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gerrit-public.fairphone.software / platform / external / webrtc / c16fa5ea692e06992c30c8b9853356b90da74e2a
  1. c16fa5e Replace all use of the VERIFY macro. by nisse · 8 years ago
  2. ff0e72f Add QP sum stats for received streams. by sakal · 8 years ago
  3. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  4. 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
  5. 338f78a RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected. by hbos · 8 years ago
  6. 3443bb7 RTCRTPStreamStats.ssrc changed type to uint32_t. by hbos · 8 years ago
  7. 87b8e9f Add missing dependency to audio_decoder_unittests. by ehmaldonado · 8 years ago
  8. a53d4e7 Reduce parallel jobs in build_aar.py to 200 when building with goma. by sakal · 8 years ago
  9. 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
  10. 585a9b1 Refactor and clean-up relating to RTCCodecStats. by hbos · 8 years ago
  11. 040f5cc Roll chromium_revision 496a750d38..70957b2671 (447619:448581) by buildbot · 8 years ago
  12. b99b596 Add chromium-junit4 tag to instrumentation test AndroidManifests. by sakal · 8 years ago
  13. e0ac5a6 Use std::unique_ptr in VideoProcessorIntegrationTest. by asapersson · 8 years ago
  14. 1b21b9b Replace occurences of string by std::string. by ehmaldonado · 8 years ago
  15. 1634e16 Remove use of selectors matching Apple private API names. by kthelgason · 8 years ago
  16. 4a9a595 Make rtcp packets copyable by danilchap · 8 years ago
  17. 1959b63 Remove Assert lint suppression. by sakal · 8 years ago
  18. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  19. 6b3fcfd Add support for extra GN args to Android build script. by kjellander · 8 years ago
  20. 6b34124 Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
  21. f748ca4 Change order of tear down/create of default audio stream, to avoid starting/stopping audio card playout unnecessarily. by solenberg · 8 years ago
  22. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  23. f9b6e5e Fix KeepsHighBitrateWhenReconfiguringSender to avoid flakiness if probing succeeds in between encoder reconfigurations. by Stefan Holmer · 8 years ago
  24. 7a2d8ca Rewrite iOS FAT libraries build script in Python by oprypin · 8 years ago
  25. 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
  26. b77c716 Enable send-side BWE by default for video in call tests. by stefan · 8 years ago
  27. 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
  28. d40b0f3 Improve and re-enable FEC end-to-end tests. by brandtr · 8 years ago
  29. cb789bb Remove NewApi lint suppression. by sakal · 8 years ago
  30. 93e1e23 Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call. by asapersson · 8 years ago
  31. 447dba9 Add debuggable=true to AppRTCMobile manifest. by henrika · 8 years ago
  32. b114e9c Camera2Session: Add return statements after reportError where needed. by sakal · 8 years ago
  33. 873fcb9 Drop the check for stray mobileprovision (no longer needed) by oprypin · 8 years ago
  34. 61202ac Ensure that AEC3 is not run in tandem with AEC2 by peah · 8 years ago
  35. 237e1bb Fix potential use after free in H264 packetizer. by sprang · 8 years ago
  36. 60f7c63 Remove temporary AddRtxInfo member function. by brandtr · 8 years ago
  37. d44ce05 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  38. 656610f Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common. by ehmaldonado · 8 years ago
  39. a7111eb Fixing an error in ANA FrameLengthController unittest. by minyue · 8 years ago
  40. e702b30 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 8 years ago
  41. d1f5fda Allow changing the minimal ICE ping timeout with PeerConnection.SetConfiguration. by skvlad · 8 years ago
  42. 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 · 8 years ago
  43. 390e64d Add VP9 full stack tests: - ConferenceMotionHd2000kbps100msLimitedQueueVP9 by jianj · 8 years ago
  44. 53b6cc3 Reland of Enable audio streams to send padding. (patchset #4 id:60001 of https://codereview.webrtc.org/2652893004/ ) by stefan · 8 years ago
  45. b11fb25 Protect APM in webkit builds. by agouaillard · 8 years ago
  46. 9d58d94 Fix and improve FlexFEC configuration for RTP/RTCP. by brandtr · 8 years ago
  47. 4cb1b75 Extends timer from 10 to 30 seconds for output volume check on Android. by henrika · 8 years ago
  48. 77ce9a5 Avoid calling PostTask in audio callbacks. by henrika · 8 years ago
  49. 5f47126 Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. by ilnik · 8 years ago
  50. 4b512d7 Fix Chromium FYI bot by skvlad · 8 years ago
  51. d030912 Pick the DTLS handshake timeout based on the ICE RTT estimate by skvlad · 8 years ago
  52. a24a9e2 Get rid of unqualified std:: types. by deadbeef · 8 years ago
  53. 6741516 Implement new PeerConnection certificate policy API in ObjC API by hnsl · 8 years ago
  54. a5d94ff Objective-C API to set the ICE check rate through RTCConfiguration. by skvlad · 8 years ago
  55. b55bd5f Don't capture variables explicitly in lambda expression. by ehmaldonado · 8 years ago
  56. 5107246 Allow applications to limit the ICE check rate through RTCConfiguration by skvlad · 8 years ago
  57. e5bd702 Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ ) by philipel · 8 years ago
  58. 8c61924 video_coding::PacketBuffer now group all H264 packets with the same timestamp into the same frame. by philipel · 8 years ago
  59. 1dffc62 Remove all occurrences of "using std::string". by ehmaldonado · 8 years ago
  60. e372d3c Add event log visualization of rtp timestamps over time. by stefan · 8 years ago
  61. a55f021 Add 120ms frame ability to ANA by michaelt · 8 years ago
  62. ed01647 Remove bad DCHECK added as part of https://codereview.webrtc.org/2452163004/ by solenberg · 8 years ago
  63. b33eed2 Fix perf issue when timinig out receiver infos in RTCP. by stefan · 8 years ago
  64. cc99bc2 Change StunMessage::AddAttribute return type from bool to void. by nisse · 8 years ago
  65. f7826d6 Remove InlinedApi lint ignore. by sakal · 8 years ago
  66. a29d5ec Make 'webrtc' target a complete static library on Linux, Android and Windows by kjellander · 8 years ago
  67. 24af663 Adding Java wrapper for DtmfSender. by deadbeef · 8 years ago
  68. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  69. 2e03c66 Adding build switch for Opus that supports 120ms ptime. by minyue · 8 years ago
  70. d3d3ba5 Revert of Enable audio streams to send padding. (patchset #4 id:60001 of https://codereview.webrtc.org/2652893004/ ) by deadbeef · 8 years ago
  71. 1cbf518 Roll chromium_revision 6b2002254c..496a750d38 (447561:447619) by buildbot · 8 years ago
  72. 353e7e1 Roll chromium_revision 9f2c537112..6b2002254c (447517:447561) by buildbot · 8 years ago
  73. e35f89a Enable audio streams to send padding. by stefan · 8 years ago
  74. 46fbb7d Roll chromium_revision ccc17b815a..9f2c537112 (447493:447517) by buildbot · 8 years ago
  75. b1ca073 Rename adaptation api methods, extended vie_encoder unit test. by sprang · 8 years ago
  76. d83b967 Replace consecutive-losses count by a calculation of first-order-FEC recoverability. by elad.alon · 8 years ago
  77. 14245cc Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  78. 77f0580 Add new step graph type to event log visualization tool. Currently used for bitrate estimate and accumulated packet count, but could in general be used for any metric that is piecewise constant. by terelius · 8 years ago
  79. a565f92 Roll chromium_revision e87481817b..ccc17b815a (447482:447493) by buildbot · 8 years ago
  80. 099110c Don't send audio packets if the network is down. by stefan · 8 years ago
  81. 4637b6a Consistent 30% improvement in audio mixer running time. by aleloi · 8 years ago
  82. 35fc2aa Revert of Drop frames until specified bitrate is achieved. (patchset #12 id:240001 of https://codereview.webrtc.org/2630333002/ ) by minyue · 8 years ago
  83. 2ad42ca Roll chromium_revision 8346af5a71..e87481817b (447464:447482) by buildbot · 8 years ago
  84. 6d4dd59 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  85. 803dc29 Enable cpplint and fix cpplint errors in webrtc/api by oprypin · 8 years ago
  86. 83399ca Drop frames until specified bitrate is achieved. by kthelgason · 8 years ago
  87. fdd9b85 Roll chromium_revision e4d460e023..8346af5a71 (447441:447464) by buildbot · 8 years ago
  88. a1cf88d Roll chromium_revision 9d90548426..e4d460e023 (447390:447441) by buildbot · 8 years ago
  89. 3f6d817 Roll chromium_revision 2ed48364ed..9d90548426 (447343:447390) by buildbot · 8 years ago
  90. dc20e26 Use correct calling convention for CreateThread callback on Windows. by deadbeef · 8 years ago
  91. 3e4ebc7 Roll chromium_revision 0851a43de7..2ed48364ed (447237:447343) by buildbot · 8 years ago
  92. ac61b74 Refactor FakeAudioDevice to have separate methods for starting recording and playout. by perkj · 8 years ago
  93. 1c05625 Fix race condition in FrameBuffer2 by philipel · 8 years ago
  94. 54340d8 Change opus min bitrate. by michaelt · 8 years ago
  95. cf34fde Roll chromium_revision 721746ebca..0851a43de7 (447221:447237) by buildbot · 8 years ago
  96. 7f08e82 Fix per regression in probing. by stefan · 8 years ago
  97. 6fb4f56 Reland of move usage of deprecated g_type_init API (patchset #1 id:1 of https://codereview.webrtc.org/2666103002/ ) by oprypin · 8 years ago
  98. d1685ab Revert of Remove usage of deprecated g_type_init API (patchset #1 id:1 of https://codereview.webrtc.org/2660823003/ ) by oprypin · 8 years ago
  99. 0fe1216 Move more calls to webrtc::field_trial::FindFullName into ctor, thereby minimizing the non-trivial cost of repeated string comparisons. by elad.alon · 8 years ago
  100. 89f281c Roll chromium_revision f74de5a3c9..721746ebca (447212:447221) by buildbot · 8 years ago