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gerrit-public.fairphone.software / platform / external / webrtc / 0e96535be97916d8fcaa9873ffab3c636539f9d8 / call / audio_send_stream.h
  1. b8c775a Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api by Tim Na · 4 years, 7 months ago
  2. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 4 years, 8 months ago
  3. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 4 years, 10 months ago
  4. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 4 years, 10 months ago
  5. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 10 months ago
  6. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 4 years, 10 months ago
  7. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 4 years, 11 months ago
  8. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  9. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  10. 6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  11. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  12. cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 5 years ago
  13. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  14. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  15. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  16. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  17. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  18. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  19. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  20. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  21. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  22. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  23. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  24. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  25. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  26. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  27. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 6 years ago
  28. 77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 6 years ago
  29. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  30. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  31. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  32. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  33. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  34. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  35. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  36. b3944f0 Media track ID visibility at BWE level by Alex Narest · 7 years ago
  37. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  38. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  39. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/audio_send_stream.h]
  40. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  41. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  42. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  43. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
  44. 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
  45. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  46. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  47. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  48. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  49. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  50. 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  51. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago[Renamed (96%) from webrtc/api/call/audio_send_stream.h]
  52. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  53. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  54. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  55. 10cbb46 Fixing config for Audio BWE. by minyue · 8 years ago
  56. 6b825df Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  57. 940b6d6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  58. 189f9b1 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) by terelius · 8 years ago
  59. 1836fd6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  60. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  61. 7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  62. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago[Renamed (96%) from webrtc/audio_send_stream.h]
  63. 86cc6ff Variable audio bitrate. by mflodman · 8 years ago
  64. 9421853 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 8 years ago
  65. 971cab0 Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 8 years ago
  66. 6806136 Remove RED support from WebRtcVoiceEngine/MediaChannel by kwiberg · 8 years ago
  67. fd8be34 Remove webrtc/base/scoped_ptr.h by kwiberg · 8 years ago
  68. 6ab3db2 Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ ) by kwiberg · 8 years ago
  69. 65fc62e Remove webrtc/base/scoped_ptr.h by kwiberg · 8 years ago
  70. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 8 years ago
  71. bfefb03 Replace scoped_ptr with unique_ptr everywhere by kwiberg · 8 years ago
  72. 8842c3e Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 8 years ago
  73. 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
  74. 8886c81 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 8 years ago
  75. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  76. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  77. a4527c8 Add comments about the Audio parts of the public Call API being WIP. by Fredrik Solenberg · 9 years ago
  78. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  79. 85a0496 Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  80. c7a8b08 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago
  81. cf18b34 Align new VoE API with design. by solenberg · 9 years ago
  82. 2d56668 Unify Transport and newapi::Transport interfaces. by pbos · 9 years ago
  83. 4fbae2b Add send transports to individual webrtc::Call streams. by solenberg · 9 years ago
  84. cd67022 Define Stream base classes by Jelena Marusic · 9 years ago
  85. 04f4931 VoE2 API draft by Fredrik Solenberg · 9 years ago