1. 224933c Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  2. bd389d5 Android, fixes crash on devices with only front cameras. by henrike@webrtc.org · 11 years ago
  3. ccee3c3 Output logs to stderr from voe_cmd_test by default. by andrew@webrtc.org · 11 years ago
  4. e412bb6 Android example apps: fixes issue where useful failure information was suppressed. by henrike@webrtc.org · 11 years ago
  5. 7262655 Potential dead lock in receive statistics by sprang@webrtc.org · 11 years ago
  6. c85c797 Fix for libtalkmobile build error bug=b/12549061 by elham@webrtc.org · 11 years ago
  7. e1b9880 Removes script for generating supplement.gypi also adds git ignore for tools/gn. by henrike@webrtc.org · 11 years ago
  8. 2752bb1 Set up receiver RTX config using a std::map. by pbos@webrtc.org · 11 years ago
  9. 64339f0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
  10. b6a78e5 Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback by henrike@webrtc.org · 11 years ago
  11. 5426f84 Implement screen enumeration and individual screen capturing for Windows. by jiayl@webrtc.org · 11 years ago
  12. ac4c7ea Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started. by henrike@webrtc.org · 11 years ago
  13. 6b84a14 Android, WebRTCDemo: fix issue where changing remote IP was not working properly. by henrike@webrtc.org · 11 years ago
  14. 1240a3d Add full path to headers by aluebs@webrtc.org · 11 years ago
  15. d8163b6 Adds back set_sample_rate_hz() when Init is called in recordings. by bjornv@webrtc.org · 11 years ago
  16. f180890 MIPS optimizations for NS audio processing module by andrew@webrtc.org · 11 years ago
  17. 08ef082 Fix crash in MouseCursor::CopyOf() by sergeyu@chromium.org · 11 years ago
  18. 0a71d11 Exclude protoc objects from merge_libs.py. by andrew@webrtc.org · 11 years ago
  19. df9f867 Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. by mallinath@webrtc.org · 11 years ago
  20. 7f1a443 Extends the ScreenCapturer interface for individual display screen cast. by jiayl@webrtc.org · 11 years ago
  21. c1792c5 Roll Chromium 238260 -> 243863 by wjia@webrtc.org · 11 years ago
  22. ef3b24e Remove empty VideoCodecGeneric struct. by pbos@webrtc.org · 11 years ago
  23. e6882db Changing to using factory methods for some classes in NetEq by henrik.lundin@webrtc.org · 11 years ago
  24. 90c6679 Temporarily disabling some more audio processing tests. by aluebs@webrtc.org · 11 years ago
  25. 282a4d8 Fix MouseCursorMonitorMac to return correct hotspot position. by sergeyu@chromium.org · 11 years ago
  26. 1c27316 Removes the remaining uses of the list wrapper class and the list wrapper class. by henrike@webrtc.org · 11 years ago
  27. 8340614 WebRTCDemo: fix out-of-bounds array read. by fischman@webrtc.org · 11 years ago
  28. b95f445 Updated Webrtc version to 3.49 by elham@webrtc.org · 11 years ago
  29. 9c8f391 Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
  30. 4861e69 Revert "Activate ACM test for Android in modules_tests." (rev5364). by andresp@webrtc.org · 11 years ago
  31. 087bcfa Temporarily disabling audio processing tests. by aluebs@webrtc.org · 11 years ago
  32. 0782a57 Increasing simulation time for NetEqPerformanceTest by henrik.lundin@webrtc.org · 11 years ago
  33. 62ca1c6 Enables robust delay validation in AEC delay logging. by bjornv@webrtc.org · 11 years ago
  34. 457e101 Minor voice engine improvements around AGC. by andrew@webrtc.org · 11 years ago
  35. 842d07a Android: Fixes crash when exiting WebRTCDemo. by henrike@webrtc.org · 11 years ago
  36. 303c52b Activate ACM test for Android in modules_tests. by turaj@webrtc.org · 11 years ago
  37. df9f099 Permitting double start/stopping of streams. by pbos@webrtc.org · 11 years ago
  38. 1cc1166 Adding NetEq performance test to webrtc_perf_tests by henrik.lundin@webrtc.org · 11 years ago
  39. 15ba589 Delay Estimator: Adds unittests for robust validation. by bjornv@webrtc.org · 11 years ago
  40. 9fd4d83 Fixing lint errors in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  41. 401e046 Make code simpler on VCMEncodedCallback. by andresp@webrtc.org · 11 years ago
  42. 080eeee Isolate register post encode callback in video coding module to simplify code and critical sections. by andresp@webrtc.org · 11 years ago
  43. da08e77 Isolate debug recording from video sender into a thread safe small class. by andresp@webrtc.org · 11 years ago
  44. c92ae91 Add another test case for AST/TOF switching. by solenberg@webrtc.org · 11 years ago
  45. 11dddc0 Delay Estimator: Converts a constant into a configurable parameter. by bjornv@webrtc.org · 11 years ago
  46. d3c0b85 Init to 16 kHz in the fixed-point profile. by andrew@webrtc.org · 11 years ago
  47. c5eb922 Ensure capture_levels_ is sized correctly at init time. by andrew@webrtc.org · 11 years ago
  48. 859b462 Now printing less output from compare_videos.py. by phoglund@webrtc.org · 11 years ago
  49. 926e88a Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  50. f45e5b2 Remove outdated DestroyVideoSendStream comment. by pbos@webrtc.org · 11 years ago
  51. ca72300 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  52. 59a26d2 Delay Estimator: robust_validation should be stored over a reset by bjornv@webrtc.org · 11 years ago
  53. efdfe16 Add include guards to forward_error_correction_internal.h by braveyao@webrtc.org · 11 years ago
  54. 9a7cb02 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  55. a3ae4d1 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  56. 9456776 Android build: make it quiet on success and not overly noisy on failure. by fischman@webrtc.org · 11 years ago
  57. 748625e Fix the android clang bot for compiling with thread annotations. by andresp@webrtc.org · 11 years ago
  58. 7e4053c Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
  59. d3f0617 If the configured start bitrate is higher than the configures max by mflodman@webrtc.org · 11 years ago
  60. 4a185e9 Race condition in ViECapturer::RegisterObserver by sprang@webrtc.org · 11 years ago
  61. e83367b Update WebRTC to version 3.48 by tnakamura@webrtc.org · 11 years ago
  62. acc2e43 Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  63. e9c9d54 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. by andresp@webrtc.org · 11 years ago
  64. cd117d2 Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  65. 0d8474d Remove metrics_unittests by kjellander@webrtc.org · 11 years ago
  66. ef1f6c3 Remove media_file from VideoEngine dependencies. by pbos@webrtc.org · 11 years ago
  67. 2a4595a cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago
  68. b409d78 Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand(). by mflodman@webrtc.org · 11 years ago
  69. f22f12a Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss. by mflodman@webrtc.org · 11 years ago
  70. cc407fd Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  71. 32a0f69 audio_processing_unittest: unbreak clang compilation. by fischman@webrtc.org · 11 years ago
  72. 6b89cba JNI Audio: remove dead members. by fischman@webrtc.org · 11 years ago
  73. 32705ce Revert "Make MouseCursor mutable" by sergeyu@chromium.org · 11 years ago
  74. 0c7efa2 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  75. 4db3691 Stop transport in test SuspendBelowMinBitrate. by pbos@webrtc.org · 11 years ago
  76. 6f43aa7 Added method for getting default module state and protect agains a by mflodman@webrtc.org · 11 years ago
  77. 620d9e5 Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
  78. 4494516 Delete capturers after destroying streams in test. by pbos@webrtc.org · 11 years ago
  79. f3aed2f Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." by asapersson@webrtc.org · 11 years ago
  80. b06cca3 Simplification of histogram normalization in delay estimator. by bjornv@webrtc.org · 11 years ago
  81. 39139dc Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  82. 2ec8a62 Adds robust validation functionality to the delay estimator by bjornv@webrtc.org · 11 years ago
  83. beb643b Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver by sprang@webrtc.org · 11 years ago
  84. 0af1d21 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  85. ee867fa Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  86. b8dc2e2 Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. by turaj@webrtc.org · 11 years ago
  87. f34e39b Measure pacer queue size based on when packets are inserted rather than captured. by stefan@webrtc.org · 11 years ago
  88. b50a841 Fix jitter buffer delay estimate. by turaj@webrtc.org · 11 years ago
  89. 7f0519e Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  90. ab6ccbc Adding REMB to receive stream configuration, the send side will always by mflodman@webrtc.org · 11 years ago
  91. 9b3321f Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." by asapersson@webrtc.org · 11 years ago
  92. 9d9f138 Merge metrics_unittests into video_engine_tests. by pbos@webrtc.org · 11 years ago
  93. d1dd1d2 Move realtime tests to webrtc_perf_tests. by pbos@webrtc.org · 11 years ago
  94. 0e4512b Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  95. e4d538a Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
  96. e6dc4ff Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
  97. 3a4fc4b Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  98. b9cf1de ACM 2 compatibility with ACM 1. by turaj@webrtc.org · 11 years ago
  99. 9b3d2bf Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  100. cde78d6 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago