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webrtc / src / 4356490b7b66714550221273479de77ace320226 / audio / audio_send_stream.cc
  1. 4356490 Revert "Reland "Only include overhead if using send side bandwidth estimation."" by Mirko Bonadei · 5 years ago
  2. 086055d Reland "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 5 years ago
  3. c709412 Revert "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 5 years ago
  4. 8c79c6e Only include overhead if using send side bandwidth estimation. by Sebastian Jansson · 5 years ago
  5. 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 5 years ago
  6. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  7. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
  8. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
  9. f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 5 years ago
  10. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  11. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  12. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  13. cd0eedb Don't allocate audio if we have no transport sequence number. by Sebastian Jansson · 5 years ago
  14. 0a6510d Removes rtp_transport checks in AudioSendStream by Sebastian Jansson · 5 years ago
  15. 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 5 years ago
  16. 0429f78 Base overhead calculation for audio priority rate on available data. by Sebastian Jansson · 5 years ago
  17. f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 5 years ago
  18. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
  19. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 6 years ago
  20. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 6 years ago
  21. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 6 years ago
  22. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 6 years ago
  23. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 6 years ago
  24. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 6 years ago
  25. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 6 years ago
  26. 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 6 years ago
  27. aa59eca Move RtpPacketSender and merge it with RtpPacketPacer. by Erik Språng · 6 years ago
  28. 0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 6 years ago
  29. 4c2c412 Set local ssrc at construction (audio) by Erik Språng · 6 years ago
  30. e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 6 years ago
  31. 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 6 years ago
  32. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 6 years ago
  33. 6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 6 years ago
  34. a352248 Add a config flag to disable the audio ALR probing request. by Christoffer Rodbro · 6 years ago
  35. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 6 years ago
  36. 8f119ca Enable experiments with audio bitrate priority. by Jonas Olsson · 6 years ago
  37. 9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
  38. 413ccc4 Stop DCHECK which occurs in ANA BitrateController when overhead is zero. by Bjorn A Mellem · 6 years ago
  39. 63658d0 Revert "Ensure that we always set values for min and max audio bitrate." by Daniel Lee · 6 years ago
  40. e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
  41. cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 6 years ago
  42. c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
  43. 741daaf Move rtc::FunctionView to the public API by Artem Titov · 6 years ago
  44. 44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 6 years ago
  45. 0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 6 years ago
  46. 8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 6 years ago
  47. ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 6 years ago
  48. 110c64b Delete unused key WebRTC-Audio-SendSideBwe-For-Video. by Christoffer Rodbro · 6 years ago
  49. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  50. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  51. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
  52. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
  53. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
  54. 14a7cf9 Adds CallEncoder to ChannelSend. by Sebastian Jansson · 6 years ago
  55. 464a557 Adds audio priority bitrate field trial parameter. by Sebastian Jansson · 6 years ago
  56. 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
  57. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
  58. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  59. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  60. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  61. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  62. e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
  63. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  64. 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
  65. 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
  66. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  67. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  68. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  69. 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
  70. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  71. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  72. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  73. 2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 6 years ago
  74. 56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 6 years ago
  75. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  76. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  77. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  78. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  79. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  80. 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
  81. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  82. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  83. b686396 Makes AudioSendStream signal that it's part of allocation. by Sebastian Jansson · 6 years ago
  84. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  85. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  86. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  87. 35fa280 Adds allocated rate without feedback to new congestion controller. by Sebastian Jansson · 6 years ago
  88. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 7 years ago
  89. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 7 years ago
  90. bbbe4e1 Better handle target audio bitrate allocation. by Alex Narest · 7 years ago
  91. bcf9180 Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream. by Alex Narest · 7 years ago
  92. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  93. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 7 years ago
  94. 867e510 Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled. by Alex Narest · 7 years ago
  95. 24ad720 Uses config struct with bitrate allocator. by Sebastian Jansson · 7 years ago
  96. abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 7 years ago
  97. 003930a Fix MID not always getting set with audio by Steve Anton · 7 years ago
  98. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 7 years ago
  99. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 7 years ago
  100. 77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 7 years ago