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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 83db9e914d25f80271328f8cd969e123e3191eba
  1. 83db9e9 Replace gtest_prod.h include with our own FRIEND_TEST macro. by andrew@webrtc.org · 11 years ago
  2. cb139b1 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 11 years ago
  3. 9c643ec Added getter for far_time_buf in AEC. Only used in AEC debug dump. by bjornv@webrtc.org · 11 years ago
  4. 38417a8 This refactoring CL moves the nlp_mode member value from aecpc_t to aec_t, since it it never used at that level. Further, I removed two suppression variables by depending on nlp_mode directly. by bjornv@webrtc.org · 11 years ago
  5. ef1346d * Name change * Removed WebRtcAec_ function name prepending on private function. by bjornv@webrtc.org · 11 years ago
  6. 24ba537 Update to codec unit test: by marpan@webrtc.org · 11 years ago
  7. 432bc1a fixing nack list size calculation by mikhal@webrtc.org · 11 years ago
  8. 39eb955 Updated version number to 3.24 by elham@webrtc.org · 11 years ago
  9. 5962e5e Remove the dependency on dxguid.lib. by tommi@webrtc.org · 11 years ago
  10. 9e3e8e5 Move directx_sdk_path definition variable into the video_render_module gyp file. by tommi@webrtc.org · 11 years ago
  11. 85e2e0e Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. by stefan@webrtc.org · 11 years ago
  12. ce3f2ca Add VoE interface to VieRTP test by mikhal@webrtc.org · 11 years ago
  13. b115f2c Increase threshold in codec unit test. by marpan@webrtc.org · 11 years ago
  14. 4db69af Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
  15. 95f5fd9 Bug fix for webrtc issue 1391. Typo in sin_length for socket address. by vikasmarwaha@webrtc.org · 11 years ago
  16. e422fa5 This refactoring CL contains an API to get low level echo metrics stats. by bjornv@webrtc.org · 11 years ago
  17. 3942fd8 This Cl includes by bjornv@webrtc.org · 11 years ago
  18. d3eadf1 Moved the actual calculations to aec_core to avoid passing up low level members. by bjornv@webrtc.org · 11 years ago
  19. a974cea Make VoiceEngineImpl inherit from VoiceEngine. by tommi@webrtc.org · 11 years ago
  20. 1aa1eec Modify SincResampler to build in webrtc. by andrew@webrtc.org · 11 years ago
  21. 3e32dd1 Duplicated sampling frequency multiplier to aecpc_t struct to avoid a getter. by bjornv@webrtc.org · 11 years ago
  22. 64506e2 Roll Chromium revision 176094:182149 by kjellander@webrtc.org · 11 years ago
  23. 8504ad3 Moved debug file handling to aec_core from echo_cancellation.c. This removes dependency on low level member variables. by bjornv@webrtc.org · 11 years ago
  24. 4a0743e Added delay estimation test to audio processing unit tests. by bjornv@webrtc.org · 11 years ago
  25. e740a7b Remove MultiStreamMode from test. by stefan@webrtc.org · 11 years ago
  26. 4c6689a Reset ssrc when calling SetSendCodec. by mflodman@webrtc.org · 11 years ago
  27. abaff53 Fixing lint warnings from previous commit by tina.legrand@webrtc.org · 11 years ago
  28. 1368a6a Import stringize_macros from Chromium. by andrew@webrtc.org · 11 years ago
  29. 260bedc Import SincResampler from Chromium. by andrew@webrtc.org · 11 years ago
  30. 9e605b2 Fix Windows x64 errors in video_codecs_test_framework by kjellander@webrtc.org · 11 years ago
  31. 894a543 Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. by turaj@webrtc.org · 11 years ago
  32. 33c6e92 Sync libvpx and its gyp wrapper from Chromium. by andrew@webrtc.org · 11 years ago
  33. 1fb8372 Increase maximum resolution to 4k x 3k. by fbarchard@google.com · 11 years ago
  34. 28166a5 VCM: Removing frame drop enable from Reset call BUG = 1387 by mikhal@webrtc.org · 11 years ago
  35. 9c4707e Android NDK build tools by kjellander@webrtc.org · 11 years ago
  36. 9cd6011 Fix perf output for audioproc and iSAC fixed-point tests by kjellander@webrtc.org · 11 years ago
  37. 4da62e0 Set SingleStream BWE in unittests. by stefan@webrtc.org · 11 years ago
  38. e3664d5 Set qpMax to 56 in for all VP8 tests. Fixes buildbot breakage. by stefan@webrtc.org · 11 years ago
  39. 6cd34e5 Updates to send side streaming mode: by mikhal@webrtc.org · 11 years ago
  40. 6bcf2ab Update version number to 3.23 by tnakamura@webrtc.org · 11 years ago
  41. 8b5ff39 Fix Win64 build breakage by henrikg@webrtc.org · 11 years ago
  42. 75e6669 Made it possible to render custom call output to file. by phoglund@webrtc.org · 11 years ago
  43. 05a655b Fixed a bug in iSAC transform functions on ARM-Neon platform. Performance unchanged. by kma@webrtc.org · 11 years ago
  44. 89c3de3 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 11 years ago
  45. 34d1110 Enable indefinitely running vie_auto_test option by kjellander@webrtc.org · 11 years ago
  46. 4484b83 Use LOG_F interface for unsupported functions. by andrew@webrtc.org · 11 years ago
  47. f9ca8e1 Added labels in transform_neon.S in iSAC-fix, so the tables be shared with other files in iOS build. Also, moved several code lines in the same file, in case register values cannot be preserved after a function call which could cause a crash in some platforms (e.g. iOS etc.). by kma@webrtc.org · 11 years ago
  48. 313e6b5 Lint-cleaned video and audio receivers. by phoglund@webrtc.org · 11 years ago
  49. db325e2 Updated version number to 3.22 by elham@webrtc.org · 11 years ago
  50. 228e708 Moved almost all payload-related stuff to the payload registry. by phoglund@webrtc.org · 11 years ago
  51. cc895d1 Fixing/disabling Windows x64 warnings by kjellander@webrtc.org · 11 years ago
  52. eae59c5 Exchange TRY by enumerating image formats in Linux video capture by braveyao@webrtc.org · 11 years ago
  53. 9320328 Fix MaxChannels test; 32 -> 100. by andrew@webrtc.org · 11 years ago
  54. 48bfaa8 Remove (in practice) the voice engine channel limit. by andrew@webrtc.org · 11 years ago
  55. d6739c8 Adding a send side API for streaming by mikhal@webrtc.org · 11 years ago
  56. a7761c7 Fix mismatch between different NACK list lengths and packet buffers. by stefan@webrtc.org · 11 years ago
  57. 3442158 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 11 years ago
  58. 7e11001 Adding three frame sizes to Opus by tina.legrand@webrtc.org · 11 years ago
  59. 4d693f9 Implementing stereo support for G.722 by henrik.lundin@webrtc.org · 11 years ago
  60. fd2dd1a Set frame length for frame converting in external renderer by braveyao@webrtc.org · 11 years ago
  61. de55d0c Replaced relative path to reference from <(webrtc_root). by bjornv@webrtc.org · 11 years ago
  62. 2569ab5 Fix propagating RED paylaod-type to ACM. by turaj@webrtc.org · 11 years ago
  63. 57c45c2 Removing a codec from NetEq database has a bug. |funcDurationEst| is not updated. by turaj@webrtc.org · 11 years ago
  64. 6637489 fix for issue 281. by turaj@webrtc.org · 11 years ago
  65. 1f1321c fix issue 1322, accept -1 as default payload-type for redundant coding (FEC). by turaj@webrtc.org · 11 years ago
  66. 62564f1 Adding a max jitter filter to the JB estimate - allowing two modes, one will return the last estimate (current setting), and another will return the max value seen, and allow setting an initial value. by mikhal@webrtc.org · 11 years ago
  67. 8d759af VP8: Making key frame interval a tunnable parameter by mikhal@webrtc.org · 11 years ago
  68. 5f8b39f Fix NetEq4 unit tests for VS2012 by henrik.lundin@webrtc.org · 11 years ago
  69. ea85f98 Removing a hack for CNG by henrik.lundin@webrtc.org · 11 years ago
  70. 1e52bc2 Adding iSAC-fb support by henrik.lundin@webrtc.org · 11 years ago
  71. 3824adf Fix audio_e2e_test command line arguments by kjellander@webrtc.org · 11 years ago
  72. f8dc257 This is a change in the iOS audio device to use VoiceProcessingIO API instead of RemoteIO. This way we don't need to use WebRTC EC and NS because it happens on the device hardware. by andrew@webrtc.org · 11 years ago
  73. 0bfd5f0 Re-committing r3428 by bjornv@webrtc.org · 11 years ago
  74. d8f84db Fixing problems in audio_decoder_unittests by henrik.lundin@webrtc.org · 11 years ago
  75. b51ee74 Disable iSAC fix test in audio_decoder_unittests by henrik.lundin@webrtc.org · 11 years ago
  76. a5b65e0 Re-enabling NetEqDecodingTest.TestBitExactness and .TestNetworkStatistics by henrik.lundin@webrtc.org · 11 years ago
  77. 6bf1c81 Enabling unit tests for NetEq4 in the bots by henrik.lundin@webrtc.org · 11 years ago
  78. 9243982 Fix a few small nits in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  79. 1cd0f31 Remove codereview.settings by henrik.lundin@webrtc.org · 11 years ago
  80. 1dd36c8 Revert 3428 by bjornv@webrtc.org · 11 years ago
  81. 7bf5944 Delay estimator wrapper API changes. This should finalize the changes to delay estimator making it work for multi-probe. by bjornv@webrtc.org · 11 years ago
  82. 11f64d3 Mac 64-bit compatibility for WebRTC. by henrike@webrtc.org · 11 years ago
  83. 54958f4 Initial upload of NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  84. b4575c1 Fix webrtc compilation errors for Chrome Win64 by andrew@webrtc.org · 11 years ago
  85. 184b91c Set working dir for test run script + update resources by kjellander@webrtc.org · 12 years ago
  86. 437f62b Add <(DEPTH) to global includes by kjellander@webrtc.org · 12 years ago
  87. a13470d Optimize NACK list creation. by stefan@webrtc.org · 12 years ago
  88. 294f055 Fix Win64 warnings by kjellander@webrtc.org · 12 years ago
  89. 534c1ce Added tests for multiple near-end support. by bjornv@webrtc.org · 12 years ago
  90. aa3af37 Short CL: only name change. by bjornv@webrtc.org · 12 years ago
  91. 16f79ea Separated far-end handling in BinaryDelayEstimator. by bjornv@webrtc.org · 12 years ago
  92. ceca869 Moving ViE test files and deleting files no longer used. by mflodman@webrtc.org · 12 years ago
  93. 1de9d16 Fix path to perf Python scripts in test.gyp by kjellander@webrtc.org · 12 years ago
  94. d32e047 Reformatted rtp_sender: made lint clean. by phoglund@webrtc.org · 12 years ago
  95. d1f6087 Test launching script by kjellander@webrtc.org · 12 years ago
  96. b29af0e Moved several function pointer declarations in iSAC to isac initialization file. by kma@webrtc.org · 12 years ago
  97. 0664d36 Fixed text relocation code related to ARM assembly code. by kma@webrtc.org · 12 years ago
  98. ad89c14 Revert 3406 by kma@webrtc.org · 12 years ago
  99. 5cd9878 Revert 3405 by niklas.enbom@webrtc.org · 12 years ago
  100. 4e3c377 Moved all function pointer declarations in iSAC to a single place. by kma@webrtc.org · 12 years ago