1. b0aac71 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate by minyue@webrtc.org · 10 years ago
  2. 0e2b7ec Remove Android.mk build files. by pbos@webrtc.org · 10 years ago
  3. 7a2cfc5 Remove former team members from OWNERS and WATCHLISTS by kjellander@webrtc.org · 10 years ago
  4. 440755a Adding online bitrate change to voe_cmd_test by minyue@webrtc.org · 10 years ago
  5. 1bfd540 Adding SetOpusMaxBandwidth in VoE and ACM by minyue@webrtc.org · 10 years ago
  6. 22c283b Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 by henrike@webrtc.org · 10 years ago
  7. b9ca3e2 Fixing two bugs in voe_cmd_test. by minyue@webrtc.org · 10 years ago
  8. d89fa97 This is related to an earlier CL of enabling Opus 48 kHz. by minyue@webrtc.org · 10 years ago
  9. 6aae61c Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 10 years ago
  10. 8f02f89 Add ExperimentalNs support in Config by aluebs@webrtc.org · 10 years ago
  11. 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
  12. eb6cd40 VoEVolumeTest: Enabled Linux flaky tests by bjornv@webrtc.org · 10 years ago
  13. 73d6d1f Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size by minyue@webrtc.org · 10 years ago
  14. 8a557b5 Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. by solenberg@webrtc.org · 10 years ago
  15. 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
  16. 265cb1b VoEVolumeTest: Adds error return tests. by bjornv@webrtc.org · 10 years ago
  17. f33a674 Make vie/voe_auto_test accept non-supported flags without error. by kjellander@webrtc.org · 10 years ago
  18. efe9461 Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 by bjornv@webrtc.org · 10 years ago
  19. 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
  20. 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
  21. 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
  22. e639a03 Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 10 years ago
  23. b8db407 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 10 years ago
  24. 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
  25. 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
  26. 945d969 Remove ACM1/ACM2 switching from VoiceEngine tests by henrik.lundin@webrtc.org · 10 years ago
  27. 692224a Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  28. 66ccaff Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  29. 1a07e42 Re-enable AGC tests: by aluebs@webrtc.org · 10 years ago
  30. 8fb9156 iOS: baby steps to being able to include_tests=1 by fischman@webrtc.org · 10 years ago
  31. 4ff0eda Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  32. 56aeb0e Make ACM2 the default in voe_cmd_test. by andrew@webrtc.org · 10 years ago
  33. 172f42a VoiceEngine(iOS & Android): removed NOT_SUPPORTED by fischman@webrtc.org · 10 years ago
  34. 2bf87a2 Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out. by solenberg@webrtc.org · 10 years ago
  35. fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  36. 650772a Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered. by henrika@webrtc.org · 10 years ago
  37. 9a82322 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. by wu@webrtc.org · 10 years ago
  38. 4845ee0 Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 10 years ago
  39. f6c4fc3 Remove obsolete voe_unit_test. by solenberg@webrtc.org · 10 years ago
  40. a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  41. 9a95ac2 Remove unused and not working voe_extended_test. by solenberg@webrtc.org · 10 years ago
  42. 4c20478 Reduce mixing threshold in test to avoid flakiness. by andrew@webrtc.org · 10 years ago
  43. 8ad10ec Restore mixing integration tests. by andrew@webrtc.org · 10 years ago
  44. 05384c0 Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents" by pbos@webrtc.org · 10 years ago
  45. 6b15b51 Fix locking in LoopBackTransport::StorePacket. by pbos@webrtc.org · 10 years ago
  46. a84ddcd Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents by marpan@webrtc.org · 10 years ago
  47. c13a537 Move out typing detection to its own class. by henrikg@webrtc.org · 10 years ago
  48. 0584274 Output logs to stderr from voe_cmd_test by default. by andrew@webrtc.org · 11 years ago
  49. b3ff385 Temporarily disabling some more audio processing tests. by aluebs@webrtc.org · 11 years ago
  50. 09b40ec Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
  51. 76dad96 Fixing broken tests in voe_auto_test extended by tina.legrand@webrtc.org · 11 years ago
  52. 9653397 Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  53. 0de0049 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
  54. e21b64b Disables RtpRtcpTest.CanTransmitExtraRtpPacketsWithoutError as it flakily breaks the waterfall. See http://chromegw.corp.google.com/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/99/steps/voe_auto_test/logs/stdio the cl triggering it was a no-change (disabled some other broken tests). by henrike@webrtc.org · 11 years ago
  55. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  56. d171544 Disable CanTransmitExtraRtpPacketsWithoutError on Windows. by pbos@webrtc.org · 11 years ago
  57. 7d82c9d Hand over loopback packets to a network thread. by pbos@webrtc.org · 11 years ago
  58. b3ada15 Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 11 years ago
  59. f3bae63 Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer. by phoglund@webrtc.org · 11 years ago
  60. 44634a6 Disabled SsrcPropagatesCorrectly on Linux. by phoglund@webrtc.org · 11 years ago
  61. 3f45c2e Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  62. 6349e17 Default constructor for RtcpAppHandler. by pbos@webrtc.org · 11 years ago
  63. 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
  64. 6a4acb9 Fix some voe_auto_test uninitialised-value errors. by pbos@webrtc.org · 11 years ago
  65. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  66. 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  67. 0e7cd85 Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 11 years ago
  68. 915ca75 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 11 years ago
  69. a80d94b Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
  70. 50a4d9f Remove #pragma once by pbos@webrtc.org · 11 years ago
  71. 5221d1c Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
  72. 471ae72 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
  73. 8510750 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
  74. ca7a9a2 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
  75. c0fc487 Allow voe_cmd_test to select Opus mono (now the default). by andrew@webrtc.org · 11 years ago
  76. 9e0d9a6 Disabled flaky test. by phoglund@webrtc.org · 11 years ago
  77. 237fe4f Remove vim/emacs modelines from .gypi files by pbos@webrtc.org · 11 years ago
  78. f272497 Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
  79. 54f03bc WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 11 years ago
  80. ef91cbf Remove WEBRTC_*_ENGINE_NETWORK_API use by pwestin@webrtc.org · 11 years ago
  81. c39749a Fix no received audio in tests. by pwestin@webrtc.org · 11 years ago
  82. 84423e9 Disabling MixingTests due to race conditions. by henrika@webrtc.org · 11 years ago
  83. e493218 Remove UDP transport API from VoE by pwestin@webrtc.org · 11 years ago
  84. 3b6f728 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  85. fa2dd22 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. by solenberg@webrtc.org · 11 years ago
  86. 2ffc8bf Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." by wu@webrtc.org · 11 years ago
  87. 365ca40 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  88. aa922de Move the VoE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
  89. 912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  90. 2daec4c Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
  91. 15a03fd Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
  92. 8665399 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. by turaj@webrtc.org · 11 years ago
  93. 1e7f77a Fixing/disabling Windows x64 warnings by kjellander@webrtc.org · 11 years ago
  94. 5ea815f Fix MaxChannels test; 32 -> 100. by andrew@webrtc.org · 11 years ago
  95. 8339bd1 Remove (in practice) the voice engine channel limit. by andrew@webrtc.org · 11 years ago
  96. 8ba1e50 Allow for some error in volume testing. by andrew@webrtc.org · 12 years ago
  97. b790741 Generalized mechanism for excluding gtests on platforms, disabled broken tests on mac. by phoglund@webrtc.org · 12 years ago
  98. fbb6e45 Change Sleep() comment in test fixture. by andrew@webrtc.org · 12 years ago
  99. c62e750 Reverting two mixing test patches: seems to introduce a persistent problem for win voe_auto_test (wrapping problem?) by phoglund@webrtc.org · 12 years ago
  100. b7663cd Fix implicit conversion error in mixing test. by andrew@webrtc.org · 12 years ago