cursor monitor by Zijie He · 7 years ago
  50. 817c8af Revert of Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting. (patchset #2 id:20001 of https://codereview.webrtc.org/2983213002/ ) by deadbeef · 7 years ago
  51. d14d9f7 Use array declaration for extension URIs. by Steve Anton · 7 years ago
  52. ad95614 Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting. by deadbeef · 7 years ago
  53. a3251dd Add parsing/serializing for MID RTP header extension. by Steve Anton · 7 years ago
  54. 3296256 Fixing lint issue by henrika · 7 years ago
  55. cfccdae Adds WebRtcAudioTrack.setAudioTrackUsageAttribute API by henrika · 7 years ago
  56. e29117e Modifies closing of AudioTrack resource on Android by henrika · 7 years ago
  57. 8ac955b Set target API to 18 for MediaCodecUtils. by sakal · 7 years ago
  58. cb79d23 Add common TLS extensions by Emad Omara · 7 years ago
  59. 3c45186 Move total audio energy and duration tracking to AudioLevel and protect with existing critial section. by zstein · 7 years ago
  60. c4a5c14 Print general usage information for event_log_analyzer by terelius · 7 years ago
  61. ad908f8 Roll chromium_revision 08a8c75946..1238950005 (488179:488218) by buildbot · 7 years ago
  62. 6d1dfa7 Roll chromium_revision d5f03e2816..08a8c75946 (488153:488179) by buildbot · 7 years ago
  63. 12d3084 Correct the calculation of discard rate. by minyue-webrtc · 7 years ago
  64. 5ed0904 Roll chromium_revision 40e1a3f583..d5f03e2816 (488128:488153) by buildbot · 7 years ago
  65. 6492338 Roll chromium_revision d23ce0ebaa..40e1a3f583 (488066:488128) by buildbot · 7 years ago
  66. 817ebf1 Roll chromium_revision 281eabf35e..d23ce0ebaa (488004:488066) by buildbot · 7 years ago
  67. 1777c5f Move temporal-layer properties to FrameConfig. by pbos · 7 years ago
  68. c40e1d3 Roll chromium_revision c86185cb26..281eabf35e (487874:488004) by buildbot · 7 years ago
  69. 398c3fd Introduce RtpTransportInternal and SrtpTransport. by zstein · 7 years ago
  70. f6a861a Remove remains of webrtc/base by ehmaldonado · 7 years ago
  71. 3c3b110 Roll chromium_revision 8779e5365c..c86185cb26 (487834:487874) by buildbot · 7 years ago
  72. e76f55e Disable flaky NoBandwidthDropAfterDtx test. by tschumim · 7 years ago
  73. a4c2117 Roll chromium_revision 6ada1228ca..8779e5365c (487809:487834) by buildbot · 7 years ago
  74. 2a06617 Roll chromium_revision fee38fe5d6..6ada1228ca (487773:487809) by buildbot · 7 years ago
  75. 9c0e0fa Fix fromAndroidGraphicsMatrix to use column-major order for output. by sakal · 7 years ago
  76. b4aa4eb Replace WEBRTC_TRACE logging in modules/audio_device/.. mac/ win/ by saza · 7 years ago
  77. c58f8c0 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 7 years ago
  78. e1d4dca Roll chromium_revision b845da41ed..fee38fe5d6 (487740:487773) by buildbot · 7 years ago
  79. dc7feb0 Roll chromium_revision 9b8266f849..b845da41ed (487695:487740) by buildbot · 7 years ago
  80. 4076576 Roll chromium_revision 0c6d79c9a3..9b8266f849 (487542:487695) by buildbot · 7 years ago
  81. 0cf9a4a Add texture support to HardwareVideoEncoder. by Bjorn Mellem · 7 years ago
  82. f80f344 Roll chromium_revision 712d1cf93f..0c6d79c9a3 (487487:487542) by buildbot · 7 years ago
  83. 8fb2361 Add texture support to HardwareVideoDecoder. by Bjorn Mellem · 7 years ago
  84. 153e204 [Webrtc] Reenable libc++ on ubsan by thomasanderson · 7 years ago
  85. 3e45cb5 Mapping screen id from DirectX capturer to GDI capturer by Zijie He · 7 years ago
  86. 8d2c235 Roll chromium_revision c09d9b5e5a..712d1cf93f (487450:487487) by buildbot · 7 years ago
  87. f549dff Fix gtest-parallel-wrapper by ehmaldonado · 7 years ago
  88. 80c829f Enable tracing on rtcstats_integrationtest.cc by ehmaldonado · 7 years ago
  89. b878575 Roll chromium_revision 03613afa1f..c09d9b5e5a (487428:487450) by buildbot · 7 years ago
  90. b5c1607 UBSan fuzzer bug in LowCutFilter::BiqueadFilter::Process by Alex Loiko · 7 years ago
  91. fcf97c3 Fix fullscreen scaling in AppRTCMobile. by sakal · 7 years ago
  92. b0b721a Increase the size of the buffer for type.name.id. by ehmaldonado · 7 years ago
  93. 43a85f0 Patch set 1: by saza · 7 years ago
  94. 0f2f169 Roll chromium_revision 11a66767dd..03613afa1f (487325:487428) by buildbot · 7 years ago
  95. a26196b Trace stats in RTCStatsCollector. by ehmaldonado · 7 years ago
  96. 16005b7 Remove potential left shift of negative value in WebRtcSpl_AnalysisQMF by Alex Loiko · 7 years ago
  97. 157cbbd Added implementation of three classes: by gnish · 7 years ago
  98. 3124cc6 [Webrtc] Reenable libc++ on gcc by thomasanderson · 7 years ago
  99. 89ac292 Roll chromium_revision 12bbaee380..11a66767dd (487259:487325) by buildbot · 7 years ago
  100. e103fc7 Roll chromium_revision 1751c38ce0..12bbaee380 (487180:487259) by buildbot · 7 years ago