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gerrit-public.fairphone.software / platform / external / webrtc / 26e084906140af8dd7b75d381a6476161bba5bb2 / audio / audio_send_stream.h
  1. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 7 years ago
  2. 763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 7 years ago
  3. 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
  4. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  5. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  6. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  7. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  8. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  9. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  10. cedd351 Do not add audio bitrate observer if TWCC sending is not supported by Alex Narest · 7 years ago
  11. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  12. 8d9c540 Deprecated BitrateController::CreateRtcpBandwidthObserver. by Sebastian Jansson · 7 years ago
  13. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  14. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_send_stream.h]
  15. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  16. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 8 years ago
  17. c58f8c0 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 8 years ago
  18. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  19. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  20. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  21. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 8 years ago
  22. 93e4522 Renaming probing_interval to bwe_period globally. by minyue · 8 years ago
  23. 3b9ff38 Have AudioSendStream register CNG payload types with the RtpRtcpModule. by ossu · 8 years ago
  24. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  25. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  26. d12a8e1 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 8 years ago
  27. 559af38 Split CongestionController into send- and receive-side classes. by nisse · 8 years ago
  28. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  29. f4caaab Fix for bwe with overhead on audio only calls. by michaelt · 8 years ago
  30. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  31. 9332b7d Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  32. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  33. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  34. 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
  35. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  36. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  37. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  38. 7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  39. 982bf89 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  40. e0729c5 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
  41. e035e2d Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets. by terelius · 8 years ago
  42. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 8 years ago
  43. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  44. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
  45. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  46. 86cc6ff Variable audio bitrate. by mflodman · 9 years ago
  47. 9421853 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 9 years ago
  48. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  49. 4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  50. 8842c3e Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 9 years ago
  51. 3ecb5c8 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ ) by solenberg · 9 years ago
  52. 8886c81 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 9 years ago
  53. fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
  54. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  55. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  56. 1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
  57. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  58. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  59. 85a0496 Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  60. c7a8b08 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago