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webrtc / src / webrtc / f65fb4c7c24e9812e198ad4ceff1fba3dc2d3bc6 / voice_engine
  1. 6045993 Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  2. 3eb926a Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common. by ehmaldonado · 8 years ago
  3. d35e102 Replace consecutive-losses count by a calculation of first-order-FEC recoverability. by elad.alon · 8 years ago
  4. 9551ae0 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  5. 90cc2a9 Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) by mbonadei · 8 years ago
  6. df5ec64 Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) by mbonadei · 8 years ago
  7. 4fdc30d Reland of "Log audio network adapter decisions in event log." by minyue · 8 years ago
  8. 5c59d8f Moving webrtc.gni up one level from build/ by mbonadei · 8 years ago
  9. 022a283 Add TransportFeedbackPacketLossTracker. by minyue · 8 years ago
  10. bc2b639 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  11. ca68fc1 UdpTransport:IsIpAddressValid: Added extra :: check for ipv6 by ossu · 8 years ago
  12. 5b7322d Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ ) by sakal · 8 years ago
  13. dc916a1 Log audio network adapter decisions in event log. by michaelt · 8 years ago
  14. 22174cb Pass event log to ANA. by michaelt · 8 years ago
  15. 756b219 Update smoothed bitrate. by michaelt · 8 years ago
  16. 1709986 Replace RTC_DCHECK(false) with RTC_NOTREACHED(). by nisse · 8 years ago
  17. 0e0db71 Make OverheadObserver::OnOverheadChanged count RTP headers only by nisse · 8 years ago
  18. c3c7f14 Move FilePlayer and FileRecorder to Voice Engine by kwiberg · 8 years ago
  19. d06add1 Drop unneeded include of media_file.h. by nisse · 8 years ago
  20. c9999ff Wire-up audio BWE with overhead. by michaelt · 8 years ago
  21. a1daebb Delete voice_engine_configurations.h by henrik.lundin · 8 years ago
  22. bbecdb6 Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h by henrik.lundin · 8 years ago
  23. d7ef04b Move functionality out from AudioFrame and into AudioFrameOperations. by aleloi · 8 years ago
  24. 0d9fb71 Deprecated SetAudioPacketSize from RTPSender and removed calls to it. by ossu · 8 years ago
  25. f292dcf Remove API-related #defines from voice_engine_configurations.h by henrik.lundin · 8 years ago
  26. d382b83 Prep to remove API-related #defines from voice_engine_configurations.h by henrik.lundin · 8 years ago
  27. 2bc91bf Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  28. 45a0bd3 Replace test_support_main by test_main and get rid of test_support_main_threaded_mac by ehmaldonado · 8 years ago
  29. c01dfe6 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  30. 3668b3d Remove 3 defines in voice_engine_configurations.h by henrik.lundin · 8 years ago
  31. cf7d22e Refactor RMSLevel and give it new functionality by henrik.lundin · 8 years ago
  32. 4cb29f6 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 8 years ago
  33. acd8db6 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 8 years ago
  34. 017ebe5 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  35. 4d9c52e Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  36. 6b56973 Smooth BWE and pass it to Audio Network Adaptor. by michaelt · 8 years ago
  37. 69d52d1 Revert of Move ADM specific Android files into modules/audio_device/android/ (patchset #2 id:20001 of https://codereview.webrtc.org/2533573002/ ) by solenberg · 8 years ago
  38. cb9420c Move ADM specific Android files into modules/audio_device/android/ by solenberg · 8 years ago
  39. 1cbcced Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ ) by magjed · 8 years ago
  40. e82ac9a Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ ) by magjed · 8 years ago
  41. 402fe33 Remove RTPPayloadStrategy and simplify RTPPayloadRegistry by magjed · 8 years ago
  42. 4c1eb05 Send audio and video codecs to RTPPayloadRegistry by magjed · 8 years ago
  43. cd1064b WebRTC: Replace ProjectRootPath by ResourcePath by ehmaldonado · 8 years ago
  44. ff9d77c Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  45. abef9e9 Remove all references to GYP by Henrik Kjellander · 8 years ago
  46. dbe2c77 Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  47. 2454551 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  48. e9523f0 Clean up abs-send-time for audio. by stefan · 8 years ago
  49. 72aebf4 Remove voe::Channel::StopReceive() and associated logic. by solenberg · 8 years ago
  50. 5206602 Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  51. b87bc5d Add a NeededFrequency() method to the AudioMixer::Source interface. by aleloi · 8 years ago
  52. 23bb454 Original CL: https://codereview.webrtc.org/2433153003/, commit 8b8d3e4c30e8ea3846b58dfd36d1fd35a7799df4. by ivoc · 8 years ago
  53. 75666cc Revert of New statistics interface for APM (patchset #11 id:200001 of https://codereview.webrtc.org/2433153003/ ) by ivoc · 8 years ago
  54. b0bdb91 New statistics interface for APM by ivoc · 8 years ago
  55. 77c17ed Stop using old AudioCodingModule::RegisterReceiveCodec overloads by kwiberg · 8 years ago
  56. fbd0246 Move audio frame memory handling inside AudioMixer. by aleloi · 8 years ago
  57. b4ac6b0 Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  58. d8e3fa0 Call OnTransportFeedback just when feedback_observer exist. by michaelt · 8 years ago
  59. a6c6623 GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  60. 8f98950 Cleanup of voice_engine includes. by aleloi · 8 years ago
  61. 5e0ea59 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  62. 0d87bce Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz. by ossu · 8 years ago
  63. 2a6e87c Hooking up audio network adaptor to VoE. by minyue · 8 years ago
  64. 6dcc486 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
  65. 5bbbed1 Prep to remove APM-related #defines from voice_engine_configurations.h by henrik.lundin · 8 years ago
  66. 19689f4 Added logging for audio send/receive stream configs. by ivoc · 8 years ago
  67. c859ded Delete old video defines in engine config. by mflodman · 8 years ago
  68. e59b6ff Moved RtcEventLog files from call/ to logging/ by skvlad · 8 years ago
  69. 36a2479 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  70. 15b966d Enable the -Wundef warning for clang by kwiberg · 8 years ago
  71. a9eeb97 Moved Gn target rtc_event_log to one directory above. by charujain · 8 years ago
  72. 2cf29b5 GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 8 years ago
  73. 4e6d4da Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ ) by danilchap · 8 years ago
  74. 1bed982 Remove unnecessary interface TelephoneEventHandler. by solenberg · 8 years ago
  75. ecd8880 Add voe_cmd_test to voice_engine/BUILD.gn (and remove it from voice_engine.gyp, together with the channel_transport gyp target) by solenberg · 8 years ago
  76. 435c7b3 GYP: Remove targets inside include_tests==1 that are converted to GN. by kjellander · 8 years ago
  77. 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 · 8 years ago
  78. 17365a3 Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ by solenberg · 8 years ago
  79. 0380718 Revert of Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ (patchset #7 id:120001 of https://codereview.webrtc.org/2319583005/ ) by solenberg · 8 years ago
  80. 1087e71 Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ by solenberg · 8 years ago
  81. 9eb213a OWNERS: Make everyone able to change *.gn,*.gni files. by Henrik Kjellander · 9 years ago
  82. f47ee70 GN: Move audio_coding to public_deps in voice engine by ehmaldonado · 9 years ago
  83. 11148c4 FilePlayer: Remove backwards compatibility stuff that we no longer need by kwiberg · 9 years ago
  84. a686d5e Moving/renaming webrtc/common.h. by solenberg · 9 years ago
  85. f69efd1 Add time line for acked bitrate. by Stefan Holmer · 9 years ago
  86. 80e6692 GN Templates: Move common_inherited_config to the template. by ehmaldonado · 9 years ago
  87. 1d49219 GN Templates: Move common_config to the template. by ehmaldonado · 9 years ago
  88. 6ae9a37 GN Templates: Add //build/config/sanitizers:deps to rtc_executable. by ehmaldonado · 9 years ago
  89. f0532f3 GN: Introduce templates. by ehmaldonado · 9 years ago
  90. c5631c3 Remove Channel::UpdatePacketDelay and some member variables by henrik.lundin · 9 years ago
  91. cf82062 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago
  92. 6817c24 Added a level indicator to new mixer. by aleloi · 9 years ago
  93. 8b8ab57 Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine by henrik.lundin · 9 years ago
  94. 09b6e84 Fix trivial lint errors in FileRecorder and FilePlayer by kwiberg · 9 years ago
  95. baf5c02 GN: Add "//build/config/sanitizers:deps" as a dependency to executable targets. by ehmaldonado · 9 years ago
  96. 7fe0c9f voice_engine: Removed old variants of Channel constructor and CreateChannel by ossu · 9 years ago
  97. 6566d38 FileRecorder + FilePlayer: Let Create functions return unique_ptr by kwiberg · 9 years ago
  98. 9b7edf0 Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ ) by kwiberg · 9 years ago
  99. 6c87fcb Move FilePlayer and FileRecorder to Voice Engine by kwiberg · 9 years ago
  100. bcb5760 Remove old methods in AudioTransport, make it pass a gn build by maxmorin · 9 years ago