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gerrit-public.fairphone.software / platform / external / webrtc / 1379f1f1e69632485d07ac7a3a575ea231e62522
  1. 1379f1f Extract the parameters for the encoder stack from the CodecManager by kwiberg · 9 years ago
  2. 30a5e56 Roll chromium_revision 3796a7a..e038f1d (361065:361088) by kjellander · 9 years ago
  3. f0a431a Exclude EndToEndTest.ReceivesTransportFeedback and TransportFeedbackNotConfigured from DrMemory. by Stefan Holmer · 9 years ago
  4. db81ffd Request keyframe if too many packets are missing and NACK is disabled. by jbauch · 9 years ago
  5. fa8ae9a Remove <iostream> include from file_audio_device.cc by kjellander@webrtc.org · 9 years ago
  6. 8becec3 talk: remove deprecated *processor.h files by tfarina · 9 years ago
  7. 87d5845 Fix androidmediadecoder_jni TS logging. by perkj · 9 years ago
  8. c3c4cdb Add Android x86 and x64 trybots to CQ. by kjellander@webrtc.org · 9 years ago
  9. d5674c3 Roll chromium_revision c29f20c..3796a7a (361043:361065) by kjellander · 9 years ago
  10. 50c5136 RTCP Bye packet moved to own file by danilchap · 9 years ago
  11. c982913 Roll chromium_revision 6018759..c29f20c (361030:361043) by kjellander · 9 years ago
  12. 485b5b0 Roll chromium_revision 4df2d47..6018759 (361029:361030) by kjellander · 9 years ago
  13. 82581a0 Roll chromium_revision 3966d2c..4df2d47 (361020:361029) by kjellander · 9 years ago
  14. b4a29d9 Roll chromium_revision b854092..3966d2c (360794:361020) by kjellander · 9 years ago
  15. 13f6b8f Increase transport feedback frequency to 20 Hz. by stefan · 9 years ago
  16. 43edf0f Require negotiation to send transport cc feedback over RTCP. by stefan · 9 years ago
  17. bd13838 Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack. by solenberg · 9 years ago
  18. 672304a NetEq: Remove overly verbose logging by henrik.lundin · 9 years ago
  19. 5def7b9 Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) by deadbeef · 9 years ago
  20. 7add058 Move some receive stream configuration into webrtc::AudioReceiveStream. by solenberg · 9 years ago
  21. 6834fa1 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) by deadbeef · 9 years ago
  22. 0a43fef Allow pacer to boost bitrate in order to meet time constraints. by sprang · 9 years ago
  23. 34911ad Improved error handling in iOS ADM to avoid race during init by henrika · 9 years ago
  24. 76a31ca Avoids hitting DCHECK in OpenSL ES player by henrika · 9 years ago
  25. 1afae74 Roll chromium_revision 5c83f4e..b854092 (360728:360794) by kjellander · 9 years ago
  26. 30e9182 This cl add support to encode from textures to MediaCodecVideoEncoder. by perkj · 9 years ago
  27. 5663b4f iOS: Set enable_protobuf=1 by default. by kjellander@webrtc.org · 9 years ago
  28. 7e63ef0 Allow default audio receive channel to receive on any unsignalled SSRC. by solenberg · 9 years ago
  29. b0ad43b Add aecdump support to audioproc_f by aluebs · 9 years ago
  30. ceb450b Roll chromium_revision c8eec9a..5c83f4e (360565:360728) by kjellander · 9 years ago
  31. 17c0aff Enable VP9 HW decoder on Exynos chips. by Alex Glaznev · 9 years ago
  32. 7593aad Re-enable mistakenly disabled PEM tests. Misc cleanup and alignment fixes. by torbjorng · 9 years ago
  33. 7755e20 Chrome has now been updated. by perkj · 9 years ago
  34. 726b1f7 Removed dummy "mediastreamsignaling.h" by perkj · 9 years ago
  35. 191c1f9 Disable all JsepPeerConnectionP2PTestClient tests on Mac due to flakiness on Mac Debug bots. by ivoc · 9 years ago
  36. 12e21a0 Remove dead code (we no longer support SILK) by kwiberg · 9 years ago
  37. ef45323 Android: Make classes non-final by Magnus Jedvert · 9 years ago
  38. 062e14e Roll chromium_revision bb7899a..c8eec9a (360504:360565) by kjellander · 9 years ago
  39. f399f21 Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 on linux due to flakiness on the Linux64 Debug bot. by ivoc · 9 years ago
  40. f22695c Remove build_with_libjingle and exclude failing iOS tests from 'All' target. by kjellander@webrtc.org · 9 years ago
  41. 1503867 Disabled several JsepPeerConnectionP2PTestClient tests on Mac, due to flakiness on Debug Mac trybots. by ivoc · 9 years ago
  42. e488a0d Fix DTLS packet boundary handling in SSLStreamAdapterTests. by jbauch · 9 years ago
  43. 87097a8 Roll chromium_revision ed2e3fb..bb7899a (360379:360504) by kjellander · 9 years ago
  44. b6755ab Revert of Adding thread timeout for audio recorer thread in Java (patchset #2 id:20001 of https://codereview.webrtc.org/1444313002/ ) by henrika · 9 years ago
  45. 488e75f Patchset 1 yet again relands without modification https://codereview.webrtc.org/1422963003/ by Per · 9 years ago
  46. 0969398 Revert of Remove android_rel from CQ since all of its machines are offline. (patchset #1 id:1 of https://codereview.webrtc.org/1459083002/ ) by kjellander · 9 years ago
  47. 392d0c2 Remove android_rel from CQ since all of its machines are offline. by Henrik Kjellander · 9 years ago
  48. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  49. 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 · 9 years ago
  50. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  51. 3c652b6 modules/audio_coding: Remove some codec include dirs by kjellander@webrtc.org · 9 years ago
  52. b7ce964 modules/video_coding/utility: Remove include by kjellander@webrtc.org · 9 years ago
  53. 1b20d81 Roll chromium_revision 64f2817..ed2e3fb (360275:360379) by kjellander · 9 years ago
  54. 0f59a88 modules/video_processing: refactor interface->include + more. by Henrik Kjellander · 9 years ago
  55. ed7d6ec WebRTC: Add compability header for video_coding refactoring. by Henrik Kjellander · 9 years ago
  56. ad948c4 Preliminary support of VP9 HW encoder on Android. by Alex Glaznev · 9 years ago
  57. 2557b86 modules/video_coding refactorings by Henrik Kjellander · 9 years ago
  58. 4dd7a65 Temporarily disable VERIFY while bug is investigated. by phoglund · 9 years ago
  59. 223692a Remove dead code by kwiberg · 9 years ago
  60. e1a27d4 Move CNG/RED payload type extraction to Rent-A-Codec by kwiberg · 9 years ago
  61. 49a6c99 Disables BitrateEstimatorTest.SwitchesToASTThenBackToTOFForVideo on win_drmemory_full due to flakiness. by ivoc · 9 years ago
  62. 2446e5a Fixed the render queue item size preallocation and verification, moved constant declarations, removed redundant queue allocation by peah · 9 years ago
  63. 0219c9b rtcp::App moved into own file and got Parse function by danilchap · 9 years ago
  64. 2aff615 Remove spammy logging of RTCP delivery failures. by Peter Boström · 9 years ago
  65. f70568c So long and thanks for all the code reviews! by andrew · 9 years ago
  66. cb50c96 Set temporal up switch bit to false for flexible mode (one temporal layer is configured currently). by asapersson · 9 years ago
  67. aa45843 Roll chromium_revision a6d9f7f..64f2817 (360123:360275) by kjellander · 9 years ago
  68. 310b093 Fix active tcp port to 9 by Guo-wei Shieh · 9 years ago
  69. 2935e01 Several Tick counter improvements try #2." by thaloun · 9 years ago
  70. c073615 Update references to TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305, etc. by davidben · 9 years ago
  71. 0a75749 Roll chromium_revision 04756fa..a6d9f7f (360053:360123) by kjellander · 9 years ago
  72. 32f3996 Re-apply change https://codereview.webrtc.org/1426673007/ by honghaiz · 9 years ago
  73. 5c489c9 Add OpenSL ES enable setting to AppRTCDemo (part 2). by henrika · 9 years ago
  74. 2be7c54 Remove ViEEncoder::ScaleInputImage. by Peter Boström · 9 years ago
  75. bd05f0b Unconditionally build VP9 support. by Peter Boström · 9 years ago
  76. 18adf0a Add UMA for send bwe and pacer bitrate. by stefan · 9 years ago
  77. d9eec76 Trace encoding/decoding time in a generic way. by pbos · 9 years ago
  78. 5a71f03 Deactivate the audio session after a call ends using the AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation constant by henrika · 9 years ago
  79. 45e998d Roll chromium_revision a2e8a40..04756fa (359987:360053) by kjellander · 9 years ago
  80. fd614c2 Adding thread timeout for audio recorer thread in Java by henrika · 9 years ago
  81. e663392 Add OpenSL ES enable setting to AppRTCDemo. by glaznev · 9 years ago
  82. 3c12f4d Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ ) by pbos · 9 years ago
  83. 192164e Preparational work before introducing the locks in order to harmonize the code: by peah · 9 years ago
  84. 4d291f7 Applied the render queueing to the agc. by peah · 9 years ago
  85. 03179cd Roll chromium_revision 6fd4bdd..a2e8a40 (359891:359987) by kjellander · 9 years ago
  86. 740c4f1 Remove packet initializer in RtpRtcpRtxNackTest. by pbos · 9 years ago
  87. 854e84c Use webrtc/base/logging.h for video coding/processing. by pbos · 9 years ago
  88. c91d173 Revert of Several Tick counter improvements. (patchset #8 id:140001 of https://codereview.webrtc.org/1415923010/ ) by thaloun · 9 years ago
  89. fa6228e Introduced the render sample queue for the aec and aecm. by peah · 9 years ago
  90. 4c27e4b Several Tick counter improvements. by Tim Haloun · 9 years ago
  91. eb8b388 Fix VP9 support in AppRTCDemo. by Alex Glaznev · 9 years ago
  92. 6f8ce06 common_video: rename interface -> include by kjellander · 9 years ago
  93. 591cb1f Roll chromium_revision c958aa7..6fd4bdd (359816:359891) by kjellander · 9 years ago
  94. b27f590 Create rtc::AtomicInt POD struct. by pbos · 9 years ago
  95. 3528a27 Flesh out webrtc/.gitignore by brucedawson · 9 years ago
  96. 482b12e Remove BundleFilter filtering of RTCP. by pbos · 9 years ago
  97. 8b85de2 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail. by solenberg · 9 years ago
  98. 9a7c838 Adding stddef.h to opus_inst.h. by minyue · 9 years ago
  99. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  100. 633a3aa ThreadUtils: Add joinUninterruptibly() with timeout by magjed · 9 years ago