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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 9e4ad86dc143a3a36bbeeba48318982008117107
  1. 9e4ad86 Remove 44.1 kHz workaround from AudioDevice on PulseAudio. by andrew@webrtc.org · 11 years ago
  2. 8cfa495 Remove 44.1 kHz workaround from AudioDevice on WASAPI. by andrew@webrtc.org · 11 years ago
  3. 27f61e2 Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert(). by sergeyu@chromium.org · 11 years ago
  4. b842af8 VCM: Updating receiver logic by mikhal@webrtc.org · 11 years ago
  5. 20f81fe Correct and update dir name by leozwang@webrtc.org · 11 years ago
  6. 52b2ee5 Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." by pbos@webrtc.org · 11 years ago
  7. 3a3a787 Get rid of some unnecessary copying when sending REMBs. by solenberg@webrtc.org · 11 years ago
  8. b31332e Formatting ACM tests by tina.legrand@webrtc.org · 11 years ago
  9. 49d151e Fix when SetMinimumPlayoutDelay is configured to 0 by pwestin@webrtc.org · 11 years ago
  10. 677ca88 Removing bad code resulting in flaky test. by pwestin@webrtc.org · 11 years ago
  11. 36bdba4 Adding trace and changing pacing constants by pwestin@webrtc.org · 11 years ago
  12. 24de1a0 Update third party license file by niklas.enbom@webrtc.org · 11 years ago
  13. 422a124 Bugfix custom call stop. by pwestin@webrtc.org · 11 years ago
  14. c0fc487 Allow voe_cmd_test to select Opus mono (now the default). by andrew@webrtc.org · 11 years ago
  15. ad9cee8 Relax VoE's max packet length threshold. by andrew@webrtc.org · 11 years ago
  16. 9e0d9a6 Disabled flaky test. by phoglund@webrtc.org · 11 years ago
  17. 2d6f0df Revert 3933 "Remove traces of deprecated WebRtc_Word types." by pbos@webrtc.org · 11 years ago
  18. e422d12 Remove traces of deprecated WebRtc_Word types. by pbos@webrtc.org · 11 years ago
  19. be6ad88 WebRTCDemo Android app to route audio to headphone when it's plugged in. by braveyao@webrtc.org · 11 years ago
  20. 4a68e95 Replace Resampler with PushResampler in transmit_mixer. by andrew@webrtc.org · 11 years ago
  21. 166153e Consolidate common_audio into a single target. by andrew@webrtc.org · 11 years ago
  22. e9c34ba Add AEC suppression level option to audioproc. by andrew@webrtc.org · 11 years ago
  23. 22b72cb Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi . by sergeyu@chromium.org · 11 years ago
  24. 9137e98 Fixes two bugs in receive statistics. by stefan@webrtc.org · 11 years ago
  25. 4d4ba47 Fixing AV sync. Increased 2 const to allow for a bigger difference in AV sync. by pwestin@webrtc.org · 11 years ago
  26. 2725aa0 VCM: Setting buffering delay in timing by mikhal@webrtc.org · 11 years ago
  27. 6ed1c8c Adding buffered mode to loopback test by mikhal@webrtc.org · 11 years ago
  28. 5f09bcc Apply Chromium C++ style to RemoteRateControl. by solenberg@webrtc.org · 11 years ago
  29. 85a1dab Add DesktopCapturer interface for desktop capturers. by sergeyu@chromium.org · 11 years ago
  30. d60f7a9 Don't reset the last je value and mode by mikhal@webrtc.org · 11 years ago
  31. b6fadb1 Add a wrapper around PushSincResampler and the old Resampler. by andrew@webrtc.org · 11 years ago
  32. c12e655 Fix two issues where we might end up busy looping in decoder_render mode. by stefan@webrtc.org · 11 years ago
  33. b8f1cf3 Enable Nack pacing. by pwestin@webrtc.org · 11 years ago
  34. 1a58dd7 Removing vie file related code from vie_custom_call by mikhal@webrtc.org · 11 years ago
  35. b13f394 Fixed remaining nits from Stefan by pwestin@webrtc.org · 11 years ago
  36. dea537b Add a push-based wrapper around SincResampler. by andrew@webrtc.org · 11 years ago
  37. 3e20f91 Add comfort noise disabling and routing mode selection to audioproc. by andrew@webrtc.org · 11 years ago
  38. 05c25a7 Removing another instance of file api by mikhal@webrtc.org · 11 years ago
  39. 3816c52 Fix the encoder pause logic. BUG=1691 by pwestin@webrtc.org · 11 years ago
  40. 40bd744 VCM: Adding API for the size(duration) of the jitter buffer. by mikhal@webrtc.org · 11 years ago
  41. b47b7e0 VCM/JB: Using last decoded state for waiting for key by mikhal@webrtc.org · 11 years ago
  42. 1886a04 VCM/JB: FrameForDecoding->IncompleteFrameForDecoding by mikhal@webrtc.org · 11 years ago
  43. 4eac481 Disabling avi file interface by mikhal@webrtc.org · 11 years ago
  44. e4ae7a2 Avoid adding duplicates in pacer lists. by pwestin@webrtc.org · 11 years ago
  45. 2dfffc3 Make sure timestamps are monotonically increasing. by stefan@webrtc.org · 11 years ago
  46. 8f97f02 Revert 3892 "VCM/JB: Using last decoded state for waiting for key" by andrew@webrtc.org · 11 years ago
  47. f292306 Adding extra options to interact with external encoder/decoder. by andresp@webrtc.org · 11 years ago
  48. a7be357 VCM/JB: Using last decoded state for waiting for key by mikhal@webrtc.org · 11 years ago
  49. d430f32 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 11 years ago
  50. 292ed1d Buf fix for r3883. by turaj@webrtc.org · 11 years ago
  51. 2788107 Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 11 years ago
  52. 9ded0b1 VP8: Avoid copying the codec struct on Reset(). by pbos@webrtc.org · 11 years ago
  53. 0dae366 BUG=1351 by mflodman@webrtc.org · 11 years ago
  54. 4123abf VCM/JB: Skip to the next complete key frame by mikhal@webrtc.org · 11 years ago
  55. f13f1fc Updated the sync module with a slow moving filter by pwestin@webrtc.org · 11 years ago
  56. 92aa25b Improve AV-sync when initial delay is set and NetEq has long buffer. by turaj@webrtc.org · 11 years ago
  57. 6cb19e1 emove desktop_capture.gypi from modules.gyp by kjellander@webrtc.org · 11 years ago
  58. c11933f Removed unused variable. by mflodman@webrtc.org · 11 years ago
  59. bea854a Fixing Coverity issues. by mflodman@webrtc.org · 11 years ago
  60. a788a4d Update iOS build script to run on bots. by kjellander@webrtc.org · 11 years ago
  61. e07ec09 Revert 3876 by mikhal@webrtc.org · 11 years ago
  62. c2c65ba VCM/Receiver: Only update render time when decoding by mikhal@webrtc.org · 11 years ago
  63. b6e175d Ensure build_demo.py run subprocesses with bash shell. by kjellander@webrtc.org · 11 years ago
  64. 4fdfb61 Add the build script of the voice engine for iOS. by sjlee@webrtc.org · 11 years ago
  65. dc6a521 revert r3871 by mikhal@webrtc.org · 11 years ago
  66. 20aee3a - Replace the BWE_MIN and BWE_MAX macros with std::min and std::max by solenberg@webrtc.org · 11 years ago
  67. e5117e7 Apply Chromium C++ style to BitRateStats. by solenberg@webrtc.org · 11 years ago
  68. 06e8026 New ViE interface. by mflodman@webrtc.org · 11 years ago
  69. 1a67b9c Add lock to prevent possible rare race condition in Win coreAudio capture implementation. by braveyao@webrtc.org · 11 years ago
  70. ea2ea7b Add desktop_capture directory for screen and window capturers. by sergeyu@chromium.org · 11 years ago
  71. e7afdc7 Updating delay for first value by mikhal@webrtc.org · 11 years ago
  72. 119bc54 Remove libvpx pre-processor conditions and conditional compile of default temporal layers files. by andresp@webrtc.org · 11 years ago
  73. 0cd2401 Revert "Updating test file contents to emmastjernloef" by kjellander@webrtc.org · 11 years ago
  74. e8bfe2a Updating test file contents to emmastjernloef by kjellander@webrtc.org · 11 years ago
  75. 28fb40d Adding Opus unit test by tina.legrand@webrtc.org · 11 years ago
  76. 570c4a5 Fix for "RTP dynamic payload type 100 is reserved" by henrika@webrtc.org · 11 years ago
  77. 6b33839 Issue 1647. Avoid unsequenced modification. by turaj@webrtc.org · 11 years ago
  78. 237fe4f Remove vim/emacs modelines from .gypi files by pbos@webrtc.org · 11 years ago
  79. 4d2a2ec Add support for multiple streams to RtpPlayer: by solenberg@webrtc.org · 11 years ago
  80. dd32d85 Start NACKing as soon as we have the first packet of a key frame. by stefan@webrtc.org · 11 years ago
  81. 7cc4a54 Change receive statistics bitrate to be provided in bps instead of kbps. by stefan@webrtc.org · 11 years ago
  82. c6f71c5 Make win_support_condition_variables_primitive global to aligned with |library| by wu@webrtc.org · 11 years ago
  83. 56e0484 Elevate NetEq short-term activity statistics to ACM level for logging. by turaj@webrtc.org · 11 years ago
  84. 8432603 Disable -Wunsequenced warning in audio_coding_module by kjellander@webrtc.org · 11 years ago
  85. 64d5d26 Partial revert of r3844 by mikhal@webrtc.org · 11 years ago
  86. ab33b46 removing redundant calls to cleanframes by mikhal@webrtc.org · 11 years ago
  87. 7bc7e02 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
  88. 9b53152 Change capture interface to use NTP capture time. by stefan@webrtc.org · 11 years ago
  89. f272497 Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
  90. 02f158e VCM/JB:Removing hybrid and setting a decodable state. by mikhal@webrtc.org · 11 years ago
  91. 42c7409 Fix issues with incorrect wrap checks when having big buffers and high bitrate. by stefan@webrtc.org · 11 years ago
  92. 1e43446 Fixes an issue where the start bitrate is stored in kbps instead of bps. by stefan@webrtc.org · 11 years ago
  93. e68605a Fix -Wstring-conversion warnings. by wu@webrtc.org · 11 years ago
  94. bda02e4 Re-write the build of the nacklist. by andresp@webrtc.org · 11 years ago
  95. 1064639 WebRTCDemo: handle stride!=width from first frame. by fischman@webrtc.org · 11 years ago
  96. de1c434 Updated WebRTC version number to 3.29 by elham@webrtc.org · 11 years ago
  97. 6e816cb WebRTCDemo: Enable making multiple calls. by fischman@webrtc.org · 11 years ago
  98. 980d8ea Add OWNERS file for channel_transport by kjellander@webrtc.org · 11 years ago
  99. 45a3434 Replace legacy G_CONST with const. by pbos@webrtc.org · 11 years ago
  100. 0486a10 Removing remaining WebRtc_Word32 not in typedefs.h by pbos@webrtc.org · 11 years ago