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gerrit-public.fairphone.software / platform / external / webrtc / 7992b409944597be058b43b506fc2a875518e82a
  1. 7992b40 (Auto)update libjingle 77953038-> 77970462 by buildbot@webrtc.org · 10 years ago
  2. b1dac33 Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." by henrike@webrtc.org · 10 years ago
  3. 5820294 Cleaning up Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  4. 0371a37 Moving creating TURN configration to the host machine instead of the bots - rtcBot by houssainy@google.com · 10 years ago
  5. f7030d4 Query Android device orientation on every camera frame received. by glaznev@webrtc.org · 10 years ago
  6. 9c58ea8 rtc_unittest: copied gtest excludes from libjingle_p2p_unittest since its tests have move to rtc_unittests. by henrike@webrtc.org · 10 years ago
  7. c221db6 Test names changed from e.g) testOneWayVideo/chrome=>chrome to testOneWayVideo/chrome-chrome. by houssainy@google.com · 10 years ago
  8. 264e66f Add encoded_timestamp to AudioEncoder base class by henrik.lundin@webrtc.org · 10 years ago
  9. 9ea6f8a New interface class AudioEncoder by henrik.lundin@webrtc.org · 10 years ago
  10. 8efaa27 Disable a bunch of Nat and Ice tests when running under DrMemory. by stefan@webrtc.org · 10 years ago
  11. 458c2c3 Improve rtcbot to load all test files at start and allow them to registerTests by andresp@webrtc.org · 10 years ago
  12. 9aed002 Add ability to include a larger time span (in addition to encode time) for measuring the processing time of a frame. by asapersson@webrtc.org · 10 years ago
  13. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
  14. 3e2f8ff Selecting bot_type changed to be specified in the test file by houssainy@google.com · 10 years ago
  15. e93cbd1 Fix data races in ThreadTest.ThreeThreadsInvoke. by pbos@webrtc.org · 10 years ago
  16. f87c0af audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> by bjornv@webrtc.org · 10 years ago
  17. f02ba9b audio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> by bjornv@webrtc.org · 10 years ago
  18. 8dc00d7 audio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> by bjornv@webrtc.org · 10 years ago
  19. 99e561f Extend AcmSwitchingOutputFrequencyOldApi with more frequencies by henrik.lundin@webrtc.org · 10 years ago
  20. 64f5611 Roll chromium_revision 2d714fa..de13cf4 (298667:299488) by kjellander@webrtc.org · 10 years ago
  21. fab5439 common_audio: Removed version API from signal_processing by bjornv@webrtc.org · 10 years ago
  22. 81ddc78 (Auto)update libjingle 77701902-> 77709729 by buildbot@webrtc.org · 10 years ago
  23. 1ecbe45 (Auto)update libjingle 77689511-> 77696841 by buildbot@webrtc.org · 10 years ago
  24. 43336b6 Remove unused (no-op) VideoOptions. by pbos@webrtc.org · 10 years ago
  25. a4351a0 libjingle: use _stricmp instead of deprecated stricmp. by henrike@webrtc.org · 10 years ago
  26. a73a678 Remove -1 from Call::Config::start_bitrate_bps. by pbos@webrtc.org · 10 years ago
  27. eb24b04 Add periodic logging of received RTP headers and estimated clock offsets for e2e delay. by stefan@webrtc.org · 10 years ago
  28. 81a7893 New ACM test to trigger audio glitch when switching output sample rate by henrik.lundin@webrtc.org · 10 years ago
  29. c216b9a Add a packet loss full stack test to the new API. by stefan@webrtc.org · 10 years ago
  30. a57678a Workarounds for a bug in VS2013.3 linker when PGO is turned on. by kwiberg@webrtc.org · 10 years ago
  31. 7fe1e03 Wire up external encoders. by pbos@webrtc.org · 10 years ago
  32. f68cc0b (Auto)update libjingle 77554188-> 77629208 by buildbot@webrtc.org · 10 years ago
  33. 82e6fa5 Move exlusion of VP9 integration tests for DrMemory by marpan@webrtc.org · 10 years ago
  34. b6af428 Adjust speech probability in NS when echo by aluebs@webrtc.org · 10 years ago
  35. 1e6a5dd Removes xmllite from talk/xmllite since webrtc/xmllite is used instead. by henrike@webrtc.org · 10 years ago
  36. 8bee130 Disable VP9 integration tests on DrMemory. by marpan@webrtc.org · 10 years ago
  37. bc1a457 common_audio: Removed macro WEBRTC_SPL_RSHIFT_W16 by bjornv@webrtc.org · 10 years ago
  38. a3722b6 iSAC tests: Type buffers as uint8_t[] to avoid casts by kwiberg@webrtc.org · 10 years ago
  39. d4fe824 audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> by bjornv@webrtc.org · 10 years ago
  40. 396a5e0 WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[] by kwiberg@webrtc.org · 10 years ago
  41. 3f7f899 WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16 by kwiberg@webrtc.org · 10 years ago
  42. 1172988 Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[] by kwiberg@webrtc.org · 10 years ago
  43. 3c16d8b (Auto)update libjingle 77414393-> 77554188 by buildbot@webrtc.org · 10 years ago
  44. c502df5 Merge the supporting to UYVY on Linux video capture in crbug/410202 to webrtc standalone. by braveyao@webrtc.org · 10 years ago
  45. 651c05e Release _inputSendPin & _outputCapturePin before _captureFilter & _sinkFilter since they should depend on the filters. by braveyao@webrtc.org · 10 years ago
  46. 7f7b0a1 Re-enable ThreadCheckerDeathTest.MethodNotAllowedOnDifferentThreadInDebug (missed when enabling other base tests). by henrike@webrtc.org · 10 years ago
  47. 4ddbbed Disable SendsAndReceivesVP9 test for now. by marpan@webrtc.org · 10 years ago
  48. c87b747 Adjust/increase rate control thresold for a vp9 test. by marpan@webrtc.org · 10 years ago
  49. 573c78e Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
  50. 3cefbc9 Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. by xians@webrtc.org · 10 years ago
  51. afede83 Cleanup scripts and suppressions for TSan v1 by kjellander@webrtc.org · 10 years ago
  52. fae6bc4 Remove talk_base from suppressions. by pbos@webrtc.org · 10 years ago
  53. e46bc77 Reland 28629004: adding new AEC dump start interface for chrome. by xians@webrtc.org · 10 years ago
  54. c5593ef Workaround deps2git issue with inline Python in DEPS. by kjellander@webrtc.org · 10 years ago
  55. c732a3e Re-enable allmost all base tests. by henrike@webrtc.org · 10 years ago
  56. 4a73519 Re-enables a bunch of base unittests part II. by henrike@webrtc.org · 10 years ago
  57. dae40dc Change setting VP8 codec specific info values by HW VP8 encoder by glaznev@webrtc.org · 10 years ago
  58. e30dab7 base/thread_unittest: wrap test was setting current thread to NULL. by henrike@webrtc.org · 10 years ago
  59. 17f8ddd Make pbos and kjellander only owners of tsan2 suppressions. by henrike@webrtc.org · 10 years ago
  60. 8768f16 Fix comments in common_types.h by henrik.lundin@webrtc.org · 10 years ago
  61. 3ff788c Increase timeout for AsyncWriteTest.TestWrite. by pbos@webrtc.org · 10 years ago
  62. 4bd2db9 Opus wrapper: Use const for inputs and uint8[] for byte streams by kwiberg@webrtc.org · 10 years ago
  63. 1bada48 Make DEPS find check_root_dir.py in legacy checkouts. by kjellander@webrtc.org · 10 years ago
  64. 2c0cdbc Estimating NTP time with a given RTT. by minyue@webrtc.org · 10 years ago
  65. c803907 Removing useless packets when inserting them (NetEq) by minyue@webrtc.org · 10 years ago
  66. 0b0ac82 Remove root_dir variable from DEPS + enforce rename. by kjellander@webrtc.org · 10 years ago
  67. 3ea35fd common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16 by bjornv@webrtc.org · 10 years ago
  68. 127ca3f Disable TestDTLSConnectWithSmallMtu on all platforms. by pbos@webrtc.org · 10 years ago
  69. 0001adc Use openmax_dl on all ARM (v7 or higher) platforms. by andrew@webrtc.org · 10 years ago
  70. 95bacfe Remove bad waiting code from video decoder release function. by glaznev@webrtc.org · 10 years ago
  71. 97abeee (Auto)update libjingle 77263371-> 77296420 by buildbot@webrtc.org · 10 years ago
  72. 536eb98 Re-enables a bunch of base unittests. by henrike@webrtc.org · 10 years ago
  73. 9ea5396 Roll chromium_revision fc668e2..2d714fa (298195:298667) by andrew@webrtc.org · 10 years ago
  74. 4165f7a Add a variable for deciding when to use openmax_dl. by andrew@webrtc.org · 10 years ago
  75. f71785c audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> by bjornv@webrtc.org · 10 years ago
  76. 575d126 Protect send_/recv_streams_ in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  77. 9c6dc46 CHECK/DCHECK: Explicitly state whether the condition can have side effects by kwiberg@webrtc.org · 10 years ago
  78. 5e3d7c7 Change name of a NetEq internal member variable by henrik.lundin@webrtc.org · 10 years ago
  79. 742922b Make the media content send only if offerToReceive is false while local streams exist. by jiayl@webrtc.org · 10 years ago
  80. d6bda09 Initialize sctp_paddrparams in OpenSctpSocket(). by pbos@webrtc.org · 10 years ago
  81. 27e5898 Explicitly unpoison FDs for MSan. by pbos@webrtc.org · 10 years ago
  82. 46ffc70 Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder. by glaznev@webrtc.org · 10 years ago
  83. 963b979 Remove potential deadlock in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  84. a9e363e Roll chromium_revision c264a05..fc668e2 (297113:298195) by kjellander@webrtc.org · 10 years ago
  85. 77d5a57 Revert "Only configure the SSL library in one place." by pbos@webrtc.org · 10 years ago
  86. 6ed1cf4 Isolate: Remove use of --ignore_broken_items by kjellander@webrtc.org · 10 years ago
  87. 9103953 Fix neteq_rtpplay so that empty SSRC is valid by henrik.lundin@webrtc.org · 10 years ago
  88. 7cbc4f9 Set NetEq playout mode through the Config struct by henrik.lundin@webrtc.org · 10 years ago
  89. 8b65d51 Add an SSRC filter to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  90. 532ed43 Prevent reading outside iSAC bitstream, if the stream is corrupted. by turaj@webrtc.org · 10 years ago
  91. 8234fa6 Only configure the SSL library in one place. by henrike@webrtc.org · 10 years ago
  92. 2fe5893 Mac: adds missing _DEBUG flag to mac debug builds. by henrike@webrtc.org · 10 years ago
  93. 528fc65 Fixing build issue with L-sdk by henrike@webrtc.org · 10 years ago
  94. 9a742b4 talk: removes empty directories base and sound. by henrike@webrtc.org · 10 years ago
  95. 5d3e7ac Check on the existence of report directory by houssainy@google.com · 10 years ago
  96. 42684be Wire up CPU adaptation in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  97. 31b75ea Moves xmllite's unittests to rtc_unittest. by henrike@webrtc.org · 10 years ago
  98. 25cc745 Switch to SW video decoder on Android after getting 2 or more by glaznev@webrtc.org · 10 years ago
  99. 4b133da Let RtpFileSource use RtpFileReader by henrik.lundin@webrtc.org · 10 years ago
  100. 348eac6 audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with >> by bjornv@webrtc.org · 10 years ago