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gerrit-public.fairphone.software / platform / external / webrtc / 63117339dc56440e233f1edc003c38f01863cfaa
  1. 6311733 Updated the sync module with a slow moving filter by pwestin@webrtc.org · 11 years ago
  2. 28d54ab Improve AV-sync when initial delay is set and NetEq has long buffer. by turaj@webrtc.org · 11 years ago
  3. 1b42771 emove desktop_capture.gypi from modules.gyp by kjellander@webrtc.org · 11 years ago
  4. 7c9e992 Removed unused variable. by mflodman@webrtc.org · 11 years ago
  5. aeff4f3 Fixing Coverity issues. by mflodman@webrtc.org · 11 years ago
  6. 8aa4a90 Set mime type on device-switch.html by tommi@webrtc.org · 11 years ago
  7. 006acc4 Update iOS build script to run on bots. by kjellander@webrtc.org · 11 years ago
  8. e0e029e Revert 3876 by mikhal@webrtc.org · 11 years ago
  9. ee184b9 VCM/Receiver: Only update render time when decoding by mikhal@webrtc.org · 11 years ago
  10. c41478f Ensure build_demo.py run subprocesses with bash shell. by kjellander@webrtc.org · 11 years ago
  11. a39a8fe Add owner to Apprtc by braveyao@webrtc.org · 11 years ago
  12. 1fea17d Add the build script of the voice engine for iOS. by sjlee@webrtc.org · 11 years ago
  13. a73d52c revert r3871 by mikhal@webrtc.org · 11 years ago
  14. 9756017 - Replace the BWE_MIN and BWE_MAX macros with std::min and std::max by solenberg@webrtc.org · 11 years ago
  15. d26457f Apply Chromium C++ style to BitRateStats. by solenberg@webrtc.org · 11 years ago
  16. 65f995a New ViE interface. by mflodman@webrtc.org · 11 years ago
  17. c14b728 Add lock to prevent possible rare race condition in Win coreAudio capture implementation. by braveyao@webrtc.org · 11 years ago
  18. ceaedc0 Remove executable bit from dc1.html. by andrew@webrtc.org · 11 years ago
  19. a0cd918 Add desktop_capture directory for screen and window capturers. by sergeyu@chromium.org · 11 years ago
  20. dbd6a6d Updating delay for first value by mikhal@webrtc.org · 11 years ago
  21. 48c5882 Remove libvpx pre-processor conditions and conditional compile of default temporal layers files. by andresp@webrtc.org · 11 years ago
  22. f090167 Revert "Updating test file contents to emmastjernloef" by kjellander@webrtc.org · 11 years ago
  23. f1bf3a0 A device switcher code example, with fake. by hta@webrtc.org · 11 years ago
  24. 11959d3 Updating test file contents to emmastjernloef by kjellander@webrtc.org · 11 years ago
  25. db11fab Adding Opus unit test by tina.legrand@webrtc.org · 11 years ago
  26. 4392d5f Fix for "RTP dynamic payload type 100 is reserved" by henrika@webrtc.org · 11 years ago
  27. f1a3b4b Issue 1647. Avoid unsequenced modification. by turaj@webrtc.org · 11 years ago
  28. 6e788df Remove vim/emacs modelines from .gypi files by pbos@webrtc.org · 11 years ago
  29. 56b5f77 Add support for multiple streams to RtpPlayer: by solenberg@webrtc.org · 11 years ago
  30. 885cd13 Start NACKing as soon as we have the first packet of a key frame. by stefan@webrtc.org · 11 years ago
  31. bdb9b97 Change receive statistics bitrate to be provided in bps instead of kbps. by stefan@webrtc.org · 11 years ago
  32. e44a064 Make win_support_condition_variables_primitive global to aligned with |library| by wu@webrtc.org · 11 years ago
  33. 92d1f07 Elevate NetEq short-term activity statistics to ACM level for logging. by turaj@webrtc.org · 11 years ago
  34. 4b8de90 Disable -Wunsequenced warning in audio_coding_module by kjellander@webrtc.org · 11 years ago
  35. f806ad2 Roll chromium_revision 182149:193311 by kjellander@webrtc.org · 11 years ago
  36. c83b356 Roll libvpx to 192165. -pick up libvpx roll to 3db60c8. by marpan@webrtc.org · 11 years ago
  37. c2a3aa7 Partial revert of r3844 by mikhal@webrtc.org · 11 years ago
  38. d6bd7cd removing redundant calls to cleanframes by mikhal@webrtc.org · 11 years ago
  39. 9f5ebb5 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
  40. b8e7f4c Change capture interface to use NTP capture time. by stefan@webrtc.org · 11 years ago
  41. d35dff7 Move to Chrome infra try server. by kjellander@webrtc.org · 11 years ago
  42. 1de0135 Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
  43. 9da7517 VCM/JB:Removing hybrid and setting a decodable state. by mikhal@webrtc.org · 11 years ago
  44. 7bc465b Fix issues with incorrect wrap checks when having big buffers and high bitrate. by stefan@webrtc.org · 11 years ago
  45. 122d209 Fixes an issue where the start bitrate is stored in kbps instead of bps. by stefan@webrtc.org · 11 years ago
  46. eac36b8 Fix -Wstring-conversion warnings. by wu@webrtc.org · 11 years ago
  47. 523f937 Re-write the build of the nacklist. by andresp@webrtc.org · 11 years ago
  48. f2a97fc WebRTCDemo: handle stride!=width from first frame. by fischman@webrtc.org · 11 years ago
  49. d40e404 Revert r3815 by marpan@webrtc.org · 11 years ago
  50. 1b2a6e0 Updated WebRTC version number to 3.29 by elham@webrtc.org · 11 years ago
  51. 6f41ca9 WebRTCDemo: Enable making multiple calls. by fischman@webrtc.org · 11 years ago
  52. 59d8889 Add OWNERS file for channel_transport by kjellander@webrtc.org · 11 years ago
  53. 6bfcbcd Roll libvpx to 192165. -pick up libvpx roll to 3db60c8. by marpan@webrtc.org · 11 years ago
  54. e4b6064 Replace legacy G_CONST with const. by pbos@webrtc.org · 11 years ago
  55. ab9202b Removing remaining WebRtc_Word32 not in typedefs.h by pbos@webrtc.org · 11 years ago
  56. 77d59fe WebRTCDemo: no-op out instead of NPEing on destroyed camera. by fischman@webrtc.org · 11 years ago
  57. dfc5bb9 WebRtc_Word32 -> int32_t in video_capture/ by pbos@webrtc.org · 11 years ago
  58. ddf94e7 WebRtc_Word32 -> int32_t in video_render/ by pbos@webrtc.org · 11 years ago
  59. b7192b8 WebRtc_Word32 -> int32_t in audio_processing/ by pbos@webrtc.org · 11 years ago
  60. 557e925 Reapply the reverted r3747. by marpan@webrtc.org · 11 years ago
  61. 806dc3b More trace events by hclam@chromium.org · 11 years ago
  62. 4d2f5de Improve how NACK lists are generated before a frame has been decoded. by stefan@webrtc.org · 11 years ago
  63. ac89162 WebRtc_Word32 -> int32_t in audio_conference_mixer/ by pbos@webrtc.org · 11 years ago
  64. b091307 WebRtc_Word32 -> int32_t in common_audio/ by pbos@webrtc.org · 11 years ago
  65. 7da3459 Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." by stefan@webrtc.org · 11 years ago
  66. b238d12 WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
  67. 1ab45f6 WebRtc_Word32 -> int32_t in video_processing/ by pbos@webrtc.org · 11 years ago
  68. afcc610 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. by stefan@webrtc.org · 11 years ago
  69. fd2bfc8 WebRtc_Word32 -> int32_t in common_video. by pbos@webrtc.org · 11 years ago
  70. c75102e WebRtc_Word32 -> int32_t in utility/ by pbos@webrtc.org · 11 years ago
  71. 0ea11c1 WebRtc_Word32 -> int32_t in media_file/ by pbos@webrtc.org · 11 years ago
  72. a701c0e Fixing the flakiness of ThreadWakesTwice. by hta@webrtc.org · 11 years ago
  73. a5f1787 WebRtc_Word32 -> int32_t in test/ by pbos@webrtc.org · 11 years ago
  74. 2550988 WebRtc_Word32 -> int32_t in audio_device/ by pbos@webrtc.org · 11 years ago
  75. 6141e13 WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 11 years ago
  76. 046deb9 WebRtc_Word32 -> int32_t in system_wrappers by pbos@webrtc.org · 11 years ago
  77. 29758de Always set render delay in ViEChannel::RegisterExternalDecoder. by pbos@webrtc.org · 11 years ago
  78. 0946a56 WebRtc_Word32 => int32_t etc. in audio_coding/ by pbos@webrtc.org · 11 years ago
  79. 6faf71d Remove the old unused udp_transport by pwestin@webrtc.org · 11 years ago
  80. 4c44fe0 Updated pranswer, dtmf demos & deleted pc1-deprecated.html. by vikasmarwaha@webrtc.org · 11 years ago
  81. 6ff76c7 Reduce execution time of rate control test. by marpan@webrtc.org · 11 years ago
  82. cf8e108 Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array. by kma@webrtc.org · 11 years ago
  83. b4a0623 Fix of lint script errors in apprtc.py by pbos@webrtc.org · 11 years ago
  84. f2e7bc6 Added maxlen=80 to CheckLongLines() call in PRESUBMIT.py by pbos@webrtc.org · 11 years ago
  85. 034f004 WebRtc_Word32 => int32_t in video_coding/ by pbos@webrtc.org · 11 years ago
  86. 2f44673 WebRtc_Word32 => int32_t for rtp_rtcp/ by pbos@webrtc.org · 11 years ago
  87. 367804c Clean packets on the network when closing + made loopback test actually run again. by mflodman@webrtc.org · 11 years ago
  88. ff7e130 WebRtc_Word32 => int32_t remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
  89. 37bf584 Show stats from both sides by hta@webrtc.org · 11 years ago
  90. 222e994 Migrating Apprtc to use new TURN service which supports time-limited TURN credentials. by vikasmarwaha@webrtc.org · 11 years ago
  91. 123b618 Fix a crash issue on WinXP where LoadLibrary(TEXT("Kernel32.dll")) may fail. by wu@webrtc.org · 11 years ago
  92. 2e6b7e9 In streaming mode it is preferable to fade to silence when sender stops sending, or long period of packet loss. by turaj@webrtc.org · 11 years ago
  93. 19da719 Resolves TSan v2 reports data races in voe_auto_test. by henrika@webrtc.org · 11 years ago
  94. 10eb920 Add GYP target for WebRTC Video demo for Android. by kjellander@webrtc.org · 11 years ago
  95. b5bf54c Permit arbitrary payload names for kVideoCodecGeneric. by pbos@webrtc.org · 11 years ago
  96. b9e402d Remove WEBRTC_*_ENGINE_NETWORK_API use by pwestin@webrtc.org · 11 years ago
  97. 79b0289 Adds event traces and counters for WebRTC receive side. by edjee@google.com · 11 years ago
  98. 835dbf4 Fix no received audio in tests. by pwestin@webrtc.org · 11 years ago
  99. aa527bb Disabling MixingTests due to race conditions. by henrika@webrtc.org · 11 years ago
  100. fcb7c38 Two more sleep calls converted to use SleepMs(). by hta@webrtc.org · 11 years ago