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webrtc / src.git / 76b0c2ce47347184fa051870234bd7698ab30f3e
  1. 76b0c2ce Revert "Reland "Port: migrate to TaskQueue."" by Mirko Bonadei · 3 years, 9 months ago
  2. 3ead44a Update WebRTC code version (2021-07-13T04:01:55). by webrtc-version-updater · 3 years, 9 months ago
  3. 3fcd31c Roll chromium_revision 2945be3cf4..2bb5b1a1b1 (900657:900780) by chromium-webrtc-autoroll · 3 years, 9 months ago
  4. f5b73a7 Roll chromium_revision ab1696a8d1..2945be3cf4 (900556:900657) by chromium-webrtc-autoroll · 3 years, 9 months ago
  5. bf15e56 dcsctp: Abandon chunks consistently by Victor Boivie · 3 years, 9 months ago
  6. 4e18a1f Roll chromium_revision e3a078c0d6..ab1696a8d1 (900427:900556) by chromium-webrtc-autoroll · 3 years, 9 months ago
  7. ba8a204 Roll chromium_revision a564f9a8f4..e3a078c0d6 (900313:900427) by chromium-webrtc-autoroll · 3 years, 9 months ago
  8. 6307809 Update WebRTC code version (2021-07-12T04:04:47). by webrtc-version-updater · 3 years, 9 months ago
  9. 3bc0f87 Roll chromium_revision 52a2811082..a564f9a8f4 (900213:900313) by chromium-webrtc-autoroll · 3 years, 9 months ago
  10. 4f3226d Update WebRTC code version (2021-07-11T04:04:45). by webrtc-version-updater · 3 years, 9 months ago
  11. 813a087 dcsctp: Add packet capture support in unit tests by Victor Boivie · 3 years, 9 months ago
  12. 5e726da dcsctp: Extract logging packet observer as utility by Victor Boivie · 3 years, 9 months ago
  13. e7e9292 Analog AGC: Add clipping rate metrics by Hanna Silen · 3 years, 9 months ago
  14. a88d5f6 Roll chromium_revision 0c7499c8ca..52a2811082 (900073:900213) by chromium-webrtc-autoroll · 3 years, 9 months ago
  15. 23a2b3b Roll chromium_revision a5d70b42f2..0c7499c8ca (899910:900073) by chromium-webrtc-autoroll · 3 years, 9 months ago
  16. fb1a0f0 Cleanup rtp utils in media/base by Danil Chapovalov · 3 years, 9 months ago
  17. e09a174 Fix ssl_certificate_fuzzer by Danil Chapovalov · 3 years, 9 months ago
  18. d6afbea Correctly set number of reference buffers in H264 encoder by Sergey Silkin · 3 years, 9 months ago
  19. e91c992 Implement nack_count metric for outbound audio rtp streams. by Jakob Ivarsson · 3 years, 9 months ago
  20. b629aee Roll chromium_revision 17c6abea92..a5d70b42f2 (899632:899910) by chromium-webrtc-autoroll · 3 years, 9 months ago
  21. 25ab322 Replace assert() with RTC_DCHECK(). by Mirko Bonadei · 3 years, 9 months ago
  22. 9b5d570 Update WebRTC code version (2021-07-09T04:05:24). by webrtc-version-updater · 3 years, 9 months ago
  23. 2bbbd66 Remove backwards compatible build targets. by Mirko Bonadei · 3 years, 9 months ago
  24. a102d8e Roll chromium_revision 0d58dd3c85..17c6abea92 (899016:899632) by chromium-webrtc-autoroll · 3 years, 9 months ago
  25. 44450a0 Support header only parsing by RtpPacket by Danil Chapovalov · 3 years, 9 months ago
  26. ea9ae5b Destroy threads and TaskQueue at the end of tests. by Mirko Bonadei · 3 years, 9 months ago
  27. 8bd26e1 dcsctp: Only reset paused streams when peer acks by Victor Boivie · 3 years, 9 months ago
  28. 706ef1b Create name->value text map for frame and video statistics by Sergey Silkin · 3 years, 9 months ago
  29. 5d70fe76 Temporarily skip tests that consistently fail on Linux MSan. by Mirko Bonadei · 3 years, 9 months ago
  30. 0c5a5ca doc: using triple backticks instead of <pre> blocks by Byoungchan Lee · 3 years, 9 months ago
  31. f715618 Use flat_map in RTCPReceiver by Victor Boivie · 3 years, 9 months ago
  32. 5900ba0 Explicitly expose EraseIf in flat_map/flat_set by Victor Boivie · 3 years, 9 months ago
  33. 1864997 Use flat_map in ReceiveStatisticsImpl by Victor Boivie · 3 years, 9 months ago
  34. ac5f2e7 Use flat_map/flat_set in RtpDemuxer by Victor Boivie · 3 years, 9 months ago
  35. 9e89c76 Roll chromium_revision 78296924f6..0d58dd3c85 (898909:899016) by chromium-webrtc-autoroll · 3 years, 9 months ago
  36. e07f8c5 Update WebRTC code version (2021-07-07T04:06:33). by webrtc-version-updater · 3 years, 9 months ago
  37. 53d3fc9 iOS: Get WebRTC building for Mac Catalyst by Jordan Rose · 3 years, 9 months ago
  38. fd954fc Import flat_map and flat_set from chromium/base/ by Victor Boivie · 3 years, 9 months ago
  39. 93f9f35 Roll chromium_revision 94a136c73d..78296924f6 (898790:898909) by chromium-webrtc-autoroll · 3 years, 9 months ago
  40. 6d92fcd Roll chromium_revision ba5ff58b6c..94a136c73d (898571:898790) by Mirko Bonadei · 3 years, 9 months ago
  41. 5a5d751 VP9 parser: undo r34393 and fix incorrect return statement. by Erik Språng · 3 years, 9 months ago
  42. 54388a8 Fix a comment in FrameDropper by Sergey Silkin · 3 years, 9 months ago
  43. c41093b Add ability to build XCFramework for iOS by Byoungchan Lee · 3 years, 9 months ago
  44. 00ca004 Unify helpers IsRtpPacket and IsRtcpPacket by Danil Chapovalov · 3 years, 9 months ago
  45. 93ce46f Update WebRTC code version (2021-07-06T04:06:32). by webrtc-version-updater · 3 years, 9 months ago
  46. 6b09c45 Silence OpenGLES deprecation warning. by Mirko Bonadei · 3 years, 9 months ago
  47. c661b3f Roll chromium_revision 2d8d6a6937..ba5ff58b6c (898461:898571) by chromium-webrtc-autoroll · 3 years, 9 months ago
  48. bac0f9f Remove x86 from build_ios_libs. by Mirko Bonadei · 3 years, 9 months ago
  49. f94f565 Roll chromium_revision 38e62a7013..2d8d6a6937 (898361:898461) by chromium-webrtc-autoroll · 3 years, 9 months ago
  50. 1121b8b Roll chromium_revision 487e5997d5..38e62a7013 (898251:898361) by chromium-webrtc-autoroll · 3 years, 9 months ago
  51. 6504fbd Roll chromium_revision 6f7025c98c..487e5997d5 (893293:898251) by chromium-webrtc-autoroll · 3 years, 9 months ago
  52. dfcc23b Remove arm32 from build_ios_libs. by Mirko Bonadei · 3 years, 9 months ago
  53. 2a4ed16 Clean up iOS 32 bits build GN configs from MB. by Mirko Bonadei · 3 years, 9 months ago
  54. 02768ae Increase iOS deployment target from 10 to 12. by Björn Terelius · 3 years, 9 months ago
  55. 94f2ef2 Run pylint on presubmit only for modified python files. by Byoungchan Lee · 3 years, 9 months ago
  56. 510c94c Return one report block per media ssrc, ignoring sender ssrc. by Danil Chapovalov · 3 years, 9 months ago
  57. 2ba604d Update upload completion check logs to make them more intuitive by Andrey Logvin · 3 years, 9 months ago
  58. 92fc021 Replace PacketView by vector of pointers in a wrapper class. by Björn Terelius · 3 years, 9 months ago
  59. d45f930 Add missing rate control settings for av1 wrapper by Jerome Jiang · 3 years, 9 months ago
  60. c0a4a09 Use default NetEq config for simulation in event log visualizer. by Jakob Ivarsson · 3 years, 9 months ago
  61. b42ced4 Prepare WebRtcVideoReceiveStream for configuration changes. by Tommi · 3 years, 9 months ago
  62. e54914a Implement nack_count metric for inbound audio rtp streams. by Jakob Ivarsson · 3 years, 9 months ago
  63. 6832ee2 Delete unneeded references to string_encode.h by Niels Möller · 3 years, 9 months ago
  64. ef83d15 Update peerconnection example to not use Win32SocketServer by Niels Möller · 3 years, 9 months ago
  65. 899b29e Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Byoungchan Lee · 3 years, 9 months ago
  66. 7d0b422 Add bionic mb builder config by Christoffer Jansson · 3 years, 9 months ago
  67. 9e5da27 Update WebRTC code version (2021-07-01T04:09:57). by webrtc-version-updater · 3 years, 9 months ago
  68. f11f0e0 openssl: Use unique_ptr to remove unnecessary goto statements from openssl_adapter by Vojin Ilic · 3 years, 9 months ago
  69. 7a3db9d Fix CPD upload script --input-results-file description by Andrey Logvin · 3 years, 9 months ago
  70. 286b1db Fix -Wunreachable-code-aggressive. by Peter Kasting · 3 years, 9 months ago
  71. c060ce4 video_replay: use abseil uint32_t for ssrc and ssrc_rtx by Philipp Hancke · 3 years, 9 months ago
  72. 2ae4ed2 Fix the last checksum by Christoffer Jansson · 3 years, 9 months ago
  73. 2c4d243 Add ARM64 for min expected PSNR score for the M1 Mac. by Christoffer Jansson · 3 years, 9 months ago
  74. 46d002c Add M1 Mac expected results for AudioDecoderIsacFixTest by Christoffer Jansson · 3 years, 9 months ago
  75. fbdd778 Update WebRTC code version (2021-06-30T04:03:59). by webrtc-version-updater · 3 years, 9 months ago
  76. 985905d Add fieldtrial to enable minimum pacing of video frames by Johannes Kron · 3 years, 9 months ago
  77. dcb9ffc DataChannel: Propagate SCTP transport errors to the channels by Florent Castelli · 3 years, 9 months ago
  78. c362eb2 dcsctp: Add mocks by Victor Boivie · 3 years, 9 months ago
  79. da9dfae Re-enable ChangeFramerateVP8 & ChangeBitrateVP8 for Android and iOS by Christoffer Jansson · 3 years, 9 months ago
  80. 06a88fe openssl: Use unique_ptr to remove unnecessary goto statements by Vojin Ilic · 3 years, 9 months ago
  81. 37477b8 Update WebRTC code version (2021-06-29T04:02:31). by webrtc-version-updater · 3 years, 9 months ago
  82. 5038c8b Revert "Reland "Trigger postsubmit tests."" by Mirko Bonadei · 3 years, 9 months ago
  83. 809f21b Reland "Trigger postsubmit tests." by Mirko Bonadei · 3 years, 9 months ago
  84. 53f1fe4 Fail instead of crashing while writing invalid dependency descriptor by Danil Chapovalov · 3 years, 9 months ago
  85. 80090467a Revert "Trigger postsubmit tests." by Mirko Bonadei · 3 years, 9 months ago
  86. 1a8c8b6 Trigger postsubmit tests. by Mirko Bonadei · 3 years, 9 months ago
  87. df400c3 Update roll_deps.py to use main branch by Christoffer Jansson · 3 years, 9 months ago
  88. e99f687 Move WebRTC to non deprecated jsoncpp APIs. by Mirko Bonadei · 3 years, 9 months ago
  89. 06b8f7e Move supported_platforms.md under g3doc/. by Mirko Bonadei · 3 years, 9 months ago
  90. e6ab520 First draft of WebRTC supported platforms/compilers documentation. by Mirko Bonadei · 3 years, 9 months ago
  91. 7208457 Same length for all ARM64 platforms Update more audio checksums for M1 by Christoffer Jansson · 3 years, 9 months ago
  92. 74543b7 PlatformThreadTest: fix flake. by Markus Handell · 3 years, 9 months ago
  93. 2b3a10e Add MAC arm64 platform and update checksums for acm unittest by Christoffer Jansson · 3 years, 9 months ago
  94. 5e82c75 Remove TODOs to remove SetAudioPlayback and SetAudioRecording by Harald Alvestrand · 3 years, 9 months ago
  95. 482b7c0 Fix -Wimplicit-int-float-conversions. by Peter Kasting · 3 years, 9 months ago
  96. 64851c0 Reland: Fix echo return loss stats and add to RTCAudioSourceStats. by Taylor Brandstetter · 3 years, 9 months ago
  97. a987429 Update WebRTC code version (2021-06-25T04:04:09). by webrtc-version-updater · 3 years, 9 months ago
  98. c830bd6 Remove ssl_certificate() accessor. by Harald Alvestrand · 3 years, 9 months ago
  99. e9a74c9 Public RtpVideoFrameAssembler by philipel · 3 years, 9 months ago
  100. 4e51334 AV1 OBU test helper. by philipel · 3 years, 9 months ago