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webrtc / src / 8d085422ed9d8ce684c4dc4fc1f95edc18743356 / audio / audio_send_stream.h
  1. 8037fc6 Migrate absl::optional to std::optional by Florent Castelli · 6 months ago
  2. 96c1b9c Add variables to lend unused audio bits to video by Dan Tan · 8 months ago
  3. c157f29 Pass Environment into audio ChannelSend by Danil Chapovalov · 10 months ago
  4. 9d9b3a3 Add option for the audio encoder to allocate a bitrate range. by Jakob Ivarsson · 1 year, 1 month ago
  5. 68e85b8 Adds WebRTC-Audio-PriorityBitrate for controlling audio/video rate allocation by Dan Tan · 1 year, 1 month ago
  6. 340d6c0 Remove packet overhead lock and cached bitrate constraints. by Jakob Ivarsson · 1 year, 2 months ago
  7. b1799b0 Cleanup usage of the rtc::TaskQueue in audio/ by Danil Chapovalov · 1 year, 2 months ago
  8. a2cf8ee Simplify handling rtcp messages in audio send channel by Danil Chapovalov · 1 year, 10 months ago
  9. 50b0a76 Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream" by Per Kjellander · 1 year, 11 months ago
  10. 73f048d Revert "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream" by Tomas Gunnarsson · 1 year, 11 months ago
  11. dd557fd [WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream by Per K · 1 year, 11 months ago
  12. e0b4cab Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead by Per Kjellander · 2 years, 3 months ago
  13. acabb36 pc: Add asynchronous RtpSender::SetParameters() call by Florent Castelli · 2 years, 4 months ago
  14. 828ef91 Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface by Per Kjellander · 2 years, 5 months ago
  15. e62c2f2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf by Jonas Oreland · 3 years ago
  16. a943e73 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 7/inf by Jonas Oreland · 3 years ago
  17. b0ea637 Use backticks not vertical bars to denote variables in comments for /audio by Artem Titov · 3 years, 8 months ago
  18. eb61b7f ModuleRtcRtcpImpl2: remove Module inheritance. by Markus Handell · 3 years, 9 months ago
  19. 3907e7b AudioSendStream: s/worker_queue_/rtp_transport_queue_/g by Markus Handell · 3 years, 10 months ago
  20. d15a575 Use SequenceChecker from public API by Artem Titov · 4 years, 1 month ago
  21. a208861 Reland "Fix data race for config_ in AudioSendStream" by Artem Titov · 4 years, 1 month ago
  22. 76a1041 Revert "Fix data race for config_ in AudioSendStream" by Henrik Boström · 4 years, 1 month ago
  23. 51e5c4b Fix data race for config_ in AudioSendStream by Artem Titov · 4 years, 1 month ago
  24. 47a03e8 Default enable sending transport sequence numbers on audio packets. by Jakob Ivarsson · 4 years, 4 months ago
  25. de95329 Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS by Niels Möller · 4 years, 6 months ago
  26. a166a35 webrtc::AudioSendStream: Add lock annotation to audio_level_ by Sam Zackrisson · 4 years, 8 months ago
  27. 6287280 Migrate audio/ to use webrtc::Mutex by Markus Handell · 4 years, 8 months ago
  28. f25761d Remove dependency from RtpRtcp on the Module interface. by Tomas Gunnarsson · 4 years, 9 months ago
  29. cf6544a Avoids unnecessary calls to audio encoder. by Erik Språng · 4 years, 10 months ago
  30. 04e1bab Replaces OverheadObserver with simple getter. by Erik Språng · 4 years, 10 months ago
  31. 9abc6bd Reduce audiosendstream dependency on rttstats (dead code). by Tommi · 4 years, 11 months ago
  32. 74dadc1 Ready to support of absolute capture timestamp header extension. by Minyue Li · 5 years ago
  33. 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 5 years ago
  34. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
  35. 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 5 years ago
  36. f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 5 years ago
  37. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
  38. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 6 years ago
  39. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 6 years ago
  40. 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 6 years ago
  41. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 6 years ago
  42. 9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
  43. 63658d0 Revert "Ensure that we always set values for min and max audio bitrate." by Daniel Lee · 6 years ago
  44. e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
  45. 44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 6 years ago
  46. 0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 6 years ago
  47. 8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 6 years ago
  48. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  49. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  50. 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
  51. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
  52. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  53. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  54. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  55. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  56. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  57. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  58. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  59. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  60. 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
  61. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  62. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  63. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 7 years ago
  64. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 7 years ago
  65. 763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 7 years ago
  66. 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
  67. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  68. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  69. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  70. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  71. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  72. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  73. cedd351 Do not add audio bitrate observer if TWCC sending is not supported by Alex Narest · 7 years ago
  74. 56d46090 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  75. 8d9c540 Deprecated BitrateController::CreateRtcpBandwidthObserver. by Sebastian Jansson · 7 years ago
  76. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  77. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_send_stream.h]
  78. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 8 years ago
  79. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 8 years ago
  80. c58f8c0 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 8 years ago
  81. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  82. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  83. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  84. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 8 years ago
  85. 93e4522 Renaming probing_interval to bwe_period globally. by minyue · 8 years ago
  86. 3b9ff38 Have AudioSendStream register CNG payload types with the RtpRtcpModule. by ossu · 8 years ago
  87. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  88. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  89. d12a8e1 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 8 years ago
  90. 559af38 Split CongestionController into send- and receive-side classes. by nisse · 8 years ago
  91. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  92. f4caaab Fix for bwe with overhead on audio only calls. by michaelt · 8 years ago
  93. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  94. 9332b7d Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  95. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  96. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  97. 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
  98. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  99. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  100. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago