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gerrit-public.fairphone.software / platform / external / webrtc / 37ca765650adb36d3dd0ab84a507e660f44766c6
  1. 37ca765 Add fir_filter to common_audio by aluebs@webrtc.org · 11 years ago
  2. af839b2 Add AIMD option to BWE API. by stefan@webrtc.org · 11 years ago
  3. ba5a6c3 ACM2/NetEq4 did not decode Opus in stereo by tina.legrand@webrtc.org · 11 years ago
  4. 152208a (Auto)update libjingle 63547048-> 63560528 by henrike@webrtc.org · 11 years ago
  5. 07bc734 Refactor in BitrateController module. by andresp@webrtc.org · 11 years ago
  6. be7e26d (Auto)update libjingle 63503990-> 63547048 by henrike@webrtc.org · 11 years ago
  7. 6f9c483 Fixing crash in video_render_tests in release mode. by henrikg@webrtc.org · 11 years ago
  8. 16b75c2 Remove locks in SendSideBandwidthEstimation since those are only accessed while owning locks in by andresp@webrtc.org · 11 years ago
  9. b28bfa7 Adding FEC support in NetEq 4. by minyue@webrtc.org · 11 years ago
  10. 0e65fda Fix "unreachable code" warnings (MSVC warning 4702) in webrtc. by pbos@webrtc.org · 11 years ago
  11. 0209e56 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 11 years ago
  12. 9c2bd2b Roll libvpx 254609:258445. by fischman@webrtc.org · 11 years ago
  13. 0c6f0f9 Revert 5737 "Add system wrapper dependency to libjingle targets." by mallinath@webrtc.org · 11 years ago
  14. 5e83c65 (Auto)update libjingle 63493960-> 63503990 by henrike@webrtc.org · 11 years ago
  15. 062e6e5 ARGBScale fix for bilinear down sampling overread when source size is odd. by fbarchard@google.com · 11 years ago
  16. a8ebdb7 Revert "(Auto)update libjingle 63363208-> 63493960" (r5740) by henrike@webrtc.org · 11 years ago
  17. 5f768ad (Auto)update libjingle 63363208-> 63493960 by henrike@webrtc.org · 11 years ago
  18. 1faef7d Use codec width/height as the encoded_image width/height. by wu@webrtc.org · 11 years ago
  19. 3ab57c5 Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets by henrik.lundin@webrtc.org · 11 years ago
  20. 979f1f8 Add system wrapper dependency to libjingle targets. by mallinath@webrtc.org · 11 years ago
  21. 8a8c3ef Add ability to configure cpu overuse options via an API. by asapersson@webrtc.org · 11 years ago
  22. d669299 Prevent playout delay wrap-around in VoiceEngine by henrik.lundin@webrtc.org · 11 years ago
  23. 800b8db Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered. by henrika@webrtc.org · 11 years ago
  24. c148079 Extend perf tests to perform rampup on single stream. by andresp@webrtc.org · 11 years ago
  25. c8ac17c Adjust the captured window rect when the window is maximized. by jiayl@webrtc.org · 11 years ago
  26. ffe2620 (Auto)update libjingle 63352036-> 63363208 by henrike@webrtc.org · 11 years ago
  27. 1639522 Properly account for retransmitted packets when not using the pacer. by stefan@webrtc.org · 11 years ago
  28. 7c6ff2d Fixes RTX related bugs. by stefan@webrtc.org · 11 years ago
  29. 9af85c4 Disabling SendsSetSimulcastSsrcs. by pbos@webrtc.org · 11 years ago
  30. 1e98a15 Revert "Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets" by henrik.lundin@webrtc.org · 11 years ago
  31. e5be877 Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets by henrik.lundin@webrtc.org · 11 years ago
  32. add4073 Disable flaky CanSwitchToUseAllSsrcs. by pbos@webrtc.org · 11 years ago
  33. 709e297 Simplify pacer interface. by pbos@webrtc.org · 11 years ago
  34. f577ae9 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 11 years ago
  35. ac4b87c Fix a deadlock in ViEEncoder::DeliverFrame. by wuchengli@chromium.org · 11 years ago
  36. 8b61011 (Auto)update libjingle 63293120-> 63352036 by henrike@webrtc.org · 11 years ago
  37. 08e2dd8 Exclude WebRtcVideoMediaChannelTest.AddRemoveCapturerMultipleSources for Valgrind on Mac by kjellander@webrtc.org · 11 years ago
  38. 886c94f Adds a method to WindowCapturer to bring a window to the front. by jiayl@webrtc.org · 11 years ago
  39. e9793ab (Auto)update libjingle 63111035-> 63293120 by henrike@webrtc.org · 11 years ago
  40. dcc301b Adding thread annotations to NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  41. 3f655aa Add #include <cstdlib> for std::abs. by pbos@webrtc.org · 11 years ago
  42. 944cbeb Resolves TSan v2 warnings in voe_auto_test. by henrika@webrtc.org · 11 years ago
  43. ed8b281 Re-comitting r5711: "Fixing a flaky test in video_engine_tests" by henrik.lundin@webrtc.org · 11 years ago
  44. 12499ff Revert 5711 "Fixing a flaky test in video_engine_tests" by turaj@webrtc.org · 11 years ago
  45. d0f0c76 Fixing a flaky test in video_engine_tests by henrik.lundin@webrtc.org · 11 years ago
  46. 4e69f78 Small refactor on send_side_bandwidth_estimation. by andresp@webrtc.org · 11 years ago
  47. ccb33a6 turn-prober: enable running headlessly and only emit output on error. by fischman@webrtc.org · 11 years ago
  48. a714eaf Refactor rampup tests: by andresp@webrtc.org · 11 years ago
  49. 44eb87e Tool to establish a loopback call via apprtc turn server. by andresp@webrtc.org · 11 years ago
  50. 26caf0e Suppresses/disables tsan/memcheck issues due to sync of 63111035. by henrike@webrtc.org · 11 years ago
  51. 18e5911 (Auto)update libjingle 63089643-> 63111035 by henrike@webrtc.org · 11 years ago
  52. cf6f46d References to includes in third_party should be relative, not absolute. by sprang@webrtc.org · 11 years ago
  53. 4375e1a Add support for YUV4MPEG file reading to tools files. (Minor fix). by mcasas@webrtc.org · 11 years ago
  54. 6e2d012 Add support for YUV4MPEG file reading to tools files. by mcasas@webrtc.org · 11 years ago
  55. 24779fe Fix a bug where network freeze during CNG causes delay by henrik.lundin@webrtc.org · 11 years ago
  56. 367000f Remove legacy weirdness in Merge::Downsample by henrik.lundin@webrtc.org · 11 years ago
  57. f45a550 (Auto)update libjingle 63019975-> 63089643 by henrike@webrtc.org · 11 years ago
  58. 54464e6 Stopping network threads before tearing down test by henrik.lundin@webrtc.org · 11 years ago
  59. 5a320fb Race condition in RTPSender by sprang@webrtc.org · 11 years ago
  60. 4168901 Add max delay to trace based filters and enhances drop tail queues with delay statistics. by stefan@webrtc.org · 11 years ago
  61. b10363f Re-landing "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  62. 3349ae0 Implement minimum transmit bitrate. by pbos@webrtc.org · 11 years ago
  63. 6ea4f63 Enable all RampUpTest.UpDownUp* tests by henrik.lundin@webrtc.org · 11 years ago
  64. b5f3029 Replace labs with std::abs. by pbos@webrtc.org · 11 years ago
  65. 827faae Fixing incorrect memset. by mallinath@webrtc.org · 11 years ago
  66. dd5d804 Disable all protobuf dependent targets when enable_protobuf=0. by andrew@webrtc.org · 11 years ago
  67. c7bec84 (Auto)update libjingle 62948689-> 63019975 by henrike@webrtc.org · 11 years ago
  68. 9269ba1 (Git)ignore all of /net. Works around issue: gclient sync, git clean -df, gclient runhooks -> failure (regression in r4466). by henrike@webrtc.org · 11 years ago
  69. c2313fb Enable VS2013 for Windows compilation by default. by kjellander@webrtc.org · 11 years ago
  70. 95153cc Remove platform-specific code from new-API tests. by pbos@webrtc.org · 11 years ago
  71. ca8cb95 Implement a test for an old corner-case in NetEq by henrik.lundin@webrtc.org · 11 years ago
  72. 04ea232 Developing NetEqImpl unit tests by henrik.lundin@webrtc.org · 11 years ago
  73. 10bd88e (Auto)update libjingle 62871616-> 62948689 by henrike@webrtc.org · 11 years ago
  74. 21df847 Disable TestOpusNewACM on Android. by andrew@webrtc.org · 11 years ago
  75. be39470 Revert "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  76. 12acd6e Reorder includes in audio_processing_impl_unittest. by andrew@webrtc.org · 11 years ago
  77. cdefc91 Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now. by braveyao@webrtc.org · 11 years ago
  78. 1598b80 Routing SuspendChange to VideoSendStream::Stats by henrik.lundin@webrtc.org · 11 years ago
  79. c3d13d3 Classes and tests for audio an classifier. The class can be used to classify whether a frame of audio contains speech or music. The classifier uses the music/speech classifier in Opus. by jan.skoglund@webrtc.org · 11 years ago
  80. a8b9737 Add tests and modify tools for new float deinterleaved interface. by andrew@webrtc.org · 11 years ago
  81. 3046b84 Adding new data files for audio classifier unit testing on Android try bots by jan.skoglund@webrtc.org · 11 years ago
  82. d3d6bce (Auto)update libjingle 62865357-> 62871616 by henrike@webrtc.org · 11 years ago
  83. d32797f Add a float interface to PushSincResampler. by andrew@webrtc.org · 11 years ago
  84. bc206ea iOS video_render: omit no-op setNeedsDisplay by fischman@webrtc.org · 11 years ago
  85. f792d17 AppRTCDemo(iOS): video support; part 1 of 2: webrtc/. by fischman@webrtc.org · 11 years ago
  86. 0537634 (Auto)update libjingle 62713454-> 62865357 by henrike@webrtc.org · 11 years ago
  87. 4a47be0 Disable CallTest.ReceivesAndRetransmitsNack for TSan by kjellander@webrtc.org · 11 years ago
  88. 36b6221 Adding a link to issue by henrik.lundin@webrtc.org · 11 years ago
  89. 6b0cbcb Roll chromium_revision 249215:255773 by kjellander@webrtc.org · 11 years ago
  90. 9b5f4d8 Fix build breakage introduce with r5665. by stefan@webrtc.org · 11 years ago
  91. f9e7c9d Add option to bwe_rtp_to_text to output arrival times only in nanoseconds. by stefan@webrtc.org · 11 years ago
  92. a01daf0 RTCPeerConnectionTest(objc): deflake by ignoring ICECompleted. by fischman@webrtc.org · 11 years ago
  93. 13320ea PeerConnectionTest(objc): expect ICE Completed state post 61460797-p10 by fischman@webrtc.org · 11 years ago
  94. 7811469 Roll libvpx 251850:254609 by marpan@webrtc.org · 11 years ago
  95. 11aab0e Populate VoiceReceiverInfo::delay_estimate_ms, jitter_buffer_ms, and jitter_buffer_preferred_ms to getStats. by jiayl@webrtc.org · 11 years ago
  96. 64e0405 Avoid crash in ViEEncoder::DeRegisterExternalEncoder(). by fischman@webrtc.org · 11 years ago
  97. cc08e3f Moves WEBRTC_POSIX define from header file to gyp-settings. by henrike@webrtc.org · 11 years ago
  98. 3ecc162 Remove std:: prefixes from C functions in webrtc/. by pbos@webrtc.org · 11 years ago
  99. 371243d Remove std:: prefixes from C functions in talk/. by pbos@webrtc.org · 11 years ago
  100. 46509c8 adding FEC support to WebRTC Opus wrapper and tests. by minyue@webrtc.org · 11 years ago