1. 0a6becd Replace gflags usages with rtc_base/flags in all targets based on test_main by oprypin · 7 years ago
  2. 94c183c Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ ) by maxmorin · 7 years ago
  3. 4282b1f Add logging host lookups made by TurnPort to the RtcEventLog. by jonaso · 7 years ago
  4. 71e1fa5 Uncomment thread-checkers in ChannelProxy by eladalon · 7 years ago
  5. 3ba7282 Delete unneeded includes of atomic32.h. by nisse · 7 years ago
  6. e0d2fcf Reland of Stop silently accepting unsupported flags in test binaries (patchset #1 id:1 of https://codereview.webrtc.org/3002963002/ ) by oprypin · 7 years ago
  7. f46c48b Revert of Stop silently accepting unsupported flags in test binaries (patchset #5 id:150001 of https://codereview.webrtc.org/2968003003/ ) by oprypin · 7 years ago
  8. 54c62fa Stop silently accepting unsupported flags in test binaries by oprypin · 7 years ago
  9. dc2ec3e Add thread annotations to AudioLevel by henrik.lundin · 7 years ago
  10. ea15d68 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 7 years ago
  11. 504c822 Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock by srte · 7 years ago
  12. 63b0b01 Renamed fields in common_types.h/RtcpStatistics. by srte · 7 years ago
  13. a207a3a Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates by eladalon · 7 years ago
  14. 31d74e4 Move total audio energy and duration tracking to AudioLevel and protect with existing critial section. by zstein · 8 years ago
  15. 2bc62f3 Remove remains of webrtc/base by ehmaldonado · 8 years ago
  16. d1701d0 Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. by zstein · 8 years ago
  17. 5f13072 TransmitMixer: Check GetSendCodec return value. by ossu · 8 years ago
  18. 59d5575 Use relative paths in GN files. by jianjun.zhu · 8 years ago
  19. 4f870fc Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 8 years ago
  20. c6c814d Remove remains of webrtc/base by ehmaldonado · 8 years ago
  21. a36569a Fix a variable naming typo by henrik.lundin · 8 years ago
  22. b7b8932 External APM usage downstream dependency support cleanup by peah · 8 years ago
  23. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  24. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  25. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  26. 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  27. 04ef6b0 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  28. 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
  29. 5c0fe6f Add SafeClamp(), which accepts args of different types by kwiberg · 8 years ago
  30. 34b78a2 Fix Channel::GetSendCodec when used together with SetEncoder. by ossu · 8 years ago
  31. 2d1de82 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 8 years ago
  32. 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
  33. 032d66e Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead. by nisse · 8 years ago
  34. b11cc61 Rename elad.alon to eladalon, to avoid confusion between repositories. by eladalon · 8 years ago
  35. eaf4d68 Replace AudioSendStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
  36. a6d8766 Replace AudioReceiveStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
  37. 433b35c Replace VideoSendStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
  38. acddd51 Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
  39. d100c35 Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend() by henrika · 8 years ago
  40. f26202b Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  41. 3e16eec Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) by mbonadei · 8 years ago
  42. fe5f71c Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  43. ce4e632 Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) by mbonadei · 8 years ago
  44. 0d80b56 Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) by mbonadei · 8 years ago
  45. b70f74c Creating webrtc/modules:module_api by mbonadei · 8 years ago
  46. 0f7c318 Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ ) by mbonadei · 8 years ago
  47. 5efa81c Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) by mbonadei · 8 years ago
  48. cf45c6e Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ ) by mbonadei · 8 years ago
  49. cddf701 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
  50. c5c89b5 Fix invalid output buffer usage by steweg · 8 years ago
  51. df45757 Don't make a top-level namespace called "voetest" by kwiberg · 8 years ago
  52. 3df925a Enable GN check for webrtc/base by kjellander · 8 years ago
  53. 41f1066 Replace Clock with timeutils in AudioEncoder. by michaelt · 8 years ago
  54. f4509f8 Removing unnecessary parameters from initializeAndroidGlobals. by deadbeef · 8 years ago
  55. 69579d7 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  56. fee642e Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  57. b5eeeee Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  58. 1aa04eb Injectable audio encoders: voice_engine/channel changes. by ossu · 8 years ago
  59. 138c150 Injectable audio encoders: BuiltinAudioEncoderFactory by ossu · 8 years ago
  60. 8c39fad Resolve cyclic dependency between audio network adaptor and event log api by michaelt · 8 years ago
  61. 4136d39 Replace use of system_wrappers/include/logging.h by base/logging.h. by nisse · 8 years ago
  62. 16f7430 Revert of Supporting 48kHz PCM file. (patchset #1 id:1 of https://codereview.webrtc.org/2790493004/ ) by lliuu · 8 years ago
  63. fb8e8e8 Supporting 48kHz PCM file. by minyue · 8 years ago
  64. 5e9233b Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  65. ab2deb1 Moves channel-dependent audio input processing to separate encoder task queue. by henrika · 8 years ago
  66. e699d57 Remove voe_auto_test cases for VoEFile. by solenberg · 8 years ago
  67. 82efb01 Delete unneeded includes of deprecated system_wrappers include files. by nisse · 8 years ago
  68. 187f726 Experiment-driven configuration of PLR/RPLR-based FecController by elad.alon · 8 years ago
  69. e153ae8 Fix UT failure by temporarily uncommenting by elad.alon · 8 years ago
  70. 8fc1f4a remove more CriticalSectionWrappers. by kthelgason · 8 years ago
  71. afd1255 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  72. 22de8f3 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  73. 52b9df6 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  74. e1b3d1c Remove voe_base_misc_test.cc. by solenberg · 8 years ago
  75. 135be28 WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  76. 1993d91 Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 8 years ago
  77. 49d1987 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 8 years ago
  78. a6a45e6 TransportFeedbackPacketLossTracker to receive std::vector<PacketFeedback> in place of the entire feedback by elad.alon · 8 years ago
  79. 8132e17 TransportFeedbackPacketLossTrackerTest cosmetic modification by elad.alon · 8 years ago
  80. 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
  81. 961dd1b Remove mixing_test.cc. by solenberg · 8 years ago
  82. 191c2da Remove VoEHardware interface. by solenberg · 8 years ago
  83. de1cc66 R/PLR calculation - time-based window by elad.alon · 8 years ago
  84. f5fbd05 Remove VoENetEqStats interface. by solenberg · 8 years ago
  85. 788c1f7 Remove VoEAudioProcessing interface. by solenberg · 8 years ago
  86. 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
  87. 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
  88. ba76094 Enable cpplint and fix cpplint errors in webrtc/*audio by oprypin · 8 years ago
  89. 07f189f Remove VoEVolumeControl interface. by solenberg · 8 years ago
  90. 2a43f3d Update formatting of AudioLevel class by henrik.lundin · 8 years ago
  91. ef54a6f Remove unused VoiceChannelTransport. by solenberg · 8 years ago
  92. 0ef22df Remove voe_cmd_test. by solenberg · 8 years ago
  93. 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
  94. d24531a Modify TransportFeedbackPacketLossTrackerTest to use parameterization by elad.alon · 8 years ago
  95. 5f25489 Packet Loss Tracker - Stream Separation by elad.alon · 8 years ago
  96. c72b1c8 Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule. by tommi · 8 years ago
  97. 517d6de Enable GN check in voice_engine/ by kjellander · 8 years ago
  98. 8c74388 VoE Utility: Fix a naming nit in RemixAndResample by henrik.lundin · 8 years ago
  99. bf77153 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  100. f21613d Support 4 channel mic in Windows Core Audio by jens.nielsen · 8 years ago