Google Git
Sign in
webrtc / src / webrtc / 9d2bc450ed496995251c99f6b14253d9eaa8d8cf / audio
  1. 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
  2. 1e2f1e5 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  3. 6ba7dfc Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ ) by pbos · 9 years ago
  4. f4d4351 Revert of Remove audio/video distinction for probe packets. (patchset #2 id:20001 of https://codereview.webrtc.org/2061193002/ ) by honghaiz · 9 years ago
  5. b0d0745 Remove audio/video distinction for probe packets. by Peter Boström · 9 years ago
  6. 784336a Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 9 years ago
  7. d9cd888 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 9 years ago
  8. ff1d51a Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 9 years ago
  9. 5a37e3e Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 9 years ago
  10. 8f99654 Remove RED support from WebRtcVoiceEngine/MediaChannel by kwiberg · 9 years ago
  11. c9da01c GN: Add video_engine_tests by Peter Boström · 9 years ago
  12. e840777 Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  13. c4921f4 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  14. d6e6c8d Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate by perkj · 9 years ago
  15. 7738985 Delete all use of tick_util.h. by Niels Möller · 9 years ago
  16. fe59b8e Revert "Reland of Remove SendPacer from ViEEncoder by perkj · 9 years ago
  17. 760fe89 Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )" by Per · 9 years ago
  18. 737cc97 Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ ) by perkj · 9 years ago
  19. 94dd955 Remove SendPacer from ViEEncoder by perkj · 9 years ago
  20. 47a40a3 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  21. 823f908 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  22. 5fb5bd2 #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  23. 0b54e5a Reland "Add check_deps rules in DEPS files." by kjellander@webrtc.org · 9 years ago
  24. de82d23 Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ ) by kjellander · 9 years ago
  25. 181e867 Add check_deps rules in DEPS files. by kjellander@webrtc.org · 9 years ago
  26. f2e3315 Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 9 years ago
  27. b9a65af Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ ) by solenberg · 9 years ago
  28. 9ef75db - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 9 years ago
  29. e38b09a Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  30. 19f0a9b GN: Update audio_sink.h location by kjellander@webrtc.org · 9 years ago
  31. caa8176 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  32. 52cf08c Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ ) by kjellander · 9 years ago
  33. 0bb951e Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  34. abb2e3e Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
  35. ead3cf2 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  36. 484e0cd Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
  37. 1262b9d Simplify CongestionController. by Stefan Holmer · 9 years ago
  38. ed50be1 Clean up of CongestionController. by Stefan Holmer · 9 years ago
  39. 74c29e2 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  40. 3dad57b Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  41. 1e5b805 Remove mutable from rtc::CriticalSection members. by pbos · 9 years ago
  42. 9b91023 Enable transport seq num extension on receive channel to suppress log warning. by stefan · 9 years ago
  43. e3f40fb Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 9 years ago
  44. cf354ef Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  45. 80590d9 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  46. 87f3db7 Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 9 years ago
  47. fa9f5a8 Misc. small cleanups. by pkasting · 9 years ago
  48. 387e90b Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  49. e8f0735 Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ ) by Peter Boström · 9 years ago
  50. a24951b Nuke TickTime::UseFakeClock. by Peter Boström · 9 years ago
  51. 32949e5 Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago
  52. 5be013d Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  53. f95302f Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  54. ae4b1f0 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  55. edbb7ba Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors. by Fredrik Solenberg · 9 years ago
  56. 21ca0a4 Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. by solenberg · 9 years ago
  57. 5bbf7f9 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
  58. 775e132 Move some receive stream configuration into webrtc::AudioReceiveStream. by solenberg · 9 years ago
  59. 03d4810 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail. by solenberg · 9 years ago
  60. ffe1ce0 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  61. b0f22c5 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  62. 36a14b5 modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  63. 4f247a6 Changed FakeVoiceEngine into a MockVoiceEngine. by Fredrik Solenberg · 9 years ago
  64. 78f65d0 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  65. 1c28f5c Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  66. 10762d3 Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  67. 0e9f679 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  68. bbb922f Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  69. f707c68 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago
  70. f863304 Log Call {audio, video} stream deletions. by pbos · 9 years ago
  71. bf9f73c Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 10 years ago