Google Git
Sign in
webrtc / src / webrtc / 53cda26ce558b6c5e03af70122f4bac51ab6632b / voice_engine / test / auto_test / standard
  1. 054582b Replace remaining gflags usages with rtc_base/flags by oprypin · 8 years ago
  2. 3ba7282 Delete unneeded includes of atomic32.h. by nisse · 8 years ago
  3. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  4. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  5. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  6. 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  7. e699d57 Remove voe_auto_test cases for VoEFile. by solenberg · 8 years ago
  8. afd1255 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  9. 52b9df6 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  10. e1b3d1c Remove voe_base_misc_test.cc. by solenberg · 8 years ago
  11. 135be28 WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  12. 961dd1b Remove mixing_test.cc. by solenberg · 8 years ago
  13. 191c2da Remove VoEHardware interface. by solenberg · 8 years ago
  14. f5fbd05 Remove VoENetEqStats interface. by solenberg · 8 years ago
  15. 788c1f7 Remove VoEAudioProcessing interface. by solenberg · 8 years ago
  16. 07f189f Remove VoEVolumeControl interface. by solenberg · 8 years ago
  17. 6bbab56 Reland of Delete class SSRCDatabase, and its global ssrc registry. (patchset #1 id:1 of https://codereview.webrtc.org/2700413002/ ) by nisse · 8 years ago
  18. 71e8df9 Remove usage of VoEAudioProcessing from WVoE/MC. by solenberg · 8 years ago
  19. 3264fc7 Revert of Delete class SSRCDatabase, and its global ssrc registry. (patchset #20 id:370001 of https://codereview.webrtc.org/2644303002/ ) by kjellander · 8 years ago
  20. 46dc8df Delete class SSRCDatabase, and its global ssrc registry, by nisse · 8 years ago
  21. ac16f1b Remove VoEVideoSync interface. by solenberg · 8 years ago
  22. bda935c Remove VoEExternalMedia interface. by solenberg · 8 years ago
  23. 660a2e0 Clean out platform specific things from voice_engine_defines.h. by solenberg · 8 years ago
  24. accd9f2 Remove unused voe_stress_test.cc by solenberg · 8 years ago
  25. 45a0bd3 Replace test_support_main by test_main and get rid of test_support_main_threaded_mac by ehmaldonado · 8 years ago
  26. ff9d77c Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  27. e9523f0 Clean up abs-send-time for audio. by stefan · 8 years ago
  28. 72aebf4 Remove voe::Channel::StopReceive() and associated logic. by solenberg · 8 years ago
  29. e59b6ff Moved RtcEventLog files from call/ to logging/ by skvlad · 8 years ago
  30. d208d89 The VoE functionality to apply receive-side processing to VoE channels is unused. I'm removing it so we can avoid instantiating a full APM per channel (and thus also for webrtc::AudioSendStream and webrtc::AudioReceiveStream), and then never use it. by solenberg · 9 years ago
  31. a194cd7 Adding comment regarding the disabling the flaky test VolumeTest.ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAgcOff by peah · 9 years ago
  32. af8f70e Disabling the test VolumeTest.ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAgcOff by peah · 9 years ago
  33. 23ea12e Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface" by ivoc · 9 years ago
  34. 822f09e Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 9 years ago
  35. 1e2f1e5 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  36. 69e722e Make mixing test die earlier on failure instead of spamming errors by phoglund · 9 years ago
  37. 47d5136 Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2018553002/ ) by Tommi · 9 years ago
  38. 51e5639 Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2014183003/ ) by tommi · 9 years ago
  39. 9f94c1b Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006313009/ ) by tommi · 9 years ago
  40. 130ee05 Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2009253004/ ) by kjellander · 9 years ago
  41. 4fcc5b2 Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006243002/ ) by tommi · 9 years ago
  42. b53ca75 Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #6 id:100001 of https://codereview.webrtc.org/2007563002/ ) by tommi · 9 years ago
  43. 049b2cd Adding a some checks and switching out a few assert for RTC_[D]CHECK. by tommi · 9 years ago
  44. 39673bb Remove the VoEDtmf interface. by solenberg · 9 years ago
  45. 484e0cd Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
  46. c8e8cb0 Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/ by tommi · 9 years ago
  47. 80590d9 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  48. 98836c0 Remove DISABLED_ON_ macros. by Peter Boström · 9 years ago
  49. 4eb3389 AudioCodingModuleImpl: Stop failing artificially for non-Opus encoders by kwiberg · 9 years ago
  50. 36a14b5 modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  51. 78f65d0 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  52. eb16b93 Add a PacketOptions struct to webrtc::Transport. by stefan · 9 years ago
  53. d5bdda3 Unify Transport and newapi::Transport interfaces. by pbos · 10 years ago
  54. bf9f73c Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 10 years ago
  55. 00a7fa4 Remove channel ids from various interfaces. by Peter Boström · 10 years ago
  56. 35fd753 Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call. by ivoc · 10 years ago
  57. a0ad248 Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 10 years ago
  58. 20d4fde Reformat existing code. There should be no functional effects. by Peter Kasting · 10 years ago
  59. 5361f04 VoE: Remove unused interfaces by Jelena Marusic · 10 years ago
  60. 2450d62 Remove rtp_rtcp/ dump functionality. by Peter Boström · 10 years ago
  61. 26b26b3 Remove VideoEngine interfaces. by Peter Boström · 10 years ago
  62. ca55fa1 Moving the pacer and the pacer thread to ChannelGroup. by Stefan Holmer · 10 years ago
  63. 0487759 Supporting Opus DTX in Voice Engine. by minyue@webrtc.org · 10 years ago
  64. 860ac53 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 10 years ago
  65. deb9dae Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
  66. 5f8d288 Adding two new stats to VoiceReceiverInfo by minyue@webrtc.org · 10 years ago
  67. fae177c Remove EventWrapper::Reset(). by pbos@webrtc.org · 10 years ago
  68. a270e21 Fix potential flakiness in voe_auto_test. by solenberg@webrtc.org · 10 years ago
  69. 7114b82 Fixing a VoE test to set correct rate for iSAC by henrik.lundin@webrtc.org · 10 years ago
  70. f924d43 Reland r8125: Modify some tests to never use DTX disable mode by henrik.lundin@webrtc.org · 10 years ago
  71. bc39115 Revert 8125 "Modify some tests to never use DTX disable mode" by kjellander@webrtc.org · 10 years ago
  72. 5bd6ea0 Modify some tests to never use DTX disable mode by henrik.lundin@webrtc.org · 10 years ago
  73. 0ab923a Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  74. 9735284 Report total bitrate for all streams in GetStats. by pbos@webrtc.org · 10 years ago
  75. c10bfea Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. by xians@webrtc.org · 10 years ago
  76. d29f51c Reland "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 11 years ago
  77. 21cb262 Revert "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 11 years ago
  78. f70e632 Remove DTMF status methods from Voice Engine by henrik.lundin@webrtc.org · 11 years ago
  79. b0aac71 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate by minyue@webrtc.org · 11 years ago
  80. 1bfd540 Adding SetOpusMaxBandwidth in VoE and ACM by minyue@webrtc.org · 11 years ago
  81. 22c283b Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 by henrike@webrtc.org · 11 years ago
  82. d89fa97 This is related to an earlier CL of enabling Opus 48 kHz. by minyue@webrtc.org · 11 years ago
  83. 6aae61c Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 11 years ago
  84. eb6cd40 VoEVolumeTest: Enabled Linux flaky tests by bjornv@webrtc.org · 11 years ago
  85. 73d6d1f Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size by minyue@webrtc.org · 11 years ago
  86. 8a557b5 Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. by solenberg@webrtc.org · 11 years ago
  87. 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 · 11 years ago
  88. 265cb1b VoEVolumeTest: Adds error return tests. by bjornv@webrtc.org · 11 years ago
  89. efe9461 Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 by bjornv@webrtc.org · 11 years ago
  90. 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 · 11 years ago
  91. 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 · 11 years ago
  92. 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 · 11 years ago
  93. e639a03 Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 11 years ago
  94. b8db407 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 11 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 · 11 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 · 11 years ago
  97. 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 · 11 years ago
  98. 66ccaff Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  99. 1a07e42 Re-enable AGC tests: by aluebs@webrtc.org · 11 years ago
  100. 8fb9156 iOS: baby steps to being able to include_tests=1 by fischman@webrtc.org · 11 years ago