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webrtc / src.git / 182748378496ebe0c2d7d0b46252fa8a50a08fe1 / audio
  1. bc1c93d Add remote-outbound stats for audio streams by Alessio Bazzica · 4 years, 1 month ago
  2. 5eda59c Replace legacy RtpRtcp::GetRemoteStat function with GetLatestReportBlockData by Danil Chapovalov · 4 years, 1 month ago
  3. 451a8af Feed the clock skew to AbsoluteCaptureTimeReceiver in audio receiver. by Minyue Li · 4 years, 1 month ago
  4. e904161 Replace RTC_DEPRECATED with ABSL_DEPRECATED by Danil Chapovalov · 4 years, 1 month ago
  5. dea374a Deliver packet info to source listeners when audio playout is disabled. by Ranveer Aggarwal · 4 years, 2 months ago
  6. d15a575 Use SequenceChecker from public API by Artem Titov · 4 years, 2 months ago
  7. 06159aa Remove deprecated thread checker by Artem Titov · 4 years, 2 months ago
  8. a208861 Reland "Fix data race for config_ in AudioSendStream" by Artem Titov · 4 years, 2 months ago
  9. ad32586 Reland "Prepare to avoid hops to worker for network events." by Tomas Gunnarsson · 4 years, 2 months ago
  10. 47ec157 Revert "Prepare to avoid hops to worker for network events." by Mirko Bonadei · 4 years, 2 months ago
  11. 76a1041 Revert "Fix data race for config_ in AudioSendStream" by Henrik Boström · 4 years, 2 months ago
  12. d48a2b1 Prepare to avoid hops to worker for network events. by Tomas Gunnarsson · 4 years, 2 months ago
  13. c8421c4 Replace rtc::ThreadChecker with webrtc::SequenceChecker by Artem Titov · 4 years, 2 months ago
  14. 51e5c4b Fix data race for config_ in AudioSendStream by Artem Titov · 4 years, 2 months ago
  15. e7c79fd Remove from chromium build targets that are not compatible with it. by Andrey Logvin · 4 years, 2 months ago
  16. 7864600 Add absl_deps field for rtc_test and rtc_executable by Andrey Logvin · 4 years, 2 months ago
  17. 49dbad0 Fixing audio timestamp stall during inactivation (under a kill switch) by Minyue Li · 4 years, 2 months ago
  18. 1cbf21e ChannelStatistics RTT test case around remote SSRC change. by Tim Na · 4 years, 2 months ago
  19. 8467cf2 Reduce redundant flags for audio stream playout state. by Tomas Gunnarsson · 4 years, 2 months ago
  20. e5f4c6b Reland "Refactor rtc_base build targets." by Mirko Bonadei · 4 years, 2 months ago
  21. 7acc2d9 Revert "Refactor rtc_base build targets." by Mirko Bonadei · 4 years, 3 months ago
  22. 507eacf Reland "ChannelStatistics used for RTP stats in VoipStatistics." by Tim Na · 4 years, 3 months ago
  23. 37827c9 Revert "ChannelStatistics used for RTP stats in VoipStatistics." by Alex Loiko · 4 years, 3 months ago
  24. 444e04b ChannelStatistics used for RTP stats in VoipStatistics. by Tim Na · 4 years, 3 months ago
  25. 2accc7d Revert "Add task queue to RtpRtcpInterface::Configuration." by Niels Moller · 4 years, 3 months ago
  26. f23e214 Add task queue to RtpRtcpInterface::Configuration. by Niels Möller · 4 years, 3 months ago
  27. 69241a9 Refactor rtc_base build targets. by Mirko Bonadei · 4 years, 3 months ago
  28. 2412602 Using absl::optional for round trip time return type handling. by Tim Na · 4 years, 4 months ago
  29. 9325d34 Enforcing return type handling on VoIP API. by Tim Na · 4 years, 4 months ago
  30. c20baf6 Remove nesting of Naggy/Strict/NiceMock by Alex Konradi · 4 years, 4 months ago
  31. 0d863f7 Cleanup of bwe_defines.h by Niels Möller · 4 years, 4 months ago
  32. 47a03e8 Default enable sending transport sequence numbers on audio packets. by Jakob Ivarsson · 4 years, 4 months ago
  33. 20e4c80 Reland "Introduce RTC_NO_UNIQUE_ADDRESS." by Mirko Bonadei · 4 years, 4 months ago
  34. b223cb6 Defining API result types on VoIP API by Tim Na · 4 years, 4 months ago
  35. 01719fb Reland "Rename FATAL() into RTC_FATAL()." by Mirko Bonadei · 4 years, 4 months ago
  36. a4fd641 Revert "Rename FATAL() into RTC_FATAL()." by Mirko Bonadei · 4 years, 4 months ago
  37. 26ce03e Locating input audio level before TaskQueue. by Tim Na · 4 years, 4 months ago
  38. 9653d26 Rename FATAL() into RTC_FATAL(). by Mirko Bonadei · 4 years, 4 months ago
  39. a58cae3 VoipVolumeControl subAPI for VoIP API by Tim Na · 4 years, 5 months ago
  40. 254ad1b Delay VoipCore initialization. by Tim Na · 4 years, 5 months ago
  41. 428432d Name change on channel and channel_id for consistency. by Tim Na · 4 years, 5 months ago
  42. 4552e8f Enable continuous audio polling from ADM after StopPlay in VoIP API by Tim Na · 4 years, 5 months ago
  43. 42cafa5 Delete legacy stats minWaitingTimeMs and medianWaitingTimeMs from ACM. by Niels Möller · 4 years, 5 months ago
  44. cd4203b Adding total duration and more test cases to VoipStatistics. by Tim Na · 4 years, 5 months ago
  45. 36274f9 Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead."" by Jakob Ivarsson · 4 years, 5 months ago
  46. 8cc6695 Reformat python files checked by pylint (part 1/2). by Mirko Bonadei · 4 years, 5 months ago
  47. f4347f7 VoipStatistics subAPI and implementation. by Tim Na · 4 years, 5 months ago
  48. d546186 Revert "Reland "Default enable WebRTC-SendSideBwe-WithOverhead."" by Björn Terelius · 4 years, 5 months ago
  49. 1dbe30c Reland "Default enable WebRTC-SendSideBwe-WithOverhead." by Jakob Ivarsson · 4 years, 5 months ago
  50. 27af3c4 Revert "Default enable WebRTC-SendSideBwe-WithOverhead." by Jakob Ivarsson · 4 years, 5 months ago
  51. 87c1950 Default enable WebRTC-SendSideBwe-WithOverhead. by Jakob Ivarsson · 4 years, 5 months ago
  52. 0abd518 Revert "Introduce RTC_NO_UNIQUE_ADDRESS." by Mirko Bonadei · 4 years, 6 months ago
  53. 16e7b51 Unit test around ProcessThread usage by Tim Na · 4 years, 6 months ago
  54. 09ceed2 Async audio processing API by Olga Sharonova · 4 years, 6 months ago
  55. f5e261a Introduce RTC_NO_UNIQUE_ADDRESS. by Mirko Bonadei · 4 years, 6 months ago
  56. de95329 Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS by Niels Möller · 4 years, 6 months ago
  57. 6154a74 Start/stop ProcessThread as first/last audio channel is added/removed. by Tim Na · 4 years, 6 months ago
  58. 77baeee Make MessageHandler be a pure virtual interface. by Tomas Gunnarsson · 4 years, 6 months ago
  59. 4461f05 Delete unused NetEq stats currentPacketLossRate, currentDiscardRate and addedSamples by Niels Möller · 4 years, 6 months ago
  60. 467b7c0 Removing logic forcing AEC within VoIP core by Tim Na · 4 years, 6 months ago
  61. 6b4d962 Fix standard GetStats to not modify NetEq state. by Niels Möller · 4 years, 7 months ago
  62. f264e70 Expand is_linux to is_linux || is_chromeos. by Hidehiko Abe · 4 years, 7 months ago
  63. bef7b05 Make AV sync robust to failures to set a desired minimum delay by Ivo Creusen · 4 years, 7 months ago
  64. 9e9c8b7 Delete obsolete method AudioReceiveStream::OnRtpPacket by Niels Möller · 4 years, 7 months ago
  65. abdb470 Make MessageHandler cleanup optional. by Tomas Gunnarsson · 4 years, 7 months ago
  66. a534729 DTMF Event Sub-API on VoIP API by Jason Long · 4 years, 7 months ago
  67. fde2b24 Reland "Call OnReceivedOverhead after audio network adaptor is created." by Jakob Ivarsson · 4 years, 7 months ago
  68. c8ac358 Revert "Call OnReceivedOverhead after audio network adaptor is created." by Erik Språng · 4 years, 8 months ago
  69. a135557 Call OnReceivedOverhead after audio network adaptor is created. by Jakob Ivarsson · 4 years, 8 months ago
  70. ed97116 Log audio network adaptor and DSCP in AudioSendStream. by Jakob Ivarsson · 4 years, 8 months ago
  71. dba1f94 Added Error Checking in Ingress/Egress and Extra Unit Tests by Jason Long · 4 years, 8 months ago
  72. 3d22108 Remove unused critical section includes. by Markus Handell · 4 years, 9 months ago
  73. a166a35 webrtc::AudioSendStream: Add lock annotation to audio_level_ by Sam Zackrisson · 4 years, 9 months ago
  74. 6287280 Migrate audio/ to use webrtc::Mutex by Markus Handell · 4 years, 9 months ago
  75. 1ff3c58 Add TimeController to the CreatePeerConnectionE2EQualityTestFixture API by Artem Titov · 4 years, 9 months ago
  76. 2b4d2f3 Removes locking in TransportFeedbackProxy. by Erik Språng · 4 years, 9 months ago
  77. edcd966 negotiate RED codec for audio by Philipp Hancke · 4 years, 9 months ago
  78. 08ce986 Switch to absl single target when building with Chromium. by Mirko Bonadei · 4 years, 10 months ago
  79. 2dcf348 Use absl_deps in order to preapre to the Abseil component build release. by Mirko Bonadei · 4 years, 10 months ago
  80. 506d4eb Add missing headers to fix chromium roll by Artem Titov · 4 years, 10 months ago
  81. f25761d Remove dependency from RtpRtcp on the Module interface. by Tomas Gunnarsson · 4 years, 10 months ago
  82. fae0562 Deprecate the static RtpRtcp::Create() method. by Tomas Gunnarsson · 4 years, 10 months ago
  83. 1a49756 fix typos in comments by Philipp Hancke · 4 years, 10 months ago
  84. 909f3a5 Rename several more tests that use EXPECT_DEATH to *DeathTest. by Tommi · 4 years, 11 months ago
  85. cf6544a Avoids unnecessary calls to audio encoder. by Erik Språng · 4 years, 11 months ago
  86. f9c6b68 In audio/ replace mock macros with unified MOCK_METHOD macro by Danil Chapovalov · 4 years, 11 months ago
  87. 04e1bab Replaces OverheadObserver with simple getter. by Erik Språng · 5 years ago
  88. c0df5fc VoIP API implementation on top of AudioIngress/Egress by Tim Na · 5 years ago
  89. d719708 Add unit tests for audio channel send frame transformer delegate. by Marina Ciocea · 5 years ago
  90. 701ccf9 Add unit tests for audio receive channel frame transformer delegate. by Marina Ciocea · 5 years ago
  91. 9abc6bd Reduce audiosendstream dependency on rttstats (dead code). by Tommi · 5 years ago
  92. cc73ed3 APM: Add build flag to allow building WebRTC without APM by Per Åhgren · 5 years ago
  93. f4ee036 [InsertableStreams] Clear callback to audio receive channel in delegate. by Marina Ciocea · 5 years ago
  94. 11f92bc Audio ingress implementation for voip api. by Tim Na · 5 years ago
  95. b9d4685 insertable streams: include rtp_timestamp offset for audio by Philipp Hancke · 5 years ago
  96. fc23cc0 [InsertableStreams] Don't include the header in the transformable payload. by Marina Ciocea · 5 years ago
  97. 4862320 Transform received audio frames in ChannelReceive. by Marina Ciocea · 5 years ago
  98. 3e9af7f Insert audio frame transformer between depacketizer and decoder. by Marina Ciocea · 5 years ago
  99. 65674d8 Transform encoded frames in ChannelSend. by Marina Ciocea · 5 years ago
  100. d2aa8f9 Insert audio frame transformer between encoder and packetizer. by Marina Ciocea · 5 years ago