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webrtc / src / 4161715e3f7e744bc9ef3d3ae437da1e8e4de38d
  1. 4161715 Remove ChangeUniqueID. by tommi@webrtc.org · 10 years ago
  2. 1ece0cb Revert "Remove <(webrtc_root) from source file entries." by kjellander@webrtc.org · 10 years ago
  3. a26f511 Remove frame copy in ViEExternalRendererImpl::RenderFrame by magjed@webrtc.org · 10 years ago
  4. a87c398 Move audio_codec_speed_tests into include_tests==1 condition. by kjellander@webrtc.org · 10 years ago
  5. 2d2a1f9 Remove <(webrtc_root) from source file entries. by kjellander@webrtc.org · 10 years ago
  6. 73ca194 Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h by kwiberg@webrtc.org · 10 years ago
  7. 43c8839 Allow rtp packet history to dynamically expand in size. by sprang@webrtc.org · 10 years ago
  8. 827d7e8 Change AsyncInvoker to store its closure in a scoped_refptr instead of using a raw pointer. by perkj@webrtc.org · 10 years ago
  9. a742cb1 Enable DTLS for peerconnection example. If it's a loopback test, then we recreate another peerconnection with DTLS off. by braveyao@webrtc.org · 10 years ago
  10. f17ee9c Add case to ApmTest.Process to test the extended filter mode by aluebs@webrtc.org · 10 years ago
  11. e7a4a12 Add arraysize() macro from Chromium, and make use of it in a few places. by pkasting@chromium.org · 10 years ago
  12. 035e912 Move channel_buffer.{h,cc} to common_audio. by kjellander@webrtc.org · 10 years ago
  13. a67ca1a Only report the first rtp packet because it indicates the media has started flowing. by honghaiz@google.com · 10 years ago
  14. a094cac Add stats for network merge. by guoweis@webrtc.org · 10 years ago
  15. 7d2b6a9 Enable Clang warning implicit-fallthrough and annotate the code. by kjellander@webrtc.org · 10 years ago
  16. a907e01 Adding constness. by tommi@webrtc.org · 10 years ago
  17. 664ccb7 Reland r8125: Modify some tests to never use DTX disable mode by henrik.lundin@webrtc.org · 10 years ago
  18. 37c0559 Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets). by asapersson@webrtc.org · 10 years ago
  19. 22c2f05 Add "score" unit to SSIM perf score output. by kjellander@webrtc.org · 10 years ago
  20. 4aecd00 Add support for 40 and 60 ms frames to AudioEncoderIlbc by henrik.lundin@webrtc.org · 10 years ago
  21. 2a6558c Make sure ByteReader<T>::Read* is properly constified. by sprang@webrtc.org · 10 years ago
  22. 7aef80c GN: Remove webrtc_base target in favor for rtc_base. by kjellander@webrtc.org · 10 years ago
  23. 9b64a6e Adjust parameter in videoprocessor_integrationtest for VP9. by marpan@webrtc.org · 10 years ago
  24. dc8a9da Adjust qp-max settinhg in VP9 wrapper. by marpan@webrtc.org · 10 years ago
  25. 922cfcd Use non-zero data in AudioRingBufferTest. by andrew@webrtc.org · 10 years ago
  26. 36401ab Update GAE API paths for join/leave. by tkchin@webrtc.org · 10 years ago
  27. 8bb32d6 Minor updates to AudioEncoderCng by henrik.lundin@webrtc.org · 10 years ago
  28. db1ebf6 Add jakehilton@gmail.com to AUTHORS by tnakamura@webrtc.org · 10 years ago
  29. 478cedc Add new methods to AudioEncoder interface by henrik.lundin@webrtc.org · 10 years ago
  30. 5614cf1 audio_processing: Use fixed aggregation window in delay metrics by bjornv@webrtc.org · 10 years ago
  31. 6e25182 Whitespace change after enabling gnumbd by kjellander@webrtc.org · 10 years ago
  32. ccd608e Whitespace change for git updater by kjellander@webrtc.org · 10 years ago
  33. 0bc73a1 Whitespace change to trigger git updater by kjellander@webrtc.org · 10 years ago
  34. f68ffca Add PRESUBMIT check for GYP files including source files above itself. by kjellander@webrtc.org · 10 years ago
  35. 76e5e20 Roll chromium_revision 4664fe0..9070a80 (312733:313233) by kjellander@webrtc.org · 10 years ago
  36. 273fbbb Update StreamDataCounter with FEC bytes. by asapersson@webrtc.org · 10 years ago
  37. 70117a8 AEC: Implements a new function for calculating delay metrics by bjornv@webrtc.org · 10 years ago
  38. fc5ad95 Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139 by magjed@webrtc.org · 10 years ago
  39. 8501ee6 Support VP8 HW decoding on devices with Exynos codec. by glaznev@webrtc.org · 10 years ago
  40. df9a41d Fix bug in GetREDStatus(): it doesn't actually return the current status. by pkasting@chromium.org · 10 years ago
  41. 82415e3 Update AppRTCDemo to use renamed GAE messages. by glaznev@webrtc.org · 10 years ago
  42. 041035b Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface. by andrew@webrtc.org · 10 years ago
  43. 4dba2e9 Consolidate anonymous namespace content and file-static methods to all be in the by pkasting@chromium.org · 10 years ago
  44. d7e34e1 Make it easier to use external libyuv + cleanup GYP files. by kjellander@webrtc.org · 10 years ago
  45. d25c034 Refactor common_audio/vad: Removed usage of macro WEBRTC_SPL_MUL_16_16() by bjornv@webrtc.org · 10 years ago
  46. 04cd466 Move ThreadChecker into rtc_base_approved. by tommi@webrtc.org · 10 years ago
  47. 38d11b8 Enable encoder multi-threading for VP9. by marpan@webrtc.org · 10 years ago
  48. 6f200b5 Temporarily revert r8147 ("Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h") by kwiberg@webrtc.org · 10 years ago
  49. b6fab2b Introduce rtc::CheckedDivExact by henrik.lundin@webrtc.org · 10 years ago
  50. 19eb4e4 Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h by kwiberg@webrtc.org · 10 years ago
  51. 995b4c9 Remove win_asan trybot from PRESUBMIT.py by kjellander@webrtc.org · 10 years ago
  52. acb8085 Roll chromium_revision c086b4e..4664fe0 (312108:312733) by kjellander@webrtc.org · 10 years ago
  53. 7519de5 Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..." by tkchin@webrtc.org · 10 years ago
  54. 0f98844 Revert 8139 "Implement elapsed time and capture start NTP time e..." by tkchin@webrtc.org · 10 years ago
  55. dacdd94 Reland r7980: by jiayl@webrtc.org · 10 years ago
  56. 8919cfe Change a GYP reference to cpufeatures.gypi by fdegans@chromium.org · 10 years ago
  57. ad3ee2c Implement elapsed time and capture start NTP time estimation. by pbos@webrtc.org · 10 years ago
  58. a02d768 Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness by kjellander@webrtc.org · 10 years ago
  59. 456f014 Re-allowing RED in voice engine. by minyue@webrtc.org · 10 years ago
  60. 182ea46 Remove frame copy in ViEExternalRendererImpl::RenderFrame by magjed@webrtc.org · 10 years ago
  61. 73ee453 Switch to use range based loops in the BWE simulation framework. by stefan@webrtc.org · 10 years ago
  62. 36d5c3c Leave BIO_METHOD non-const. by davidben@webrtc.org · 10 years ago
  63. 586f2ed Change GetStreamBySsrc to not copy StreamParams. by tommi@webrtc.org · 10 years ago
  64. 7e5b380 Fix a crash in AllocationSequence. Internal bug 19074679. by jiayl@webrtc.org · 10 years ago
  65. ff108fe Revert 8125 "Modify some tests to never use DTX disable mode" by kjellander@webrtc.org · 10 years ago
  66. b40c7bb Change sprintf use in talk samples to snprintf by jlmiller@webrtc.org · 10 years ago
  67. ea1c842 Correct GetDriveType error handling. by jlmiller@webrtc.org · 10 years ago
  68. 043db24 Modify some tests to never use DTX disable mode by henrik.lundin@webrtc.org · 10 years ago
  69. e5251ad Integrate send-side BWE into simulation framework. by stefan@webrtc.org · 10 years ago
  70. cfd82df Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. by asapersson@webrtc.org · 10 years ago
  71. 3dd33a6 Fix bug in thresholds for bitrate probing and adjust thresholds to allow a larger dispersion and concentration for successful probes. by stefan@webrtc.org · 10 years ago
  72. fbd37bd Make iSAC SWB own its decoder by henrik.lundin@webrtc.org · 10 years ago
  73. cceb166 Fix a use-after-free when sending queued messages is aborted for blocked channel. by jiayl@webrtc.org · 10 years ago
  74. e65d9d9 Fix an unitialized variable warning. by andrew@webrtc.org · 10 years ago
  75. c429b82 GN: Prepare to remove webrtc_base target by kjellander@webrtc.org · 10 years ago
  76. c78d81a Re-land "Support 48kHz in AEC" by aluebs@webrtc.org · 10 years ago
  77. e81c5d6 Fix TransientDetectorTest in modules_unittests on Android ARM64 by aluebs@webrtc.org · 10 years ago
  78. 11af039 Disable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_voip on ARM64. by minyue@webrtc.org · 10 years ago
  79. df7b65b Change CreateOrGetReportBlockInformation to have one return path. by asapersson@webrtc.org · 10 years ago
  80. f938922 Simplify and guard access to WindowsRealTimeClock. by pbos@webrtc.org · 10 years ago
  81. 4fb7e25 Update StatsReport and by extension StatsCollector to reduce data copying. by tommi@webrtc.org · 10 years ago
  82. f66a6b2 Remove unnecessary dependencies from webrtc_all target. by kjellander@webrtc.org · 10 years ago
  83. e7358ea Only report fraction of lost packets if report_block_stats has been updated. by asapersson@webrtc.org · 10 years ago
  84. 9ffd8fe Indentation changes. by asapersson@webrtc.org · 10 years ago
  85. fedb9ea Correct the class name in peerconnection_jni.cc. by braveyao@webrtc.org · 10 years ago
  86. 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
  87. cbacd9e Bump to version 41. by tnakamura@webrtc.org · 10 years ago
  88. 7dba786 Setting Opus target application. by minyue@webrtc.org · 10 years ago
  89. 853049f Move internal capture+render to build_with_chromium==0 condition by kjellander@webrtc.org · 10 years ago
  90. 511ab3e Roll chromium_revision a6eafec..c086b4e by kjellander@webrtc.org · 10 years ago
  91. ee0c100 Revert 8080 "Support 48kHz in AEC" by tina.legrand@webrtc.org · 10 years ago
  92. f88f88e Remove webrtc/base/compile_assert.h by kwiberg@webrtc.org · 10 years ago
  93. 9691b36 Cleanup for Rtp Rtcp API test. by changbin.shao@intel.com · 10 years ago
  94. 8e327c4 Update StatsCollector's interface in preparation of more changes. by tommi@webrtc.org · 10 years ago
  95. 43e54e3 Revert 8095 "Update StatsCollector's interface in preparation of..." by tommi@webrtc.org · 10 years ago
  96. 5b76fd7 Update StatsCollector's interface in preparation of more changes. by tommi@webrtc.org · 10 years ago
  97. 474e36e Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps. by stefan@webrtc.org · 10 years ago
  98. f9d3555 Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test. by phoglund@webrtc.org · 10 years ago
  99. ce3ac53 Adding TRYSERVER_PROJECT to codereview.settings. by kjellander@webrtc.org · 10 years ago
  100. 018c087 Add /talk/examples/androidtests/{bin,gen} to .gitignore. by kjellander@webrtc.org · 10 years ago