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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 0e9c399746f45ceaf46f12b11ba93c09cca0c2bb
  1. 3f39c00 ACM2 integration with NetEq 4. by turaj@webrtc.org · 11 years ago
  2. a3351c4 Adding Ami to the video renderer and capturer modules. by mallinath@webrtc.org · 11 years ago
  3. bc375b5 The video render module for iOS. by fischman@webrtc.org · 11 years ago
  4. 4489c51 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
  5. 66bfae2 Make PCM16 available in Chromium builds. by andrew@webrtc.org · 11 years ago
  6. 5e3379e Make the destructor of AudioCodingModule public. by andrew@webrtc.org · 11 years ago
  7. 0fd885e Fix unsigned/signed comparison error due to r4729. by andrew@webrtc.org · 11 years ago
  8. f5556f2 Reduce frequency of high audio delay warning logs. by andrew@webrtc.org · 11 years ago
  9. 9fea95a Removes function that is not used anywhere but somehow still causing library load issues on Android Release build. by henrike@webrtc.org · 11 years ago
  10. bfad17e Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  11. 8fdce8e OpenSl: fixes crashes externally reported in issue 2361 and 2362. by henrike@webrtc.org · 11 years ago
  12. 66dbbd9 Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. by turaj@webrtc.org · 11 years ago
  13. f2982c9 Remove FrameForStorage:Follow up on r4688 by mikhal@webrtc.org · 11 years ago
  14. 990c5e3 Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  15. f0adedc Reset jitter buffer and timing if frames are getting too much delay. by stefan@webrtc.org · 11 years ago
  16. 054bc03 Remove repeated conditions key. by andrew@webrtc.org · 11 years ago
  17. f46fff6 OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
  18. dadb2a1 Fix format string in video_quality_analysis.cc. by pbos@webrtc.org · 11 years ago
  19. 7b30ce3 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 11 years ago
  20. eb2d9dd Test that VideoSendStream responds to NACK. by pbos@webrtc.org · 11 years ago
  21. 3524ade Convert printing in video quality tests to Chromium's perf format. by kjellander@webrtc.org · 11 years ago
  22. b676ac7 Lock RTPSender statistics. by pbos@webrtc.org · 11 years ago
  23. fa996f2 Split up EngineTests and RampupTests. by pbos@webrtc.org · 11 years ago
  24. 5cf83f4 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
  25. 0920142 Updated WebRTC version to 3.41 by elham@webrtc.org · 11 years ago
  26. 6b4698e Lock use of _packetRequestCallback in VCM. by pbos@webrtc.org · 11 years ago
  27. 0245bee Remove include_dirs from video_engine_core.gypi. by pbos@webrtc.org · 11 years ago
  28. 4e7777b Break out RTCPSender dependency on ModuleRtpRtcpImpl. by pbos@webrtc.org · 11 years ago
  29. bf6d572 Rename VideoCall to Call. by pbos@webrtc.org · 11 years ago
  30. 6a79c9f Re-enable tests for Remote Bitrate Estimator by solenberg@webrtc.org · 11 years ago
  31. 618a0ec ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 11 years ago
  32. e97b69f Handle empty RTP video packets agnostic to codec. by pbos@webrtc.org · 11 years ago
  33. 11a8868 Reduce cost of PushSincResampler::Resample(). by andrew@webrtc.org · 11 years ago
  34. ca20f3d Clamp camera id to legal values. by fischman@webrtc.org · 11 years ago
  35. 7dc1790 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 11 years ago
  36. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  37. 4014302 Add temporal layer factory. by andresp@webrtc.org · 11 years ago
  38. 31a8ce7 Removing FrameForStorage by mikhal@webrtc.org · 11 years ago
  39. e41c6b2 Make unittest log printouts opt-in with a --logs flag. by andrew@webrtc.org · 11 years ago
  40. f2ef20c Pre-multiply images for MouseCursorShape. by alexeypa@chromium.org · 11 years ago
  41. 6f458ed Recognize armv7 target_arch for ios support in webrtc common.gyp by fischman@webrtc.org · 11 years ago
  42. 06eaa54 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
  43. 0020858 Remove send and receive streams when destroyed. by pbos@webrtc.org · 11 years ago
  44. 0e2cb29 Add clockdrift to RtpGenerator by henrik.lundin@webrtc.org · 11 years ago
  45. 4998966 Allow unknown flags in test_main.cc. by pbos@webrtc.org · 11 years ago
  46. 787364c NetEq4: Small change to reduce allocs in AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
  47. c77dcb0 Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter. by mflodman@webrtc.org · 11 years ago
  48. 7a21a64 Clean capture timestamp code. by andresp@webrtc.org · 11 years ago
  49. 1cd055c Disable EngineTest.ReceivesPliAndRecoversWithNack. by mflodman@webrtc.org · 11 years ago
  50. 00c95bf Protecting Bitrate to avoid data race found by tsan. by mflodman@webrtc.org · 11 years ago
  51. 0f62690 Revert 4671 "Enable SetInitialPlayoutDelay on Android." by mflodman@webrtc.org · 11 years ago
  52. 0fe8944 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  53. 0987043 Don't force cont' when enabling kWithErrors by mikhal@webrtc.org · 11 years ago
  54. 9787291 Removing some TODO's from libyuv by mikhal@webrtc.org · 11 years ago
  55. 4dae3c6 Removing non decodable count from session info: This value isn't used, and therfore can be removed. This is a step towards the refactor of the session info to use maps. by mikhal@webrtc.org · 11 years ago
  56. 1cc93a2 Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule. by mflodman@webrtc.org · 11 years ago
  57. 9e70940 Add FakeEncoder to VideoSendStream tests. by pbos@webrtc.org · 11 years ago
  58. 1c9d3fe Correcting two nits in InputAudioFile by henrik.lundin@webrtc.org · 11 years ago
  59. 324a016 Changed method name. by mflodman@webrtc.org · 11 years ago
  60. 94ef274 Renamed method. by mflodman@webrtc.org · 11 years ago
  61. 710d2e1 Function name change. by mflodman@webrtc.org · 11 years ago
  62. a594db2 Fixing capture frame race in ViECapturer. by mflodman@webrtc.org · 11 years ago
  63. 88a2327 Disable all LS_VERBOSE logging in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  64. 797eb64 NetEq4: Make the algorithm buffer a member variable by henrik.lundin@webrtc.org · 11 years ago
  65. ce9de71 Overuse detection based on capture-input jitter. by pbos@webrtc.org · 11 years ago
  66. 42baf15 Removing JPEG as it is not used. by mikhal@webrtc.org · 11 years ago
  67. 1aa0938 Zero comfort noise for stereo insted of assertion. by turaj@webrtc.org · 11 years ago
  68. b22fe00 Reorder and add critical section to the public method NetEqImpl::PacketBufferStatistics(). by turaj@webrtc.org · 11 years ago
  69. d808778 Fix typo in InvertedDesktopFrame by sergeyu@chromium.org · 11 years ago
  70. 27103e5 Fix fileutils.cc for tests running under Win memory tools. by kjellander@webrtc.org · 11 years ago
  71. e0e5a6a Disabling CondVarTest for TSan v2 (take 2) by kjellander@webrtc.org · 11 years ago
  72. c766a74 Update SSRC in RtpRtcp for audio channel so that it can have NTP values for further AV sync. by dwkang@webrtc.org · 11 years ago
  73. 42758b3 update neteq 4 to facilitate NACK by minyue@webrtc.org · 11 years ago
  74. e879919 Fix metrics_unittests on Android. by kjellander@webrtc.org · 11 years ago
  75. 8c6633c Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 11 years ago
  76. 584b688 Re-organizing ACM tests by tina.legrand@webrtc.org · 11 years ago
  77. 9b7bdee Revert r4562 by elham@webrtc.org · 11 years ago
  78. fd8cc12 Fix image flipping for OpenGL-based screen capturer on Mac. by sergeyu@chromium.org · 11 years ago
  79. e8acc7c Enable ObjC build by default and reenable 64-bit mac libjingle build by fischman@webrtc.org · 11 years ago
  80. 6203090 Updated WebRTC version to 3.40 by elham@webrtc.org · 11 years ago
  81. d9416e6 VCM:Accounting for bounds when inserting packets. We currently receive indicators to the first and last packets of the frame, but not have any sanity to verify that all packets are indeed within the bounds (when available). This cl attempts to fix that, by mikhal@webrtc.org · 11 years ago
  82. e2e033a Relanding 4597 - Don't force key frame when decoding with errors. by mikhal@webrtc.org · 11 years ago
  83. e562e02 WindowCapturer implementation for Linux. by sergeyu@chromium.org · 11 years ago
  84. e21b64b Disables RtpRtcpTest.CanTransmitExtraRtpPacketsWithoutError as it flakily breaks the waterfall. See http://chromegw.corp.google.com/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/99/steps/voe_auto_test/logs/stdio the cl triggering it was a no-change (disabled some other broken tests). by henrike@webrtc.org · 11 years ago
  85. c179706 Remove newapi:: namespace for typenames without overlap. by pbos@webrtc.org · 11 years ago
  86. f83a872 Revert 4597 "Don't force key frame when decoding with errors" by henrike@webrtc.org · 11 years ago
  87. bb89390 Implement window capturer for OS X. by sergeyu@chromium.org · 11 years ago
  88. c5fc6e0 Don't force key frame when decoding with errors by mikhal@webrtc.org · 11 years ago
  89. 0f911c9 Remove template usage of typeless enum in fake_encoder. by pbos@webrtc.org · 11 years ago
  90. 206c4a5 Enabling and testing RTCP CNAME in new API. by pbos@webrtc.org · 11 years ago
  91. 55afdbe Adds two tests for verifying padding and ramp-up behavior. by stefan@webrtc.org · 11 years ago
  92. 3540c82 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  93. 5f199d9 Android audio opensles: random deadlock in stopRecording(). by braveyao@webrtc.org · 11 years ago
  94. a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  95. 9e8a66c Follow-up changes to kSelectiveErrors by mikhal@webrtc.org · 11 years ago
  96. 3ded8c9 Disables ReceivesPliAndRecoversWithNack and NoPacketLoss as they break the bots. by henrike@webrtc.org · 11 years ago
  97. c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  98. efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
  99. f96e534 Call SetExecutablePath from test_main.cc by pbos@webrtc.org · 11 years ago
  100. 7deb335 Make FrameGeneratorCapturer own frame_generator. by pbos@webrtc.org · 11 years ago