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gerrit-public.fairphone.software / platform / external / webrtc / 67a39ac511f627f6dd9a557297018434e0e06ed3 / audio
  1. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  2. 31d8b52 Delete unneeded includes of rtc_base/stringutils.h. by Niels Möller · 6 years ago
  3. 3d2ed19 Remove Transport implementation from ChannelSend by Fredrik Solenberg · 6 years ago
  4. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  5. 2a977cf For audio receive channel use default max reordering threshold instead of 0 by Danil Chapovalov · 6 years ago
  6. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  7. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  8. e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
  9. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  10. 8ce0d2b In ReceiveStatistic require callbacks during construction by Danil Chapovalov · 6 years ago
  11. e3abb81 Decouple //rtc_base:rtc_base_tests_utils from gunit. by Mirko Bonadei · 6 years ago
  12. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  13. 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
  14. 8b5d9d8 Remove the audio/video split for the RTCP report intervals. by Jiawei Ou · 6 years ago
  15. 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
  16. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  17. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  18. 985a1f3 Add const or GUARDED_BY on a few ChannelSend members by Niels Möller · 6 years ago
  19. 26e88b0 Replace RTC_DCHECK by RTC_DCHECK_RUN_ON for worker thread. by Niels Möller · 6 years ago
  20. eb13484 Remove ChannelSendState by Fredrik Solenberg · 6 years ago
  21. c5e8be3 Remove ChannelReceiveState by Fredrik Solenberg · 6 years ago
  22. 78e88fe Move NetworkStatistics and AudioDecodingCallStats from common_types.h by Fredrik Solenberg · 6 years ago
  23. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  24. 179a392 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. by Piotr (Peter) Slatala · 6 years ago
  25. 1eebec9 Fix data race in channel_send.cc by Piotr (Peter) Slatala · 6 years ago
  26. 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
  27. 2681523 Tweak ChannelSend interface, to make it closer to ChannelSendProxy by Niels Möller · 6 years ago
  28. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  29. 8fb5746 Delete obsolete interface class RtpData by Niels Möller · 6 years ago
  30. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  31. 80c6762 Tweak ChannelReceive interface, to make it closer to ChannelReceiveProxy by Niels Möller · 6 years ago
  32. b768e88 Reland "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  33. 61c6e56 Revert "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  34. a7f77a7 Isolating APM API build target: making :api an actual target. by Alessio Bazzica · 6 years ago
  35. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  36. 967f7d5 Add audio level to CSRC class by Jonas Oreland · 6 years ago
  37. fd1a2fb Set RtpRtcp config receive_only in voe::ChannelReceive by Niels Möller · 6 years ago
  38. 273d029 Implement data channel methods in LoopbackMediaTransport. by Bjorn Mellem · 6 years ago
  39. 2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 6 years ago
  40. 56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 6 years ago
  41. 21cddff Harmonize paths to dependent targets. by Yves Gerey · 6 years ago
  42. fcc3981 Revert "Use only first payload timestamp for RTCP SR generation for audio" by Ilya Nikolaevskiy · 6 years ago
  43. 9a0662a Use only first payload timestamp for RTCP SR generation for audio by Ilya Nikolaevskiy · 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. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  50. 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
  51. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  52. 2dfa998 Reland "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
  53. c538fc7 Revert "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
  54. 5ccdc13 Prefix flag macros with WEBRTC_. by Mirko Bonadei · 6 years ago
  55. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  56. 40a7a35 Extract functionality of test_main into separate library. by Artem Titov · 6 years ago
  57. b686396 Makes AudioSendStream signal that it's part of allocation. by Sebastian Jansson · 6 years ago
  58. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
  59. 2e00abc Reland "[cleanup] Remove useless includes." by Yves Gerey · 6 years ago
  60. 64be7fa Move FecController to RtpVideoSender. by Stefan Holmer · 6 years ago
  61. 96a0f61 Revert "[cleanup] Remove useless includes." by Oleh Prypin · 6 years ago
  62. be8b534 [cleanup] Remove useless includes. by Yves Gerey · 6 years ago
  63. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 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. 4a72ba9 Delete RtpReceiver and related code. by Niels Möller · 6 years ago
  67. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  68. 35fa280 Adds allocated rate without feedback to new congestion controller. by Sebastian Jansson · 6 years ago
  69. 1f3206c Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1 by Niels Möller · 6 years ago
  70. 17f4878 Remove deprecated field_trial_default and metrics_default. by Mirko Bonadei · 6 years ago
  71. 7988e5c Remove echo_cancellation() and echo_control_mobile() interface access outside APM by Sam Zackrisson · 6 years ago
  72. 637b0b5 Make Python-based performance tests output an empty result output.json by Oleh Prypin · 6 years ago
  73. db12856 Cleanup modules_common_types by Danil Chapovalov · 6 years ago
  74. 6151828 Delete always true member voe::Channel::pacing_enabled_ by Niels Möller · 6 years ago
  75. 5304a32 Delete StreamStatistician::IsRetransmitOfOldPacket by Niels Möller · 6 years ago
  76. 8fdcac3 Remove clang:find_bad_constructs suppression from call:call. by Mirko Bonadei · 6 years ago
  77. 2370b08 Revert "Update packetsLost and jitter stats any time a packet is received." by Qingsi Wang · 6 years ago
  78. 4e199e9 Mark DirectTransport subclasses ctors as deprecated and switch from them by Artem Titov · 6 years ago
  79. 46c4e60 Introduce SimulatedNetworkReceiverInterface. by Artem Titov · 6 years ago
  80. fa2b2d6 Delete use of RtpPayloadRegistry. by Niels Möller · 6 years ago
  81. 30b4839 Refactor voe::Channel to not use RtpReceiver. by Niels Möller · 6 years ago
  82. 9701e0c Makes treatment of received reports of packets lost signed. by Sebastian Jansson · 6 years ago
  83. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  84. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
  85. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
  86. 264bee8 Remove memcheck. by Mirko Bonadei · 6 years ago
  87. ab4a530 Delete telephone-event handling from RTPReceiverAudio. by Niels Möller · 6 years ago
  88. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  89. 3890262 Reland "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
  90. a61f7db Revert "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
  91. 06f66c7 Removing unneeded dependency. by Mirko Bonadei · 6 years ago
  92. bbbe4e1 Better handle target audio bitrate allocation. by Alex Narest · 6 years ago
  93. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  94. 64b17c2 Remove StreamStatistician::IsPacketInOrder by Danil Chapovalov · 7 years ago
  95. bcf9180 Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream. by Alex Narest · 7 years ago
  96. 8491693 Update packetsLost and jitter stats any time a packet is received. by Taylor Brandstetter · 7 years ago
  97. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  98. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 7 years ago
  99. 867e510 Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled. by Alex Narest · 7 years ago
  100. f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 7 years ago