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gerrit-public.fairphone.software / platform / external / webrtc / c3a0983d4bedea07b6d1a74dc165b56fca590e82
  1. c3a0983 Roll chromium_revision a8e5140..c6076f2 (372922:372974) incl. update to Opus v.1.1.2 by kjellander · 9 years ago
  2. a7ad7c3 Get the adapter type information from Android OS. by honghaiz · 9 years ago
  3. ae695e9 Refactor RtpSender and SSRCDatabase. by tommi · 9 years ago
  4. 040b79f Add helper macros for calling a histogram with different names. by asapersson · 9 years ago
  5. ed3277b Deprecate VideoDecoder::Reset() and remove calls. by Peter Boström · 9 years ago
  6. ce23bee Remove SendStreamFormat and ViewRequests. by Peter Boström · 9 years ago
  7. d467a91 Roll chromium_revision f41a54b..a8e5140 (372876:372922) by kjellander · 9 years ago
  8. c61635c PRESUBMIT: Exclude supplement.gypi from _CheckNoSourcesAboveGyp check. by kjellander · 9 years ago
  9. c5a39c2 H264: Thread-safe InitializeFFmpeg. Flag to control if InitializeFFmpeg should be called. by hbos · 9 years ago
  10. 799379e Let a minimum time interval pass (one bucket size) after initialization before reporting rates (to avoid rates being based on too short time intervals). by asapersson · 9 years ago
  11. 6fd26b6 Roll chromium_revision f6e3d46..f41a54b (372710:372876) by kjellander · 9 years ago
  12. c463e20 Reset TURN port NONCE when a new socket is created. by honghaiz · 9 years ago
  13. 9429148 Extra logging for HW codec. by glaznev · 9 years ago
  14. 3668cf0 Roll chromium_revision 65f9b34..f6e3d46 (372637:372710) by kjellander · 9 years ago
  15. a6c39d9 Remove unimplemented VideoChannel code. by Peter Boström · 9 years ago
  16. 6f7557e Disable useless BWE tests. by stefan · 9 years ago
  17. e37a2d1 Reland "Removing webrtc::AudioFrame::energy_." by minyue · 9 years ago
  18. d8de115 Remove mutable from rtc::CriticalSections. by pbos · 9 years ago
  19. 34877ee Revert of Added validation between RTP and RTCP timestamps (patchset #7 id:120001 of https://codereview.webrtc.org/1633843003/ ) by danilchap · 9 years ago
  20. 74451a5 Prevent zero division in VCMJitterEstimator. by Peter Boström · 9 years ago
  21. b46c333 Roll chromium_revision 126e210..65f9b34 (372588:372637) by kjellander · 9 years ago
  22. bba9dec Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  23. 1f611fa Fixed minor issue: added missing semicolons to metric_recorder.cc by cesar.ilharco · 9 years ago
  24. d06d4d8 Add linux_ubsan_vptr to default trybots. by kjellander · 9 years ago
  25. e1f2f1f Unwrap timestamps in VideoAnalyzer by sprang · 9 years ago
  26. 44efbec Converting picture_id to bitstring pushed from WithPictureId to Create function. by danilchap · 9 years ago
  27. 6b231e0 Roll chromium_revision d785e7c..126e210 (372580:372588) by kjellander · 9 years ago
  28. 25f17d7 Roll chromium_revision 2084e1d..d785e7c (372575:372580) by kjellander · 9 years ago
  29. ac53c88 Roll chromium_revision 750447f..2084e1d (372566:372575) by kjellander · 9 years ago
  30. 430e400 Roll chromium_revision f5d1a9c..750447f (372546:372566) by kjellander · 9 years ago
  31. 3f70562 Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015). by conceptgenesis · 9 years ago
  32. c97c886 Roll chromium_revision f07b6b8..f5d1a9c (372524:372546) by kjellander · 9 years ago
  33. f0269a6 Roll chromium_revision e9e4e90..f07b6b8 (372389:372524) by kjellander · 9 years ago
  34. eee86a6 Add option to disable particular HW video codec from app. by Alex Glaznev · 9 years ago
  35. 9dfed79 Stop processing any incoming packets when turn port is disconnected. by honghaiz · 9 years ago
  36. 083b8e9 Roll chromium_revision 3784ca9..e9e4e90 (372326:372389) by kjellander · 9 years ago
  37. de13882 rtcp::ExtenededReports packet class got Parse function by danilchap · 9 years ago
  38. ff63ed2 Format changes achieved by running clang-format -i -style=Chromium by peah · 9 years ago
  39. f5b804b Fix implicit bool casts in producer_fec_fuzzer.cc. by Peter Boström · 9 years ago
  40. 3a8cac8 Roll chromium_revision 105cb5f..3784ca9 (372268:372326) by kjellander · 9 years ago
  41. b163c3f Delete unused members from VideoOptions by nisse · 9 years ago
  42. a37babe Roll chromium_revision ffa6c99..105cb5f (372122:372268) by kjellander · 9 years ago
  43. 378dc77 Consolidate setters into SetRecvParameters. by pbos · 9 years ago
  44. 5e8351b Prevent division-by-zero in VCMFecMethod. by Peter Boström · 9 years ago
  45. 46eed76 Removing "candidates" attribute from TransportDescription. by deadbeef · 9 years ago
  46. e8f0836 Roll chromium_revision ea1b30c..ffa6c99 (371978:372122) by kjellander · 9 years ago
  47. fb15270 Replace const-reference with pointer in SendData. by Peter Boström · 9 years ago
  48. f4b9c77 Changed test to validate rtp timstamps not just in RTP packets but also in RTCP Sender Reports. by danilchap · 9 years ago
  49. 55b97fe clang-format -i -style=file webrtc/voice_engine/channel.* by kwiberg · 9 years ago
  50. 6043f2e Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #5 id:80001 of https://codereview.webrtc.org/1581693006/ ) by terelius · 9 years ago
  51. e73afba New rtc::VideoSinkInterface. by nisse · 9 years ago
  52. 533a4e4 Switch critical section locks out for atomic operations by tommi · 9 years ago
  53. bec70ab https://github.com/w3c/webrtc-stats/pull/10/files added mediaType to the tracks. The closest in the current stats is the ssrc type. by fippo · 9 years ago
  54. 6a062bd Deleted method AudioTrackInterface::GetRenderer. by nisse · 9 years ago
  55. edc978d Roll chromium_revision da1acd5..ea1b30c (371832:371978) by kjellander · 9 years ago
  56. ab8f82f Make ECDSA default for RTCPeerConnection by tkchin · 9 years ago
  57. 691b836 Using buffered signal to calculate the level of echo cancellation. by minyue · 9 years ago
  58. d162a5e Add shouldDisableBuffering to RTCFileLogger. by tkchin · 9 years ago
  59. 919ff75 Use high QP threshold for HW VP8 encoder frame downscaling. by glaznev · 9 years ago
  60. da2183c Update API for Objective-C RTCDataChannelConfiguration. by hjon · 9 years ago
  61. 08a6eab Adding "first packet received" notification to PeerConnectionObserver. by Taylor Brandstetter · 9 years ago
  62. d7a75d7 Roll chromium_revision c6ec25c..da1acd5 (371549:371832) by kjellander · 9 years ago
  63. 7b3c72f Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #4 id:60001 of https://codereview.webrtc.org/1581693006/ ) by deadbeef · 9 years ago
  64. 42265a8 Adding "first packet received" notification to PeerConnectionObserver. by Taylor Brandstetter · 9 years ago
  65. 80f1db9 Include relay protocol type when computing the turn candidate foundation. by Honghai Zhang · 9 years ago
  66. 3afc8c4 Consolidate SetSendParameters into one setter. by Peter Boström · 9 years ago
  67. ec2922f Change PeerConnectionFactory.setVideoHwAccelerationOptions to create shared Egl context for harware encoders and decoders. by Per · 9 years ago
  68. 2098fca Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ ) by nisse · 9 years ago
  69. a862d45 New rtc::VideoSinkInterface. by Niels Möller · 9 years ago
  70. f5dca48 Add transport sequence number on the non-pacer path of the rtp sender. by Stefan Holmer · 9 years ago
  71. 1c39098 Use rtc::time for all your timing needs! by Erik Språng · 9 years ago
  72. d673b0f [rtp_rtcp] Fix potentional time difference between rtp and rtcp packets. by Danil Chapovalov · 9 years ago
  73. b11e97a Move talk/media/webrtc/OWNERS to talk/media. by Peter Boström · 9 years ago
  74. 0b518bf Remove incorrect cast to AsyncSocketAdapter. by Peter Boström · 9 years ago
  75. bab934b H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding. by hbos · 9 years ago
  76. fab0a28 Fix BasicNetworkManager not to spam logs when internet is unreacheable. by Sergey Ulanov · 9 years ago
  77. 3ea1852 Sync build_ios_libs.sh script with http://webrtc.org/native-code/ios/ by hjon · 9 years ago
  78. 4cb3e39 Fix compilation if HAVE_WEBRTC_VIDEO is not defined. by jbauch · 9 years ago
  79. 6d49a8e Update API for Objective-C RTCConfiguration. by hjon · 9 years ago
  80. 7b582a2 Roll chromium_revision 2ca77c1..c6ec25c (371488:371549) by kjellander · 9 years ago
  81. a2c5523 Allow packets to be reordered in the fake network pipe. by philipel · 9 years ago
  82. 7fd8817 Fix type of local encoded length variable from uint32_t to size_t. by asapersson · 9 years ago
  83. 59b2d3e Remove zero-divide in VCMContentMetricsProcessing. by Peter Boström · 9 years ago
  84. 8327713 AudioCodingModuleImpl: Put CodecManager and Rent-A-Codec in a separate struct by kwiberg · 9 years ago
  85. d0c7bba [rtp_rtcp] Dlrr::SubBlock struct renamed to ReceiveTimeInfo by Danil Chapovalov · 9 years ago
  86. 5c7f110 Roll chromium_revision fb2e77c..2ca77c1 (371273:371488) by kjellander · 9 years ago
  87. 6a07f12 AudioCodingModuleImpl: Initialize encoder_stack_ to nullptr by kwiberg · 9 years ago
  88. 2bdcfad Revert of Removing webrtc::AudioFrame::energy_. (patchset #2 id:20001 of https://codereview.webrtc.org/1589953002/ ) by terelius · 9 years ago
  89. ffa3fdc Reallocate encoded buffer size if needed for VP8. Initially set to the input image size. by asapersson · 9 years ago
  90. e791ffd Remove non-monotonic clock support by sprang · 9 years ago
  91. 4fd6cda Add tracing to VCMGenericEncoder::Release. by Peter Boström · 9 years ago
  92. 86956de Small cleanup in VP9EncoderImpl::GetEncodedLayerFrame. by asapersson · 9 years ago
  93. bacae81 Remove webrtc::AudioFrame::energy_. by minyue · 9 years ago
  94. 58a80b5 Roll chromium_revision 717238e..fb2e77c (370438:371273) by kjellander · 9 years ago
  95. 85b22e2 Remove vp8_factory.{cc,h}. by Peter Boström · 9 years ago
  96. b332e5d Roll chromium_revision 6a04368..717238e (370362:370438) + tcmalloc by primiano · 9 years ago
  97. 28ba927 Switch to use new implementation in metrics.h. by asapersson · 9 years ago
  98. 5ad935c Remove mutable from rtc::CriticalSection members. by pbos · 9 years ago
  99. 7d0d0e0 Remove dead code from webrtc/base/timing.* by tommi · 9 years ago
  100. 9de632a Deleted unused enums MediaChannelOptions and VoiceMediaChannelOptions, by nisse · 9 years ago