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gerrit-public.fairphone.software / platform / external / webrtc / 04e1bab1b324e08b8dec4bf617e2afb13b6bde3b / audio / audio_send_stream_unittest.cc
  1. 04e1bab Replaces OverheadObserver with simple getter. by Erik Språng · 4 years, 5 months ago
  2. 9abc6bd Reduce audiosendstream dependency on rttstats (dead code). by Tommi · 4 years, 5 months ago
  3. cc73ed3 APM: Add build flag to allow building WebRTC without APM by Per Åhgren · 4 years, 5 months ago
  4. d2aa8f9 Insert audio frame transformer between encoder and packetizer. by Marina Ciocea · 4 years, 6 months ago
  5. d14525e Make sure that the audio stream is allocated with the correct overhead. by Jakob Ivarsson · 4 years, 7 months ago
  6. cad3e0e Replace DataSize and DataRate factories with newer versions by Danil Chapovalov · 4 years, 8 months ago
  7. 0c626af Use newer version of TimeDelta and TimeStamp factories in webrtc by Danil Chapovalov · 4 years, 8 months ago
  8. c3eb9fd Reland "Reland "Only include overhead if using send side bandwidth estimation."" by Sebastian Jansson · 4 years, 8 months ago
  9. 4356490 Revert "Reland "Only include overhead if using send side bandwidth estimation."" by Mirko Bonadei · 4 years, 8 months ago
  10. 086055d Reland "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 4 years, 8 months ago
  11. c709412 Revert "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 4 years, 8 months ago
  12. 8c79c6e Only include overhead if using send side bandwidth estimation. by Sebastian Jansson · 4 years, 9 months ago
  13. 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 4 years, 9 months ago
  14. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 4 years, 10 months ago
  15. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
  16. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  17. eb90e6f Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest by Danil Chapovalov · 5 years ago
  18. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  19. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  20. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
  21. 40de3cc Propagating TargetRate struct to BitrateAllocator. by Sebastian Jansson · 5 years ago
  22. 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 5 years ago
  23. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  24. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  25. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  26. 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 5 years ago
  27. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  28. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  29. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  30. 9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 5 years ago
  31. 413ccc4 Stop DCHECK which occurs in ANA BitrateController when overhead is zero. by Bjorn A Mellem · 5 years ago
  32. 63658d0 Revert "Ensure that we always set values for min and max audio bitrate." by Daniel Lee · 5 years ago
  33. e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 5 years ago
  34. 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 5 years ago
  35. 31660fd Avoid using global task queue factory in audio/ unittests by Danil Chapovalov · 6 years ago
  36. 44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 6 years ago
  37. 0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 6 years ago
  38. ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 6 years ago
  39. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  40. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  41. da6806c Injecting Clock into BitrateAllocator. by Sebastian Jansson · 6 years ago
  42. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
  43. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
  44. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
  45. 14a7cf9 Adds CallEncoder to ChannelSend. by Sebastian Jansson · 6 years ago
  46. 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
  47. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  48. e7d08df Fix chromium roll into WebRTC. by Artem Titov · 6 years ago
  49. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  50. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  51. e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
  52. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  53. 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
  54. 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
  55. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  56. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  57. 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
  58. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  59. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  60. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  61. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
  62. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  63. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  64. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  65. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  66. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  67. 35fa280 Adds allocated rate without feedback to new congestion controller. by Sebastian Jansson · 6 years ago
  68. 9701e0c Makes treatment of received reports of packets lost signed. by Sebastian Jansson · 6 years ago
  69. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  70. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
  71. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  72. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  73. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  74. 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
  75. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 7 years ago
  76. fe617a3 Adding has_packet_feedback to LimitObserver callback. by Sebastian Jansson · 7 years ago
  77. d6fbf2a Tests: Pass codec ID argument to audio codecs by Karl Wiberg · 7 years ago
  78. f69e768 Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 1. by philipel · 7 years ago
  79. ef9daee Using mock transport controller in audio unit tests. by Sebastian Jansson · 7 years ago
  80. 41f16be Silencing warnings in audio send stream unit tests. by Sebastian Jansson · 7 years ago
  81. 97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
  82. 1896cec Removed dependencies from audio send stream unit test by Sebastian Jansson · 7 years ago
  83. 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
  84. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  85. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  86. f85e31b Don't (re-)configure BitrateObserver unless already sending by Oskar Sundbom · 7 years ago
  87. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  88. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  89. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  90. 2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 7 years ago
  91. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  92. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  93. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  94. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_send_stream_unittest.cc]
  95. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  96. 5c8942a Move PacedSender ownership to RtpTransportControllerSend. by Stefan Holmer · 7 years ago
  97. 8de1826 Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by minyue-webrtc · 7 years ago
  98. 7df370b Revert "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by Minyue Li · 7 years ago
  99. 4a88120 Allow AudioSendStream to reconfig AudioNetworkAdaptor by minyue-webrtc · 7 years ago
  100. 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