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gerrit-public.fairphone.software / platform / external / webrtc / 10fc0e6385b812981f6b64f611259887ca14de51
  1. 10fc0e6 Delay based logging. by philipel · 7 years ago
  2. 64e739a Add content type information to Encoded Images and add corresponding RTP extension header. by ilnik · 7 years ago
  3. 93cda2e APM-QA tool, renaming noise generators into input-reference generators. by alessiob · 7 years ago
  4. 9765370 Resolve dependency between rtc_event_log_api and remote_bitrate_estimator by michaelt · 7 years ago
  5. 810eecf Roll chromium_revision c57654688e..860f7b94d4 (463520:463558) by buildbot · 7 years ago
  6. 7fb7bbd Revert of Add first part of the network_tester functionality. (patchset #13 id:260001 of https://codereview.webrtc.org/2779233002/ ) by michaelt · 7 years ago
  7. 333d0ff Add first part of the network_tester functionality. by michaelt · 7 years ago
  8. e0ab0ad Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT by kjellander · 7 years ago
  9. 0d4e068 Make safe_cmp::* constexpr by kwiberg · 7 years ago
  10. d491109 Roll chromium_revision 1af3c1a4a8..c57654688e (463476:463520) by buildbot · 7 years ago
  11. 4a9d08f Roll chromium_revision d3a2a83fbf..1af3c1a4a8 (463418:463476) by buildbot · 7 years ago
  12. 8c459f9 Restore old (deprecated) signature of initializeAndroidGlobals. by deadbeef · 7 years ago
  13. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 7 years ago
  14. d9ce764 Make RtpTransport actually implement RtpTransportInterface by zstein · 7 years ago
  15. 0f92c79 Roll chromium_revision 5d7042a87c..d3a2a83fbf (463209:463418) by buildbot · 7 years ago
  16. b4fc73a Removing unnecessary parameters from initializeAndroidGlobals. by deadbeef · 7 years ago
  17. 6799553 Add information about microphone gain changes to AEC3 by peah · 7 years ago
  18. 6d822ad Added forced zero AEC output after call startup and echo path changes by peah · 7 years ago
  19. ca31f17 Remove deprecated RTPPayloadStrategy by danilchap · 7 years ago
  20. a1ef71f Add parser to visualise the ana dump by michaelt · 7 years ago
  21. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 7 years ago
  22. b0f7e39 Move IsIntlike to type_traits.h by kwiberg · 7 years ago
  23. 37e99fd Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 7 years ago
  24. 2fa97fd Roll chromium_revision 0a53e4a670..5d7042a87c (463181:463209) by buildbot · 7 years ago
  25. 0642b32 Remove duplicate entries from AUTHORS file by henrik.lundin · 7 years ago
  26. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 7 years ago
  27. 925e9d7 Removed workaround for the WARN_UNUSED_RESULT issue. by peah · 7 years ago
  28. 4fb651d Event log cleanup in tests. by philipel · 7 years ago
  29. fca900a Fix two invalid DCHECKs related to audio BWE. by stefan · 7 years ago
  30. 49cad02 Ignore some UBSan errors by kwiberg · 7 years ago
  31. 9f2c18e Changed OLA window for neteq. Old code didnt work well with 48khz by soren · 7 years ago
  32. c547e84 Allow rtp::Packet::*RawExtension to take 0 as an extension id by danilchap · 7 years ago
  33. 02465b8 Add some unit tests to vie_encoder. by asapersson · 7 years ago
  34. 36e6a8f WavReaderAdaptor is a simple adaptor of the existing class WavReader from webrtc/common_audio/wav_file.h. The adaptor was mainly needed to use dependency injection and easily test the MultiEndCall class (see https://codereview.webrtc.org/2761853002/). by alessiob · 7 years ago
  35. 2042c16 Revert of Delete class ScopedPtrCollection. Replaced with vector of unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2808463002/ ) by nisse · 7 years ago
  36. 64c93c3 Roll chromium_revision 1ab7c6059c..0a53e4a670 (463170:463181) by buildbot · 7 years ago
  37. 188596f Delete class ScopedPtrCollection. Replaced with vector of unique_ptr. by nisse · 7 years ago
  38. bd1a681 Roll chromium_revision c8a0f6b4c5..1ab7c6059c (463161:463170) by buildbot · 7 years ago
  39. 1b0882d Roll chromium_revision c30f6366d7..c8a0f6b4c5 (463148:463161) by buildbot · 7 years ago
  40. 1d9ef47 Roll chromium_revision b0bf8e8ed3..c30f6366d7 (463140:463148) by buildbot · 7 years ago
  41. a3d4d94 Roll chromium_revision 5e27d4b8d1..b0bf8e8ed3 (463138:463140) by buildbot · 7 years ago
  42. 4b37127 Fix compilation issues of std::unique_ptr by steweg · 7 years ago
  43. 8cc642b Roll chromium_revision d5df028c94..5e27d4b8d1 (463137:463138) by buildbot · 7 years ago
  44. 89292d1 Roll chromium_revision 270af5af87..d5df028c94 (463130:463137) by buildbot · 7 years ago
  45. 5253411 Roll chromium_revision 9d8fbbd04c..270af5af87 (463126:463130) by buildbot · 7 years ago
  46. 01f2793 Roll chromium_revision a35a1e2ce2..9d8fbbd04c (463121:463126) by buildbot · 7 years ago
  47. 50ddc63 Roll chromium_revision 98c8321fe9..a35a1e2ce2 (463081:463121) by buildbot · 7 years ago
  48. 9d8052d Roll chromium_revision 762665735a..98c8321fe9 (462871:463081) by buildbot · 7 years ago
  49. 66e9f76 Adjust parameter in vp9 videoprocessor_integration test. by jianj · 7 years ago
  50. 8d23c05 MultiEndCall::CheckTiming() verifies that a set of audio tracks and timing information is valid to simulate conversational speech. Unordered turns are rejected. Self cross-talk and cross-talk with 3 or more speakers are not permitted since it would require mixing at the simulation step. by alessiob · 7 years ago
  51. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 7 years ago
  52. bb16a48 Roll chromium_revision 61577bba5a..762665735a (462822:462871) by buildbot · 7 years ago
  53. 8942045 Adding support for handling highly reverberant echoes in AEC3. by peah · 7 years ago
  54. 38415b2 Reland of Adding PRESUBMIT check on google::protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2791583002/ ) by mbonadei · 7 years ago
  55. 423f106 Add support for 64-bit architectures in build_aar.py. by sakal · 7 years ago
  56. 2ce640f Fixing sample-rate dependent band-split filter issues in AEC3 by peah · 7 years ago
  57. ea44ad4 Roll chromium_revision d65c1d7370..61577bba5a (462801:462822) by buildbot · 7 years ago
  58. 7c2c843 Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ ) by mbonadei · 7 years ago
  59. 0944a80 Update stats for cpu/quality adaptation changes to excluded time when video is suspended. by asapersson · 7 years ago
  60. 533b7ac Roll chromium_revision c405301b70..d65c1d7370 (462743:462801) by buildbot · 7 years ago
  61. 7749286 Make AudioProcessing::GetConfig() pure virtual by henrik.lundin · 7 years ago
  62. abd101b Support multiple connected Android devices in low bandwidth audio test by oprypin · 7 years ago
  63. 225bfc0 Make PacketTransportInternal inherit from PacketTransportInterface. by deadbeef · 7 years ago
  64. 81c899d Roll chromium_revision fbddb8a080..c405301b70 (462718:462743) by buildbot · 7 years ago
  65. f49011b Roll chromium_revision 300bd53cd8..fbddb8a080 (462634:462718) by buildbot · 7 years ago
  66. 4b572d0 Correction of the AEC3 underrun behavior and minor other corrections by peah · 7 years ago
  67. 86afe9d Major updates to the echo removal functionality in AEC3 by peah · 7 years ago
  68. f51517a Roll chromium_revision bb330d1b8f..300bd53cd8 (462561:462634) by buildbot · 7 years ago
  69. 8cd1cce Roll chromium_revision 86718dcdf2..bb330d1b8f (462494:462561) by buildbot · 7 years ago
  70. 1ffbd6c Injectable audio encoders: voice_engine/channel changes. by ossu · 7 years ago
  71. 5f4aaeb Re-enable FullStackTest.ScreenshareSlidesVP9_2SL test. by marpan · 7 years ago
  72. 4b62001 Adding AudioDeviceDataObserver interface by Lu Liu · 7 years ago
  73. a1a040a Injectable audio encoders: BuiltinAudioEncoderFactory by ossu · 7 years ago
  74. ac4bbdf Roll chromium_revision 3014f8b41e..86718dcdf2 (462374:462494) by buildbot · 7 years ago
  75. cab8d88 Revert of Enable rtc_unittests on iOS simulator (patchset #2 id:20001 of https://codereview.webrtc.org/2799033004/ ) by kjellander · 7 years ago
  76. 6167b26 Make RtpTransportControllerSend::send_side_cc_ a direct member. by nisse · 7 years ago
  77. cde46b7 Resolve cyclic dependency between audio network adaptor and event log api by michaelt · 7 years ago
  78. 28dc285 Adding cbr support for Opus by soren · 7 years ago
  79. 388fe42 Make WARN_UNUSED_RESULT a no-op on gcc by kwiberg · 7 years ago
  80. 177b17e Move AndroidVideoTrackSourceObserver from API to src by magjed · 7 years ago
  81. 639d46a Delete system_wrappers logging facility. by nisse · 7 years ago
  82. be77920 Revert of CQ: Remove Linux ARM64 Debug trybot from default set. (patchset #1 id:1 of https://codereview.webrtc.org/2790263003/ ) by kjellander · 7 years ago
  83. 2418001 ACM: Change test output files from PCM to WAV by henrik.lundin · 7 years ago
  84. 4fcfdd8 Enable rtc_unittests on iOS simulator by kjellander · 7 years ago
  85. a280f7c Added integer parsing functions in base/string_to_number.h by ossu · 7 years ago
  86. b1e3fc4 Enable tools_unittests and rtc_stats_unittests on iOS Simulator by kjellander · 7 years ago
  87. e24991d Adds AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex unittest. by henrika · 7 years ago
  88. 978504e Move rtp header extension length check from Packet::FindExtension to ExtensionT::Parse by danilchap · 7 years ago
  89. ed6343d Roll chromium_revision 875e8893e9..3014f8b41e (462360:462374) by buildbot · 7 years ago
  90. 251eb27 Roll chromium_revision 75820eb165..875e8893e9 (460410:462360) by kjellander · 7 years ago
  91. f6a4f37 Reland of Fixed error for missing explict class initialization error on iOS WebRTC buildbots (patchset #1 id:1 of https://codereview.webrtc.org/2803933002/ ) by guidou · 7 years ago
  92. 854e507 Revert of Fixed error for missing explict class initialization error on iOS WebRTC buildbots (patchset #1 id:1 of https://codereview.webrtc.org/2799813002/ ) by guidou · 7 years ago
  93. 5ac18af Fixed error for missing explicit class initialization error on iOS buildbots by peah · 7 years ago
  94. 7343c8e DirectX capturer may crash after switching shared screen by zijiehe · 7 years ago
  95. cf02cf1 Major AEC3 render pipeline changes by peah · 7 years ago
  96. 4aceaf2 Android: Move Histogram from api to src. by sakal · 7 years ago
  97. c522e75 Use new RTCCameraVideoCapturer in AppRTCMobile. by sakal · 7 years ago
  98. 1ba21eb Add [c]begin() and [c]end() member functions to rtc::Buffer by kwiberg · 7 years ago
  99. dea682d This CL fixes the following: by alessiob · 7 years ago
  100. 129fc9c Enabling 'gn check' on //webrtc/tools. by mbonadei · 7 years ago