Gitiles
Code Review Sign In
gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / da3ad0887393306d461577c0e30dbbe486bda421
  1. da3ad08 Updating test file contents to emmastjernloef by kjellander@webrtc.org · 11 years ago
  2. e5210af Adding Opus unit test by tina.legrand@webrtc.org · 11 years ago
  3. 4eb4487 Fix for "RTP dynamic payload type 100 is reserved" by henrika@webrtc.org · 11 years ago
  4. faff94b Issue 1647. Avoid unsequenced modification. by turaj@webrtc.org · 11 years ago
  5. e45d9af Remove vim/emacs modelines from .gypi files by pbos@webrtc.org · 11 years ago
  6. c4cd83d Add support for multiple streams to RtpPlayer: by solenberg@webrtc.org · 11 years ago
  7. cc543b1 Start NACKing as soon as we have the first packet of a key frame. by stefan@webrtc.org · 11 years ago
  8. d2c7357 Change receive statistics bitrate to be provided in bps instead of kbps. by stefan@webrtc.org · 11 years ago
  9. e3acc78 Make win_support_condition_variables_primitive global to aligned with |library| by wu@webrtc.org · 11 years ago
  10. 1411d54 Elevate NetEq short-term activity statistics to ACM level for logging. by turaj@webrtc.org · 11 years ago
  11. 9685136 Disable -Wunsequenced warning in audio_coding_module by kjellander@webrtc.org · 11 years ago
  12. 76f9c60 Partial revert of r3844 by mikhal@webrtc.org · 11 years ago
  13. 7133809 removing redundant calls to cleanframes by mikhal@webrtc.org · 11 years ago
  14. 06077c9 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
  15. c4c16bf Change capture interface to use NTP capture time. by stefan@webrtc.org · 11 years ago
  16. e90a0af Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
  17. 73e1779 VCM/JB:Removing hybrid and setting a decodable state. by mikhal@webrtc.org · 11 years ago
  18. 6cb8d9d Fix issues with incorrect wrap checks when having big buffers and high bitrate. by stefan@webrtc.org · 11 years ago
  19. 58dfa66 Fixes an issue where the start bitrate is stored in kbps instead of bps. by stefan@webrtc.org · 11 years ago
  20. 08be23b Fix -Wstring-conversion warnings. by wu@webrtc.org · 11 years ago
  21. 6a145d7 Re-write the build of the nacklist. by andresp@webrtc.org · 11 years ago
  22. 2939d14 WebRTCDemo: handle stride!=width from first frame. by fischman@webrtc.org · 11 years ago
  23. 8129077 Updated WebRTC version number to 3.29 by elham@webrtc.org · 11 years ago
  24. b28e522 WebRTCDemo: Enable making multiple calls. by fischman@webrtc.org · 11 years ago
  25. 7793e44 Add OWNERS file for channel_transport by kjellander@webrtc.org · 11 years ago
  26. bb48e9c Replace legacy G_CONST with const. by pbos@webrtc.org · 11 years ago
  27. 76076ec Removing remaining WebRtc_Word32 not in typedefs.h by pbos@webrtc.org · 11 years ago
  28. 919738e WebRTCDemo: no-op out instead of NPEing on destroyed camera. by fischman@webrtc.org · 11 years ago
  29. e0e4035 WebRtc_Word32 -> int32_t in video_capture/ by pbos@webrtc.org · 11 years ago
  30. 470cb87 WebRtc_Word32 -> int32_t in video_render/ by pbos@webrtc.org · 11 years ago
  31. 211b771 WebRtc_Word32 -> int32_t in audio_processing/ by pbos@webrtc.org · 11 years ago
  32. b28b83e Reapply the reverted r3747. by marpan@webrtc.org · 11 years ago
  33. bffd956 More trace events by hclam@chromium.org · 11 years ago
  34. d1c6dde Improve how NACK lists are generated before a frame has been decoded. by stefan@webrtc.org · 11 years ago
  35. 4b41852 WebRtc_Word32 -> int32_t in audio_conference_mixer/ by pbos@webrtc.org · 11 years ago
  36. 1727dc7 WebRtc_Word32 -> int32_t in common_audio/ by pbos@webrtc.org · 11 years ago
  37. 41e3677 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
  38. 2a5d229 WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
  39. 5691648 WebRtc_Word32 -> int32_t in video_processing/ by pbos@webrtc.org · 11 years ago
  40. 82e0d35 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
  41. 66a0ab3 WebRtc_Word32 -> int32_t in common_video. by pbos@webrtc.org · 11 years ago
  42. 896b1e1 WebRtc_Word32 -> int32_t in utility/ by pbos@webrtc.org · 11 years ago
  43. 14353cc WebRtc_Word32 -> int32_t in media_file/ by pbos@webrtc.org · 11 years ago
  44. 6c604ea Fixing the flakiness of ThreadWakesTwice. by hta@webrtc.org · 11 years ago
  45. b9ada57 WebRtc_Word32 -> int32_t in test/ by pbos@webrtc.org · 11 years ago
  46. c404426 WebRtc_Word32 -> int32_t in audio_device/ by pbos@webrtc.org · 11 years ago
  47. 1d46b92 WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 11 years ago
  48. acf4b69 WebRtc_Word32 -> int32_t in system_wrappers by pbos@webrtc.org · 11 years ago
  49. 51868ad Always set render delay in ViEChannel::RegisterExternalDecoder. by pbos@webrtc.org · 11 years ago
  50. 3e3f84a WebRtc_Word32 => int32_t etc. in audio_coding/ by pbos@webrtc.org · 11 years ago
  51. 713488f Remove the old unused udp_transport by pwestin@webrtc.org · 11 years ago
  52. d51934d Reduce execution time of rate control test. by marpan@webrtc.org · 11 years ago
  53. ef32e92 Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array. by kma@webrtc.org · 11 years ago
  54. 2708412 WebRtc_Word32 => int32_t in video_coding/ by pbos@webrtc.org · 11 years ago
  55. 771774f WebRtc_Word32 => int32_t for rtp_rtcp/ by pbos@webrtc.org · 11 years ago
  56. 98e70d4 Clean packets on the network when closing + made loopback test actually run again. by mflodman@webrtc.org · 11 years ago
  57. 14d016a WebRtc_Word32 => int32_t remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
  58. dd78d46 Fix a crash issue on WinXP where LoadLibrary(TEXT("Kernel32.dll")) may fail. by wu@webrtc.org · 11 years ago
  59. 50fb4ed 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
  60. 2cc0155 Resolves TSan v2 reports data races in voe_auto_test. by henrika@webrtc.org · 11 years ago
  61. ad45772 Add GYP target for WebRTC Video demo for Android. by kjellander@webrtc.org · 11 years ago
  62. 3c48614 Permit arbitrary payload names for kVideoCodecGeneric. by pbos@webrtc.org · 11 years ago
  63. 47e4f00 Remove WEBRTC_*_ENGINE_NETWORK_API use by pwestin@webrtc.org · 11 years ago
  64. 0b8adb4 Adds event traces and counters for WebRTC receive side. by edjee@google.com · 11 years ago
  65. 34dac64 Fix no received audio in tests. by pwestin@webrtc.org · 11 years ago
  66. fe3a907 Disabling MixingTests due to race conditions. by henrika@webrtc.org · 11 years ago
  67. 5bea712 Two more sleep calls converted to use SleepMs(). by hta@webrtc.org · 11 years ago
  68. ebc0331 TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC by henrika@webrtc.org · 11 years ago
  69. 22f789f Remove UDP transport API from VoE by pwestin@webrtc.org · 11 years ago
  70. 25dda04 Fixes memory leak in AudioLevel class reported by memory try bots. by henrika@webrtc.org · 11 years ago
  71. 63ef6e2 Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer by henrika@webrtc.org · 11 years ago
  72. e561f8c Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
  73. 45d75a4 Webrtc_Word32 => int32_t in video_coding/main/ by pbos@webrtc.org · 11 years ago
  74. 1562c72 Revert of r3747. by henrike@webrtc.org · 11 years ago
  75. d393127 Two more sleep calls converted to use SleepMs(). by hta@webrtc.org · 11 years ago
  76. d042a17 Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer by henrika@webrtc.org · 11 years ago
  77. d8322b9 Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher by justinlin@chromium.org · 11 years ago
  78. 435b50c For VGA (640x360), currently 1 thread is used. This change increases it to 2 threads. For HD, 4 threads are enabled. by fbarchard@google.com · 11 years ago
  79. 2379013 Updated Webrtc version to 3.28 by elham@webrtc.org · 11 years ago
  80. bbf5086 Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549 by marpan@webrtc.org · 11 years ago
  81. bcce6df Fixes build break in previous cl (https://code.google.com/p/webrtc/source/detail?r=3739) found by Android bots. by henrike@webrtc.org · 11 years ago
  82. 18881d5 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  83. 1ca9d42 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. by solenberg@webrtc.org · 11 years ago
  84. e148532 Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." by wu@webrtc.org · 11 years ago
  85. 90edf85 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  86. fece2f5 Fix broken audio. by leozwang@webrtc.org · 11 years ago
  87. 11552e9 G722-stereo has been missing when creating AudioDecoder. by turaj@webrtc.org · 11 years ago
  88. 3e00311 NetEq4 fails if the first packets inserted in are out-of-band DTMFs. by turaj@webrtc.org · 11 years ago
  89. c3ab830 Fix flakiness in network up/down event tests when running under memcheck. by stefan@webrtc.org · 11 years ago
  90. 09e8463 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. by fischman@webrtc.org · 11 years ago
  91. e3eea1b Add interface to signal a network down event. by stefan@webrtc.org · 11 years ago
  92. fb6a7c4 Split condition_variable_win.cc into native (for Vista and newer OS versions) and generic implementation (based on events). by henrike@webrtc.org · 11 years ago
  93. 41419d9 Remove VoE's default call in Trace::SetLevelFilter. by andrew@webrtc.org · 11 years ago
  94. eefab4e Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust. by solenberg@webrtc.org · 11 years ago
  95. 6fc92b4 Alphabetize include order in fake_voe_external_media.h. by andrew@webrtc.org · 11 years ago
  96. 6666b90 Restart Android capture after orientation change. Also prevent an NPE on exit. by fischman@webrtc.org · 11 years ago
  97. 58a5924 Add some VoE and AudioProcessing mocks. by andrew@webrtc.org · 11 years ago
  98. f658278 Refactor unittest trace printouts to a separate class. by andrew@webrtc.org · 11 years ago
  99. 8cfba7e Enable the below APIs for iOS. by sjlee@webrtc.org · 11 years ago
  100. 60c8100 Introduced pause and resume to the pacer by pwestin@webrtc.org · 11 years ago