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webrtc / src / 00b8f6b3643332cce1ee711715f7fbb824d793ca / webrtc / voice_engine
  1. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
  2. ac2d27d Fix style violations in common_types.h and config.h by kwiberg@webrtc.org · 10 years ago
  3. d324546 Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : by pkasting@chromium.org · 10 years ago
  4. c0bd7be Adding two new stats to VoiceReceiverInfo by minyue@webrtc.org · 10 years ago
  5. e9facf8 Add range checks in a variety of places where the values will subsequently be by pkasting@chromium.org · 10 years ago
  6. 0a7d4ee Remove usage of BitrateController in VoiceEngine. by mflodman@webrtc.org · 10 years ago
  7. d5ce2e6 Remove EventWrapper::Reset(). by pbos@webrtc.org · 10 years ago
  8. 8db5854 Fix potential flakiness in voe_auto_test. by solenberg@webrtc.org · 10 years ago
  9. 63da1dd audio_processing: Now records mic volume level also when using new AGC by bjornv@webrtc.org · 10 years ago
  10. 0c3e12b Revamp the ProcessThreadImpl implementation. by tommi@webrtc.org · 10 years ago
  11. cc64a9c voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric by bjornv@webrtc.org · 10 years ago
  12. 875c97e Remove SetNotAlive method from the thread class. by tommi@webrtc.org · 10 years ago
  13. a671f4b Fixing a VoE test to set correct rate for iSAC by henrik.lundin@webrtc.org · 10 years ago
  14. 4161715 Remove ChangeUniqueID. by tommi@webrtc.org · 10 years ago
  15. 664ccb7 Reland r8125: Modify some tests to never use DTX disable mode by henrik.lundin@webrtc.org · 10 years ago
  16. df9a41d Fix bug in GetREDStatus(): it doesn't actually return the current status. by pkasting@chromium.org · 10 years ago
  17. d7e34e1 Make it easier to use external libyuv + cleanup GYP files. by kjellander@webrtc.org · 10 years ago
  18. 456f014 Re-allowing RED in voice engine. by minyue@webrtc.org · 10 years ago
  19. ff108fe Revert 8125 "Modify some tests to never use DTX disable mode" by kjellander@webrtc.org · 10 years ago
  20. 043db24 Modify some tests to never use DTX disable mode by henrik.lundin@webrtc.org · 10 years ago
  21. 8315d7d Remove dual stream functionality in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  22. 86e1e48 Move system_wrappers.gyp files to the proper directory. by andresp@webrtc.org · 10 years ago
  23. 46323b3 Remove useless AudioProcessing::Create() overload. by andrew@webrtc.org · 10 years ago
  24. 16825b1 Use int64_t more consistently for times, in particular for RTT values. by pkasting@chromium.org · 10 years ago
  25. 8649fed GN: Fix Windows build. by kjellander@webrtc.org · 10 years ago
  26. d16e839 Rtp-Rtcp sender cleanup. by pbos@webrtc.org · 10 years ago
  27. ce4e9a3 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
  28. 0b1534c Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. by pkasting@chromium.org · 10 years ago
  29. 6fd9308 Suppressing warnings in GetRTT() in VoE. by minyue@webrtc.org · 10 years ago
  30. a954c07 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer by henrika@webrtc.org · 10 years ago
  31. 4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  32. ece3890 Report total bitrate for all streams in GetStats. by pbos@webrtc.org · 10 years ago
  33. 52bb521 Update isolate files for Android APK tests. by kjellander@webrtc.org · 10 years ago
  34. 6a364fe Remove uses of build date/time. by pbos@webrtc.org · 10 years ago
  35. 78c222b Update all .isolate files for the new format. by kjellander@webrtc.org · 10 years ago
  36. aada86b Add a simple AudioConverter class. by andrew@webrtc.org · 10 years ago
  37. 580d367 Add macros and APIs for webrtc histograms. by asapersson@webrtc.org · 10 years ago
  38. 3cefbc9 Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. by xians@webrtc.org · 10 years ago
  39. 2c0cdbc Estimating NTP time with a given RTT. by minyue@webrtc.org · 10 years ago
  40. 4cebd84 Reland "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 10 years ago
  41. f21ea91 GN: Add common configs to all targets. by kjellander@webrtc.org · 11 years ago
  42. ca110b8 Mark virtual overrides of ViENetwork and VoENetwork as such. by henrikg@webrtc.org · 11 years ago
  43. 3987f10 Revert "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 11 years ago
  44. bf7b9e0 Remove DTMF status methods from Voice Engine by henrik.lundin@webrtc.org · 11 years ago
  45. 64a2f10 Remove Get/SetNetEQPlayoutMode APIs by henrik.lundin@webrtc.org · 11 years ago
  46. 2b58a44 Calculating round-trip-time in send-only channel in VoE. by minyue@webrtc.org · 11 years ago
  47. 1972ff8 Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. by henrik.lundin@webrtc.org · 11 years ago
  48. 0372b93 Partial revert of r7014 (Android APK refactor) by kjellander@webrtc.org · 11 years ago
  49. adee8f9 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate by minyue@webrtc.org · 11 years ago
  50. 8e24d87 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 · 11 years ago
  51. 3bd4156 Android APK tests built from a normal WebRTC checkout. by kjellander@webrtc.org · 11 years ago
  52. 524b8f7 GN: Implement voice engine, common audio, audio coding and audio processing by kjellander@webrtc.org · 11 years ago
  53. 047a46f Remove Android.mk build files. by pbos@webrtc.org · 11 years ago
  54. b96ea2a Remove former team members from OWNERS and WATCHLISTS by kjellander@webrtc.org · 11 years ago
  55. 4521e2d Adding online bitrate change to voe_cmd_test by minyue@webrtc.org · 11 years ago
  56. 6aac93b Adding SetOpusMaxBandwidth in VoE and ACM by minyue@webrtc.org · 11 years ago
  57. 6ac22e6 Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 by henrike@webrtc.org · 11 years ago
  58. 2a8df7c Fixing two bugs in voe_cmd_test. by minyue@webrtc.org · 11 years ago
  59. 1ebd2e9 Remove timestamp retreival warning/error. by turaj@webrtc.org · 11 years ago
  60. 026859b This is related to an earlier CL of enabling Opus 48 kHz. by minyue@webrtc.org · 11 years ago
  61. 74aaf29 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 · 11 years ago
  62. eec6ecd Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. --- by tommi@webrtc.org · 11 years ago
  63. 4ef438e Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 11 years ago
  64. 62bafae Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 11 years ago
  65. 9825afc Add ExperimentalNs support in Config by aluebs@webrtc.org · 11 years ago
  66. 1227ab8 GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 11 years ago
  67. a1bfc50 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 11 years ago
  68. 7b82c18 Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 11 years ago
  69. 94454b7 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 11 years ago
  70. ef92755 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 11 years ago
  71. e6e1391 Android: cleanup gtest_target_type conditions. by henrike@webrtc.org · 11 years ago
  72. 1fddd61 Add a Reset() method to AudioFrame. by andrew@webrtc.org · 11 years ago
  73. c1a40a7 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 · 11 years ago
  74. aa5ea1c 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 · 11 years ago
  75. 88fbb2d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 11 years ago
  76. 2fa7f79 Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 11 years ago
  77. 82c4b85 Calculate capture ntp timestamp in local timebase for decoded audio frame. by wu@webrtc.org · 11 years ago
  78. 125ffd7 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 11 years ago
  79. f3e1341 VoEVolumeTest: Enabled Linux flaky tests by bjornv@webrtc.org · 11 years ago
  80. 2db9f45 Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size by minyue@webrtc.org · 11 years ago
  81. cb711f77 Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 11 years ago
  82. 57e0602 Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. by solenberg@webrtc.org · 11 years ago
  83. 21299d4 Remove the use of AudioFrame::energy_ from AudioProcessing and VoE. by andrew@webrtc.org · 11 years ago
  84. a36ad69 Add webrtc field trials API. by andresp@webrtc.org · 11 years ago
  85. 9f27735 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 · 11 years ago
  86. f383a1b 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
  87. 06c1d6f VoEVolumeTest: Adds error return tests. by bjornv@webrtc.org · 11 years ago
  88. 98c76a1 Make vie/voe_auto_test accept non-supported flags without error. by kjellander@webrtc.org · 11 years ago
  89. 8d63d0e Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 by bjornv@webrtc.org · 11 years ago
  90. 6b02eea 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. 1cec395 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. 66021e0 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. 3b76627 Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 11 years ago
  94. 3106b70 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 11 years ago
  95. 9de3d84 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. 382c0c2 Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 11 years ago
  97. 7f3a041 Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test. by henrika@webrtc.org · 11 years ago
  98. e44a84d Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 11 years ago
  99. 8f69330 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 11 years ago
  100. 93fd25c * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. by wu@webrtc.org · 11 years ago