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webrtc / src / 1bcfce5ff2cda67aeaf796cbb4779afaace2847e / webrtc
  1. 1bcfce5 Deactivated the intelligibility enhancement functionality by default by peah · 9 years ago
  2. 7d67e45 Revert of Added functionality for specifying the initial signal level to use for the gain estimation in the l… (patchset #8 id:160001 of https://codereview.webrtc.org/2254973003/ ) by peah · 9 years ago
  3. fe1d191 MB: Flip Linux bots to GN by default. by kjellander · 9 years ago
  4. a897f26 AbsoluteSendTime rtp header extension publish MsTo24Bit conversion by danilchap · 9 years ago
  5. 84bc985 Removed virtual from several methods in DecoderDatabase to minimize by ossu · 9 years ago
  6. 57fec1d This CL adds functionality in the level controller to by peah · 9 years ago
  7. 1e8ed4a Replace calls to assert() with RTC_DCHECK_*() in .c code by kwiberg · 9 years ago
  8. 073ece4 Skip unit test if GYP_DEFINES="rtc_use_h264=1" is not set. by johan · 9 years ago
  9. c766804 NetEq: Update CNG code to accommodate 48 kHz sample rate by henrik.lundin · 9 years ago
  10. 5085b0c Adding AecDump functionality to AppRTCDemo for iOS by peah · 9 years ago
  11. bad33bf Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 9 years ago
  12. 91b03b0 Revert of Delete method cricket::VideoFrame::Copy. (patchset #3 id:210001 of https://codereview.webrtc.org/2275243002/ ) by philipel · 9 years ago
  13. f715f98 Reland of Delete method cricket::VideoFrame::Copy. (patchset #1 id:1 of https://codereview.webrtc.org/2087923004/ ) by nisse · 9 years ago
  14. 6bf62f7 Avoids java.lang.NullPointerException in WebRtcAudioRecord by henrika · 9 years ago
  15. 4805231 Moved format_macros.h from rtc_base to rtc_base_approved. by ivoc · 9 years ago
  16. 4bc4d27 GN: Fix Windows Clang errors by ehmaldonado · 9 years ago
  17. 3f746ea Fix error when accumulating floats in an int. by maxmorin · 9 years ago
  18. e29352b Refactor certificate stats collection, added SSLCertificateStats. by hbos · 9 years ago
  19. 2ab012c Implement CVO for iOS capturer by magjed · 9 years ago
  20. 19319a3 Add missing "//build/config/sanitizers:deps" to executable targets. by ehmaldonado · 9 years ago
  21. 00e45bb Move InsertZeroColumns and CopyColumn to ::internal. by brandtr · 9 years ago
  22. 7a770e0 GN build rules for four audio processing test executables by kwiberg · 9 years ago
  23. 8a6a600 Make neteq_rtpplay parse RTP header extensions by henrik.lundin · 9 years ago
  24. 5f09980 Removed inline definitions and added destructors to fix chromium-style. by aleloi · 9 years ago
  25. 549d80b NetEq: only update current_rtp_payload_type_ when validated by henrik.lundin · 9 years ago
  26. fe8f489 Fix setting the MTU for SCTP. by deadbeef · 9 years ago
  27. b60a819 Fixing inconsistency with behavior of `ClearGettingPorts`. by deadbeef · 9 years ago
  28. 824f586 Fixing segfault caused by TurnServer. by deadbeef · 9 years ago
  29. 1d7a637 Fixing off-by-one error with max SCTP id. by Taylor Brandstetter · 9 years ago
  30. fcada90 Fixing timestamp comparison assert. by deadbeef · 9 years ago
  31. 36a06a9 Increase QP threshold for H.264 encoder QP based scaling. by glaznev · 9 years ago
  32. 1184025 Restart capture session if needed on active. by tkchin · 9 years ago
  33. 5fac3f0 NetEq: Don't check sample rate and frame size upon error by henrik.lundin · 9 years ago
  34. d1a10a0 Make FakeDecodeFromFile handle codec-internal CNG by henrik.lundin · 9 years ago
  35. f02207d MB: Flip Mac bots to GN by default. by kjellander · 9 years ago
  36. b0b0edb Roll chromium_revision e3860bd297..938114be1e (412289:414059) by ehmaldonado · 9 years ago
  37. 28a0ffd GN: Synchronize resources between Android and iOS. by kjellander · 9 years ago
  38. 2ec45b9 Make dependency of audio_device of ApplicationServices explicit. by maxmorin · 9 years ago
  39. 4e7e8d7 Now probe for x3 and x6 of the initial start bitrate and allow for faster receive bitrates when calculating probing estimates. by philipel · 9 years ago
  40. 2c670db Added GN target for webrtc_opus_fec_test. by ivoc · 9 years ago
  41. 7a0ff2f Disable examples for GYP Android bots. by ehmaldonado · 9 years ago
  42. 98468bb Revert of GN build rules for four audio processing test executables (patchset #3 id:40001 of https://codereview.webrtc.org/2267403003/ ) by sakal · 9 years ago
  43. 538b560 GN build rules for four audio processing test executables by kwiberg · 9 years ago
  44. 0561bdf Only use payload size within the know send/receive interval for probing calculations. by philipel · 9 years ago
  45. 619a211 iLBC: Handle a case of bad input data by kwiberg · 9 years ago
  46. 0aa9d18 Set send side bitrate estimate on successful probing attempt. by philipel · 9 years ago
  47. f944c35 GN: Add resources for webrtc_perf_tests on Android by kjellander · 9 years ago
  48. e51b41a Added GN target for isac_test. by ivoc · 9 years ago
  49. 5d167d6 Removals and renamings in the new audio mixer. by aleloi · 9 years ago
  50. 76f91cd Add ThreadChecker to the TimestampAligner class. by nisse · 9 years ago
  51. 665d181 Increased column width for python tool rtp_analyzer.py. by aleloi · 9 years ago
  52. 30be5d7 Updated mixer unittests and fixed a related bug in the new mixer. by aleloi · 9 years ago
  53. 615d301 RTCStats and RTCStatsReport added (webrtc/stats). by hbos · 9 years ago
  54. 616df1e Added a level indicator to new mixer. by aleloi · 9 years ago
  55. 1f77905 Remove outdated symlink by kthelgason · 9 years ago
  56. a53fa0a Fix AppRTC Android Demo GN build when is_component_build=true. by sakal · 9 years ago
  57. 4c8adb1 MB: Flip Android bots to GN by default. by kjellander · 9 years ago
  58. b246a29 Define a protobuf format for representing plots. Add code to convert the C-representation generated by the RtcEventLog analysis tool, to the new protobuf format. by terelius · 9 years ago
  59. 6addf49 Adds function for computing moving average to visualization tool. by terelius · 9 years ago
  60. 5048f57 Add logs and small change in BasicPortAllocator. by Honghai Zhang · 9 years ago
  61. f99a9de ProbingEstimator: Erase history based on time threshold by Irfan Sheriff · 9 years ago
  62. 185ba29 Extract library from the RTCEventLog visualizer by skvlad · 9 years ago
  63. 5bed20f Do not update stats for WebRTC.Call.EstimatedSendBitrateInKbps if we are not sending video. by Per · 9 years ago
  64. b37c45c GN: Add libjingle_peerconnection_java to peerconnection_unittests. by kjellander · 9 years ago
  65. a246cfb Don't include RTP headers in send-side BWE. by Stefan Holmer · 9 years ago
  66. 9a11784 Migrated GN target :g722_test by aleloi · 9 years ago
  67. 16f55a1 Migrated GN target :g711_test by aleloi · 9 years ago
  68. 649a21a Disable RampUpTest.UpDownUpThreeStreams. by philipel · 9 years ago
  69. 2e48646 RTC_CHECK and RTC_DCHECK macros for C by kwiberg · 9 years ago
  70. 7924697 Refactor WebRtcVideoCapturer. by nisse · 9 years ago
  71. d8dd190 GN: Fix test_support_unittests and MIPS compile issue. by kjellander · 9 years ago
  72. 84c03ba Add rtc_media as a direct dependency of rtc_media_unittests. by nisse · 9 years ago
  73. 0d1ad32 Add histogram for percentage of incoming frames that are limited in resolution due to cpu: by asapersson · 9 years ago
  74. 14cf12b Fixing TSan data race warning in video end-to-end tests. by Taylor Brandstetter · 9 years ago
  75. 23d947d Some cleanup in BaseChannel RTCP mux code. by deadbeef · 9 years ago
  76. b3f1c5d Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine by henrik.lundin · 9 years ago
  77. e131ea5 Adding deadbeef and honghaiz as owners of webrtc/pc. by deadbeef · 9 years ago
  78. 72a5645 Removed the deactivation of the level controller when there is a built-in AGC available by peah · 9 years ago
  79. 8c16520 Method to parse event log directly from a string. by terelius · 9 years ago
  80. 6c46eaa Add gtest as a dependency for neteq_quality_test_support. by ehmaldonado · 9 years ago
  81. d48717b Fix issue where the number of packets reported in tests/simulations sometimes are negative. by stefan · 9 years ago
  82. 4ec01d9 Fix trivial lint errors in FileRecorder and FilePlayer by kwiberg · 9 years ago
  83. 853ecb2 Style cleanup in UpdateTmmbr: by danilchap · 9 years ago
  84. 7f82fc9 WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left shift of 1 by 31 overflows) by kwiberg · 9 years ago
  85. 642e3bc [rtcp] TransportFeedback adjusted to match other rtcp packets: by danilchap · 9 years ago
  86. 4981051 [Reland] Cleanup of the AudioDeviceBuffer class. by henrika · 9 years ago
  87. 83d79cd Revert of Add pps id and sps id parsing to the h.264 depacketizer. (patchset #5 id:80001 of https://codereview.webrtc.org/2238253002/ ) by kjellander · 9 years ago
  88. 4381700 WebRtcVideoFrame constructor without transport_frame_id. by nisse · 9 years ago
  89. e5b4141 Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData by danilchap · 9 years ago
  90. ff101d6 iOS: add PlistBuddy location to path to avoid build errors by vopatop.skam · 9 years ago
  91. 4905f06 Disable the software noise suppressor on iOS devices as that by peah · 9 years ago
  92. abcc3de Add pps id and sps id parsing to the h.264 depacketizer. by stefan · 9 years ago
  93. 86ccd7b Revert of Add field_trial_default dependency to libjingle_peerconnection (patchset #3 id:40001 of https://codereview.webrtc.org/2120673004/ ) by sakal · 9 years ago
  94. a7a01df Add field_trial_default dependency to libjingle_peerconnection by arlolra · 9 years ago
  95. 8177452 iOS H264VideoToolBoxEncoder: Stop scaling native CVPixelBuffers by magjed · 9 years ago
  96. d7a89db Revert of Cleanup of the AudioDeviceBuffer class (patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ ) by henrika · 9 years ago
  97. cf327b4 Cleanup of the AudioDeviceBuffer class. by henrika · 9 years ago
  98. da161d7 Reformat rtcp_receiver git cl format --full by danilchap · 9 years ago
  99. 861da3c Refactor neteq_test_support. by ehmaldonado · 9 years ago
  100. 294fb05 Add a timeout for starting the camera on CameraCapturer. by sakal · 9 years ago