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gerrit.openfyde.cn / webrtc.googlesource.com / src / a1a475a5b66f7576c3a8013605ea409fa26b04d3 / audio / audio_send_stream.cc
  1. 78609d5 Reland of BWE allocation strategy by Alex Narest · 8 years ago
  2. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 8 years ago
  3. a5fbc23 BWE allocation strategy by Alex Narest · 8 years ago
  4. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 8 years ago
  5. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 8 years ago
  6. b3944f0 Media track ID visibility at BWE level by Alex Narest · 8 years ago
  7. 90e1f53 Fix potentional race in AudioSendStream constructor by Danil Chapovalov · 8 years ago
  8. 1c239d4 Remove voe::Statistics. by solenberg · 8 years ago
  9. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  10. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/audio/audio_send_stream.cc]
  11. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 8 years ago
  12. 3651fdd Uncomment thread-check in AudioSendStream::OnPacketFeedbackVector() by eladalon · 8 years ago
  13. 8de1826 Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by minyue-webrtc · 8 years ago
  14. 7df370b Revert "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by Minyue Li · 8 years ago
  15. 4a88120 Allow AudioSendStream to reconfig AudioNetworkAdaptor by minyue-webrtc · 8 years ago
  16. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 8 years ago
  17. c58f8c0 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 8 years ago
  18. e76bd3a 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
  19. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  20. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  21. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  22. 1129df2 Always ResetSenderCongestionControlObjects before RegisterEtc... by ossu · 8 years ago
  23. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  24. edd6eea Rename elad.alon to eladalon, to avoid confusion between repositories. by eladalon · 8 years ago
  25. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 8 years ago
  26. 93e4522 Renaming probing_interval to bwe_period globally. by minyue · 8 years ago
  27. 48368ad Fixing video loopback test with encoder factory. by minyue · 8 years ago
  28. 3b9ff38 Have AudioSendStream register CNG payload types with the RtpRtcpModule. by ossu · 8 years ago
  29. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  30. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
  31. fca900a Fix two invalid DCHECKs related to audio BWE. by stefan · 8 years ago
  32. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  33. 4e76451 Fix UT failure by temporarily uncommenting by elad.alon · 8 years ago
  34. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  35. dadb4dc Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 8 years ago
  36. d12a8e1 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 8 years ago
  37. 559af38 Split CongestionController into send- and receive-side classes. by nisse · 8 years ago
  38. 5bbf43f Move delay_based_bwe_ into CongestionController by elad.alon · 8 years ago
  39. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  40. 06f240b Clean out platform specific things from voice_engine_defines.h. by solenberg · 8 years ago
  41. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  42. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  43. 55d1ebb Enable periodic bitrate probing when application limited for audio BWE. by stefan · 8 years ago
  44. f4caaab Fix for bwe with overhead on audio only calls. by michaelt · 9 years ago
  45. 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 9 years ago
  46. 9332b7d Reland "Update rtt on audio only calls". by michaelt · 9 years ago
  47. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 9 years ago
  48. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 9 years ago
  49. 6287e82 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 9 years ago
  50. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 9 years ago
  51. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 9 years ago
  52. d4adce4 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 9 years ago
  53. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 9 years ago
  54. 7aba029 Make use of new APM statistics interface. by ivoc · 9 years ago
  55. 6f0b9fd Allowing resetting of AudioNetworkAdaptor in AudioSendStream. by minyue · 9 years ago
  56. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 9 years ago
  57. 10cbb46 Fixing config for Audio BWE. by minyue · 9 years ago
  58. 572ae12 Fix crash when registering abs-send-time to AudioSend/ReceiveStream. by stefan · 9 years ago
  59. b521aa7 Clean up abs-send-time for audio. by stefan · 9 years ago
  60. 6b825df Using AudioOption to enable audio network adaptor. by minyue · 9 years ago
  61. 940b6d6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 9 years ago
  62. 189f9b1 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) by terelius · 9 years ago
  63. 1836fd6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 9 years ago
  64. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 9 years ago
  65. 7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 9 years ago
  66. 982bf89 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 9 years ago
  67. e0729c5 Add RtcpRttStats to AudioStream by michaelt · 9 years ago
  68. e035e2d Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets. by terelius · 9 years ago
  69. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 9 years ago
  70. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 9 years ago
  71. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 9 years ago
  72. 86cc6ff Variable audio bitrate. by mflodman · 9 years ago
  73. 9421853 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 9 years ago
  74. 971cab0 Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 9 years ago
  75. 6806136 Remove RED support from WebRtcVoiceEngine/MediaChannel by kwiberg · 9 years ago
  76. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  77. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  78. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  79. 8842c3e Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 9 years ago
  80. 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
  81. 8886c81 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 9 years ago
  82. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  83. fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
  84. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  85. b7f89d6 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
  86. bba9dec Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  87. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 10 years ago
  88. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 10 years ago
  89. 358057b Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. by solenberg · 10 years ago
  90. 1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 10 years ago
  91. 8b85de2 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail. by solenberg · 10 years ago
  92. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 10 years ago
  93. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 10 years ago
  94. 85a0496 Implement AudioSendStream::GetStats(). by solenberg · 10 years ago
  95. c7a8b08 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 10 years ago