Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 08f6a6c672d3be5f2614729dc0eaaf2bcdb2c272 / audio
  1. d8d3248 Reland "Delete test/constants.h" by Elad Alon · 6 years ago
  2. 4f36b7a Revert "Delete test/constants.h" by Oleh Prypin · 6 years ago
  3. e049eba Revert "Add Sender and Receiver interfaces for MediaTransport audio" by Sergey Silkin · 6 years ago
  4. 0d8eed6 Add Sender and Receiver interfaces for MediaTransport audio by Niels Möller · 6 years ago
  5. afb5dbb Update ACM to use RTPHeader instead of WebRtcRTPHeader by Niels Möller · 6 years ago
  6. 389b167 Delete test/constants.h by Elad Alon · 6 years ago
  7. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
  8. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
  9. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
  10. 14a7cf9 Adds CallEncoder to ChannelSend. by Sebastian Jansson · 6 years ago
  11. 464a557 Adds audio priority bitrate field trial parameter. by Sebastian Jansson · 6 years ago
  12. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
  13. 1c54605 [clang-tidy] Apply performance-move-const-arg fixes (misc). by Mirko Bonadei · 6 years ago
  14. 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
  15. 80a8687 [clang-tidy] Apply performance-move-const-arg fixes (mutable lambdas). by Mirko Bonadei · 6 years ago
  16. 432c833 Remove redundant check in channel_receive.cc. by Ruslan Burakov · 6 years ago
  17. 01dc691 Delete sequence number save and restore in ChannelSend. by Niels Möller · 6 years ago
  18. c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 6 years ago
  19. fe055c1 [clang-tidy] Apply modernize-use-override fixes. by Mirko Bonadei · 6 years ago
  20. b4977de Receive-side ready for multiple channels. by Alex Loiko · 6 years ago
  21. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  22. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
  23. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  24. 5c2f1f0 Add some missing includes and dependencies. by Bjorn Terelius · 6 years ago
  25. e7d08df Fix chromium roll into WebRTC. by Artem Titov · 6 years ago
  26. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  27. 36faf0b Delete setting of unused variable nack_window_ms by Niels Möller · 6 years ago
  28. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  29. ba50223 Make voiceengine/audio transport stop using voice_detection() interface by Sam Zackrisson · 6 years ago
  30. 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
  31. ac63ac7 Update refcounting of AudioState to use rtc::RefCountedObject by Niels Möller · 6 years ago
  32. 40d5533 Include absl/memory/memory.h if absl::make_unique is used by Steve Anton · 6 years ago
  33. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  34. 31d8b52 Delete unneeded includes of rtc_base/stringutils.h. by Niels Möller · 6 years ago
  35. 3d2ed19 Remove Transport implementation from ChannelSend by Fredrik Solenberg · 6 years ago
  36. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  37. 2a977cf For audio receive channel use default max reordering threshold instead of 0 by Danil Chapovalov · 6 years ago
  38. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  39. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  40. e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
  41. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  42. 8ce0d2b In ReceiveStatistic require callbacks during construction by Danil Chapovalov · 6 years ago
  43. e3abb81 Decouple //rtc_base:rtc_base_tests_utils from gunit. by Mirko Bonadei · 6 years ago
  44. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  45. 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
  46. 8b5d9d8 Remove the audio/video split for the RTCP report intervals. by Jiawei Ou · 6 years ago
  47. 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
  48. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  49. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  50. 985a1f3 Add const or GUARDED_BY on a few ChannelSend members by Niels Möller · 6 years ago
  51. 26e88b0 Replace RTC_DCHECK by RTC_DCHECK_RUN_ON for worker thread. by Niels Möller · 6 years ago
  52. eb13484 Remove ChannelSendState by Fredrik Solenberg · 6 years ago
  53. c5e8be3 Remove ChannelReceiveState by Fredrik Solenberg · 6 years ago
  54. 78e88fe Move NetworkStatistics and AudioDecodingCallStats from common_types.h by Fredrik Solenberg · 6 years ago
  55. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  56. 179a392 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. by Piotr (Peter) Slatala · 6 years ago
  57. 1eebec9 Fix data race in channel_send.cc by Piotr (Peter) Slatala · 6 years ago
  58. 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
  59. 2681523 Tweak ChannelSend interface, to make it closer to ChannelSendProxy by Niels Möller · 6 years ago
  60. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  61. 8fb5746 Delete obsolete interface class RtpData by Niels Möller · 6 years ago
  62. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  63. 80c6762 Tweak ChannelReceive interface, to make it closer to ChannelReceiveProxy by Niels Möller · 6 years ago
  64. b768e88 Reland "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  65. 61c6e56 Revert "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  66. a7f77a7 Isolating APM API build target: making :api an actual target. by Alessio Bazzica · 6 years ago
  67. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  68. 967f7d5 Add audio level to CSRC class by Jonas Oreland · 6 years ago
  69. fd1a2fb Set RtpRtcp config receive_only in voe::ChannelReceive by Niels Möller · 6 years ago
  70. 273d029 Implement data channel methods in LoopbackMediaTransport. by Bjorn Mellem · 6 years ago
  71. 2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 6 years ago
  72. 56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 6 years ago
  73. 21cddff Harmonize paths to dependent targets. by Yves Gerey · 6 years ago
  74. fcc3981 Revert "Use only first payload timestamp for RTCP SR generation for audio" by Ilya Nikolaevskiy · 6 years ago
  75. 9a0662a Use only first payload timestamp for RTCP SR generation for audio by Ilya Nikolaevskiy · 6 years ago
  76. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  77. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  78. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
  79. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  80. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  81. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  82. 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
  83. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  84. 2dfa998 Reland "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
  85. c538fc7 Revert "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
  86. 5ccdc13 Prefix flag macros with WEBRTC_. by Mirko Bonadei · 6 years ago
  87. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  88. 40a7a35 Extract functionality of test_main into separate library. by Artem Titov · 6 years ago
  89. b686396 Makes AudioSendStream signal that it's part of allocation. by Sebastian Jansson · 6 years ago
  90. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
  91. 2e00abc Reland "[cleanup] Remove useless includes." by Yves Gerey · 6 years ago
  92. 64be7fa Move FecController to RtpVideoSender. by Stefan Holmer · 6 years ago
  93. 96a0f61 Revert "[cleanup] Remove useless includes." by Oleh Prypin · 6 years ago
  94. be8b534 [cleanup] Remove useless includes. by Yves Gerey · 6 years ago
  95. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
  96. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  97. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  98. 4a72ba9 Delete RtpReceiver and related code. by Niels Möller · 6 years ago
  99. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  100. 35fa280 Adds allocated rate without feedback to new congestion controller. by Sebastian Jansson · 6 years ago