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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 11e96c736207147b6c79963073f89112a8dbefb4
  1. 11e96c7 Rebase webrtc/base 6163:6216 (svn diff -r 6163:6216 http://webrtc.googlecode.com/svn/trunk/talk/base, apply diff manually) by henrike@webrtc.org · 10 years ago
  2. f85a9af Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h. by henrike@webrtc.org · 10 years ago
  3. b45df1c Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect. by jiayl@webrtc.org · 10 years ago
  4. 0a9ed7c Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 10 years ago
  5. 8520b33 Revert 6208 "Patch from henrike@webrtc.org" by mcasas@webrtc.org · 10 years ago
  6. 1528532 Patch from henrike@webrtc.org by mcasas@webrtc.org · 10 years ago
  7. dbd03e5 WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker by braveyao@webrtc.org · 10 years ago
  8. 881a32d Calculate capture ntp timestamp in local timebase for decoded audio frame. by wu@webrtc.org · 10 years ago
  9. da3266d Enabling NetEq bit-exactness test for Win x64 by henrik.lundin@webrtc.org · 10 years ago
  10. 28b7c07 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  11. b139e8a Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky. by stefan@webrtc.org · 10 years ago
  12. b5e74d6 Revert r6198 "Expose the original packet length in in the RTP play tools." by stefan@webrtc.org · 10 years ago
  13. 2ffdd60 Expose the original packet length in in the RTP play tools. by stefan@webrtc.org · 10 years ago
  14. 294081c Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8. by stefan@webrtc.org · 10 years ago
  15. 3d12566 Suppress GMOCK printouts from TestVideoSenderWithVp8 by henrik.lundin@webrtc.org · 10 years ago
  16. eb6cd40 VoEVolumeTest: Enabled Linux flaky tests by bjornv@webrtc.org · 10 years ago
  17. b5b8648 Add NACK and RPSI packet types to RTCP packet builder. by asapersson@webrtc.org · 10 years ago
  18. 73d6d1f Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size by minyue@webrtc.org · 10 years ago
  19. d4e20db Remove IOKit linkage from iOS builds. by tkchin@webrtc.org · 10 years ago
  20. 22f69bd Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 10 years ago
  21. 5359a2e Avoid NACK-list flush error on keyframe packets. by pbos@webrtc.org · 10 years ago
  22. 07818d1 Don't crash if a frame returned from the decoder is too old. by stefan@webrtc.org · 10 years ago
  23. e4ba5ce Use the new gyp_var_prefix local variable set by gyp instead of the by michaelbai@google.com · 10 years ago
  24. 5199d1d libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict. by henrike@webrtc.org · 10 years ago
  25. e3acf53 common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16 by bjornv@webrtc.org · 10 years ago
  26. 8a557b5 Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. by solenberg@webrtc.org · 10 years ago
  27. bd49ac2 Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main} by andresp@webrtc.org · 10 years ago
  28. f4d24ed common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16 by bjornv@webrtc.org · 10 years ago
  29. e3d3f0b common_audio/signal_processing: Removed macro WEBRTC_SPL_MAX_SEED_USED by bjornv@webrtc.org · 10 years ago
  30. 752b879 * Implement WindowsRealTimeClock::CurrentTimeVal with GetSystemTimeAsFileTime as it supposes to return a POSIX gettimeofday, so that later it can be converted to NTP timee correctly. by wu@webrtc.org · 10 years ago
  31. e6c8658 removed webrtc_base_tests_utils from merge libs as it was breaking some builds. by henrike@webrtc.org · 10 years ago
  32. b08a990 Made the presubmit script accept license headers back to 2003 by henrike@webrtc.org · 10 years ago
  33. 698ee5a Rebase webrtc/base 6129:6163 (svn diff -r 6129:6163 http://webrtc.googlecode.com/svn/trunk/talk/base apply diff manually) by henrike@webrtc.org · 10 years ago
  34. da7c539 Fix Windows debug compile of overrides/ logging. by pbos@webrtc.org · 10 years ago
  35. e9b4340 Revert "Revert "Audio processing: Feed each processing step its choice by mflodman@webrtc.org · 10 years ago
  36. 66b1bf8 Fix Win VideoSendStream::...::ToString() compiles. by pbos@webrtc.org · 10 years ago
  37. 7e68693 Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 10 years ago
  38. 3362d42 common_audio: Removes unused macros by bjornv@webrtc.org · 10 years ago
  39. c17094a Re-enable almost all NetEqDecodingTests for Android by henrik.lundin@webrtc.org · 10 years ago
  40. ca2c70f WebRTCDemo: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process. by braveyao@webrtc.org · 10 years ago
  41. dd0f8b2 Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log. by wu@webrtc.org · 10 years ago
  42. 7d20dda Remove the use of AudioFrame::energy_ from AudioProcessing and VoE. by andrew@webrtc.org · 10 years ago
  43. b2eea5c Added namespace rtc to some base classes and functions. It was causing linker error in the FYI bots: http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Android%20Builder%20%28dbg%29/builds/1808/steps/compile/logs/stdio but also, not doing it pollutes the global namespace. by henrike@webrtc.org · 10 years ago
  44. 2fae0d1 Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe. by wu@webrtc.org · 10 years ago
  45. bc57e0f Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 10 years ago
  46. 0b8a1c4 Add webrtc field trials API. by andresp@webrtc.org · 10 years ago
  47. c4e54b6 Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  48. 7b2651a Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  49. 54c9b21 Re-enable NetEqExternalDecoderTest for Android by henrik.lundin@webrtc.org · 10 years ago
  50. 8a35df1 Re-enable NetEQ DecoderDatabase test for Android by henrik.lundin@webrtc.org · 10 years ago
  51. 339626b Revert "Audio processing: Feed each processing step its choice of int or float data" by mflodman@webrtc.org · 10 years ago
  52. ff4e210 Re-enable the BitrateEstimatorTest cases for the Call API. by solenberg@webrtc.org · 10 years ago
  53. 3e4630d Remove all use of AudioFrame::energy_ from AudioCodingModule by henrik.lundin@webrtc.org · 10 years ago
  54. 265cb1b VoEVolumeTest: Adds error return tests. by bjornv@webrtc.org · 10 years ago
  55. ef3ff93 Audio processing: Feed each processing step its choice of int or float data by kwiberg@webrtc.org · 10 years ago
  56. fcdc5b5 Remove WEBRTC_TRACE use in video_capture/ by pbos@webrtc.org · 10 years ago
  57. 3468f20 Remove WEBRTC_TRACE uses in video_engine/ by pbos@webrtc.org · 10 years ago
  58. f33a674 Make vie/voe_auto_test accept non-supported flags without error. by kjellander@webrtc.org · 10 years ago
  59. 47be73b Adds a modified copy of talk/base to webrtc/base. It is the first step in by henrike@webrtc.org · 10 years ago
  60. efe9461 Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 by bjornv@webrtc.org · 10 years ago
  61. 8378f1e Revert "FieldTrial implementation for webrtc." (rev 6089) by andresp@webrtc.org · 10 years ago
  62. c773ded Reduced kMaxSampleDiffMs (limit to 22fps). by asapersson@webrtc.org · 10 years ago
  63. 11de507 Move gflags usage to video_loopback. by pbos@webrtc.org · 10 years ago
  64. ae9a21f Deleting all NetEq3 files by henrik.lundin@webrtc.org · 10 years ago
  65. ad230ee The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy. by henrik.lundin@webrtc.org · 10 years ago
  66. 50daa53 Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..." by perkj@webrtc.org · 10 years ago
  67. c9ccea3 Deleting all ACM1 files by henrik.lundin@webrtc.org · 10 years ago
  68. 068cd6f Fix failing test introduced with r6111. by stefan@webrtc.org · 10 years ago
  69. b50d671 Fixes log spam introduced with r6041. by stefan@webrtc.org · 10 years ago
  70. 04e6703 Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. by henrike@webrtc.org · 10 years ago
  71. 7f5e297 Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  72. d2632a0 Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  73. 3f0b9bf Echo cancellation functions docs: Follow style guide w.r.t. placement of * by kwiberg@webrtc.org · 10 years ago
  74. 12884ba Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  75. 6ca6896 One of the NetEq methods needs to be virtual. by turaj@webrtc.org · 10 years ago
  76. 53c1d3c Modifying neteq.gyp by turaj@webrtc.org · 10 years ago
  77. e639a03 Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 10 years ago
  78. b8db407 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 10 years ago
  79. a4943ea Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  80. d88b46f FieldTrial implementation for webrtc. by andresp@webrtc.org · 10 years ago
  81. 6ecc773 Raise kViEMaxNumberOfChannels from 32 to 64 by wu@webrtc.org · 10 years ago
  82. e1f0419 Updated WebRTC version to 3.53 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  83. ecbc55f AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_ by kwiberg@webrtc.org · 10 years ago
  84. d2fb259 Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. by wu@webrtc.org · 10 years ago
  85. 519f74e Fix odd codes in video_capture on Mac. by braveyao@webrtc.org · 10 years ago
  86. a4bb5f2 video_render.gypi: clean up some libraries directives to be more specific. by fischman@webrtc.org · 10 years ago
  87. 39d9fa5 Remove timestamp_extrapolator's dependency to Clock and vcm defines. by wu@webrtc.org · 10 years ago
  88. 3cd0f7c Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 10 years ago
  89. 60c62f8 Remove ALLOW_UNUSED. by andrew@webrtc.org · 10 years ago
  90. 1cbc360 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. by wu@webrtc.org · 10 years ago
  91. 6b6e3ea Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
  92. 1f415cb Revert 6048 "Implement the Windows screen capturer using the Mag..." by tina.legrand@webrtc.org · 10 years ago
  93. c9d0634 WebRTCDemo: correct set trace filter operation. by braveyao@webrtc.org · 10 years ago
  94. 66a2eae Add ALLOW_UNUSED and update COMPILE_ASSERT to Chromium's latest. by andrew@webrtc.org · 10 years ago
  95. 547a7cd Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
  96. 99ec896 Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test. by henrika@webrtc.org · 10 years ago
  97. d885109 Pointers were not dereferenced in GetRtpStatistics. by asapersson@webrtc.org · 10 years ago
  98. 4d61a36 Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. by stefan@webrtc.org · 10 years ago
  99. c1878ac Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 10 years ago
  100. 785c2fd Fix constness of AudioBuffer accessors. by andrew@webrtc.org · 10 years ago