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gerrit-public.fairphone.software / platform / external / webrtc / 7268f28520a4b011abb26e4fffa0dde7fcc30f64
  1. 7268f28 General cleanup on AppRTC Android Demo codebase by sakal · 8 years ago
  2. 2036135 AudioEncoderOpus: Default to 32 kbit/s for mono, 64 kbit/s for stereo by kwiberg · 8 years ago
  3. 3fc8e8d Fix UBSan errors (left shift of negative value) by kwiberg · 8 years ago
  4. f8312cb Convert LooperExecutorTest in AppRTC Demo into JUnit test by sakal · 8 years ago
  5. 2d285ca Remove AppRTCUtils.NonThreadSafe in AppRTC Android Demo by sakal · 8 years ago
  6. 7cf1147 Replace LooperExecutor with built-in class in Android AppRTC Demo by sakal · 8 years ago
  7. 04ebea3 Delete obsolete cricket::VideoFrame methods. by nisse · 8 years ago
  8. 94616b3 Disabled flaky test TestFrameBuffer2.OneLayerStreamReordered by philipel · 8 years ago
  9. 3062995 Fix UBSan errors (left shift on negative value) by minyue · 8 years ago
  10. 03d45b0 base.gyp: Add conditions for NaCl builds. by kjellander@webrtc.org · 8 years ago
  11. 6c47031 Roll chromium_revision 21279e564e..9db5a0e3f5 (394725:394957) by buildbot · 8 years ago
  12. 2b3bf6b Re-enabling socket tests that were previously flaky. by Taylor Brandstetter · 8 years ago
  13. 7d01331 Only initialize usrsctp when it's used and uninitialize when it's not being used. by Tommi · 8 years ago
  14. e725f7c Turned AudioDecoderFactory into a RefCounted thing to use with scoped_refptr. by ossu · 8 years ago
  15. 6e8224f Reduce flakyness of timing dependent tests for TestFrameBuffer2.*. by philipel · 8 years ago
  16. 00b9d21 Set ViEEncoder sink_ on construction. by Peter Boström · 8 years ago
  17. 8e572f0 Adds macros to annotate variables and functions used from same thread or queue. by danilchap · 8 years ago
  18. 604abe0 VideoAdapter: Drop frames based on actual fps instead of expected fps by magjed · 8 years ago
  19. 0026dd8 Fix bug in gyp_webrtc.py when DEPOT_TOOLS_WIN_TOOLCHAIN=0 by kjellander@webrtc.org · 8 years ago
  20. bd76607 Revert of Android: Make base interface for camera1 and camera2 (patchset #1 id:1 of https://codereview.webrtc.org/1994893002/ ) by magjed · 8 years ago
  21. efafd7f Roll chromium_revision 36d2aa2331..21279e564e (394634:394725) by buildbot · 8 years ago
  22. d269b02 Reland of Android: Make base interface for camera1 and camera2 (patchset #1 id:1 of https://codereview.webrtc.org/1979583002/ ) by magjed · 8 years ago
  23. be7a9e5 Revert "Revert of FrameBuffer for the new jitter buffer. (patchset #9 id:160001 of https://codereview.webrtc.org/1969403007/ )" by philipel · 8 years ago
  24. 84f9b98 Update AppRTCDemo AppRTC URL's on iOS to appr.tc by jansson · 8 years ago
  25. f910ecd Add test for DirectRTCClient in AppRTC Demo for Android by sakal · 8 years ago
  26. fadb43e Roll chromium_revision cccd8082d7..36d2aa2331 (394463:394634) by buildbot · 8 years ago
  27. 91dd567 Only use PortAllocator on the network thread. by deadbeef · 8 years ago
  28. be0c96f Add ice_candidate_pool_size to Obj-C and Java RTCConfiguration. by deadbeef · 8 years ago
  29. b711f10 Revert of FrameBuffer for the new jitter buffer. (patchset #9 id:160001 of https://codereview.webrtc.org/1969403007/ ) by honghaiz · 8 years ago
  30. e230d36 Roll chromium_revision 34689ee3d1..cccd8082d7 (394374:394463) by buildbot · 8 years ago
  31. a376e70 FrameBuffer for the new jitter buffer. by philipel · 8 years ago
  32. a89ab96 Enable muted state by default in VoE by henrik.lundin · 8 years ago
  33. a3c2c3e Remove unneccessary log in PacedSender. by Stefan Holmer · 8 years ago
  34. 83e8286 AEC: Add UMA logging of buffer re-alignment by henrik.lundin · 8 years ago
  35. 42dda50 Propagate muted info from VoE Channel to AudioConferenceMixer by henrik.lundin · 8 years ago
  36. 84f8df7 Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ ) by kjellander · 8 years ago
  37. 2f5d600 Revert of Remove Android x86 compilation trybot from CQ. (patchset #1 id:1 of https://codereview.webrtc.org/1959923002/ ) by kjellander · 8 years ago
  38. 99111bb Roll chromium_revision 91b474353e..34689ee3d1 (394313:394374) by buildbot · 8 years ago
  39. 7bb6e75 Add missing headers and fix some missing dependencies by kjellander · 8 years ago
  40. 299ccde Direct IP connect functionality for AppRTC Android demo. by sakal · 8 years ago
  41. 5a216d0 Add muted parameter to audio_frame_manipulator methods by henrik.lundin · 8 years ago
  42. 89f237c Fix UBSan errors (left shift of negative value, left shift overflows int) by kwiberg · 8 years ago
  43. 60200d1 Add FrameAndMuteInfo to AudioConferenceMixer by henrik.lundin · 8 years ago
  44. e0615b7 GN: Disable checks in WebRTC tree due to too many errors. by Henrik Kjellander · 8 years ago
  45. 3c5a294 Fix Info.plist path in build_ios_libs.sh by tkchin · 8 years ago
  46. 837dde9 Remove DesktopFrame::shape(). by Sergey Ulanov · 8 years ago
  47. 62f6da3 Roll chromium_revision 7ae70b716a..91b474353e (394093:394313) by buildbot · 8 years ago
  48. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 8 years ago
  49. 3a0a0f4 AudioConfMixer: Add muted variable to ParticipantFramePair by henrik.lundin · 8 years ago
  50. d4ccb00 Propagate muted parameter to VoE::Channel by henrik.lundin · 8 years ago
  51. e305d95 Remove runtime NEON detection by pasko · 8 years ago
  52. de8739c Disable libyuv jpeg support on Android by magjed · 8 years ago
  53. 2b1f651 Potential fix for rtx/red issue where red is removed only from the remote description. by Stefan Holmer · 8 years ago
  54. 9b2228f Fix UBSan errors (left shift of negative value) by kwiberg · 8 years ago
  55. aa08ce6 Roll chromium_revision 667cccbd62..7ae70b716a (394017:394093) by buildbot · 8 years ago
  56. c9c142f Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ ) by nisse · 8 years ago
  57. ff27439 Separate building and enabling libevent. by phoglund · 8 years ago
  58. d98f6e0 Fixed typo. KT_DEFAULT different based on WEBRTC_CHROMIUM_BUILD by Henrik Boström · 8 years ago
  59. a73ca56 Polishing code to handle certificate generation failure in .mm files. by hbos · 8 years ago
  60. ee37326 JUnit test framework for AppRTC Android demo. by sakal · 8 years ago
  61. 2ccfbdf Undeprecate CreatePeerConnectionFactory which do not use network thread. by Danil Chapovalov · 8 years ago
  62. e9021a3 Propogate network-worker thread split to api by danilchap · 8 years ago
  63. c561479 Revert of CQ: Disable win_x64_clang_dbg trybot (patchset #1 id:1 of https://codereview.webrtc.org/1896003004/ ) by kjellander · 8 years ago
  64. 744494f Make the FakeWebRtcVideoCaptureModule class initialize frame data. by nisse · 8 years ago
  65. 9359274 Roll chromium_revision dbbc7ddf2e..667cccbd62 (393813:394017) by buildbot · 8 years ago
  66. c9b0c26 Surface the IntelligibilityEnhancer on MediaConstraints by Alejandro Luebs · 8 years ago
  67. 2abe427 Revert of Increase the stun ping interval. (patchset #5 id:80001 of https://codereview.webrtc.org/1944003002/ ) by Taylor Brandstetter · 8 years ago
  68. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 8 years ago
  69. 8bce67b Added UMA logging for audio codec usage. A histogram statistic named "WebRTC.Audio.Encoder.CodecType" has been created. by aleloi · 8 years ago
  70. 970567c Fixes a bug where AppRTC Android Demo crashes with empty roomId. by Sami Kalliomäki · 8 years ago
  71. 57f95dc New UI for AppRTC Android Demo that is easier to use and better follows by Sami Kalliomäki · 8 years ago
  72. ff66d6b Roll chromium_revision d240f3cb44..dbbc7ddf2e (393782:393813) by buildbot · 8 years ago
  73. 8ae8ab4 Makes ECDSA the default certificate to use (generated if no other certificates by hbos · 8 years ago
  74. 4b2ffe2 Roll chromium_revision c77d596b47..d240f3cb44 (393773:393782) by buildbot · 8 years ago
  75. 09f4e87 Roll chromium_revision edd509dc82..c77d596b47 (393769:393773) by buildbot · 8 years ago
  76. 0bcbbd3 Fix component build in chrome after recent TaskQueue cl by Tommi · 8 years ago
  77. e1220d9 Roll chromium_revision 68a109ab91..edd509dc82 (393760:393769) by buildbot · 8 years ago
  78. 8885a4f Roll chromium_revision 001bec1f78..68a109ab91 (393735:393760) by buildbot · 8 years ago
  79. fd8be34 Remove webrtc/base/scoped_ptr.h by kwiberg · 8 years ago
  80. c06b133 Reland of New task queueing primitive for async tasks: TaskQueue. by tommi · 8 years ago
  81. 353ca76 Disable SendsAndReceivesH264 on Memcheck. by Peter Boström · 8 years ago
  82. 52541ba Roll chromium_revision 133b4c13ca..001bec1f78 (393728:393735) by buildbot · 8 years ago
  83. 5ce1a2a Reland of Allow the localhost IP address even if it does not match the tcp port address (patchset #1 id:1 of https://codereview.webrtc.org/1979463003/ ) by tommi · 8 years ago
  84. d49c30c Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #2 id:290001 of https://codereview.webrtc.org/1963413004/ ) by tommi · 8 years ago
  85. a102507 Use generic CPU-overuse thresholds for iOS. by pbos · 8 years ago
  86. fea9309 This reland https://codereview.webrtc.org/1932683002/. by perkj · 8 years ago
  87. 256692f Roll chromium_revision 5f1d704d67..133b4c13ca (392277:393728) by buildbot · 8 years ago
  88. 1299615 Make sure WebRTC works without libvpx VP9 support. by Peter Boström · 8 years ago
  89. dae07ba Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 8 years ago
  90. 99f8cd0 CURSORINFO.flags should be checked before capturing its bitmap by zijiehe · 8 years ago
  91. 3f90087 Revert of New task queueing primitive for async tasks: TaskQueue. (patchset #8 id:330001 of https://codereview.webrtc.org/1927133004/ ) by tommi · 8 years ago
  92. 65d1f2a Reland of New task queueing primitive for async tasks: TaskQueue. (patchset #1 id:1 of https://codereview.webrtc.org/1935483002/ ) by tommi · 8 years ago
  93. 8f7a5aa Increase the stun ping interval. by zhihuang · 8 years ago
  94. 6ba3b19 Filter out some variables with initial -1 in the stats report. by zhihuang · 8 years ago
  95. 1f53452 Unify hardware and software QP thresholds. by pbos · 8 years ago
  96. fb1dd43 Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #2 id:20001 of https://codereview.webrtc.org/1973313002/ ) by kjellander · 8 years ago
  97. 709f73c VideoAdapter: Add cropping based on OnOutputFormatRequest() by magjed · 8 years ago
  98. c8d848b Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 8 years ago
  99. c1513ee Add a parameter to set a maximum file size when starting an RTC event log on the PeerConnectionFactory API. by ivoc · 8 years ago
  100. a1c3035 Relanding: Implement RTCConfiguration.iceCandidatePoolSize. by Taylor Brandstetter · 8 years ago