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gerrit-public.fairphone.software / platform / external / webrtc / 5b74f8d7ff8a8d4f3b953da1ed57498d7fff92f6 / audio / audio_send_stream_unittest.cc
  1. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  2. eb90e6f Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest by Danil Chapovalov · 5 years ago
  3. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  4. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  5. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
  6. 40de3cc Propagating TargetRate struct to BitrateAllocator. by Sebastian Jansson · 5 years ago
  7. 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 5 years ago
  8. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  9. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  10. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  11. 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 5 years ago
  12. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  13. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  14. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  15. 9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 5 years ago
  16. 413ccc4 Stop DCHECK which occurs in ANA BitrateController when overhead is zero. by Bjorn A Mellem · 5 years ago
  17. 63658d0 Revert "Ensure that we always set values for min and max audio bitrate." by Daniel Lee · 5 years ago
  18. e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 5 years ago
  19. 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 5 years ago
  20. 31660fd Avoid using global task queue factory in audio/ unittests by Danil Chapovalov · 5 years ago
  21. 44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 6 years ago
  22. 0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 6 years ago
  23. ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 6 years ago
  24. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  25. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  26. da6806c Injecting Clock into BitrateAllocator. by Sebastian Jansson · 6 years ago
  27. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
  28. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
  29. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
  30. 14a7cf9 Adds CallEncoder to ChannelSend. by Sebastian Jansson · 6 years ago
  31. 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
  32. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  33. e7d08df Fix chromium roll into WebRTC. by Artem Titov · 6 years ago
  34. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  35. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  36. e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
  37. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  38. 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
  39. 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
  40. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  41. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  42. 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
  43. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  44. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  45. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  46. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
  47. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  48. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  49. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  50. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  51. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  52. 35fa280 Adds allocated rate without feedback to new congestion controller. by Sebastian Jansson · 6 years ago
  53. 9701e0c Makes treatment of received reports of packets lost signed. by Sebastian Jansson · 6 years ago
  54. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  55. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
  56. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  57. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  58. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  59. 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
  60. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 6 years ago
  61. fe617a3 Adding has_packet_feedback to LimitObserver callback. by Sebastian Jansson · 6 years ago
  62. d6fbf2a Tests: Pass codec ID argument to audio codecs by Karl Wiberg · 7 years ago
  63. f69e768 Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 1. by philipel · 7 years ago
  64. ef9daee Using mock transport controller in audio unit tests. by Sebastian Jansson · 7 years ago
  65. 41f16be Silencing warnings in audio send stream unit tests. by Sebastian Jansson · 7 years ago
  66. 97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
  67. 1896cec Removed dependencies from audio send stream unit test by Sebastian Jansson · 7 years ago
  68. 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
  69. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  70. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  71. f85e31b Don't (re-)configure BitrateObserver unless already sending by Oskar Sundbom · 7 years ago
  72. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  73. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  74. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  75. 2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 7 years ago
  76. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  77. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  78. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  79. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_send_stream_unittest.cc]
  80. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  81. 5c8942a Move PacedSender ownership to RtpTransportControllerSend. by Stefan Holmer · 7 years ago
  82. 8de1826 Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by minyue-webrtc · 7 years ago
  83. 7df370b Revert "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by Minyue Li · 7 years ago
  84. 4a88120 Allow AudioSendStream to reconfig AudioNetworkAdaptor by minyue-webrtc · 7 years ago
  85. 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
  86. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  87. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  88. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  89. 1129df2 Always ResetSenderCongestionControlObjects before RegisterEtc... by ossu · 7 years ago
  90. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  91. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 7 years ago
  92. 8c96a14 Simple tests for Call::SetBitrateConfig. by zstein · 7 years ago
  93. 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
  94. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  95. 3b9ff38 Have AudioSendStream register CNG payload types with the RtpRtcpModule. by ossu · 7 years ago
  96. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  97. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 7 years ago
  98. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 7 years ago
  99. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 7 years ago
  100. 40854ea Reland of Delete class MockCongestionController. (patchset #1 id:1 of https://codereview.webrtc.org/2762133003/ ) by nisse · 7 years ago