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gerrit-public.fairphone.software / platform / external / webrtc / aeb0c28193a012f7431edd36f96937510a555fc8
  1. aeb0c28 Update PRESUBMIT.py's list of "DO_NOT_SUBMIT_FILES". by henrike@webrtc.org · 11 years ago
  2. e57ae02 WebRtcAecm_Process: Reduce code duplication by kwiberg@webrtc.org · 11 years ago
  3. d2f366f StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16 by kwiberg@webrtc.org · 11 years ago
  4. 6680348 Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  5. 0f73755 Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  6. e2e9abb Revert "Make VoiceEngine choose ACM2 by default" by henrik.lundin@webrtc.org · 11 years ago
  7. 0b3c6c3 (Auto)update libjingle 65086785-> 65104022 by buildbot@webrtc.org · 11 years ago
  8. adaf809 Removing AudioCoding duplicate tests by henrik.lundin@webrtc.org · 11 years ago
  9. 6cec07f Make VoiceEngine choose ACM2 by default by henrik.lundin@webrtc.org · 11 years ago
  10. c0a15b7 Fix crashes due to dangling external decoder pointer. by fischman@webrtc.org · 11 years ago
  11. 39b868b (Auto)update libjingle 65055925-> 65086785 by buildbot@webrtc.org · 11 years ago
  12. 8f88f20 Expand the test max wait time from 1000ms to 2000ms. by jiayl@webrtc.org · 11 years ago
  13. c187291 Set include_internal_video_capture=1 for video_capture_tests by kjellander@webrtc.org · 11 years ago
  14. f927fd6 Re-enable AGC tests: by aluebs@webrtc.org · 11 years ago
  15. 7de47bc Remove use of tmpnam. by kjellander@webrtc.org · 11 years ago
  16. 2c3f1ab Replace flooding logs in rtp_sender.cc with a comment. by andrew@webrtc.org · 11 years ago
  17. 36eda7c Workaround for https://bugzilla.mozilla.org/show_bug.cgi?id=996329, where the m line from firefox have a space at the end. by wu@webrtc.org · 11 years ago
  18. ca539bb iOS: baby steps to being able to include_tests=1 by fischman@webrtc.org · 11 years ago
  19. 7c6e3d1 Moved voe_neteq_stats_unittest to audio_coding_module_unittest by henrik.lundin@webrtc.org · 11 years ago
  20. 6c75c98 Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 11 years ago
  21. 1fd5b45 (Auto)update libjingle 64956819-> 64982143 by buildbot@webrtc.org · 11 years ago
  22. 2f8d5f3 Check if a header extension is registered before updating it and fail silently if it's not. by stefan@webrtc.org · 11 years ago
  23. 190b72a Make libjingle Android example build without sourcing envsetup.sh by kjellander@webrtc.org · 11 years ago
  24. 6e105ed Make WebRTC Android examples build without sourcing envsetup.sh by kjellander@webrtc.org · 11 years ago
  25. ad4440a In shared socket mode, use udp port as default receiver even if by mallinath@webrtc.org · 11 years ago
  26. 505f400 (Auto)update libjingle 64909599-> 64919255 by buildbot@webrtc.org · 11 years ago
  27. e98598d Make everyone an OWNER for .gyp/.gypi add/delete purposes, talk/ edition. by fischman@webrtc.org · 11 years ago
  28. 2c89b5c Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 11 years ago
  29. 35ead38 Adding a config struct to NetEq by henrik.lundin@webrtc.org · 11 years ago
  30. 810acbc New Packet and PacketSource classes for NetEq tests by henrik.lundin@webrtc.org · 11 years ago
  31. 1da6047 (Auto)update libjingle 64813990-> 64909599 by buildbot@webrtc.org · 11 years ago
  32. cf0b46c iosdeviceinfo.cc: remove unnecessary file by fischman@webrtc.org · 11 years ago
  33. 5cf7396 Fix gyp for video_capture/ensure_initialized.cc. by primiano@chromium.org · 11 years ago
  34. f875f15 (Auto)update libjingle 64709629-> 64813990 by buildbot@webrtc.org · 11 years ago
  35. b9309be Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  36. 5692531 Added a new OnMoreData() interface which will not feed the playout data to APM. by xians@webrtc.org · 11 years ago
  37. a956ec2 Add win_drmemory_light trybot to default trybot list. by kjellander@webrtc.org · 11 years ago
  38. 940894d DrMemory: Excluding failing tests for Dr Memory Full by kjellander@webrtc.org · 11 years ago
  39. ef79fd7 DrMemory: Excluding failing tests for Dr Memory Full by kjellander@webrtc.org · 11 years ago
  40. 41e8774 DrMemory: Excluding failing tests for Dr Memory Full by kjellander@webrtc.org · 11 years ago
  41. 8ce7c72 Fix the captured screen rect conversion. by jiayl@webrtc.org · 11 years ago
  42. 8d1cdaa NetEq changes. by turaj@webrtc.org · 11 years ago
  43. ffd2424 DrMemory: Suppress and exclude more tests to green up the full build. by kjellander@webrtc.org · 11 years ago
  44. 34c5da6 Cleaned up logging in video_coding. by stefan@webrtc.org · 11 years ago
  45. 8b2ec15 Convert WEBRTC_TRACE to LOG in utility. by asapersson@webrtc.org · 11 years ago
  46. b884eb6 (Auto)update libjingle 64630087-> 64709629 by wu@webrtc.org · 11 years ago
  47. 8dce41b Remove erronuous commit message from auto sync. by henrike@webrtc.org · 11 years ago
  48. 22cf747 Disable UsesTraceCallback by pbos@webrtc.org · 11 years ago
  49. e6013bb Fix loopback test for case where no constraint is given. by andresp@webrtc.org · 11 years ago
  50. 2a77082 Remove usage of webrtc trace in video processing modules. by asapersson@webrtc.org · 11 years ago
  51. 0273fa9 Add ability to control peer connection constraints for the loopback test. by andresp@webrtc.org · 11 years ago
  52. 15192f9 (Auto)update libjingle 64594651-> 64630087 by buildbot@webrtc.org · 11 years ago
  53. f930214 Remove self-assignment hacks that were added to avoid unused variable warnings. by fischman@webrtc.org · 11 years ago
  54. 0569d93 Move a chatty creation log in neteq to LS_VERBOSE. by andrew@webrtc.org · 11 years ago
  55. 8f89497 Remove erronuous commit message. by henrike@webrtc.org · 11 years ago
  56. f4357f3 Make Android-APK compile in release again. by solenberg@webrtc.org · 11 years ago
  57. 52fd65b Partial revert of "Removing samples directory following move to Github" by kjellander@webrtc.org · 11 years ago
  58. 8883a0f (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds by henrika@webrtc.org · 11 years ago
  59. 7ecc142 Removing samples directory following move to Github by dutton@google.com · 11 years ago
  60. 61c1b8e (Auto)update libjingle 64585415-> 64594651 by buildbot@webrtc.org · 11 years ago
  61. 2e9d89c Unbreak android APK buildbots by emptying the video_capture_tests_apk target. by fischman@webrtc.org · 11 years ago
  62. b0b135e VideoCaptureAndroid: support multiple frame-rates per resolution. by fischman@webrtc.org · 11 years ago
  63. 74f6074 Fix DesktopSize::is_empty() for the case when only width or only height is 0. by sergeyu@chromium.org · 11 years ago
  64. a78a41f Move output_mixer_unittest.cc to utility_unittest.cc. by andrew@webrtc.org · 11 years ago
  65. f4c9444 VideoCaptureAndroid: stop referencing ViERenderer by fischman@webrtc.org · 11 years ago
  66. f824fde (Auto)update libjingle 64326665-> 64585415 by henrike@webrtc.org · 11 years ago
  67. 984e4fb video_capture(iOS): move stopCapture to background thread by fischman@webrtc.org · 11 years ago
  68. 2a03498 Implement FEC support in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  69. dc80bae Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 11 years ago
  70. b287d96 New NetEq test to verify correct timestamp propagation by henrik.lundin@webrtc.org · 11 years ago
  71. 74a7c48 Removes unused thread causing compiler warnings. by henrike@webrtc.org · 11 years ago
  72. 4e39307 Compare the answer's media type against offer to make sure they are match. Otherwise we should return failure. by wu@webrtc.org · 11 years ago
  73. 413d001 Removed the disabling of include_tests from r2729. by henrike@webrtc.org · 11 years ago
  74. 9337c83 Updated WebRTC version to 3.52 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  75. b08db28 Clean up traces and logs in RemoteBitrateEstimator. by stefan@webrtc.org · 11 years ago
  76. 5574dac Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 11 years ago
  77. e8d1865 Disable more tests for DrMemory to speed up execution. by kjellander@webrtc.org · 11 years ago
  78. 36947bb Fix logging calls in bitrate_controller module. by andresp@webrtc.org · 11 years ago
  79. 9f57404 Excluding and suppressing Dr Memory test failures. by kjellander@webrtc.org · 11 years ago
  80. 0fefb10 Remove WEBRTC_TRACE use in common_video/ by pbos@webrtc.org · 11 years ago
  81. 09b0c10 Talk: fixes warning: local variable is initialized but not referenced due to only using the variable in question for asserts. by henrike@webrtc.org · 11 years ago
  82. d1fe6b7 AppRTCDemo(android): fix a couple of SDP-related regressions. by fischman@webrtc.org · 11 years ago
  83. f040bd8 Fix a crash in WindowCapturereMac when capture() fails. by jiayl@webrtc.org · 11 years ago
  84. f5bebd4 (Auto)update libjingle 64247466-> 64326665 by henrike@webrtc.org · 11 years ago
  85. 653c325 Fix the library path for android 64-bit build by michaelbai@google.com · 11 years ago
  86. 40ee3d0 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 11 years ago
  87. cca888a Removed rehydrate.html by dutton@google.com · 11 years ago
  88. be8e8ee Remove bad *s from filename. by andrew@webrtc.org · 11 years ago
  89. c7b8b2f PRESUBMIT.py: use new way to specify default try builders by kjellander@webrtc.org · 11 years ago
  90. fe165de Added warning for Github move ****THESE_FILES_ARE_MOVING**** by dutton@google.com · 11 years ago
  91. 240eec3 Delay Estimator: Minor refactoring and added a setter function. by bjornv@webrtc.org · 11 years ago
  92. 1481491 (Auto)update libjingle 64147530-> 64247466 by wu@webrtc.org · 11 years ago
  93. 5e760e7 Check the return value of the FromString call and return failure when then value is invalid. I.e. uses by wu@webrtc.org · 11 years ago
  94. e387771 Remove webrtc_unittest.cc from talk presubmit script. by wu@webrtc.org · 11 years ago
  95. 184b913 Rename RTPanalyze to rtp_analyze and remove old version by henrik.lundin@webrtc.org · 11 years ago
  96. c7c432a Remove AudioDevice::{Microphone,Speaker}IsAvailable. by andrew@webrtc.org · 11 years ago
  97. 7549ff4 This is to get rid of a bug relating to the return of NULL in calling GetDecoder when there are DTMF packets. by minyue@webrtc.org · 11 years ago
  98. 1092ea0 Add format specification to output file names by henrik.lundin@webrtc.org · 11 years ago
  99. 620d444 Extends max sample rate from 96kHz to 192kHz on the input side. by henrika@webrtc.org · 11 years ago
  100. 790385f sink_filter_ds.cc: add lock to Receive procedure to Pause(). by braveyao@webrtc.org · 11 years ago