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gerrit-public.fairphone.software / platform / external / webrtc / 64c4a7ecfc86307136e3dd07ac46462b56d60fda
  1. 64c4a7e Refactor webrtc/modules/audio_processing for GN check by ehmaldonado · 8 years ago
  2. b0a1111 Decode h264 fmtp sprop-parameter-sets to binary. by johan · 8 years ago
  3. c95d810 Roll chromium_revision b4a2bcaef4..bd4fdcd8d3 (437198:437221) by buildbot · 8 years ago
  4. 623427c Injectable output rate calculater for AudioMixer. by aleloi · 8 years ago
  5. 9abd275 Remove unused arguments and variable in MediaOptimization. by asapersson · 8 years ago
  6. 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
  7. 710c335 Add ability to scale to arbitrary factors by kthelgason · 8 years ago
  8. b39db84 Refactoring: Declare cricket::Codec constructors protected. by hta · 8 years ago
  9. 06a6984 Android classreferenceholder.h: Reorder function declaration keywords by magjed · 8 years ago
  10. 7e1070d Roll chromium_revision 3cd4e5505f..b4a2bcaef4 (436604:437198) by kjellander · 8 years ago
  11. 2cd872a Log BitBlt failure by zijiehe · 8 years ago
  12. c4adabf Create the Java Wrapper of RtpReceiverObserverInterface. by zhihuang · 8 years ago
  13. dd33e63 Translate isolated-script-test-output to dump_json_test_results in gtest-parallel-wrapper.py by ehmaldonado · 8 years ago
  14. 157a34d Add kwiberg@ to gtest-parallel OWNERS. by Peter Boström · 8 years ago
  15. 676e08f Refactor webrtc/{api,audio} and modules/audio_coding for GN check by kjellander · 8 years ago
  16. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  17. 7495c8c Clean up redundant include of ../webrtc_overrides by ehmaldonado · 8 years ago
  18. 5a4eb51 Roll gtest-parallel. by ehmaldonado · 8 years ago
  19. eca373f Adding OnReceivedOverhead to AudioEncoder. by minyue · 8 years ago
  20. ac382f3 Make ostream<< for enum class H264PacketizationMode by hta · 8 years ago
  21. e36c46e Use SSL_CTX_set_max_proto_version instead of SSL_CTX_set_max_version. by davidben · 8 years ago
  22. d3de4ab Remove deprecated comments by zijiehe · 8 years ago
  23. 49f34fd Relanding: Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  24. 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
  25. dd87d58 Add File::Open / Create functions to take an rtc::Pathname by zijiehe · 8 years ago
  26. bd28681 Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  27. ebbe4f2 Set the preferred DSCP value for Rtp data channel to be DSCP_AF41. by zhihuang · 8 years ago
  28. 66f99a4 Roll chromium_revision d8bf23963e..3cd4e5505f (436579:436604) by buildbot · 8 years ago
  29. 6d314c7 Reject XR TargetBitrate items with unsupported layer indices by sprang · 8 years ago
  30. 99cc10f Roll chromium_revision 092e72e03b..d8bf23963e (436556:436579) by buildbot · 8 years ago
  31. 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
  32. 28c2487 Add unit tests for avfoundation format mapper functions. by denicija · 8 years ago
  33. 768c648 Move /webrtc/api/android files to /webrtc/sdk/android by magjed · 8 years ago
  34. 45bb513 Reland of APM: Change 3 UMA metrics to fewer but linearly distributed buckets (patchset #1 id:1 of https://codereview.webrtc.org/2548333002/ ) by henrik.lundin · 8 years ago
  35. 243a0a7 Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ ) by hta · 8 years ago
  36. dbc960c The level controller complexity tests have lately been by peah · 8 years ago
  37. c9badd5 Add comment to metrics.h by henrik.lundin · 8 years ago
  38. 68d3213 RTPPayloadRegistry: Stop using the rate to keep track of receive codecs by kwiberg · 8 years ago
  39. e59647b This approach passes packetization mode to the encoder as part of by hta · 8 years ago
  40. 5f0ddf2 Roll chromium_revision 4f26eb8a56..092e72e03b (436542:436556) by buildbot · 8 years ago
  41. 406616f Fix spelling mistake in rtp_rtcp.h. by brandtr · 8 years ago
  42. 6612cb7 Roll chromium_revision c4e9c3c5de..4f26eb8a56 (436513:436542) by buildbot · 8 years ago
  43. 7439f97 Split targets mixing .c and .cc sources. by kjellander · 8 years ago
  44. 5f7a9dc Roll chromium_revision c9600cd059..c4e9c3c5de (436473:436513) by buildbot · 8 years ago
  45. ea7a33d Roll chromium_revision f5e72ebfb7..c9600cd059 (436420:436473) by buildbot · 8 years ago
  46. 548d321 Roll chromium_revision 1376f2e121..f5e72ebfb7 (436364:436420) by buildbot · 8 years ago
  47. c9f9500 Expose audio_jitter_buffer_fast_accelerate config to objc wrapper by haysc · 8 years ago
  48. 5fe4d49 Remove unsupported mac framework target. by kthelgason · 8 years ago
  49. 61a2af3 Roll chromium_revision d43cbc46d5..1376f2e121 (436297:436364) by buildbot · 8 years ago
  50. bd681b9 AGC: Route clipping parameter from webrtc::Config to AGC by henrik.lundin · 8 years ago
  51. db752f9 Revert "Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )" by stefan · 8 years ago
  52. 53e0980 Roll chromium_revision 49f56a2e7b..d43cbc46d5 (436268:436297) by buildbot · 8 years ago
  53. 37535bf Refactor fileutils.cc/h and fileutils_unittests.cc into their own targets. by ehmaldonado · 8 years ago
  54. 1d08100 Use RTC_DISALLOW_COPY_AND_ASSIGN in webrtc/base/sigslottester.h by kjellander · 8 years ago
  55. d654a9b Reduce number of FlexFEC VideoSendStreamTests and lower packet loss. by brandtr · 8 years ago
  56. 63407a9 Revert of APM: Change 3 UMA metrics to fewer but linearly distributed buckets (patchset #2 id:20001 of https://codereview.webrtc.org/2547593002/ ) by henrik.lundin · 8 years ago
  57. dbf729c Roll chromium_revision a253a1bb31..49f56a2e7b (436254:436268) by buildbot · 8 years ago
  58. 49715fe APM: Change 3 UMA metrics to fewer but linearly distributed buckets by henrik.lundin · 8 years ago
  59. fa07b91 Delete unused spreadsort implementation. by nisse · 8 years ago
  60. 29c9dda Roll chromium_revision 60f4f97afb..a253a1bb31 (436236:436254) by buildbot · 8 years ago
  61. e545e5d RtpPacketizer::NextPacket fills RtpPacket instead of just payload. by danilchap · 8 years ago
  62. f00082d Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h by henrik.lundin · 8 years ago
  63. 6321b49 Move functionality out from AudioFrame and into AudioFrameOperations. by aleloi · 8 years ago
  64. bd9bdf6 Roll chromium_revision b5ed348a58..60f4f97afb (436228:436236) by buildbot · 8 years ago
  65. deb1f8a Roll chromium_revision cfcd3337a6..b5ed348a58 (436209:436228) by buildbot · 8 years ago
  66. 280be7d Roll chromium_revision 5f3032e780..cfcd3337a6 (436201:436209) by buildbot · 8 years ago
  67. 595d15c Roll chromium_revision dfb4c71778..5f3032e780 (436198:436201) by buildbot · 8 years ago
  68. a887f7c Roll chromium_revision b0dfc121a4..dfb4c71778 (436194:436198) by buildbot · 8 years ago
  69. 347908f Roll chromium_revision 0740b0dfd2..b0dfc121a4 (436192:436194) by buildbot · 8 years ago
  70. a99e52f Roll chromium_revision 5bbd80fb56..0740b0dfd2 (436191:436192) by buildbot · 8 years ago
  71. d3b79f5 Roll chromium_revision 62ef852f1d..5bbd80fb56 (436188:436191) by buildbot · 8 years ago
  72. 6b5789c Roll chromium_revision 4c53995f0b..62ef852f1d (436184:436188) by buildbot · 8 years ago
  73. aa5c7b9 Roll chromium_revision 1b019f5fd3..4c53995f0b (436175:436184) by buildbot · 8 years ago
  74. c46e0b7 Roll chromium_revision 8cbbe0ffc3..1b019f5fd3 (436165:436175) by buildbot · 8 years ago
  75. c191bd3 Roll chromium_revision 0a4f767df7..8cbbe0ffc3 (436162:436165) by buildbot · 8 years ago
  76. 7ba6e02 Roll chromium_revision 64637ac52e..0a4f767df7 (436108:436162) by buildbot · 8 years ago
  77. 4c6696c Revert of Disable P2PTestConductor.LocalP2PTestDtlsBundleInIceRestart under msan (patchset #1 id:1 of https://codereview.webrtc.org/2546913003/ ) by deadbeef · 8 years ago
  78. a59b012 Roll chromium_revision f857831468..64637ac52e (436038:436108) by buildbot · 8 years ago
  79. c23efae Allow locally rendered video to be downscaled in end-to-end tests. by deadbeef · 8 years ago
  80. c63b894 Modify the parameter type of PeerConnectionObserver callback OnAddTrack. by zhihuang · 8 years ago
  81. d5c93e0 Roll chromium_revision 5e077354e9..f857831468 (435981:436038) by buildbot · 8 years ago
  82. 2e59a02 Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ ) by deadbeef · 8 years ago
  83. ef0cd56 Roll chromium_revision 58c06a5f1b..5e077354e9 (435946:435981) by buildbot · 8 years ago
  84. 9cef11b Fix exponential probing in ProbeController. by sergeyu · 8 years ago
  85. 5e6c52c Roll chromium_revision dd10720676..58c06a5f1b (435924:435946) by buildbot · 8 years ago
  86. 4eb03c7 Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled. by asapersson · 8 years ago
  87. 5932149 Use different restrictions of acked bitrate lag depending on operating point. by stefan · 8 years ago
  88. a790d83 Wire up rtcp xr target bitrate on receive side. by sprang · 8 years ago
  89. 9afbfc0 Roll chromium_revision f50152dfc4..dd10720676 (435897:435924) by buildbot · 8 years ago
  90. 0f01c7f Added tool for reference less video analysis (go/refless-video-analysis) by charujain · 8 years ago
  91. b2250e5 New gn target video_frame_api. by nisse · 8 years ago
  92. 969b12f Remove xdisplaycheck by kjellander · 8 years ago
  93. df28e47 fix coding and documentary ambiguity in AimdRateControl::TimeToReduceFurther. by howtofly · 8 years ago
  94. a28a1b9 Disable P2PTestConductor.LocalP2PTestDtlsBundleInIceRestart under msan by henrik.lundin · 8 years ago
  95. 5c71166 VP8DecoderImpl: Fix uninitialized memory crash by magjed · 8 years ago
  96. 00bceb1 Deprecated SetAudioPacketSize from RTPSender and removed calls to it. by ossu · 8 years ago
  97. e066b30 Remove API-related #defines from voice_engine_configurations.h by henrik.lundin · 8 years ago
  98. 15cf9f9 Roll chromium_revision 601e4f48a3..f50152dfc4 (435870:435897) by buildbot · 8 years ago
  99. b336392 Sanity check parsed QP values from H264 bitstream by kthelgason · 8 years ago
  100. 864f58b Roll chromium_revision 353f713f3d..601e4f48a3 (435846:435870) by buildbot · 8 years ago