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gerrit-public.fairphone.software / platform / external / webrtc / 654d54c07385cf46f17693906aacb7716a2a3e14
  1. 654d54c Use std::unique_ptr in VideoProcessor. by asapersson · 8 years ago
  2. c29988c Roll chromium_revision 033d9e89ed..7c7d51c594 (449543:449558) by buildbot · 8 years ago
  3. 8a49763 Roll chromium_revision 18995b57d5..033d9e89ed (449497:449543) by buildbot · 8 years ago
  4. a03d438 Roll chromium_revision 5cd331dec1..18995b57d5 (449426:449497) by buildbot · 8 years ago
  5. faedf7f Getting rid of "benign blocking error" log spam. by deadbeef · 8 years ago
  6. 22e3970 Roll chromium_revision ba3e8dd8fa..5cd331dec1 (449304:449426) by buildbot · 8 years ago
  7. 5d52bf7 Roll chromium_revision baaeb3f30e..ba3e8dd8fa (449272:449304) by kjellander · 8 years ago
  8. 3795376 replace NtpTime->Clock with Clock->NtpTime dependency by danilchap · 8 years ago
  9. 1e1c84d Fixing typo by ilnik · 8 years ago
  10. 85d5ac7 Fix bug in recv-bwe tests introduced when switching to send-side bwe by default in tests. by Stefan Holmer · 8 years ago
  11. 86a6617 Roll chromium_revision cbcac91f7f..baaeb3f30e (449250:449272) by buildbot · 8 years ago
  12. 8443238 Remove rtcp_utility as mostly unused. by danilchap · 8 years ago
  13. 9def800 Added a flag to AudioCodecSpec to indicate adaptive bitrate support. by ossu · 8 years ago
  14. 0289364 Remove unused voe_stress_test.cc by solenberg · 8 years ago
  15. 3dd5ad9 Reland of Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. (patchset #2 id:150001 of https://codereview.webrtc.org/2687073002/ ) by ilnik · 8 years ago
  16. cc452e1 Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ ) by sakal · 8 years ago
  17. e67c59e Revert of Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. (patchset #5 id:80001 of https://codereview.webrtc.org/2668763004/ ) by ilnik · 8 years ago
  18. ae81217 Roll chromium_revision 2019b9e075..cbcac91f7f (449230:449250) by buildbot · 8 years ago
  19. 1752a10 Remove unused voe_cpu_test.cc. by solenberg · 8 years ago
  20. a48e1b6 Fix for left shift of potentially negative value. by ivoc · 8 years ago
  21. 2324b35 Remove unused voe_output_test.cc. by solenberg · 8 years ago
  22. 3029210 Move Android video quality loopback script. by kjellander · 8 years ago
  23. 234accd Roll chromium_revision cf2dce6a6d..2019b9e075 (448969:449230) by kjellander · 8 years ago
  24. 94a2f21 Increase STUN RTOs to work better on poor networks, such as 2G networks. by pthatcher · 8 years ago
  25. 1749bc3 Use fake clock in some more networks tests. by pthatcher · 8 years ago
  26. 4da058c Create an Obj-C wrapper of the RtpReceiverObserverInterface. by zhihuang · 8 years ago
  27. bb46b95 Add option to print information about configured SSRCs from RTC event logs. by terelius · 8 years ago
  28. ed1850a Log information (at level LS_INFO) about which overuse estimator is used. by terelius · 8 years ago
  29. 273f31b Fix for flaky RemoveOverheadFromBandwidth test. by michaelt · 8 years ago
  30. 87d11cd Reland of Avoid calling PostTask in audio callbacks (patchset #1 id:1 of https://codereview.webrtc.org/2684913003/ ) by henrika · 8 years ago
  31. 5d83780 Fix flaky test introduced by r16478 by stefan · 8 years ago
  32. 0e3213a Fix bug in BitrateProber where an old probe added at a high bitrate will stay active indefinitely if the bandwidth estimate becomes too low to probe at that bitrate. by Stefan Holmer · 8 years ago
  33. 488c5dc Add new target direct_transport and remove fake_network and direct_transport from test_common. by perkj · 8 years ago
  34. 91873b7 Roll chromium_revision 70957b2671..cf2dce6a6d (448581:448969) by buildbot · 8 years ago
  35. e525d6a Revert Make the new jitter buffer the default jitter buffer. by stefan · 8 years ago
  36. 498ee8e Remove repeat flag from SendRTCP by danilchap · 8 years ago
  37. fd8f102 Revert of Avoid calling PostTask in audio callbacks (patchset #6 id:100001 of https://codereview.webrtc.org/2663383004/ ) by henrika · 8 years ago
  38. 2192089 Adding full initial version of delay estimation functionality in echo by peah · 8 years ago
  39. d4ed7f5 New tool for printing basic packet information from an RTC event log to stdout. by terelius · 8 years ago
  40. abcef5d Replace std::tr1::tuple by ::testing::tuple. by ehmaldonado · 8 years ago
  41. 5e5a072 iOS: Fix breakage caused by buildbot recipe update by Henrik Kjellander · 8 years ago
  42. b10f32f Adding more comments to every header file in api/ subdirectory. by deadbeef · 8 years ago
  43. 76e02cd Reland of ll chromium_revision 496a750d38..70957b2671 (447619:448581) (patchset #1 id:1 of https://codereview.webrtc.org/2680743003/ ) by kjellander · 8 years ago
  44. 54b6e98 Added gn target for rtc_event_log2rtp_dump. by ivoc · 8 years ago
  45. 7798501 Fix the Chrome crash caused by RtcEventLog by zhihuang · 8 years ago
  46. 9dd77ba Clarifying error messages in ParseIceServerUrl for invalid transport parameters. by zstein · 8 years ago
  47. 69fb2cc Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ ) by skvlad · 8 years ago
  48. ed02c6d Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ ) by skvlad · 8 years ago
  49. 76bc8e8 Delete VideoReceiveStream::Config::pre_render_callback. by nisse · 8 years ago
  50. cd195be RTCInboundRTPStreamStats.qpSum collected. by hbos · 8 years ago
  51. c16fa5e Replace all use of the VERIFY macro. by nisse · 8 years ago
  52. ff0e72f Add QP sum stats for received streams. by sakal · 8 years ago
  53. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  54. 2bc6864 Reland of Drop frames until specified bitrate is achieved. (patchset #1 id:1 of https://codereview.webrtc.org/2666303002/ ) by kthelgason · 8 years ago
  55. 338f78a RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected. by hbos · 8 years ago
  56. 3443bb7 RTCRTPStreamStats.ssrc changed type to uint32_t. by hbos · 8 years ago
  57. 87b8e9f Add missing dependency to audio_decoder_unittests. by ehmaldonado · 8 years ago
  58. a53d4e7 Reduce parallel jobs in build_aar.py to 200 when building with goma. by sakal · 8 years ago
  59. f81be0a Revert of Roll chromium_revision 496a750d38..70957b2671 (447619:448581) (patchset #1 id:1 of https://codereview.webrtc.org/2683593002/ ) by kjellander · 8 years ago
  60. 585a9b1 Refactor and clean-up relating to RTCCodecStats. by hbos · 8 years ago
  61. 040f5cc Roll chromium_revision 496a750d38..70957b2671 (447619:448581) by buildbot · 8 years ago
  62. b99b596 Add chromium-junit4 tag to instrumentation test AndroidManifests. by sakal · 8 years ago
  63. e0ac5a6 Use std::unique_ptr in VideoProcessorIntegrationTest. by asapersson · 8 years ago
  64. 1b21b9b Replace occurences of string by std::string. by ehmaldonado · 8 years ago
  65. 1634e16 Remove use of selectors matching Apple private API names. by kthelgason · 8 years ago
  66. 4a9a595 Make rtcp packets copyable by danilchap · 8 years ago
  67. 1959b63 Remove Assert lint suppression. by sakal · 8 years ago
  68. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  69. 6b3fcfd Add support for extra GN args to Android build script. by kjellander · 8 years ago
  70. 6b34124 Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
  71. f748ca4 Change order of tear down/create of default audio stream, to avoid starting/stopping audio card playout unnecessarily. by solenberg · 8 years ago
  72. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  73. f9b6e5e Fix KeepsHighBitrateWhenReconfiguringSender to avoid flakiness if probing succeeds in between encoder reconfigurations. by Stefan Holmer · 8 years ago
  74. 7a2d8ca Rewrite iOS FAT libraries build script in Python by oprypin · 8 years ago
  75. 1134b7b Reland of Improve and re-enable FEC end-to-end tests. (patchset #1 id:1 of https://codereview.webrtc.org/2672373002/ ) by brandtr · 8 years ago
  76. b77c716 Enable send-side BWE by default for video in call tests. by stefan · 8 years ago
  77. fd8d265 Revert of Improve and re-enable FEC end-to-end tests. (patchset #3 id:40001 of https://codereview.webrtc.org/2675573004/ ) by brandtr · 8 years ago
  78. d40b0f3 Improve and re-enable FEC end-to-end tests. by brandtr · 8 years ago
  79. cb789bb Remove NewApi lint suppression. by sakal · 8 years ago
  80. 93e1e23 Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call. by asapersson · 8 years ago
  81. 447dba9 Add debuggable=true to AppRTCMobile manifest. by henrika · 8 years ago
  82. b114e9c Camera2Session: Add return statements after reportError where needed. by sakal · 8 years ago
  83. 873fcb9 Drop the check for stray mobileprovision (no longer needed) by oprypin · 8 years ago
  84. 61202ac Ensure that AEC3 is not run in tandem with AEC2 by peah · 8 years ago
  85. 237e1bb Fix potential use after free in H264 packetizer. by sprang · 8 years ago
  86. 60f7c63 Remove temporary AddRtxInfo member function. by brandtr · 8 years ago
  87. d44ce05 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  88. 656610f Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common. by ehmaldonado · 8 years ago
  89. a7111eb Fixing an error in ANA FrameLengthController unittest. by minyue · 8 years ago
  90. e702b30 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 9 years ago
  91. d1f5fda Allow changing the minimal ICE ping timeout with PeerConnection.SetConfiguration. by skvlad · 9 years ago
  92. 98c4374 Allow passing network config constraint as base64 encoded string to preserve values of serialized protos. The values are a serialized byte stream packed into a std::string. To be represented as a NSString they must be base64 encoded or bytes outside of the ASCII range will be encoded into multi byte UTF8 sequences by default. by haysc · 9 years ago
  93. 390e64d Add VP9 full stack tests: - ConferenceMotionHd2000kbps100msLimitedQueueVP9 by jianj · 9 years ago
  94. 53b6cc3 Reland of Enable audio streams to send padding. (patchset #4 id:60001 of https://codereview.webrtc.org/2652893004/ ) by stefan · 9 years ago
  95. b11fb25 Protect APM in webkit builds. by agouaillard · 9 years ago
  96. 9d58d94 Fix and improve FlexFEC configuration for RTP/RTCP. by brandtr · 9 years ago
  97. 4cb1b75 Extends timer from 10 to 30 seconds for output volume check on Android. by henrika · 9 years ago
  98. 77ce9a5 Avoid calling PostTask in audio callbacks. by henrika · 9 years ago
  99. 5f47126 Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. by ilnik · 9 years ago
  100. 4b512d7 Fix Chromium FYI bot by skvlad · 9 years ago