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webrtc / src.git / cb56065c62a31d83919abcd4e343ea3dbe029e9f / webrtc / audio
  1. 86cc6ff Variable audio bitrate. by mflodman · 9 years ago
  2. 14d5dbe 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
  3. 9e03c3b 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
  4. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  5. 2169d8b Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ ) by pbos · 9 years ago
  6. 17bde8c Revert of Remove audio/video distinction for probe packets. (patchset #2 id:20001 of https://codereview.webrtc.org/2061193002/ ) by honghaiz · 9 years ago
  7. a7d88d3 Remove audio/video distinction for probe packets. by Peter Boström · 9 years ago
  8. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 9 years ago
  9. 9421853 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 9 years ago
  10. 8189b02 Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 9 years ago
  11. 971cab0 Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 9 years ago
  12. 6806136 Remove RED support from WebRtcVoiceEngine/MediaChannel by kwiberg · 9 years ago
  13. 0208322 GN: Add video_engine_tests by Peter Boström · 9 years ago
  14. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  15. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  16. ec81bcd Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate by perkj · 9 years ago
  17. d28db7f Delete all use of tick_util.h. by Niels Möller · 9 years ago
  18. e30c272 Revert "Reland of Remove SendPacer from ViEEncoder by perkj · 9 years ago
  19. 28a4456 Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )" by Per · 9 years ago
  20. 825eb58 Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ ) by perkj · 9 years ago
  21. 857c5cc Remove SendPacer from ViEEncoder by perkj · 9 years ago
  22. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  23. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  24. 4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  25. 94a23f0 Reland "Add check_deps rules in DEPS files." by kjellander@webrtc.org · 9 years ago
  26. 56cf60e Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ ) by kjellander · 9 years ago
  27. 086f851 Add check_deps rules in DEPS files. by kjellander@webrtc.org · 9 years ago
  28. 8842c3e Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 9 years ago
  29. 3ecb5c8 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ ) by solenberg · 9 years ago
  30. 8886c81 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 9 years ago
  31. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  32. 50772f1 GN: Update audio_sink.h location by kjellander@webrtc.org · 9 years ago
  33. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  34. 7324eb9 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
  35. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  36. fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
  37. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  38. b7f89d6 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
  39. 789ba92 Simplify CongestionController. by Stefan Holmer · 9 years ago
  40. 58c664c Clean up of CongestionController. by Stefan Holmer · 9 years ago
  41. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  42. bba9dec Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  43. 5ad935c Remove mutable from rtc::CriticalSection members. by pbos · 9 years ago
  44. 3313ec9 Enable transport seq num extension on receive channel to suppress log warning. by stefan · 9 years ago
  45. 2d110be 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
  46. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  47. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  48. 3842c5c Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 9 years ago
  49. 25702cb Misc. small cleanups. by pkasting · 9 years ago
  50. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  51. 7623ce4 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
  52. d3c9447 Nuke TickTime::UseFakeClock. by Peter Boström · 9 years ago
  53. 8237abf Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago
  54. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  55. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  56. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  57. ea07373 Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors. by Fredrik Solenberg · 9 years ago
  58. 358057b Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. by solenberg · 9 years ago
  59. 1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
  60. 7add058 Move some receive stream configuration into webrtc::AudioReceiveStream. by solenberg · 9 years ago
  61. 8b85de2 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail. by solenberg · 9 years ago
  62. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  63. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  64. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  65. 0ccae13 Changed FakeVoiceEngine into a MockVoiceEngine. by Fredrik Solenberg · 9 years ago
  66. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  67. 85a0496 Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  68. 4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  69. 43e83d4 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  70. a457752 Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  71. c7a8b08 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago
  72. a2f30de Log Call {audio, video} stream deletions. by pbos · 9 years ago
  73. 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 10 years ago