Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / c01367db40c8c0ed266dad70771ba2b8a5b63c93 / audio
  1. c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
  2. 31660fd Avoid using global task queue factory in audio/ unittests by Danil Chapovalov · 6 years ago
  3. 741daaf Move rtc::FunctionView to the public API by Artem Titov · 6 years ago
  4. 94b57c0 Cleanup BUILD.gn files from imports like foo:foo by Artem Titov · 6 years ago
  5. 53de725 Fix outdated android sdk path in tests. by Oleksandr Iakovenko · 6 years ago
  6. ef1052a Reland "Move api/rtp_headers.h to its own build target." by Niels Möller · 6 years ago
  7. 2baef35 Revert "Move api/rtp_headers.h to its own build target." by Steve Anton · 6 years ago
  8. a67050d Move api/rtp_headers.h to its own build target. by Niels Möller · 6 years ago
  9. c936cb6 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h by Niels Möller · 6 years ago
  10. f0b8dee Qualify cmath functions. by Mirko Bonadei · 6 years ago
  11. 17b050f Fixes ClangTidy errors in audio/ by Benjamin Wright · 6 years ago
  12. 471783f Remove rtc::QueuedTask alias, use webrtc::QueuedTask directly by Danil Chapovalov · 6 years ago
  13. 9ffb5df Removes unused mock_bitrate_controller. by Sebastian Jansson · 6 years ago
  14. ad89528 Reland "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" by Danil Chapovalov · 6 years ago
  15. 42d8c93 Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" by Yves Gerey · 6 years ago
  16. 44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 6 years ago
  17. 304e9d2 Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current by Danil Chapovalov · 6 years ago
  18. d5af402 Add overhead observers to MediaTransportInterface by Niels Möller · 6 years ago
  19. 87e2d78 Prepare for splitting FrameType into AudioFrameType and VideoFrameType by Niels Möller · 6 years ago
  20. 0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 6 years ago
  21. 8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 6 years ago
  22. ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 6 years ago
  23. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 6 years ago
  24. c44f6cc Modernize RtpRtcp factory function: use unique_ptr as return type by Danil Chapovalov · 6 years ago
  25. 110c64b Delete unused key WebRTC-Audio-SendSideBwe-For-Video. by Christoffer Rodbro · 6 years ago
  26. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  27. 7949f21 Revert "Removes lock from ChannelSend." by Sebastian Jansson · 6 years ago
  28. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  29. 9b93447 Removes lock from ChannelSend. by Sebastian Jansson · 6 years ago
  30. da6806c Injecting Clock into BitrateAllocator. by Sebastian Jansson · 6 years ago
  31. fc52b91 Implicitly suppress //build/config/clang:find_bad_constructs. by Mirko Bonadei · 6 years ago
  32. 3cdd4d5 Fix: Ignore empty frames in Media Transport by Piotr (Peter) Slatala · 6 years ago
  33. d8d3248 Reland "Delete test/constants.h" by Elad Alon · 6 years ago
  34. 4f36b7a Revert "Delete test/constants.h" by Oleh Prypin · 6 years ago
  35. e049eba Revert "Add Sender and Receiver interfaces for MediaTransport audio" by Sergey Silkin · 6 years ago
  36. 0d8eed6 Add Sender and Receiver interfaces for MediaTransport audio by Niels Möller · 6 years ago
  37. afb5dbb Update ACM to use RTPHeader instead of WebRtcRTPHeader by Niels Möller · 6 years ago
  38. 389b167 Delete test/constants.h by Elad Alon · 6 years ago
  39. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
  40. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
  41. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
  42. 14a7cf9 Adds CallEncoder to ChannelSend. by Sebastian Jansson · 6 years ago
  43. 464a557 Adds audio priority bitrate field trial parameter. by Sebastian Jansson · 6 years ago
  44. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
  45. 1c54605 [clang-tidy] Apply performance-move-const-arg fixes (misc). by Mirko Bonadei · 6 years ago
  46. 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
  47. 80a8687 [clang-tidy] Apply performance-move-const-arg fixes (mutable lambdas). by Mirko Bonadei · 6 years ago
  48. 432c833 Remove redundant check in channel_receive.cc. by Ruslan Burakov · 6 years ago
  49. 01dc691 Delete sequence number save and restore in ChannelSend. by Niels Möller · 6 years ago
  50. c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 6 years ago
  51. fe055c1 [clang-tidy] Apply modernize-use-override fixes. by Mirko Bonadei · 6 years ago
  52. b4977de Receive-side ready for multiple channels. by Alex Loiko · 6 years ago
  53. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  54. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
  55. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  56. 5c2f1f0 Add some missing includes and dependencies. by Bjorn Terelius · 6 years ago
  57. e7d08df Fix chromium roll into WebRTC. by Artem Titov · 6 years ago
  58. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  59. 36faf0b Delete setting of unused variable nack_window_ms by Niels Möller · 6 years ago
  60. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  61. ba50223 Make voiceengine/audio transport stop using voice_detection() interface by Sam Zackrisson · 6 years ago
  62. 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
  63. ac63ac7 Update refcounting of AudioState to use rtc::RefCountedObject by Niels Möller · 6 years ago
  64. 40d5533 Include absl/memory/memory.h if absl::make_unique is used by Steve Anton · 6 years ago
  65. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  66. 31d8b52 Delete unneeded includes of rtc_base/stringutils.h. by Niels Möller · 6 years ago
  67. 3d2ed19 Remove Transport implementation from ChannelSend by Fredrik Solenberg · 6 years ago
  68. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  69. 2a977cf For audio receive channel use default max reordering threshold instead of 0 by Danil Chapovalov · 6 years ago
  70. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  71. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  72. e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
  73. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  74. 8ce0d2b In ReceiveStatistic require callbacks during construction by Danil Chapovalov · 6 years ago
  75. e3abb81 Decouple //rtc_base:rtc_base_tests_utils from gunit. by Mirko Bonadei · 6 years ago
  76. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  77. 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
  78. 8b5d9d8 Remove the audio/video split for the RTCP report intervals. by Jiawei Ou · 6 years ago
  79. 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
  80. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  81. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  82. 985a1f3 Add const or GUARDED_BY on a few ChannelSend members by Niels Möller · 6 years ago
  83. 26e88b0 Replace RTC_DCHECK by RTC_DCHECK_RUN_ON for worker thread. by Niels Möller · 6 years ago
  84. eb13484 Remove ChannelSendState by Fredrik Solenberg · 6 years ago
  85. c5e8be3 Remove ChannelReceiveState by Fredrik Solenberg · 6 years ago
  86. 78e88fe Move NetworkStatistics and AudioDecodingCallStats from common_types.h by Fredrik Solenberg · 6 years ago
  87. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  88. 179a392 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. by Piotr (Peter) Slatala · 6 years ago
  89. 1eebec9 Fix data race in channel_send.cc by Piotr (Peter) Slatala · 6 years ago
  90. 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
  91. 2681523 Tweak ChannelSend interface, to make it closer to ChannelSendProxy by Niels Möller · 6 years ago
  92. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  93. 8fb5746 Delete obsolete interface class RtpData by Niels Möller · 6 years ago
  94. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  95. 80c6762 Tweak ChannelReceive interface, to make it closer to ChannelReceiveProxy by Niels Möller · 6 years ago
  96. b768e88 Reland "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  97. 61c6e56 Revert "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  98. a7f77a7 Isolating APM API build target: making :api an actual target. by Alessio Bazzica · 6 years ago
  99. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  100. 967f7d5 Add audio level to CSRC class by Jonas Oreland · 6 years ago