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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / f89ce467d1977ea1d867be4e5be2ccd9a206113c
  1. e066d34 Fix a bug preventing FilePlayer from playing encoded wav files by henrik.lundin@webrtc.org · 10 years ago
  2. 99153ba First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class. by asapersson@webrtc.org · 10 years ago
  3. 01d8e22 vie_autotest_android.cc: stop referring to undefined functions. by fischman@webrtc.org · 10 years ago
  4. b2e746e Rebase webrtc/base with r6232: by henrike@webrtc.org · 10 years ago
  5. 0631e37 Thread: delete racy API (Release()) and fix racy code (started()). by fischman@webrtc.org · 10 years ago
  6. 6db65e3 PRESUBMIT.py: accept variants on the copyright message that are present in the codebase. by fischman@webrtc.org · 10 years ago
  7. 13978fc Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix. by turaj@webrtc.org · 10 years ago
  8. 91c0a25 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED by minyue@webrtc.org · 10 years ago
  9. c78d40d Revert "Revert "Remove VideoSendStreamInput::PutFrame."" by pbos@webrtc.org · 10 years ago
  10. f3f2dba Revert "Remove VideoSendStreamInput::PutFrame." by pbos@webrtc.org · 10 years ago
  11. 0c60ee8 Remove VideoSendStreamInput::PutFrame. by pbos@webrtc.org · 10 years ago
  12. 7f54561 Fix deadlock in RegisterPreDecodeImageCallback. by pbos@webrtc.org · 10 years ago
  13. dc1a607 Bump WebRTC version number to 3.54 TBR=wu@webrtc.org by tnakamura@webrtc.org · 10 years ago
  14. e38123c Adds missing include of assert header. by henrike@webrtc.org · 10 years ago
  15. 8d385d8 WebRTCDemo: move the deletion of CritSect to end of the dtor to fix a crash in Android video renderer. by braveyao@webrtc.org · 10 years ago
  16. 774b3d3 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  17. 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
  18. 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
  19. b45df1c Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect. by jiayl@webrtc.org · 10 years ago
  20. 0a9ed7c Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 10 years ago
  21. 8520b33 Revert 6208 "Patch from henrike@webrtc.org" by mcasas@webrtc.org · 10 years ago
  22. 1528532 Patch from henrike@webrtc.org by mcasas@webrtc.org · 10 years ago
  23. 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
  24. 881a32d Calculate capture ntp timestamp in local timebase for decoded audio frame. by wu@webrtc.org · 10 years ago
  25. da3266d Enabling NetEq bit-exactness test for Win x64 by henrik.lundin@webrtc.org · 10 years ago
  26. 28b7c07 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  27. b139e8a Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky. by stefan@webrtc.org · 10 years ago
  28. b5e74d6 Revert r6198 "Expose the original packet length in in the RTP play tools." by stefan@webrtc.org · 10 years ago
  29. 2ffdd60 Expose the original packet length in in the RTP play tools. by stefan@webrtc.org · 10 years ago
  30. 294081c Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8. by stefan@webrtc.org · 10 years ago
  31. 3d12566 Suppress GMOCK printouts from TestVideoSenderWithVp8 by henrik.lundin@webrtc.org · 10 years ago
  32. eb6cd40 VoEVolumeTest: Enabled Linux flaky tests by bjornv@webrtc.org · 10 years ago
  33. b5b8648 Add NACK and RPSI packet types to RTCP packet builder. by asapersson@webrtc.org · 10 years ago
  34. 73d6d1f Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size by minyue@webrtc.org · 10 years ago
  35. d4e20db Remove IOKit linkage from iOS builds. by tkchin@webrtc.org · 10 years ago
  36. 22f69bd Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 10 years ago
  37. 5359a2e Avoid NACK-list flush error on keyframe packets. by pbos@webrtc.org · 10 years ago
  38. 07818d1 Don't crash if a frame returned from the decoder is too old. by stefan@webrtc.org · 10 years ago
  39. e4ba5ce Use the new gyp_var_prefix local variable set by gyp instead of the by michaelbai@google.com · 10 years ago
  40. 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
  41. e3acf53 common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16 by bjornv@webrtc.org · 10 years ago
  42. 8a557b5 Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. by solenberg@webrtc.org · 10 years ago
  43. 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
  44. f4d24ed common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16 by bjornv@webrtc.org · 10 years ago
  45. e3d3f0b common_audio/signal_processing: Removed macro WEBRTC_SPL_MAX_SEED_USED by bjornv@webrtc.org · 10 years ago
  46. 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
  47. e6c8658 removed webrtc_base_tests_utils from merge libs as it was breaking some builds. by henrike@webrtc.org · 10 years ago
  48. b08a990 Made the presubmit script accept license headers back to 2003 by henrike@webrtc.org · 10 years ago
  49. 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
  50. da7c539 Fix Windows debug compile of overrides/ logging. by pbos@webrtc.org · 10 years ago
  51. e9b4340 Revert "Revert "Audio processing: Feed each processing step its choice by mflodman@webrtc.org · 10 years ago
  52. 66b1bf8 Fix Win VideoSendStream::...::ToString() compiles. by pbos@webrtc.org · 10 years ago
  53. 7e68693 Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 10 years ago
  54. 3362d42 common_audio: Removes unused macros by bjornv@webrtc.org · 10 years ago
  55. c17094a Re-enable almost all NetEqDecodingTests for Android by henrik.lundin@webrtc.org · 10 years ago
  56. 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
  57. 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
  58. 7d20dda Remove the use of AudioFrame::energy_ from AudioProcessing and VoE. by andrew@webrtc.org · 10 years ago
  59. 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
  60. 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
  61. bc57e0f Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 10 years ago
  62. 0b8a1c4 Add webrtc field trials API. by andresp@webrtc.org · 10 years ago
  63. 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
  64. 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
  65. 54c9b21 Re-enable NetEqExternalDecoderTest for Android by henrik.lundin@webrtc.org · 10 years ago
  66. 8a35df1 Re-enable NetEQ DecoderDatabase test for Android by henrik.lundin@webrtc.org · 10 years ago
  67. 339626b Revert "Audio processing: Feed each processing step its choice of int or float data" by mflodman@webrtc.org · 10 years ago
  68. ff4e210 Re-enable the BitrateEstimatorTest cases for the Call API. by solenberg@webrtc.org · 10 years ago
  69. 3e4630d Remove all use of AudioFrame::energy_ from AudioCodingModule by henrik.lundin@webrtc.org · 10 years ago
  70. 265cb1b VoEVolumeTest: Adds error return tests. by bjornv@webrtc.org · 10 years ago
  71. ef3ff93 Audio processing: Feed each processing step its choice of int or float data by kwiberg@webrtc.org · 10 years ago
  72. fcdc5b5 Remove WEBRTC_TRACE use in video_capture/ by pbos@webrtc.org · 10 years ago
  73. 3468f20 Remove WEBRTC_TRACE uses in video_engine/ by pbos@webrtc.org · 10 years ago
  74. f33a674 Make vie/voe_auto_test accept non-supported flags without error. by kjellander@webrtc.org · 10 years ago
  75. 47be73b Adds a modified copy of talk/base to webrtc/base. It is the first step in by henrike@webrtc.org · 10 years ago
  76. efe9461 Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 by bjornv@webrtc.org · 10 years ago
  77. 8378f1e Revert "FieldTrial implementation for webrtc." (rev 6089) by andresp@webrtc.org · 10 years ago
  78. c773ded Reduced kMaxSampleDiffMs (limit to 22fps). by asapersson@webrtc.org · 10 years ago
  79. 11de507 Move gflags usage to video_loopback. by pbos@webrtc.org · 10 years ago
  80. ae9a21f Deleting all NetEq3 files by henrik.lundin@webrtc.org · 10 years ago
  81. 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
  82. 50daa53 Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..." by perkj@webrtc.org · 10 years ago
  83. c9ccea3 Deleting all ACM1 files by henrik.lundin@webrtc.org · 10 years ago
  84. 068cd6f Fix failing test introduced with r6111. by stefan@webrtc.org · 10 years ago
  85. b50d671 Fixes log spam introduced with r6041. by stefan@webrtc.org · 10 years ago
  86. 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
  87. 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
  88. 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
  89. 3f0b9bf Echo cancellation functions docs: Follow style guide w.r.t. placement of * by kwiberg@webrtc.org · 10 years ago
  90. 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
  91. 6ca6896 One of the NetEq methods needs to be virtual. by turaj@webrtc.org · 10 years ago
  92. 53c1d3c Modifying neteq.gyp by turaj@webrtc.org · 10 years ago
  93. e639a03 Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 10 years ago
  94. b8db407 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 10 years ago
  95. 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
  96. d88b46f FieldTrial implementation for webrtc. by andresp@webrtc.org · 10 years ago
  97. 6ecc773 Raise kViEMaxNumberOfChannels from 32 to 64 by wu@webrtc.org · 10 years ago
  98. e1f0419 Updated WebRTC version to 3.53 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  99. ecbc55f AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_ by kwiberg@webrtc.org · 10 years ago
  100. 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