Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 392d0c2701ef0112035e496cafcd2e7f0453547f
  1. 392d0c2 Remove android_rel from CQ since all of its machines are offline. by Henrik Kjellander · 10 years ago
  2. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 10 years ago
  3. 318166b Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ ) by guoweis · 10 years ago
  4. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 10 years ago
  5. 3c652b6 modules/audio_coding: Remove some codec include dirs by kjellander@webrtc.org · 10 years ago
  6. b7ce964 modules/video_coding/utility: Remove include by kjellander@webrtc.org · 10 years ago
  7. 1b20d81 Roll chromium_revision 64f2817..ed2e3fb (360275:360379) by kjellander · 10 years ago
  8. 0f59a88 modules/video_processing: refactor interface->include + more. by Henrik Kjellander · 10 years ago
  9. ed7d6ec WebRTC: Add compability header for video_coding refactoring. by Henrik Kjellander · 10 years ago
  10. ad948c4 Preliminary support of VP9 HW encoder on Android. by Alex Glaznev · 10 years ago
  11. 2557b86 modules/video_coding refactorings by Henrik Kjellander · 10 years ago
  12. 4dd7a65 Temporarily disable VERIFY while bug is investigated. by phoglund · 10 years ago
  13. 223692a Remove dead code by kwiberg · 10 years ago
  14. e1a27d4 Move CNG/RED payload type extraction to Rent-A-Codec by kwiberg · 10 years ago
  15. 49a6c99 Disables BitrateEstimatorTest.SwitchesToASTThenBackToTOFForVideo on win_drmemory_full due to flakiness. by ivoc · 10 years ago
  16. 2446e5a Fixed the render queue item size preallocation and verification, moved constant declarations, removed redundant queue allocation by peah · 10 years ago
  17. 0219c9b rtcp::App moved into own file and got Parse function by danilchap · 10 years ago
  18. 2aff615 Remove spammy logging of RTCP delivery failures. by Peter Boström · 10 years ago
  19. f70568c So long and thanks for all the code reviews! by andrew · 10 years ago
  20. cb50c96 Set temporal up switch bit to false for flexible mode (one temporal layer is configured currently). by asapersson · 10 years ago
  21. aa45843 Roll chromium_revision a6d9f7f..64f2817 (360123:360275) by kjellander · 10 years ago
  22. 310b093 Fix active tcp port to 9 by Guo-wei Shieh · 10 years ago
  23. 2935e01 Several Tick counter improvements try #2." by thaloun · 10 years ago
  24. c073615 Update references to TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305, etc. by davidben · 10 years ago
  25. 0a75749 Roll chromium_revision 04756fa..a6d9f7f (360053:360123) by kjellander · 10 years ago
  26. 32f3996 Re-apply change https://codereview.webrtc.org/1426673007/ by honghaiz · 10 years ago
  27. 5c489c9 Add OpenSL ES enable setting to AppRTCDemo (part 2). by henrika · 10 years ago
  28. 2be7c54 Remove ViEEncoder::ScaleInputImage. by Peter Boström · 10 years ago
  29. bd05f0b Unconditionally build VP9 support. by Peter Boström · 10 years ago
  30. 18adf0a Add UMA for send bwe and pacer bitrate. by stefan · 10 years ago
  31. d9eec76 Trace encoding/decoding time in a generic way. by pbos · 10 years ago
  32. 5a71f03 Deactivate the audio session after a call ends using the AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation constant by henrika · 10 years ago
  33. 45e998d Roll chromium_revision a2e8a40..04756fa (359987:360053) by kjellander · 10 years ago
  34. fd614c2 Adding thread timeout for audio recorer thread in Java by henrika · 10 years ago
  35. e663392 Add OpenSL ES enable setting to AppRTCDemo. by glaznev · 10 years ago
  36. 3c12f4d Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ ) by pbos · 10 years ago
  37. 192164e Preparational work before introducing the locks in order to harmonize the code: by peah · 10 years ago
  38. 4d291f7 Applied the render queueing to the agc. by peah · 10 years ago
  39. 03179cd Roll chromium_revision 6fd4bdd..a2e8a40 (359891:359987) by kjellander · 10 years ago
  40. 740c4f1 Remove packet initializer in RtpRtcpRtxNackTest. by pbos · 10 years ago
  41. 854e84c Use webrtc/base/logging.h for video coding/processing. by pbos · 10 years ago
  42. c91d173 Revert of Several Tick counter improvements. (patchset #8 id:140001 of https://codereview.webrtc.org/1415923010/ ) by thaloun · 10 years ago
  43. fa6228e Introduced the render sample queue for the aec and aecm. by peah · 10 years ago
  44. 4c27e4b Several Tick counter improvements. by Tim Haloun · 10 years ago
  45. eb8b388 Fix VP9 support in AppRTCDemo. by Alex Glaznev · 10 years ago
  46. 6f8ce06 common_video: rename interface -> include by kjellander · 10 years ago
  47. 591cb1f Roll chromium_revision c958aa7..6fd4bdd (359816:359891) by kjellander · 10 years ago
  48. b27f590 Create rtc::AtomicInt POD struct. by pbos · 10 years ago
  49. 3528a27 Flesh out webrtc/.gitignore by brucedawson · 10 years ago
  50. 482b12e Remove BundleFilter filtering of RTCP. by pbos · 10 years ago
  51. 8b85de2 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail. by solenberg · 10 years ago
  52. 9a7c838 Adding stddef.h to opus_inst.h. by minyue · 10 years ago
  53. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 10 years ago
  54. 633a3aa ThreadUtils: Add joinUninterruptibly() with timeout by magjed · 10 years ago
  55. e155ae6 Move CNG and RED management into the Rent-A-Codec by kwiberg · 10 years ago
  56. 54e9232 Revert of Do not delete the turn port entry right away when the respective connection is deleted. (patchset #5 id:260001 of https://codereview.webrtc.org/1426673007/ ) by tommi · 10 years ago
  57. 2a654fa Roll chromium_revision cad2987..c958aa7 (359796:359816) by kjellander · 10 years ago
  58. 0b9e29c Remove include dirs from modules/{media_file,pacing} by Henrik Kjellander · 10 years ago
  59. 3e0f602 Android EglBase: Add support for creating EGLSurface from Surface, not SurfaceHolder by magjed · 10 years ago
  60. d9b75be Fix a data race in the thread unit tests. by nisse · 10 years ago
  61. 6f14be8 Add limit for minimum number of required samples before recording input and sent framerate stats. by asapersson · 10 years ago
  62. 3c735f4 Roll chromium_revision b77e5bb..cad2987 (359767:359796) by kjellander · 10 years ago
  63. 8c64860 Roll chromium_revision 3b7968d..b77e5bb (359482:359767) by kjellander · 10 years ago
  64. e58fe8e Do not delete the turn port entry right away when the respective connection is deleted. by honghaiz · 10 years ago
  65. fa5d0db cleanup: get rid of basicdefs.h include by tfarina · 10 years ago
  66. a4845ef Fix flaky tests by honghaiz · 10 years ago
  67. 4a41361 Android SurfaceViewRenderer: Never hold a pending frame indefinitely by magjed · 10 years ago
  68. c01c254 Revert of Android MediaCodecVideoDecoder: Manage lifetime of texture frames (patchset #12 id:320001 of https://codereview.webrtc.org/1422963003/ ) by Per · 10 years ago
  69. f8506cb rtcp::Ij renamed to rtcp::ExtendedJitterReport by danilchap · 10 years ago
  70. cbe9f51 Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https://codereview.webrtc.org/1416093010/ ) by phoglund · 10 years ago
  71. 0fa9b22 Remove scoped_ptrs for VCM sender_ and receiver_. by pbos · 10 years ago
  72. df948f0 rtcp::ReportBlock refactored to contain parsing by danilchap · 10 years ago
  73. 0a41893 Remove BitrateController dependency fromVideoReceiveStream. by mflodman · 10 years ago
  74. 464c087 Rename screenshare test. by philipel · 10 years ago
  75. 0e7e259 Move BitrateAllocator from BitrateController logic to Call. by mflodman · 10 years ago
  76. 69191ed Roll chromium_revision 4771dd5..3b7968d (359351:359482) by kjellander · 10 years ago
  77. faac497 Fix for scenario where m-line is revived after being set to port 0. by deadbeef · 10 years ago
  78. 69d0d46 Roll chromium_revision e658ee0..4771dd5 (359300:359351) by kjellander · 10 years ago
  79. 2cd7afe Do not delete a connection until it has not received anything for 30 seconds. by Honghai Zhang · 10 years ago
  80. 8597543 Schedule a CreatePermissionRequest after the success of a previous request by Honghai Zhang · 10 years ago
  81. 68876f9 Introduces Android API level linting, fixes all current API lint errors. by Patrik Höglund · 10 years ago
  82. 56a34df Re-add a thread check in Call::Call that was removed by mistake in a rebase. by solenberg · 10 years ago
  83. 9576e54 Reland "Prepare MediaCodecVideoEncoder for surface textures."" by perkj · 10 years ago
  84. 8093d54 Change default SSRC for RTCP receiver reports to not collide with video. by solenberg · 10 years ago
  85. dfe434e Roll chromium_revision b0415d9..e658ee0 (359214:359300) by kjellander · 10 years ago
  86. 5dda80a Remove webrtc/modules/video_{capture,render}/include by Henrik Kjellander · 10 years ago
  87. e71b24e OpenSL ES stability improvements. by henrika · 10 years ago
  88. fc6affc Android SurfaceViewRenderer: Call glClear() for every frame to avoid bad GL state by magjed · 10 years ago
  89. 9683964 Trivial initialization fix in AudioDeviceIOS by henrika · 10 years ago
  90. 31c8167 Roll chromium_revision 7e059f9..b0415d9 (359143:359214) by kjellander · 10 years ago
  91. a8e9f5e A little cleanup in p2ptransportchannel and transportchannel. by honghaiz · 10 years ago
  92. 066ded9 Relax the stun ping check on valid result. by guoweis · 10 years ago
  93. 33daa7e Roll chromium_revision 4a38519..7e059f9 (359080:359143) by kjellander · 10 years ago
  94. 6b14f93 Adjust parameter for VP9 resize unittest. by Marco · 10 years ago
  95. 9b5ee9c Send back ping response if the ping comes from an unknown address. by honghaiz · 10 years ago
  96. 653b8e0 Reland of Adding the ability to change ICE servers through SetConfiguration. (patchset #1 id:1 of https://codereview.webrtc.org/1424803004/ ) by deadbeef · 10 years ago
  97. 9b72af9 Remove webrtc/modules/audio_processing/{aec,aecm,ns}/include by Henrik Kjellander · 10 years ago
  98. e03cab9 When running this code in chromium on a machine with IPv6 disabled, the RTC_DCHECK fails and in release build, it could leak to further crash in chromium's rtc_peer_connection_hanlder.cc. by Guo-wei Shieh · 10 years ago
  99. ee2bac2 AcmReceiver::InsertPacket and NetEq::InsertPacket: Take ArrayView arguments by kwiberg · 10 years ago
  100. 91d9260 Add receive bitrate UMA stats. by stefan · 10 years ago