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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b7e5b2741f9cbd68e7be54e3445d28d266477c92 / voice_engine
  1. b7e5b27 Update makefiles after merge of Chromium at 457b0a1c9412 by Android Chromium Automerger · 10 years ago
  2. cb45b28 Update makefiles after merge of Chromium at 041843cbf814 by Android Chromium Automerger · 10 years ago
  3. f1234f3 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 69370488385c14d73e6ae8a3d5001c42884f9275 by Android Chromium Automerger · 10 years ago
  4. 5191730 Partial revert of r7014 (Android APK refactor) by kjellander@webrtc.org · 10 years ago
  5. b0aac71 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate by minyue@webrtc.org · 10 years ago
  6. 237d079 Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator from both the audio playback thread and the network thread without locking. by stefan@webrtc.org · 10 years ago
  7. 95d2195 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at f8698ce1dacfdcf804809638483adb702760469c by Android Chromium Automerger · 10 years ago
  8. b9d6b2b Android APK tests built from a normal WebRTC checkout. by kjellander@webrtc.org · 10 years ago
  9. 05f7eb6 GN: Implement voice engine, common audio, audio coding and audio processing by kjellander@webrtc.org · 10 years ago
  10. 2b1b7b7 Update makefiles after merge of Chromium at b241671f0248 by Android Chromium Automerger · 10 years ago
  11. 0e2b7ec Remove Android.mk build files. by pbos@webrtc.org · 10 years ago
  12. 7a2cfc5 Remove former team members from OWNERS and WATCHLISTS by kjellander@webrtc.org · 10 years ago
  13. 68fe1fc Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c1696da9a74c7ed4ed793ce993352bd370cfc414 by Torne (Richard Coles) · 10 years ago
  14. f694796 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c2ef523233552340785557abce1129a0f61537eb by Android Chromium Automerger · 10 years ago
  15. 440755a Adding online bitrate change to voe_cmd_test by minyue@webrtc.org · 10 years ago
  16. 5f19242 Update makefiles after merge of Chromium at 288938 by Android Chromium Automerger · 10 years ago
  17. 1bfd540 Adding SetOpusMaxBandwidth in VoE and ACM by minyue@webrtc.org · 10 years ago
  18. 22c283b Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 by henrike@webrtc.org · 10 years ago
  19. ed01936 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at a288b8cbb568cbf1735e6d5d0012524f4f8e5f74 by Android Chromium Automerger · 10 years ago
  20. b9ca3e2 Fixing two bugs in voe_cmd_test. by minyue@webrtc.org · 10 years ago
  21. 8661714 Update makefiles after merge of Chromium at 287308 by Android Chromium Automerger · 10 years ago
  22. fa50854 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4a1b3e3a69d349b0d3e91f607f24e02d8b975688 by Android Chromium Automerger · 10 years ago
  23. ca4bc68 Remove timestamp retreival warning/error. by turaj@webrtc.org · 10 years ago
  24. d89fa97 This is related to an earlier CL of enabling Opus 48 kHz. by minyue@webrtc.org · 10 years ago
  25. 31b38da Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module. by minyue@webrtc.org · 10 years ago
  26. f3d2702 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 82383d9b14ff8e5fedf5a70229eb0ac6b512909a by Android Chromium Automerger · 10 years ago
  27. 9fbd3ec Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. --- by tommi@webrtc.org · 10 years ago
  28. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
  29. 477e6bc Update makefiles after merge of Chromium at 282385 by Android Chromium Automerger · 10 years ago
  30. 10b9861 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 138adbb0bcdab60afda25a8727e5a071abc4ae36 by Android Chromium Automerger · 10 years ago
  31. 6aae61c Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 10 years ago
  32. c7343a3 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at d13c3753199496aeddc73ec88548da73283c312f by Android Chromium Automerger · 10 years ago
  33. 8f02f89 Add ExperimentalNs support in Config by aluebs@webrtc.org · 10 years ago
  34. 841f8c8 Update makefiles after merge of Chromium at 279716 by Android Chromium Automerger · 10 years ago
  35. 4c21d3a Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c34b9e5d5cd44c31c4f9da649b71d0d3132cf516 by Android Chromium Automerger · 10 years ago
  36. 3610f63 GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 10 years ago
  37. ad3bcf4 Update makefiles after merge of Chromium at 278252 by Android Chromium Automerger · 10 years ago
  38. e5a0f26 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 847dfa535730a30d57cf26d788d31070b70a02af by Android Chromium Automerger · 10 years ago
  39. cb4fdd1 Update makefiles after merge of Chromium at 277428 by Android Chromium Automerger · 10 years ago
  40. c7fcada Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 5fcef2b6df45ceab39ee96a616ab0a4d3c63b83a by Android Chromium Automerger · 10 years ago
  41. 00dffd7 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
  42. adda09e Update makefiles after merge of Chromium at 276202 by Android Chromium Automerger · 10 years ago
  43. f006e8d Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 10 years ago
  44. 8097a46 Update makefiles after merge of Chromium at 275833 by Android Chromium Automerger · 10 years ago
  45. 20d9f00 Update makefiles after merge of Chromium at 275661 by Android Chromium Automerger · 10 years ago
  46. 431772f Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 81f8df9af96c6b4bf43234f2a0162146a5da6112 by Android Chromium Automerger · 10 years ago
  47. 6e6292d Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 00d9c49cb076626f711988332749a0ebe8d2a32f by Android Chromium Automerger · 10 years ago
  48. 81f8df9 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 10 years ago
  49. 903e746 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 10 years ago
  50. 00d9c49 Android: cleanup gtest_target_type conditions. by henrike@webrtc.org · 10 years ago
  51. 6038f4c Update makefiles after merge of Chromium at 274467 by Android Chromium Automerger · 10 years ago
  52. 35af59e Add a Reset() method to AudioFrame. by andrew@webrtc.org · 10 years ago
  53. 52dfe97 Update makefiles after merge of Chromium at 273259 by Android Chromium Automerger · 10 years ago
  54. 47475b8 Update makefiles after merge of Chromium at 273188 by Android Chromium Automerger · 10 years ago
  55. dd671de This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate. by minyue@webrtc.org · 10 years ago
  56. 98e1ef1 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at e066d34bb747f730084f1726408ca8348ff25da7 by Android Chromium Automerger · 10 years ago
  57. f290b35 Update makefiles after merge of Chromium at 272740 by Android Chromium Automerger · 10 years ago
  58. 271bf09 Update makefiles after merge of Chromium at 272566 by Android Chromium Automerger · 10 years ago
  59. 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
  60. 774b3d3 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  61. 0a9ed7c Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 10 years ago
  62. 881a32d Calculate capture ntp timestamp in local timebase for decoded audio frame. by wu@webrtc.org · 10 years ago
  63. 28b7c07 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  64. eb6cd40 VoEVolumeTest: Enabled Linux flaky tests by bjornv@webrtc.org · 10 years ago
  65. 73d6d1f Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size by minyue@webrtc.org · 10 years ago
  66. 2681883 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 22f69bd27abc89979460df6d01de8685cb058aab by Android Chromium Automerger · 10 years ago
  67. 5c9cc90 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 07818d1134a8fc4272ca0dd108d8f35d1753f9c3 by Android Chromium Automerger · 10 years ago
  68. 22f69bd Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 10 years ago
  69. 0af662d Update makefiles after merge of Chromium at 271215 by Android Chromium Automerger · 10 years ago
  70. 618be31 Update makefiles after merge of Chromium at 270770 by Torne (Richard Coles) · 10 years ago
  71. 4eff397 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at da7c539c377367da25fc913d4399c5f0f69764ad by Torne (Richard Coles) · 10 years ago
  72. 8a557b5 Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. by solenberg@webrtc.org · 10 years ago
  73. 7d20dda Remove the use of AudioFrame::energy_ from AudioProcessing and VoE. by andrew@webrtc.org · 10 years ago
  74. 0b8a1c4 Add webrtc field trials API. by andresp@webrtc.org · 10 years ago
  75. 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
  76. 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
  77. d658c11 Update makefiles after merge of Chromium at 269467 by Torne (Richard Coles) · 10 years ago
  78. 265cb1b VoEVolumeTest: Adds error return tests. by bjornv@webrtc.org · 10 years ago
  79. f33a674 Make vie/voe_auto_test accept non-supported flags without error. by kjellander@webrtc.org · 10 years ago
  80. efe9461 Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 by bjornv@webrtc.org · 10 years ago
  81. 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
  82. 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
  83. 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
  84. e639a03 Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 10 years ago
  85. b8db407 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 10 years ago
  86. 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
  87. 5b371a8 Update makefiles after merge of Chromium at 269041 by Android Chromium Automerger · 10 years ago
  88. 0a2aa9a Update makefiles after merge of Chromium at 269030 by Android Chromium Automerger · 10 years ago
  89. 625ace9 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 39d9fa5486157fb4b3ab28ae403aeaa6d651e92b by Android Chromium Automerger · 10 years ago
  90. 0de5f22 Update makefiles after merge of Chromium at 268379 by Android Chromium Automerger · 10 years ago
  91. 3cd0f7c Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 10 years ago
  92. 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
  93. 3a802b9 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at cfed80f78803395dd066261afb5c5d99e5048d5d by Android Chromium Automerger · 10 years ago
  94. c1878ac Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 10 years ago
  95. 109a4e7 Update makefiles after merge of Chromium at 266543 by Android Chromium Automerger · 10 years ago
  96. ba47616 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  97. 0c9f7d3 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 1d95c5aecd0a4924f39f24834fb06d06e61f181e by Android Chromium Automerger · 10 years ago
  98. 47e54ba * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. by wu@webrtc.org · 10 years ago
  99. 0197db2 Update makefiles after merge of Chromium at 265680 by Android Chromium Automerger · 10 years ago
  100. 91734bd Update makefiles after merge of Chromium at 265607 by Android Chromium Automerger · 10 years ago