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gerrit-public.fairphone.software / platform / external / webrtc / 757146baf1fa6157ff8d34fc8e8b6c33d6f84a51
  1. 757146b Remove PC factory options param from LocalAudioSource::Create. by deadbeef · 8 years ago
  2. 3f35e48 Roll chromium_revision 0dce9fc553..799869657a (449806:449840) by buildbot · 8 years ago
  3. 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 8 years ago
  4. 2c87d99 Roll chromium_revision 037b19bb21..0dce9fc553 (449745:449806) by buildbot · 8 years ago
  5. 81baed3 Add ability to return moved value from FunctorMessageHandler, Optional. by deadbeef · 8 years ago
  6. a4549d6 Fix SDP parsing crash due to missing track ID in "a=msid". by deadbeef · 8 years ago
  7. ef35b17 Roll chromium_revision cc96c42adf..037b19bb21 (449672:449745) by buildbot · 8 years ago
  8. abdc857 Update list of supported Android codecs based on field trial dynamically. by glaznev · 8 years ago
  9. 9238245 Fix nr of bytes sent to Opus decoder in DTX mode by flim · 8 years ago
  10. 90f1e1e Fixing SDP parsing crash due to invalid port numbers. by deadbeef · 8 years ago
  11. 612a497 Roll chromium_revision 3f4b691682..cc96c42adf (449626:449672) by buildbot · 8 years ago
  12. 5bd5ca3 Rename "PacketTransportInterface" to "PacketTransportInternal". by deadbeef · 8 years ago
  13. 640a200 Roll chromium_revision 182bfc3b6c..3f4b691682 (449577:449626) by buildbot · 8 years ago
  14. 88df0bc Make functions in fileutils.h use "const std::string&". by ehmaldonado · 8 years ago
  15. 46a0021 Retransmitted packets are now counted in receive time by ilnik · 8 years ago
  16. adb374b Remove henrik.lundin from webrtc/common_video/OWNERS by henrik.lundin · 8 years ago
  17. 087bd34 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 8 years ago
  18. 84a3759 Change rtc::VideoSinkWants to have target and a max pixel count by sprang · 8 years ago
  19. e9ad271 Increase the send-time history to 60 seconds. by stefan · 8 years ago
  20. 4ca1869 Allow residual echo detector to be enabled/disabled using AudioOptions, and no longer disable it on mobile platforms. by ivoc · 8 years ago
  21. bd344e5 Roll chromium_revision 7c7d51c594..182bfc3b6c (449558:449577) by buildbot · 8 years ago
  22. 0d729b3 Check for use_x11 before runnig desktop_capture_modules_tests on linux. by ehmaldonado · 8 years ago
  23. 38e9324 Add script for plotting statistics from webrtc integration test logs. by asapersson · 8 years ago
  24. 55c5be0 Remove unused methods in WebRtcVoiceEngine and VoiceMediaChannel. by solenberg · 8 years ago
  25. 654d54c Use std::unique_ptr in VideoProcessor. by asapersson · 8 years ago
  26. c29988c Roll chromium_revision 033d9e89ed..7c7d51c594 (449543:449558) by buildbot · 8 years ago
  27. 8a49763 Roll chromium_revision 18995b57d5..033d9e89ed (449497:449543) by buildbot · 8 years ago
  28. a03d438 Roll chromium_revision 5cd331dec1..18995b57d5 (449426:449497) by buildbot · 8 years ago
  29. faedf7f Getting rid of "benign blocking error" log spam. by deadbeef · 8 years ago
  30. 22e3970 Roll chromium_revision ba3e8dd8fa..5cd331dec1 (449304:449426) by buildbot · 8 years ago
  31. 5d52bf7 Roll chromium_revision baaeb3f30e..ba3e8dd8fa (449272:449304) by kjellander · 8 years ago
  32. 3795376 replace NtpTime->Clock with Clock->NtpTime dependency by danilchap · 8 years ago
  33. 1e1c84d Fixing typo by ilnik · 8 years ago
  34. 85d5ac7 Fix bug in recv-bwe tests introduced when switching to send-side bwe by default in tests. by Stefan Holmer · 8 years ago
  35. 86a6617 Roll chromium_revision cbcac91f7f..baaeb3f30e (449250:449272) by buildbot · 8 years ago
  36. 8443238 Remove rtcp_utility as mostly unused. by danilchap · 8 years ago
  37. 9def800 Added a flag to AudioCodecSpec to indicate adaptive bitrate support. by ossu · 8 years ago
  38. 0289364 Remove unused voe_stress_test.cc by solenberg · 8 years ago
  39. 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
  40. 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
  41. 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
  42. ae81217 Roll chromium_revision 2019b9e075..cbcac91f7f (449230:449250) by buildbot · 8 years ago
  43. 1752a10 Remove unused voe_cpu_test.cc. by solenberg · 8 years ago
  44. a48e1b6 Fix for left shift of potentially negative value. by ivoc · 8 years ago
  45. 2324b35 Remove unused voe_output_test.cc. by solenberg · 8 years ago
  46. 3029210 Move Android video quality loopback script. by kjellander · 8 years ago
  47. 234accd Roll chromium_revision cf2dce6a6d..2019b9e075 (448969:449230) by kjellander · 8 years ago
  48. 94a2f21 Increase STUN RTOs to work better on poor networks, such as 2G networks. by pthatcher · 8 years ago
  49. 1749bc3 Use fake clock in some more networks tests. by pthatcher · 8 years ago
  50. 4da058c Create an Obj-C wrapper of the RtpReceiverObserverInterface. by zhihuang · 8 years ago
  51. bb46b95 Add option to print information about configured SSRCs from RTC event logs. by terelius · 8 years ago
  52. ed1850a Log information (at level LS_INFO) about which overuse estimator is used. by terelius · 8 years ago
  53. 273f31b Fix for flaky RemoveOverheadFromBandwidth test. by michaelt · 8 years ago
  54. 87d11cd Reland of Avoid calling PostTask in audio callbacks (patchset #1 id:1 of https://codereview.webrtc.org/2684913003/ ) by henrika · 8 years ago
  55. 5d83780 Fix flaky test introduced by r16478 by stefan · 8 years ago
  56. 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
  57. 488c5dc Add new target direct_transport and remove fake_network and direct_transport from test_common. by perkj · 8 years ago
  58. 91873b7 Roll chromium_revision 70957b2671..cf2dce6a6d (448581:448969) by buildbot · 8 years ago
  59. e525d6a Revert Make the new jitter buffer the default jitter buffer. by stefan · 8 years ago
  60. 498ee8e Remove repeat flag from SendRTCP by danilchap · 8 years ago
  61. fd8f102 Revert of Avoid calling PostTask in audio callbacks (patchset #6 id:100001 of https://codereview.webrtc.org/2663383004/ ) by henrika · 8 years ago
  62. 2192089 Adding full initial version of delay estimation functionality in echo by peah · 8 years ago
  63. d4ed7f5 New tool for printing basic packet information from an RTC event log to stdout. by terelius · 8 years ago
  64. abcef5d Replace std::tr1::tuple by ::testing::tuple. by ehmaldonado · 8 years ago
  65. 5e5a072 iOS: Fix breakage caused by buildbot recipe update by Henrik Kjellander · 8 years ago
  66. b10f32f Adding more comments to every header file in api/ subdirectory. by deadbeef · 8 years ago
  67. 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
  68. 54b6e98 Added gn target for rtc_event_log2rtp_dump. by ivoc · 8 years ago
  69. 7798501 Fix the Chrome crash caused by RtcEventLog by zhihuang · 8 years ago
  70. 9dd77ba Clarifying error messages in ParseIceServerUrl for invalid transport parameters. by zstein · 8 years ago
  71. 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
  72. ed02c6d Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ ) by skvlad · 8 years ago
  73. 76bc8e8 Delete VideoReceiveStream::Config::pre_render_callback. by nisse · 8 years ago
  74. cd195be RTCInboundRTPStreamStats.qpSum collected. by hbos · 8 years ago
  75. c16fa5e Replace all use of the VERIFY macro. by nisse · 8 years ago
  76. ff0e72f Add QP sum stats for received streams. by sakal · 8 years ago
  77. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  78. 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
  79. 338f78a RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected. by hbos · 8 years ago
  80. 3443bb7 RTCRTPStreamStats.ssrc changed type to uint32_t. by hbos · 8 years ago
  81. 87b8e9f Add missing dependency to audio_decoder_unittests. by ehmaldonado · 8 years ago
  82. a53d4e7 Reduce parallel jobs in build_aar.py to 200 when building with goma. by sakal · 8 years ago
  83. 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
  84. 585a9b1 Refactor and clean-up relating to RTCCodecStats. by hbos · 8 years ago
  85. 040f5cc Roll chromium_revision 496a750d38..70957b2671 (447619:448581) by buildbot · 8 years ago
  86. b99b596 Add chromium-junit4 tag to instrumentation test AndroidManifests. by sakal · 8 years ago
  87. e0ac5a6 Use std::unique_ptr in VideoProcessorIntegrationTest. by asapersson · 8 years ago
  88. 1b21b9b Replace occurences of string by std::string. by ehmaldonado · 8 years ago
  89. 1634e16 Remove use of selectors matching Apple private API names. by kthelgason · 8 years ago
  90. 4a9a595 Make rtcp packets copyable by danilchap · 8 years ago
  91. 1959b63 Remove Assert lint suppression. by sakal · 8 years ago
  92. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  93. 6b3fcfd Add support for extra GN args to Android build script. by kjellander · 8 years ago
  94. 6b34124 Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
  95. f748ca4 Change order of tear down/create of default audio stream, to avoid starting/stopping audio card playout unnecessarily. by solenberg · 8 years ago
  96. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  97. f9b6e5e Fix KeepsHighBitrateWhenReconfiguringSender to avoid flakiness if probing succeeds in between encoder reconfigurations. by Stefan Holmer · 8 years ago
  98. 7a2d8ca Rewrite iOS FAT libraries build script in Python by oprypin · 8 years ago
  99. 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
  100. b77c716 Enable send-side BWE by default for video in call tests. by stefan · 8 years ago