1. b7b8932 External APM usage downstream dependency support cleanup by peah · 8 years ago
  2. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  3. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  4. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  5. 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  6. f4509f8 Removing unnecessary parameters from initializeAndroidGlobals. by deadbeef · 8 years ago
  7. 191c2da Remove VoEHardware interface. by solenberg · 8 years ago
  8. f5fbd05 Remove VoENetEqStats interface. by solenberg · 8 years ago
  9. 788c1f7 Remove VoEAudioProcessing interface. by solenberg · 8 years ago
  10. 07f189f Remove VoEVolumeControl interface. by solenberg · 8 years ago
  11. 2633f5d Remove saturation warning support from TransmitMixer. by tommi · 8 years ago
  12. 71e8df9 Remove usage of VoEAudioProcessing from WVoE/MC. by solenberg · 8 years ago
  13. ac16f1b Remove VoEVideoSync interface. by solenberg · 8 years ago
  14. bda935c Remove VoEExternalMedia interface. by solenberg · 8 years ago
  15. 7eb7de8 Remove the unused and untested functions from VoERTP_RTCP. by solenberg · 8 years ago
  16. f7c7480 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 8 years ago
  17. 017ebe5 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  18. e9523f0 Clean up abs-send-time for audio. by stefan · 8 years ago
  19. 72aebf4 Remove voe::Channel::StopReceive() and associated logic. by solenberg · 8 years ago
  20. 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
  21. a686d5e Moving/renaming webrtc/common.h. by solenberg · 8 years ago
  22. 8111a41 VoERTP_RTCP: Remove GetREDStatus and SetREDStatus by kwiberg · 8 years ago
  23. bbc45b5 Regression test for issue where Opus DTX status was being forgotten. by ivoc · 8 years ago
  24. 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
  25. 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
  26. 1e2f1e5 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  27. 1ac1c3f Moved CreateBuiltinDecoderFactory out to VoEBaseImpl. by ossu · 9 years ago
  28. d25cdbb - Add temporary VoEBase::audio_device_module() method. by solenberg · 9 years ago
  29. cf2fc20 Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code. by solenberg · 9 years ago
  30. 9a75859 Remove unused method OutputMixer::PlayDtmfTone() and infrastructure. by solenberg · 9 years ago
  31. 39673bb Remove the VoEDtmf interface. by solenberg · 9 years ago
  32. 3741297 Removing some unnecessary string manipulation code from VoEBase::GetVersion(). by solenberg · 9 years ago
  33. 4f247a6 Changed FakeVoiceEngine into a MockVoiceEngine. by Fredrik Solenberg · 9 years ago
  34. a90d7f5 Delete a chain of methods in ViE, VoE and ACM by henrik.lundin · 9 years ago
  35. d5bdda3 Unify Transport and newapi::Transport interfaces. by pbos · 9 years ago
  36. e98aa80 Improving support for Android Audio Effects in WebRTC. by henrika · 9 years ago
  37. 35fd753 Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call. by ivoc · 9 years ago
  38. a0ad248 Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 9 years ago
  39. 5361f04 VoE: Remove unused interfaces by Jelena Marusic · 10 years ago
  40. 2450d62 Remove rtp_rtcp/ dump functionality. by Peter Boström · 10 years ago
  41. 1103af5 Propagating RTT from send-only channel to receive-only channel. by Minyue · 10 years ago
  42. 26b26b3 Remove VideoEngine interfaces. by Peter Boström · 10 years ago
  43. 9e50d7e VoE: cleanup VoENetwork implementation by Jelena Marusic · 10 years ago
  44. 0705a02 VoE: apply new style guide on VoE interfaces and their implementations by Jelena Marusic · 10 years ago
  45. f0b18c6 Added SetBitRate function to VoE API to allow changing the audio bitrate. by Ivo Creusen · 10 years ago
  46. 7ac069e VoE: VoEBase unit test by Jelena Marusic · 10 years ago
  47. c49056d VoE: move mock directory 1 level up by Jelena Marusic · 10 years ago
  48. 0487759 Supporting Opus DTX in Voice Engine. by minyue@webrtc.org · 10 years ago
  49. 61b3feb Adds C++/JNI/Java unit test for audio device module on Android. by henrika@webrtc.org · 10 years ago
  50. deb9dae Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
  51. 49da516 voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric by bjornv@webrtc.org · 10 years ago
  52. c1aed49 Remove dual stream functionality in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  53. a3166c4 Use int64_t more consistently for times, in particular for RTT values. by pkasting@chromium.org · 10 years ago
  54. db5c754 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
  55. 0ab923a Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  56. 9735284 Report total bitrate for all streams in GetStats. by pbos@webrtc.org · 10 years ago
  57. c10bfea Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. by xians@webrtc.org · 10 years ago
  58. d29f51c Reland "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 10 years ago
  59. 21cb262 Revert "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 10 years ago
  60. f70e632 Remove DTMF status methods from Voice Engine by henrik.lundin@webrtc.org · 10 years ago
  61. c747067 Remove Get/SetNetEQPlayoutMode APIs by henrik.lundin@webrtc.org · 10 years ago
  62. b0aac71 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate by minyue@webrtc.org · 10 years ago
  63. 1bfd540 Adding SetOpusMaxBandwidth in VoE and ACM by minyue@webrtc.org · 10 years ago
  64. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 11 years ago
  65. dd671de 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
  66. 22f69bd Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 11 years ago
  67. c4e54b6 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
  68. 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
  69. 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
  70. 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
  71. 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
  72. e639a03 Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 11 years ago
  73. b8db407 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 11 years ago
  74. 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
  75. ba47616 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 11 years ago
  76. 47e54ba * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. by wu@webrtc.org · 11 years ago
  77. 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
  78. 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
  79. 4ff0eda Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  80. 2e4c621 (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds by henrika@webrtc.org · 11 years ago
  81. fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 11 years ago
  82. 9a82322 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. by wu@webrtc.org · 11 years ago
  83. 4845ee0 Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 11 years ago
  84. a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 11 years ago
  85. 87c8b86 Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 11 years ago
  86. 942ba53 Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc. by xians@webrtc.org · 11 years ago
  87. 79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  88. 7b72264 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  89. b43ac9f Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
  90. 39e22a1 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 11 years ago
  91. 4489c51 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
  92. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  93. b3ada15 Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 11 years ago
  94. 0ba496b Revert r4301 by tnakamura@webrtc.org · 12 years ago
  95. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 12 years ago
  96. 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 12 years ago
  97. 2753b76 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 12 years ago
  98. d557734 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 12 years ago
  99. 471ae72 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 12 years ago
  100. 8510750 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 12 years ago