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gerrit-public.fairphone.software / platform / external / webrtc / 7d76a31f3dfb0de235da6d4757359deee6d08d41 / audio / audio_send_stream.cc
  1. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  2. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  3. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  4. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  5. 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
  6. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  7. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  8. b686396 Makes AudioSendStream signal that it's part of allocation. by Sebastian Jansson · 6 years ago
  9. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  10. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  11. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  12. 35fa280 Adds allocated rate without feedback to new congestion controller. by Sebastian Jansson · 6 years ago
  13. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  14. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
  15. bbbe4e1 Better handle target audio bitrate allocation. by Alex Narest · 6 years ago
  16. 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 · 6 years ago
  17. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  18. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  19. 867e510 Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled. by Alex Narest · 6 years ago
  20. 24ad720 Uses config struct with bitrate allocator. by Sebastian Jansson · 6 years ago
  21. abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 6 years ago
  22. 003930a Fix MID not always getting set with audio by Steve Anton · 6 years ago
  23. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 6 years ago
  24. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 6 years ago
  25. 77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 6 years ago
  26. 763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 6 years ago
  27. 98cd810 Production code: Pass codec ID argument to audio codecs by Karl Wiberg · 7 years ago
  28. 3c24ea8 Removed SetTransportOverhead in transport controller. by Sebastian Jansson · 7 years ago
  29. 4c1ffb8 Removing access to pacer in rtp controller. by Sebastian Jansson · 7 years ago
  30. e4be6da Removing access to send side cc in rtp controller. by Sebastian Jansson · 7 years ago
  31. 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
  32. 24ea822 Remove logging in audio/* from release builds. by Jonas Olsson · 7 years ago
  33. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  34. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  35. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  36. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  37. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  38. f85e31b Don't (re-)configure BitrateObserver unless already sending by Oskar Sundbom · 7 years ago
  39. aaedf75 Replace VoEBase::[Start/Stop]Send(). by Fredrik Solenberg · 7 years ago
  40. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  41. cedd351 Do not add audio bitrate observer if TWCC sending is not supported by Alex Narest · 7 years ago
  42. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  43. 2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 7 years ago
  44. 8d9c540 Deprecated BitrateController::CreateRtcpBandwidthObserver. by Sebastian Jansson · 7 years ago
  45. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  46. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  47. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  48. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  49. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
  50. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
  51. b3944f0 Media track ID visibility at BWE level by Alex Narest · 7 years ago
  52. 90e1f53 Fix potentional race in AudioSendStream constructor by Danil Chapovalov · 7 years ago
  53. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  54. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  55. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_send_stream.cc]
  56. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  57. 3651fdd Uncomment thread-check in AudioSendStream::OnPacketFeedbackVector() by eladalon · 7 years ago
  58. 8de1826 Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by minyue-webrtc · 7 years ago
  59. 7df370b Revert "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by Minyue Li · 7 years ago
  60. 4a88120 Allow AudioSendStream to reconfig AudioNetworkAdaptor by minyue-webrtc · 7 years ago
  61. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
  62. c58f8c0 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 7 years ago
  63. 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 · 7 years ago
  64. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  65. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  66. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  67. 1129df2 Always ResetSenderCongestionControlObjects before RegisterEtc... by ossu · 7 years ago
  68. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  69. edd6eea Rename elad.alon to eladalon, to avoid confusion between repositories. by eladalon · 7 years ago
  70. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 7 years ago
  71. 93e4522 Renaming probing_interval to bwe_period globally. by minyue · 7 years ago
  72. 48368ad Fixing video loopback test with encoder factory. by minyue · 7 years ago
  73. 3b9ff38 Have AudioSendStream register CNG payload types with the RtpRtcpModule. by ossu · 7 years ago
  74. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  75. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 7 years ago
  76. fca900a Fix two invalid DCHECKs related to audio BWE. by stefan · 7 years ago
  77. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 7 years ago
  78. 4e76451 Fix UT failure by temporarily uncommenting by elad.alon · 7 years ago
  79. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 7 years ago
  80. dadb4dc Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 7 years ago
  81. d12a8e1 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 7 years ago
  82. 559af38 Split CongestionController into send- and receive-side classes. by nisse · 7 years ago
  83. 5bbf43f Move delay_based_bwe_ into CongestionController by elad.alon · 8 years ago
  84. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  85. 06f240b Clean out platform specific things from voice_engine_defines.h. by solenberg · 8 years ago
  86. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  87. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  88. 55d1ebb Enable periodic bitrate probing when application limited for audio BWE. by stefan · 8 years ago
  89. f4caaab Fix for bwe with overhead on audio only calls. by michaelt · 8 years ago
  90. 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  91. 9332b7d Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  92. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  93. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  94. 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
  95. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  96. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  97. d4adce4 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 8 years ago
  98. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  99. 7aba029 Make use of new APM statistics interface. by ivoc · 8 years ago
  100. 6f0b9fd Allowing resetting of AudioNetworkAdaptor in AudioSendStream. by minyue · 8 years ago