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gerrit-public.fairphone.software / platform / external / webrtc / 55718120e684bd4861f7a94e362a8c1a2e1c7a66 / audio / audio_send_stream_unittest.cc
  1. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  2. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  3. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  4. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
  5. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  6. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  7. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  8. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  9. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  10. 35fa280 Adds allocated rate without feedback to new congestion controller. by Sebastian Jansson · 6 years ago
  11. 9701e0c Makes treatment of received reports of packets lost signed. by Sebastian Jansson · 6 years ago
  12. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  13. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
  14. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  15. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  16. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  17. 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
  18. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 7 years ago
  19. fe617a3 Adding has_packet_feedback to LimitObserver callback. by Sebastian Jansson · 7 years ago
  20. d6fbf2a Tests: Pass codec ID argument to audio codecs by Karl Wiberg · 7 years ago
  21. f69e768 Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 1. by philipel · 7 years ago
  22. ef9daee Using mock transport controller in audio unit tests. by Sebastian Jansson · 7 years ago
  23. 41f16be Silencing warnings in audio send stream unit tests. by Sebastian Jansson · 7 years ago
  24. 97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
  25. 1896cec Removed dependencies from audio send stream unit test by Sebastian Jansson · 7 years ago
  26. 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
  27. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  28. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  29. f85e31b Don't (re-)configure BitrateObserver unless already sending by Oskar Sundbom · 7 years ago
  30. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  31. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  32. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  33. 2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 7 years ago
  34. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  35. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  36. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  37. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_send_stream_unittest.cc]
  38. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  39. 5c8942a Move PacedSender ownership to RtpTransportControllerSend. by Stefan Holmer · 7 years ago
  40. 8de1826 Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by minyue-webrtc · 7 years ago
  41. 7df370b Revert "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by Minyue Li · 7 years ago
  42. 4a88120 Allow AudioSendStream to reconfig AudioNetworkAdaptor by minyue-webrtc · 7 years ago
  43. 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
  44. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  45. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  46. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  47. 1129df2 Always ResetSenderCongestionControlObjects before RegisterEtc... by ossu · 7 years ago
  48. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  49. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 7 years ago
  50. 8c96a14 Simple tests for Call::SetBitrateConfig. by zstein · 7 years ago
  51. 7cb69d5 This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 7 years ago
  52. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  53. 3b9ff38 Have AudioSendStream register CNG payload types with the RtpRtcpModule. by ossu · 7 years ago
  54. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  55. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 7 years ago
  56. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  57. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  58. 40854ea Reland of Delete class MockCongestionController. (patchset #1 id:1 of https://codereview.webrtc.org/2762133003/ ) by nisse · 8 years ago
  59. e27f1e7 Revert of Delete class MockCongestionController. (patchset #4 id:60001 of https://codereview.webrtc.org/2762023004/ ) by skvlad · 8 years ago
  60. d19bcb7 Delete class MockCongestionController. by nisse · 8 years ago
  61. 559af38 Split CongestionController into send- and receive-side classes. by nisse · 8 years ago
  62. 5bbf43f Move delay_based_bwe_ into CongestionController by elad.alon · 8 years ago
  63. fb1fa44 Remove MockRemoteBitrateObserver (unused) by elad.alon · 8 years ago
  64. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  65. 922246a Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  66. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  67. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  68. 9332b7d Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  69. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  70. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  71. 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
  72. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  73. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  74. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  75. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  76. 7aba029 Make use of new APM statistics interface. by ivoc · 8 years ago
  77. 6f0b9fd Allowing resetting of AudioNetworkAdaptor in AudioSendStream. by minyue · 8 years ago
  78. 10cbb46 Fixing config for Audio BWE. by minyue · 8 years ago
  79. b521aa7 Clean up abs-send-time for audio. by stefan · 8 years ago
  80. 6b825df Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  81. 940b6d6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  82. 189f9b1 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) by terelius · 8 years ago
  83. 2d81eb3 Fix BWE simulations so that it uses the delay based BWE. by terelius · 8 years ago
  84. 1836fd6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  85. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  86. 7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  87. 982bf89 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  88. e0729c5 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
  89. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  90. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  91. e035e2d Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets. by terelius · 8 years ago
  92. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 8 years ago
  93. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
  94. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  95. 86cc6ff Variable audio bitrate. by mflodman · 8 years ago
  96. 14d5dbe Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface" by ivoc · 8 years ago
  97. 9e03c3b Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 8 years ago
  98. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 8 years ago
  99. 9421853 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 8 years ago
  100. 971cab0 Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 8 years ago