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webrtc / src / / c951d1b0f6f450ab668ecea6635f2e5249beff2a / audio / audio_send_stream.h
  1. a2cf8ee Simplify handling rtcp messages in audio send channel by Danil Chapovalov · 1 year, 11 months ago
  2. 50b0a76 Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream" by Per Kjellander · 2 years ago
  3. 73f048d Revert "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream" by Tomas Gunnarsson · 2 years ago
  4. dd557fd [WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream by Per K · 2 years ago
  5. e0b4cab Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead by Per Kjellander · 2 years, 4 months ago
  6. acabb36 pc: Add asynchronous RtpSender::SetParameters() call by Florent Castelli · 2 years, 5 months ago
  7. 828ef91 Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface by Per Kjellander · 2 years, 6 months ago
  8. e62c2f2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf by Jonas Oreland · 3 years ago
  9. a943e73 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 7/inf by Jonas Oreland · 3 years ago
  10. b0ea637 Use backticks not vertical bars to denote variables in comments for /audio by Artem Titov · 3 years, 8 months ago
  11. eb61b7f ModuleRtcRtcpImpl2: remove Module inheritance. by Markus Handell · 3 years, 9 months ago
  12. 3907e7b AudioSendStream: s/worker_queue_/rtp_transport_queue_/g by Markus Handell · 3 years, 10 months ago
  13. d15a575 Use SequenceChecker from public API by Artem Titov · 4 years, 2 months ago
  14. a208861 Reland "Fix data race for config_ in AudioSendStream" by Artem Titov · 4 years, 2 months ago
  15. 76a1041 Revert "Fix data race for config_ in AudioSendStream" by Henrik Boström · 4 years, 2 months ago
  16. 51e5c4b Fix data race for config_ in AudioSendStream by Artem Titov · 4 years, 2 months ago
  17. 47a03e8 Default enable sending transport sequence numbers on audio packets. by Jakob Ivarsson · 4 years, 4 months ago
  18. de95329 Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS by Niels Möller · 4 years, 6 months ago
  19. a166a35 webrtc::AudioSendStream: Add lock annotation to audio_level_ by Sam Zackrisson · 4 years, 9 months ago
  20. 6287280 Migrate audio/ to use webrtc::Mutex by Markus Handell · 4 years, 9 months ago
  21. f25761d Remove dependency from RtpRtcp on the Module interface. by Tomas Gunnarsson · 4 years, 10 months ago
  22. cf6544a Avoids unnecessary calls to audio encoder. by Erik Språng · 4 years, 11 months ago
  23. 04e1bab Replaces OverheadObserver with simple getter. by Erik Språng · 5 years ago
  24. 9abc6bd Reduce audiosendstream dependency on rttstats (dead code). by Tommi · 5 years ago
  25. 74dadc1 Ready to support of absolute capture timestamp header extension. by Minyue Li · 5 years ago
  26. 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 5 years ago
  27. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
  28. 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 5 years ago
  29. f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 5 years ago
  30. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
  31. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 6 years ago
  32. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 6 years ago
  33. 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 6 years ago
  34. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 6 years ago
  35. 9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
  36. 63658d0 Revert "Ensure that we always set values for min and max audio bitrate." by Daniel Lee · 6 years ago
  37. e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
  38. 44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 6 years ago
  39. 0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 6 years ago
  40. 8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 6 years ago
  41. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  42. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  43. 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
  44. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
  45. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  46. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  47. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  48. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  49. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  50. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  51. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  52. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  53. 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
  54. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  55. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  56. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 7 years ago
  57. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 7 years ago
  58. 763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 7 years ago
  59. 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
  60. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  61. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  62. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  63. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  64. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  65. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  66. cedd351 Do not add audio bitrate observer if TWCC sending is not supported by Alex Narest · 7 years ago
  67. 56d46090 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  68. 8d9c540 Deprecated BitrateController::CreateRtcpBandwidthObserver. by Sebastian Jansson · 7 years ago
  69. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  70. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/audio/audio_send_stream.h]
  71. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 8 years ago
  72. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 8 years ago
  73. c58f8c0 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 8 years ago
  74. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  75. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  76. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  77. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 8 years ago
  78. 93e4522 Renaming probing_interval to bwe_period globally. by minyue · 8 years ago
  79. 3b9ff38 Have AudioSendStream register CNG payload types with the RtpRtcpModule. by ossu · 8 years ago
  80. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  81. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  82. d12a8e1 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 8 years ago
  83. 559af38 Split CongestionController into send- and receive-side classes. by nisse · 8 years ago
  84. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  85. f4caaab Fix for bwe with overhead on audio only calls. by michaelt · 8 years ago
  86. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  87. 9332b7d Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  88. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  89. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  90. 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
  91. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  92. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  93. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  94. 7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  95. 982bf89 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  96. e0729c5 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
  97. e035e2d Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets. by terelius · 9 years ago
  98. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 9 years ago
  99. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago
  100. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 9 years ago