1. 00d802b Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ ) by ilnik · 7 years ago
  2. 10d095d Revert of Change NetEq::InsertPacket to take an RTPHeader (patchset #2 id:20001 of https://codereview.webrtc.org/2807273004/ ) by henrik.lundin · 7 years ago
  3. 80ff00c Improve USB device reset logic by jansson · 7 years ago
  4. b213a16 Finalized the SSE2 optimizations for the matched filter in AEC3 by peah · 7 years ago
  5. c0d74d9 Roll chromium_revision 419b6b1a41..aa2ca13eaa (463589:463610) by buildbot · 7 years ago
  6. 27c46e2 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ ) by ilnik · 7 years ago
  7. 4d02757 Change NetEq::InsertPacket to take an RTPHeader by henrik.lundin · 7 years ago
  8. 774f6b4 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 7 years ago
  9. 268862c Address denicija's comments for AppRTCMobile video codec setting. by sakal · 7 years ago
  10. 24da37b ObjC: RTCVideoSource cleanup by magjed · 7 years ago
  11. 29dbb19 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ ) by ilnik · 7 years ago
  12. dee5eb1 Android Logging.java: Load native library only when needed by magjed · 7 years ago
  13. 382f2b2 Fix swarming tests not running in parallel by kjellander · 7 years ago
  14. 4fa0c4f Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 7 years ago
  15. 5721866 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ ) by ilnik · 7 years ago
  16. cabc25c Roll chromium_revision 860f7b94d4..419b6b1a41 (463558:463589) by buildbot · 7 years ago
  17. cde2528 Enabling 'gn check' on //webrtc/ortc. by mbonadei · 7 years ago
  18. 10fc0e6 Delay based logging. by philipel · 7 years ago
  19. 64e739a Add content type information to Encoded Images and add corresponding RTP extension header. by ilnik · 7 years ago
  20. 93cda2e APM-QA tool, renaming noise generators into input-reference generators. by alessiob · 7 years ago
  21. 9765370 Resolve dependency between rtc_event_log_api and remote_bitrate_estimator by michaelt · 7 years ago
  22. 810eecf Roll chromium_revision c57654688e..860f7b94d4 (463520:463558) by buildbot · 7 years ago
  23. 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
  24. 333d0ff Add first part of the network_tester functionality. by michaelt · 7 years ago
  25. e0ab0ad Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT by kjellander · 7 years ago
  26. 0d4e068 Make safe_cmp::* constexpr by kwiberg · 7 years ago
  27. d491109 Roll chromium_revision 1af3c1a4a8..c57654688e (463476:463520) by buildbot · 7 years ago
  28. 4a9d08f Roll chromium_revision d3a2a83fbf..1af3c1a4a8 (463418:463476) by buildbot · 7 years ago
  29. 8c459f9 Restore old (deprecated) signature of initializeAndroidGlobals. by deadbeef · 7 years ago
  30. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 7 years ago
  31. d9ce764 Make RtpTransport actually implement RtpTransportInterface by zstein · 7 years ago
  32. 0f92c79 Roll chromium_revision 5d7042a87c..d3a2a83fbf (463209:463418) by buildbot · 7 years ago
  33. b4fc73a Removing unnecessary parameters from initializeAndroidGlobals. by deadbeef · 7 years ago
  34. 6799553 Add information about microphone gain changes to AEC3 by peah · 7 years ago
  35. 6d822ad Added forced zero AEC output after call startup and echo path changes by peah · 7 years ago
  36. ca31f17 Remove deprecated RTPPayloadStrategy by danilchap · 7 years ago
  37. a1ef71f Add parser to visualise the ana dump by michaelt · 7 years ago
  38. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 7 years ago
  39. b0f7e39 Move IsIntlike to type_traits.h by kwiberg · 7 years ago
  40. 37e99fd Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 7 years ago
  41. 2fa97fd Roll chromium_revision 0a53e4a670..5d7042a87c (463181:463209) by buildbot · 7 years ago
  42. 0642b32 Remove duplicate entries from AUTHORS file by henrik.lundin · 7 years ago
  43. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 7 years ago
  44. 925e9d7 Removed workaround for the WARN_UNUSED_RESULT issue. by peah · 7 years ago
  45. 4fb651d Event log cleanup in tests. by philipel · 7 years ago
  46. fca900a Fix two invalid DCHECKs related to audio BWE. by stefan · 7 years ago
  47. 49cad02 Ignore some UBSan errors by kwiberg · 7 years ago
  48. 9f2c18e Changed OLA window for neteq. Old code didnt work well with 48khz by soren · 7 years ago
  49. c547e84 Allow rtp::Packet::*RawExtension to take 0 as an extension id by danilchap · 7 years ago
  50. 02465b8 Add some unit tests to vie_encoder. by asapersson · 7 years ago
  51. 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
  52. 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
  53. 64c93c3 Roll chromium_revision 1ab7c6059c..0a53e4a670 (463170:463181) by buildbot · 7 years ago
  54. 188596f Delete class ScopedPtrCollection. Replaced with vector of unique_ptr. by nisse · 7 years ago
  55. bd1a681 Roll chromium_revision c8a0f6b4c5..1ab7c6059c (463161:463170) by buildbot · 7 years ago
  56. 1b0882d Roll chromium_revision c30f6366d7..c8a0f6b4c5 (463148:463161) by buildbot · 7 years ago
  57. 1d9ef47 Roll chromium_revision b0bf8e8ed3..c30f6366d7 (463140:463148) by buildbot · 7 years ago
  58. a3d4d94 Roll chromium_revision 5e27d4b8d1..b0bf8e8ed3 (463138:463140) by buildbot · 7 years ago
  59. 4b37127 Fix compilation issues of std::unique_ptr by steweg · 7 years ago
  60. 8cc642b Roll chromium_revision d5df028c94..5e27d4b8d1 (463137:463138) by buildbot · 7 years ago
  61. 89292d1 Roll chromium_revision 270af5af87..d5df028c94 (463130:463137) by buildbot · 7 years ago
  62. 5253411 Roll chromium_revision 9d8fbbd04c..270af5af87 (463126:463130) by buildbot · 7 years ago
  63. 01f2793 Roll chromium_revision a35a1e2ce2..9d8fbbd04c (463121:463126) by buildbot · 7 years ago
  64. 50ddc63 Roll chromium_revision 98c8321fe9..a35a1e2ce2 (463081:463121) by buildbot · 7 years ago
  65. 9d8052d Roll chromium_revision 762665735a..98c8321fe9 (462871:463081) by buildbot · 7 years ago
  66. 66e9f76 Adjust parameter in vp9 videoprocessor_integration test. by jianj · 7 years ago
  67. 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
  68. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 7 years ago
  69. bb16a48 Roll chromium_revision 61577bba5a..762665735a (462822:462871) by buildbot · 7 years ago
  70. 8942045 Adding support for handling highly reverberant echoes in AEC3. by peah · 7 years ago
  71. 38415b2 Reland of Adding PRESUBMIT check on google::protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2791583002/ ) by mbonadei · 7 years ago
  72. 423f106 Add support for 64-bit architectures in build_aar.py. by sakal · 7 years ago
  73. 2ce640f Fixing sample-rate dependent band-split filter issues in AEC3 by peah · 7 years ago
  74. ea44ad4 Roll chromium_revision d65c1d7370..61577bba5a (462801:462822) by buildbot · 7 years ago
  75. 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
  76. 0944a80 Update stats for cpu/quality adaptation changes to excluded time when video is suspended. by asapersson · 7 years ago
  77. 533b7ac Roll chromium_revision c405301b70..d65c1d7370 (462743:462801) by buildbot · 7 years ago
  78. 7749286 Make AudioProcessing::GetConfig() pure virtual by henrik.lundin · 7 years ago
  79. abd101b Support multiple connected Android devices in low bandwidth audio test by oprypin · 7 years ago
  80. 225bfc0 Make PacketTransportInternal inherit from PacketTransportInterface. by deadbeef · 7 years ago
  81. 81c899d Roll chromium_revision fbddb8a080..c405301b70 (462718:462743) by buildbot · 7 years ago
  82. f49011b Roll chromium_revision 300bd53cd8..fbddb8a080 (462634:462718) by buildbot · 7 years ago
  83. 4b572d0 Correction of the AEC3 underrun behavior and minor other corrections by peah · 7 years ago
  84. 86afe9d Major updates to the echo removal functionality in AEC3 by peah · 7 years ago
  85. f51517a Roll chromium_revision bb330d1b8f..300bd53cd8 (462561:462634) by buildbot · 7 years ago
  86. 8cd1cce Roll chromium_revision 86718dcdf2..bb330d1b8f (462494:462561) by buildbot · 7 years ago
  87. 1ffbd6c Injectable audio encoders: voice_engine/channel changes. by ossu · 7 years ago
  88. 5f4aaeb Re-enable FullStackTest.ScreenshareSlidesVP9_2SL test. by marpan · 7 years ago
  89. 4b62001 Adding AudioDeviceDataObserver interface by Lu Liu · 7 years ago
  90. a1a040a Injectable audio encoders: BuiltinAudioEncoderFactory by ossu · 7 years ago
  91. ac4bbdf Roll chromium_revision 3014f8b41e..86718dcdf2 (462374:462494) by buildbot · 7 years ago
  92. cab8d88 Revert of Enable rtc_unittests on iOS simulator (patchset #2 id:20001 of https://codereview.webrtc.org/2799033004/ ) by kjellander · 7 years ago
  93. 6167b26 Make RtpTransportControllerSend::send_side_cc_ a direct member. by nisse · 7 years ago
  94. cde46b7 Resolve cyclic dependency between audio network adaptor and event log api by michaelt · 7 years ago
  95. 28dc285 Adding cbr support for Opus by soren · 7 years ago
  96. 388fe42 Make WARN_UNUSED_RESULT a no-op on gcc by kwiberg · 7 years ago
  97. 177b17e Move AndroidVideoTrackSourceObserver from API to src by magjed · 7 years ago
  98. 639d46a Delete system_wrappers logging facility. by nisse · 7 years ago
  99. 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
  100. 2418001 ACM: Change test output files from PCM to WAV by henrik.lundin · 7 years ago