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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / caba49f3fd20b3eef3531d5d7a341c31377462a0
  1. caba49f Add an option to override the TestToStderr trace printout time. by andrew@webrtc.org · 11 years ago
  2. cff84ec Consolidate all third party licenses in LICENSE_THIRD_PARTY. by andrew@webrtc.org · 11 years ago
  3. e0aad3c Updated WebRTC version number to 3.30 by elham@webrtc.org · 11 years ago
  4. 25d4818 VCM/JB: Porting jitter_buffer_test to gtest. by mikhal@webrtc.org · 11 years ago
  5. 92b5ea0 Remove 44.1 kHz workaround from AudioDevice on PulseAudio. by andrew@webrtc.org · 11 years ago
  6. 6cd8727 Remove 44.1 kHz workaround from AudioDevice on WASAPI. by andrew@webrtc.org · 11 years ago
  7. 8e35807 Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert(). by sergeyu@chromium.org · 11 years ago
  8. cb0cb7d VCM: Updating receiver logic by mikhal@webrtc.org · 11 years ago
  9. 5398063 Correct and update dir name by leozwang@webrtc.org · 11 years ago
  10. c22830f Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." by pbos@webrtc.org · 11 years ago
  11. a257915 Get rid of some unnecessary copying when sending REMBs. by solenberg@webrtc.org · 11 years ago
  12. 7ee60d0 Formatting ACM tests by tina.legrand@webrtc.org · 11 years ago
  13. 3794539 Fix when SetMinimumPlayoutDelay is configured to 0 by pwestin@webrtc.org · 11 years ago
  14. 76318c5 Removing bad code resulting in flaky test. by pwestin@webrtc.org · 11 years ago
  15. c06da8c Adding trace and changing pacing constants by pwestin@webrtc.org · 11 years ago
  16. 6c98c86 Update third party license file by niklas.enbom@webrtc.org · 11 years ago
  17. 8f5edba Bugfix custom call stop. by pwestin@webrtc.org · 11 years ago
  18. c77e4da Allow voe_cmd_test to select Opus mono (now the default). by andrew@webrtc.org · 11 years ago
  19. c875f20 Relax VoE's max packet length threshold. by andrew@webrtc.org · 11 years ago
  20. 86f267b Disabled flaky test. by phoglund@webrtc.org · 11 years ago
  21. 2d6a699 Revert 3933 "Remove traces of deprecated WebRtc_Word types." by pbos@webrtc.org · 11 years ago
  22. bbb54b3 Remove traces of deprecated WebRtc_Word types. by pbos@webrtc.org · 11 years ago
  23. 74161fc WebRTCDemo Android app to route audio to headphone when it's plugged in. by braveyao@webrtc.org · 11 years ago
  24. b1c40c5 Replace Resampler with PushResampler in transmit_mixer. by andrew@webrtc.org · 11 years ago
  25. a23b051 Consolidate common_audio into a single target. by andrew@webrtc.org · 11 years ago
  26. 22a3795 Add AEC suppression level option to audioproc. by andrew@webrtc.org · 11 years ago
  27. 10600ab Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi . by sergeyu@chromium.org · 11 years ago
  28. 9e65a61 Fixes two bugs in receive statistics. by stefan@webrtc.org · 11 years ago
  29. c5fbd58 Fixing AV sync. Increased 2 const to allow for a bigger difference in AV sync. by pwestin@webrtc.org · 11 years ago
  30. 6677d3c VCM: Setting buffering delay in timing by mikhal@webrtc.org · 11 years ago
  31. faec77d Adding buffered mode to loopback test by mikhal@webrtc.org · 11 years ago
  32. 48e6258 Apply Chromium C++ style to RemoteRateControl. by solenberg@webrtc.org · 11 years ago
  33. dfffece Add DesktopCapturer interface for desktop capturers. by sergeyu@chromium.org · 11 years ago
  34. 78a4a73 Don't reset the last je value and mode by mikhal@webrtc.org · 11 years ago
  35. 45325fd Add a wrapper around PushSincResampler and the old Resampler. by andrew@webrtc.org · 11 years ago
  36. 45115c8 Fix two issues where we might end up busy looping in decoder_render mode. by stefan@webrtc.org · 11 years ago
  37. 8517f00 Enable Nack pacing. by pwestin@webrtc.org · 11 years ago
  38. 8a159ad Removing vie file related code from vie_custom_call by mikhal@webrtc.org · 11 years ago
  39. fd7a1b7 Fixed remaining nits from Stefan by pwestin@webrtc.org · 11 years ago
  40. 8fdc74b Add a push-based wrapper around SincResampler. by andrew@webrtc.org · 11 years ago
  41. b537037 Add comfort noise disabling and routing mode selection to audioproc. by andrew@webrtc.org · 11 years ago
  42. c960e32 Removing another instance of file api by mikhal@webrtc.org · 11 years ago
  43. 71645c8 Fix the encoder pause logic. BUG=1691 by pwestin@webrtc.org · 11 years ago
  44. 1c42057 VCM: Adding API for the size(duration) of the jitter buffer. by mikhal@webrtc.org · 11 years ago
  45. 96535b0 VCM/JB: Using last decoded state for waiting for key by mikhal@webrtc.org · 11 years ago
  46. a4d060f VCM/JB: FrameForDecoding->IncompleteFrameForDecoding by mikhal@webrtc.org · 11 years ago
  47. 2423690 Disabling avi file interface by mikhal@webrtc.org · 11 years ago
  48. 911d567 Avoid adding duplicates in pacer lists. by pwestin@webrtc.org · 11 years ago
  49. 9ad7cce Make sure timestamps are monotonically increasing. by stefan@webrtc.org · 11 years ago
  50. 39ea6be Revert 3892 "VCM/JB: Using last decoded state for waiting for key" by andrew@webrtc.org · 11 years ago
  51. 1efb11c Adding extra options to interact with external encoder/decoder. by andresp@webrtc.org · 11 years ago
  52. 3d0d20f VCM/JB: Using last decoded state for waiting for key by mikhal@webrtc.org · 11 years ago
  53. b35efcc Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 11 years ago
  54. a09c0c4 Buf fix for r3883. by turaj@webrtc.org · 11 years ago
  55. 65e6f91 Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 11 years ago
  56. f97287b VP8: Avoid copying the codec struct on Reset(). by pbos@webrtc.org · 11 years ago
  57. 38617bc BUG=1351 by mflodman@webrtc.org · 11 years ago
  58. 18a4532 VCM/JB: Skip to the next complete key frame by mikhal@webrtc.org · 11 years ago
  59. 7a14b35 Updated the sync module with a slow moving filter by pwestin@webrtc.org · 11 years ago
  60. e471f0c Improve AV-sync when initial delay is set and NetEq has long buffer. by turaj@webrtc.org · 11 years ago
  61. 7070d8f emove desktop_capture.gypi from modules.gyp by kjellander@webrtc.org · 11 years ago
  62. fe2bce3 Removed unused variable. by mflodman@webrtc.org · 11 years ago
  63. fb5b5cb Fixing Coverity issues. by mflodman@webrtc.org · 11 years ago
  64. f06fff3 Update iOS build script to run on bots. by kjellander@webrtc.org · 11 years ago
  65. 437dfaf Revert 3876 by mikhal@webrtc.org · 11 years ago
  66. 742f3e9 VCM/Receiver: Only update render time when decoding by mikhal@webrtc.org · 11 years ago
  67. 1ccedf6 Ensure build_demo.py run subprocesses with bash shell. by kjellander@webrtc.org · 11 years ago
  68. 1b4af0d Add the build script of the voice engine for iOS. by sjlee@webrtc.org · 11 years ago
  69. 64823d1 revert r3871 by mikhal@webrtc.org · 11 years ago
  70. e8a772d - Replace the BWE_MIN and BWE_MAX macros with std::min and std::max by solenberg@webrtc.org · 11 years ago
  71. 342ee85 Apply Chromium C++ style to BitRateStats. by solenberg@webrtc.org · 11 years ago
  72. 69b0d2c New ViE interface. by mflodman@webrtc.org · 11 years ago
  73. 27334a1 Add lock to prevent possible rare race condition in Win coreAudio capture implementation. by braveyao@webrtc.org · 11 years ago
  74. 5b10324 Add desktop_capture directory for screen and window capturers. by sergeyu@chromium.org · 11 years ago
  75. 42a1a30 Updating delay for first value by mikhal@webrtc.org · 11 years ago
  76. 95d5bc6 Remove libvpx pre-processor conditions and conditional compile of default temporal layers files. by andresp@webrtc.org · 11 years ago
  77. 0a884c0 Revert "Updating test file contents to emmastjernloef" by kjellander@webrtc.org · 11 years ago
  78. da3ad08 Updating test file contents to emmastjernloef by kjellander@webrtc.org · 11 years ago
  79. e5210af Adding Opus unit test by tina.legrand@webrtc.org · 11 years ago
  80. 4eb4487 Fix for "RTP dynamic payload type 100 is reserved" by henrika@webrtc.org · 11 years ago
  81. faff94b Issue 1647. Avoid unsequenced modification. by turaj@webrtc.org · 11 years ago
  82. e45d9af Remove vim/emacs modelines from .gypi files by pbos@webrtc.org · 11 years ago
  83. c4cd83d Add support for multiple streams to RtpPlayer: by solenberg@webrtc.org · 11 years ago
  84. cc543b1 Start NACKing as soon as we have the first packet of a key frame. by stefan@webrtc.org · 11 years ago
  85. d2c7357 Change receive statistics bitrate to be provided in bps instead of kbps. by stefan@webrtc.org · 11 years ago
  86. e3acc78 Make win_support_condition_variables_primitive global to aligned with |library| by wu@webrtc.org · 11 years ago
  87. 1411d54 Elevate NetEq short-term activity statistics to ACM level for logging. by turaj@webrtc.org · 11 years ago
  88. 9685136 Disable -Wunsequenced warning in audio_coding_module by kjellander@webrtc.org · 11 years ago
  89. 76f9c60 Partial revert of r3844 by mikhal@webrtc.org · 11 years ago
  90. 7133809 removing redundant calls to cleanframes by mikhal@webrtc.org · 11 years ago
  91. 06077c9 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
  92. c4c16bf Change capture interface to use NTP capture time. by stefan@webrtc.org · 11 years ago
  93. e90a0af Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
  94. 73e1779 VCM/JB:Removing hybrid and setting a decodable state. by mikhal@webrtc.org · 11 years ago
  95. 6cb8d9d Fix issues with incorrect wrap checks when having big buffers and high bitrate. by stefan@webrtc.org · 11 years ago
  96. 58dfa66 Fixes an issue where the start bitrate is stored in kbps instead of bps. by stefan@webrtc.org · 11 years ago
  97. 08be23b Fix -Wstring-conversion warnings. by wu@webrtc.org · 11 years ago
  98. 6a145d7 Re-write the build of the nacklist. by andresp@webrtc.org · 11 years ago
  99. 2939d14 WebRTCDemo: handle stride!=width from first frame. by fischman@webrtc.org · 11 years ago
  100. 8129077 Updated WebRTC version number to 3.29 by elham@webrtc.org · 11 years ago