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webrtc / src / webrtc / 4be637239ca30b974270b3ad3b553f2f2493de62 / media / engine / webrtcvoiceengine.cc
  1. 4be6372 Add SSRC to RtpEncodingParameters for audio. by deadbeef · 8 years ago
  2. 5917368 Wire up x-google-{min,start,max}-bitrate to WebRtcVoiceMediaChannel. by stefan · 8 years ago
  3. 9750537 This CL introduces the new functionality for setting by peah · 8 years ago
  4. 5595e3b In VoiceEngine, the settings for APM are applied in such a way that by peah · 8 years ago
  5. 9a1d49f Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  6. d85b51d Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  7. ff9d77c Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  8. cb8ce03 Support receiving DTMF for multiple RTP clock rates. by solenberg · 8 years ago
  9. ab81d98 Rename the adapt audio bitrate experiment. by stefan · 8 years ago
  10. ca1d233 Add AudioOption for residual echo detector, and enable the echo detector by default on non-mobile platforms. by ivoc · 8 years ago
  11. dbe2c77 Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  12. 2454551 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  13. a1912ef Fixing config for Audio BWE. by minyue · 8 years ago
  14. 1c8e57b Revert of Removed the legacy behavior of stopping playout when setting new receive codecs. (patchset #1 id:1 of https://codereview.webrtc.org/2409483003/ ) by kwiberg · 8 years ago
  15. e9523f0 Clean up abs-send-time for audio. by stefan · 8 years ago
  16. 5206602 Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  17. 3ce8e91 - Replace FakeAudioProcessing in WVoE unittest with MockAudioProcessing. by solenberg · 8 years ago
  18. dfb640f Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  19. bffc190 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  20. f272997 Added a level controller initialization value to MediaConstraints. by aleloi · 8 years ago
  21. 0556d95 Set min and max rate on caller and on callee side. by michaelt · 8 years ago
  22. a7151c0 Removed the legacy behavior of stopping playout when setting new receive codecs. by solenberg · 8 years ago
  23. d0dcb3b Restarting channel when swapping AudioReceiveStreams in WebrtcVoE. by aleloi · 8 years ago
  24. 3a2c5fb Change thread check to race check. Also, add comment to explain implementation of RaceChecker. by solenberg · 8 years ago
  25. 4450c27 Add new decoding statistics for muted output by henrik.lundin · 8 years ago
  26. 514a0ea Relaxed unnecessarily stringent thread checking in WebRtcAudioSendStream::OnData(). by solenberg · 8 years ago
  27. 1dd0a8c The current scheme for setting parameters and specifying the by peah · 9 years ago
  28. c7e4000 Revert of Introduced new scheme for controlling the functionality inside the audio processing module (patchset #12 id:260001 of https://codereview.webrtc.org/2292863002/ ) by kjellander · 9 years ago
  29. ff765cc The current scheme for setting parameters and specifying the behavior by peah · 9 years ago
  30. a686d5e Moving/renaming webrtc/common.h. by solenberg · 9 years ago
  31. cf82062 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago
  32. 7475231 Deactivated the intelligibility enhancement functionality by default by peah · 9 years ago
  33. 87366ac Removed the deactivation of the level controller when there is a built-in AGC available by peah · 9 years ago
  34. 2b13b88 Disable the software noise suppressor on iOS devices as that by peah · 9 years ago
  35. 4a731ed Updated AudioDecoderFactory to list AudioCodecSpecs instead of SdpAudioFormats. by ossu · 9 years ago
  36. a6d67c6 WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs. by ossu · 9 years ago
  37. bcb5760 Remove old methods in AudioTransport, make it pass a gn build by maxmorin · 9 years ago
  38. 762f0f9 Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine. by aleloi · 9 years ago
  39. f29ccf6 Variable audio bitrate. by mflodman · 9 years ago
  40. 4187c26 Revert of WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs. (patchset #10 id:200001 of https://codereview.webrtc.org/2072753002/ ) by ossu · 9 years ago
  41. 45cd266 WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs. by ossu · 9 years ago
  42. 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
  43. 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
  44. 3ead781 This CL adds activation logic of the new APM level control by peah · 9 years ago
  45. 1e2f1e5 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  46. 784336a Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 9 years ago
  47. d056573 - Remove use of VoERTP_RTCP::SetLocalSSRC() for receive streams; recreate them instead. by solenberg · 9 years ago
  48. d9cd888 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 9 years ago
  49. 8e3caa1 WebRtcVoiceCodecs: Eliminate some useless copying by kwiberg · 9 years ago
  50. ff1d51a Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 9 years ago
  51. 5a37e3e Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 9 years ago
  52. e4800a7 Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface. by solenberg · 9 years ago
  53. 8f99654 Remove RED support from WebRtcVoiceEngine/MediaChannel by kwiberg · 9 years ago
  54. c0c552c Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 9 years ago
  55. e840777 Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  56. c4921f4 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  57. 3c59b82 Surface the IntelligibilityEnhancer on MediaConstraints by Alejandro Luebs · 9 years ago
  58. 80b957b Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
  59. 18d8284 Add a parameter to set a maximum file size when starting an RTC event log on the PeerConnectionFactory API. by ivoc · 9 years ago
  60. b4dc913 Enable NACK for audio even if there are no send streams. by deadbeef · 9 years ago
  61. c0457bb Set rtcp_send_transport for AudioReceiveStreams. This was forgotten in https://codereview.webrtc.org/1909333002/. by solenberg · 9 years ago
  62. a0b25f3 Support RtpEncodingParameters::active in voice engine. by Taylor Brandstetter · 9 years ago
  63. 06751cb Remove usage of VoENetwork from VoEWrapper and FakeWebRtcVoiceEngine. by solenberg · 9 years ago
  64. 823f908 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  65. 76979ca Cap the send bitrate for opus and iSAC before passing down to VoE. by deadbeef · 9 years ago
  66. 5fb5bd2 #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  67. 867a835 Adding codecs to the RtpParameters returned by an RtpSender. by Taylor Brandstetter · 9 years ago
  68. 512897c Update the call when the network route changes by Honghai Zhang · 9 years ago
  69. 9a13c57 Remove unused cricket::AudioFrame class. by solenberg · 9 years ago
  70. a528b35 Early initialize recording on the ADM from WebRtcVoiceMediaChannel. by solenberg · 9 years ago
  71. 397934d Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
  72. dfa4189 Fix bug causing audio to stop being sent when AudioSendStreams are recreated. by solenberg · 9 years ago
  73. f332d89 Replace a few calls to VoEHardware with direct calls on the ADM, in WVoMC. by solenberg · 9 years ago
  74. 46c4295 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 9 years ago
  75. d25cdbb - Add temporary VoEBase::audio_device_module() method. by solenberg · 9 years ago
  76. 82688e9 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
  77. 12cfa58 Remove all uses of the HAVE_CONFIG_H define. by Henrik Kjellander · 9 years ago
  78. ee1e4ed Make the audio channel communicate network state changes to the call. by skvlad · 9 years ago
  79. bb6decf Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  80. 6a4e627 Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ ) by kjellander · 9 years ago
  81. 0912ecc Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  82. 962f7ea Remove DeviceManager and DeviceInfo. by solenberg · 9 years ago
  83. 1075220 Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 9 years ago
  84. f2e3315 Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 9 years ago
  85. 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
  86. 9ef75db - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 9 years ago
  87. c20887e Add tracing to interesting media-related methods. by Peter Boström · 9 years ago
  88. e38b09a Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  89. be8166d On WVoMC::SetSendParameters(), figure out send codec settings ONCE, not for each send stream. by solenberg · 9 years ago
  90. 5ec6244 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago
  91. caa8176 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  92. 306b1ad Replace scoped_ptr with unique_ptr in webrtc/media/ by kwiberg · 9 years ago
  93. 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
  94. 0bb951e Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  95. 2f9a5de Remove the 'audioDebugRecording' media constraint and the aec_dump AudioOptions flag. by solenberg · 9 years ago
  96. abb2e3e Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
  97. e1a2ad1 Introduce struct MediaConfig, with construction-time settings. by nisse · 9 years ago
  98. c8ca0e2 Rename webrtc/media/webrtc -> webrtc/media/engine by kjellander@webrtc.org · 9 years ago[Renamed (99%) from media/webrtc/webrtcvoiceengine.cc]
  99. 71ec0dc Change to WebRTC license in webrtc/media by kjellander · 9 years ago
  100. e1bbb30 Move talk/media to webrtc/media by kjellander · 9 years ago