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webrtc / src / webrtc / 6552e11ceb974b972169aaeaeee941fdda50f1f5 / voice_engine
  1. 4f870fc Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 8 years ago
  2. c6c814d Remove remains of webrtc/base by ehmaldonado · 8 years ago
  3. a36569a Fix a variable naming typo by henrik.lundin · 8 years ago
  4. b7b8932 External APM usage downstream dependency support cleanup by peah · 8 years ago
  5. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  6. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  7. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  8. 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  9. 04ef6b0 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  10. 7d97307 Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide: by yujo · 8 years ago
  11. 5c0fe6f Add SafeClamp(), which accepts args of different types by kwiberg · 8 years ago
  12. 34b78a2 Fix Channel::GetSendCodec when used together with SetEncoder. by ossu · 8 years ago
  13. 2d1de82 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 8 years ago
  14. 33f05e7 This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport. by perkj · 8 years ago
  15. 032d66e Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead. by nisse · 8 years ago
  16. b11cc61 Rename elad.alon to eladalon, to avoid confusion between repositories. by eladalon · 8 years ago
  17. eaf4d68 Replace AudioSendStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
  18. a6d8766 Replace AudioReceiveStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
  19. 433b35c Replace VideoSendStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
  20. acddd51 Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
  21. d100c35 Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend() by henrika · 8 years ago
  22. f26202b Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  23. 3e16eec Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) by mbonadei · 8 years ago
  24. fe5f71c Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  25. ce4e632 Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) by mbonadei · 8 years ago
  26. 0d80b56 Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) by mbonadei · 8 years ago
  27. b70f74c Creating webrtc/modules:module_api by mbonadei · 8 years ago
  28. 0f7c318 Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ ) by mbonadei · 8 years ago
  29. 5efa81c Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) by mbonadei · 8 years ago
  30. cf45c6e Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ ) by mbonadei · 8 years ago
  31. cddf701 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
  32. c5c89b5 Fix invalid output buffer usage by steweg · 8 years ago
  33. df45757 Don't make a top-level namespace called "voetest" by kwiberg · 8 years ago
  34. 3df925a Enable GN check for webrtc/base by kjellander · 8 years ago
  35. 41f1066 Replace Clock with timeutils in AudioEncoder. by michaelt · 8 years ago
  36. f4509f8 Removing unnecessary parameters from initializeAndroidGlobals. by deadbeef · 8 years ago
  37. 69579d7 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  38. fee642e Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  39. b5eeeee Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  40. 1aa04eb Injectable audio encoders: voice_engine/channel changes. by ossu · 8 years ago
  41. 138c150 Injectable audio encoders: BuiltinAudioEncoderFactory by ossu · 8 years ago
  42. 8c39fad Resolve cyclic dependency between audio network adaptor and event log api by michaelt · 8 years ago
  43. 4136d39 Replace use of system_wrappers/include/logging.h by base/logging.h. by nisse · 8 years ago
  44. 16f7430 Revert of Supporting 48kHz PCM file. (patchset #1 id:1 of https://codereview.webrtc.org/2790493004/ ) by lliuu · 8 years ago
  45. fb8e8e8 Supporting 48kHz PCM file. by minyue · 8 years ago
  46. 5e9233b Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  47. ab2deb1 Moves channel-dependent audio input processing to separate encoder task queue. by henrika · 8 years ago
  48. e699d57 Remove voe_auto_test cases for VoEFile. by solenberg · 8 years ago
  49. 82efb01 Delete unneeded includes of deprecated system_wrappers include files. by nisse · 8 years ago
  50. 187f726 Experiment-driven configuration of PLR/RPLR-based FecController by elad.alon · 8 years ago
  51. e153ae8 Fix UT failure by temporarily uncommenting by elad.alon · 8 years ago
  52. 8fc1f4a remove more CriticalSectionWrappers. by kthelgason · 8 years ago
  53. afd1255 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  54. 22de8f3 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  55. 52b9df6 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  56. e1b3d1c Remove voe_base_misc_test.cc. by solenberg · 8 years ago
  57. 135be28 WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  58. 1993d91 Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 8 years ago
  59. 49d1987 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 8 years ago
  60. a6a45e6 TransportFeedbackPacketLossTracker to receive std::vector<PacketFeedback> in place of the entire feedback by elad.alon · 8 years ago
  61. 8132e17 TransportFeedbackPacketLossTrackerTest cosmetic modification by elad.alon · 8 years ago
  62. 3d8ed00 Add thread check to ModuleProcessThread::DeRegisterModule and remove all unnecessary locking that was there due to one implementation calling from a different thread. by tommi · 8 years ago
  63. 961dd1b Remove mixing_test.cc. by solenberg · 8 years ago
  64. 191c2da Remove VoEHardware interface. by solenberg · 8 years ago
  65. de1cc66 R/PLR calculation - time-based window by elad.alon · 8 years ago
  66. f5fbd05 Remove VoENetEqStats interface. by solenberg · 8 years ago
  67. 788c1f7 Remove VoEAudioProcessing interface. by solenberg · 8 years ago
  68. 0f20d58 Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ ) by oprypin · 8 years ago
  69. 899c143 Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ ) by oprypin · 8 years ago
  70. ba76094 Enable cpplint and fix cpplint errors in webrtc/*audio by oprypin · 8 years ago
  71. 07f189f Remove VoEVolumeControl interface. by solenberg · 8 years ago
  72. 2a43f3d Update formatting of AudioLevel class by henrik.lundin · 8 years ago
  73. ef54a6f Remove unused VoiceChannelTransport. by solenberg · 8 years ago
  74. 0ef22df Remove voe_cmd_test. by solenberg · 8 years ago
  75. 17d9d86 Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo. by elad.alon · 8 years ago
  76. d24531a Modify TransportFeedbackPacketLossTrackerTest to use parameterization by elad.alon · 8 years ago
  77. 5f25489 Packet Loss Tracker - Stream Separation by elad.alon · 8 years ago
  78. c72b1c8 Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule. by tommi · 8 years ago
  79. 517d6de Enable GN check in voice_engine/ by kjellander · 8 years ago
  80. 8c74388 VoE Utility: Fix a naming nit in RemixAndResample by henrik.lundin · 8 years ago
  81. bf77153 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  82. f21613d Support 4 channel mic in Windows Core Audio by jens.nielsen · 8 years ago
  83. aca8476 Remove workaround for bug 6986 by kwiberg · 8 years ago
  84. 2633f5d Remove saturation warning support from TransmitMixer. by tommi · 8 years ago
  85. 3528419 Fix TSAN race in webrtc::voe::Channel. by hbos · 8 years ago
  86. faee334 Simplify webrtc::voe::MonitorModule and remove the .cc file. by tommi · 8 years ago
  87. 1c1bd4d Add probe logging to RtcEventLog. by philipel · 8 years ago
  88. 110da32 Roll chromium_revision 33a7a547b9..7e40b4199b (452838:452938) by kjellander · 8 years ago
  89. 1f8727f Propagate packet pacing information to SendTimeHistory. by philipel · 8 years ago
  90. 25435fe Introduce a new constructor to PlatformThread. by tommi · 8 years ago
  91. b9ca78e Fix flaky test WebRtcMediaRecorderTest.PeerConnection by solenberg · 8 years ago
  92. 95a13b4 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  93. 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
  94. 71e8df9 Remove usage of VoEAudioProcessing from WVoE/MC. by solenberg · 8 years ago
  95. 223ac94 Rename some variables and methods in RTC event log. by terelius · 8 years ago
  96. 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
  97. 964397f Remove the Windows Wave audio device implementation. by Tommi · 8 years ago
  98. 46dc8df Delete class SSRCDatabase, and its global ssrc registry, by nisse · 8 years ago
  99. 60d2fc1 Add logging of delay-based bandwidth estimate. by terelius · 8 years ago
  100. ac16f1b Remove VoEVideoSync interface. by solenberg · 8 years ago