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gerrit-public.fairphone.software / platform / external / webrtc / fc29b0ad4658ccf68a51f2f170ed5641e354bf0a / audio / audio_send_stream.h
  1. 74dadc1 Ready to support of absolute capture timestamp header extension. by Minyue Li · 4 years, 6 months ago
  2. 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 4 years, 7 months ago
  3. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 4 years, 10 months ago
  4. 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 4 years, 11 months ago
  5. f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 5 years ago
  6. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
  7. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 5 years ago
  8. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 5 years ago
  9. 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 5 years ago
  10. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  11. 9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 5 years ago
  12. 63658d0 Revert "Ensure that we always set values for min and max audio bitrate." by Daniel Lee · 5 years ago
  13. e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 5 years ago
  14. 44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 5 years ago
  15. 0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 5 years ago
  16. 8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 5 years ago
  17. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 5 years ago
  18. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 5 years ago
  19. 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
  20. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
  21. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  22. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  23. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  24. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  25. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  26. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  27. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  28. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  29. 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
  30. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  31. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  32. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  33. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 6 years ago
  34. 763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 6 years ago
  35. 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
  36. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  37. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  38. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  39. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  40. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  41. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  42. cedd351 Do not add audio bitrate observer if TWCC sending is not supported by Alex Narest · 7 years ago
  43. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  44. 8d9c540 Deprecated BitrateController::CreateRtcpBandwidthObserver. by Sebastian Jansson · 7 years ago
  45. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  46. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_send_stream.h]
  47. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  48. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
  49. c58f8c0 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 7 years ago
  50. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  51. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  52. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  53. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 7 years ago
  54. 93e4522 Renaming probing_interval to bwe_period globally. by minyue · 7 years ago
  55. 3b9ff38 Have AudioSendStream register CNG payload types with the RtpRtcpModule. by ossu · 7 years ago
  56. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  57. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 7 years ago
  58. d12a8e1 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 7 years ago
  59. 559af38 Split CongestionController into send- and receive-side classes. by nisse · 7 years ago
  60. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  61. f4caaab Fix for bwe with overhead on audio only calls. by michaelt · 8 years ago
  62. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  63. 9332b7d Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  64. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  65. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  66. 6287e82 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 8 years ago
  67. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  68. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  69. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  70. 7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  71. 982bf89 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  72. e0729c5 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
  73. e035e2d Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets. by terelius · 8 years ago
  74. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 8 years ago
  75. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  76. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
  77. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  78. 86cc6ff Variable audio bitrate. by mflodman · 8 years ago
  79. 9421853 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 8 years ago
  80. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 8 years ago
  81. 4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 8 years ago
  82. 8842c3e Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 8 years ago
  83. 3ecb5c8 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ ) by solenberg · 8 years ago
  84. 8886c81 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 8 years ago
  85. fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
  86. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  87. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  88. 1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
  89. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  90. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  91. 85a0496 Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  92. c7a8b08 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago