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gerrit-public.fairphone.software / platform / external / webrtc / 32c26eb90b16fc6ea25d5756fe605aa4ddd8c313
  1. 32c26eb Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback by henrike@webrtc.org · 11 years ago
  2. 4985927 Implement screen enumeration and individual screen capturing for Windows. by jiayl@webrtc.org · 11 years ago
  3. ead202b Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started. by henrike@webrtc.org · 11 years ago
  4. 2ce9a64 Talk: Removes deprecated example apps and moves the server apps to trunk/talk/examples. by henrike@webrtc.org · 11 years ago
  5. 0af1ffa Android, WebRTCDemo: fix issue where changing remote IP was not working properly. by henrike@webrtc.org · 11 years ago
  6. 4ffd9c7 Add full path to headers by aluebs@webrtc.org · 11 years ago
  7. 6a94734 Adds back set_sample_rate_hz() when Init is called in recordings. by bjornv@webrtc.org · 11 years ago
  8. ea9392d MIPS optimizations for NS audio processing module by andrew@webrtc.org · 11 years ago
  9. fb4e256 Fix crash in MouseCursor::CopyOf() by sergeyu@chromium.org · 11 years ago
  10. 8f35afa Exclude protoc objects from merge_libs.py. by andrew@webrtc.org · 11 years ago
  11. 4b26e2e Update libjingle to 59676287 by sergeyu@chromium.org · 11 years ago
  12. 7a2ca7c Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. by mallinath@webrtc.org · 11 years ago
  13. 8f19cb9 Revert 5387 "Re-enable webrtcvoice/videoengine unittests." by wu@webrtc.org · 11 years ago
  14. eda6823 Re-enable webrtcvoice/videoengine unittests. by wu@webrtc.org · 11 years ago
  15. 017b619 Extends the ScreenCapturer interface for individual display screen cast. by jiayl@webrtc.org · 11 years ago
  16. 03cfde2 Roll Chromium 238260 -> 243863 by wjia@webrtc.org · 11 years ago
  17. 8c5b27d Allow to skip turn by passing ts=false to apprtc. by andresp@webrtc.org · 11 years ago
  18. 39fcfd7 Remove empty VideoCodecGeneric struct. by pbos@webrtc.org · 11 years ago
  19. d9faa46 Changing to using factory methods for some classes in NetEq by henrik.lundin@webrtc.org · 11 years ago
  20. aebb1ad pRevert 5371 "Revert 5367 "Update talk to 59410372."" by henrika@webrtc.org · 11 years ago
  21. 4371d46 Temporarily disabling some more audio processing tests. by aluebs@webrtc.org · 11 years ago
  22. eb31b45 Fix MouseCursorMonitorMac to return correct hotspot position. by sergeyu@chromium.org · 11 years ago
  23. 3907c2e Removes the remaining uses of the list wrapper class and the list wrapper class. by henrike@webrtc.org · 11 years ago
  24. dde7aee WebRTCDemo: fix out-of-bounds array read. by fischman@webrtc.org · 11 years ago
  25. d7568a0 PeerConnection(java): Add OnRenegotiationNeeded support by fischman@webrtc.org · 11 years ago
  26. ad1863d Updated Webrtc version to 3.49 by elham@webrtc.org · 11 years ago
  27. 79cf3ac Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
  28. d0b436a Revert "Activate ACM test for Android in modules_tests." (rev5364). by andresp@webrtc.org · 11 years ago
  29. 44461fa Revert 5367 "Update talk to 59410372." by henrika@webrtc.org · 11 years ago
  30. 8bc4fcf Temporarily disabling audio processing tests. by aluebs@webrtc.org · 11 years ago
  31. 2c03bf1 Increasing simulation time for NetEqPerformanceTest by henrik.lundin@webrtc.org · 11 years ago
  32. bbd47fc Enables robust delay validation in AEC delay logging. by bjornv@webrtc.org · 11 years ago
  33. 0f3356e Update talk to 59410372. by mallinath@webrtc.org · 11 years ago
  34. 023cc5a Minor voice engine improvements around AGC. by andrew@webrtc.org · 11 years ago
  35. 573a1b4 Android: Fixes crash when exiting WebRTCDemo. by henrike@webrtc.org · 11 years ago
  36. 7cc64b3 Activate ACM test for Android in modules_tests. by turaj@webrtc.org · 11 years ago
  37. f777cf2 Permitting double start/stopping of streams. by pbos@webrtc.org · 11 years ago
  38. a366e81 Adding NetEq performance test to webrtc_perf_tests by henrik.lundin@webrtc.org · 11 years ago
  39. fa8d534 Delay Estimator: Adds unittests for robust validation. by bjornv@webrtc.org · 11 years ago
  40. 4625df3 Fix NaCl compilation by sergeyu@chromium.org · 11 years ago
  41. e7ce437 Fixing lint errors in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  42. c5aeb2a Make code simpler on VCMEncodedCallback. by andresp@webrtc.org · 11 years ago
  43. 1df9dc3 Isolate register post encode callback in video coding module to simplify code and critical sections. by andresp@webrtc.org · 11 years ago
  44. bb0de3c Updated Demos so they work on FF, changed the third argument in CreateOffer to null as it doesnot really require sdpConstraints. by vikasmarwaha@webrtc.org · 11 years ago
  45. 4177615 PeerConnection(java): replace ScopedLocalRef with ScopedLocalRefFrame and fix a local reference leak in OnMessage. by fischman@webrtc.org · 11 years ago
  46. 1794693 AppRTCDemo(android): close() the throw-away DataChannel. by fischman@webrtc.org · 11 years ago
  47. b08a12d Isolate debug recording from video sender into a thread safe small class. by andresp@webrtc.org · 11 years ago
  48. ab24051 Add another test case for AST/TOF switching. by solenberg@webrtc.org · 11 years ago
  49. bccd53d Delay Estimator: Converts a constant into a configurable parameter. by bjornv@webrtc.org · 11 years ago
  50. e00265e Fix a compile error on Android on sctpdataengine.cc. by wu@webrtc.org · 11 years ago
  51. d335094 Init to 16 kHz in the fixed-point profile. by andrew@webrtc.org · 11 years ago
  52. b6541ca Ensure capture_levels_ is sized correctly at init time. by andrew@webrtc.org · 11 years ago
  53. cf9d364 Now printing less output from compare_videos.py. by phoglund@webrtc.org · 11 years ago
  54. 60730cf Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  55. 39669c5 Remove outdated DestroyVideoSendStream comment. by pbos@webrtc.org · 11 years ago
  56. ccd4284 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  57. 0b7d8e6 AppRTC: Alert the user to failure to acquire TURN server. by fischman@webrtc.org · 11 years ago
  58. acc05ac Roll libvpx 232686:241571 by marpan@webrtc.org · 11 years ago
  59. a9bdee6 Add Christophe Dumez to AUTHORS. by andrew@webrtc.org · 11 years ago
  60. 7bdaf83 Updated PeerConnection samples so they run on FF. by vikasmarwaha@webrtc.org · 11 years ago
  61. f6d6ed0 Update talk to 59039880. by wu@webrtc.org · 11 years ago
  62. e667234 libyuv r949 includes changes to allow any width, mainly relating to fixed point math overflows. by fbarchard@google.com · 11 years ago
  63. a89d17d Delay Estimator: robust_validation should be stored over a reset by bjornv@webrtc.org · 11 years ago
  64. 2240763 libyuv r930 for RGB24ToUV_NEON improved color accuracy to avoid red tint, and use malloc with variable sized row buffers to avoid stack overflow and relax width restrictions. Previously was limited to 4k on x86 and 1080p on arm. In practice the new limitation is 32767 pixels wide. by fbarchard@google.com · 11 years ago
  65. 2fb72cf Add include guards to forward_error_correction_internal.h by braveyao@webrtc.org · 11 years ago
  66. 0062a6d Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  67. a7cfa67 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  68. 000dde9 Android build: make it quiet on success and not overly noisy on failure. by fischman@webrtc.org · 11 years ago
  69. a63fc87 Fix JS error in adapter.js for FF for the case when ?transport=xxx is missing in TURN url. by vikasmarwaha@webrtc.org · 11 years ago
  70. f6acf98 Fix the android clang bot for compiling with thread annotations. by andresp@webrtc.org · 11 years ago
  71. cf2b3ac Update Android trybots in the default try job list. by kjellander@webrtc.org · 11 years ago
  72. 7fb75ec Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
  73. 6031001 If the configured start bitrate is higher than the configures max by mflodman@webrtc.org · 11 years ago
  74. 8dbca8d Race condition in ViECapturer::RegisterObserver by sprang@webrtc.org · 11 years ago
  75. a463d73 Update WebRTC to version 3.48 by tnakamura@webrtc.org · 11 years ago
  76. 54ae4ff Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  77. e682aa5 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. by andresp@webrtc.org · 11 years ago
  78. faada6e Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  79. 8f99a18 Port scale and compare functions to pepper_33 and mips. by fbarchard@google.com · 11 years ago
  80. 5fe2d65 Remove metrics_unittests by kjellander@webrtc.org · 11 years ago
  81. 8a54417 Remove media_file from VideoEngine dependencies. by pbos@webrtc.org · 11 years ago
  82. b429e51 cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago
  83. bcd124c 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
  84. 1fa41be Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss. by mflodman@webrtc.org · 11 years ago
  85. 8ae7256 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  86. f8be8df audio_processing_unittest: unbreak clang compilation. by fischman@webrtc.org · 11 years ago
  87. 179908c JNI Audio: remove dead members. by fischman@webrtc.org · 11 years ago
  88. e4c9272 Revert "Make MouseCursor mutable" by sergeyu@chromium.org · 11 years ago
  89. 8fd1d26 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  90. af320fd The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation. by fischman@webrtc.org · 11 years ago
  91. 50f7b2d roll libyuv to r915 for webview jpeg build fix and NaCL pepper_33 initial support. by fbarchard@google.com · 11 years ago
  92. 052fa62 Stop transport in test SuspendBelowMinBitrate. by pbos@webrtc.org · 11 years ago
  93. e6b871b Added method for getting default module state and protect agains a by mflodman@webrtc.org · 11 years ago
  94. 9df6674 Scale down by 4x with box filter. Fix for 1 pixel wide bilinear filter. Fix for I420ToARGB overread on V plane that causes valgrind fail. by fbarchard@google.com · 11 years ago
  95. eb7b7bc Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
  96. 5b3c67e objc/README: Remove outdated advice about target_os. by fischman@webrtc.org · 11 years ago
  97. 919f87f Delete capturers after destroying streams in test. by pbos@webrtc.org · 11 years ago
  98. e7b1e11 Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." by asapersson@webrtc.org · 11 years ago
  99. 1e7d612 Simplification of histogram normalization in delay estimator. by bjornv@webrtc.org · 11 years ago
  100. 5ab7567 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago