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webrtc / src / 9b87037073f1811bcdae30967167f30b364e6879 / audio
  1. 0abd518 Revert "Introduce RTC_NO_UNIQUE_ADDRESS." by Mirko Bonadei · 4 years, 6 months ago
  2. 16e7b51 Unit test around ProcessThread usage by Tim Na · 4 years, 6 months ago
  3. 09ceed2 Async audio processing API by Olga Sharonova · 4 years, 6 months ago
  4. f5e261a Introduce RTC_NO_UNIQUE_ADDRESS. by Mirko Bonadei · 4 years, 6 months ago
  5. de95329 Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS by Niels Möller · 4 years, 6 months ago
  6. 6154a74 Start/stop ProcessThread as first/last audio channel is added/removed. by Tim Na · 4 years, 6 months ago
  7. 77baeee Make MessageHandler be a pure virtual interface. by Tomas Gunnarsson · 4 years, 6 months ago
  8. 4461f05 Delete unused NetEq stats currentPacketLossRate, currentDiscardRate and addedSamples by Niels Möller · 4 years, 6 months ago
  9. 467b7c0 Removing logic forcing AEC within VoIP core by Tim Na · 4 years, 6 months ago
  10. 6b4d962 Fix standard GetStats to not modify NetEq state. by Niels Möller · 4 years, 6 months ago
  11. f264e70 Expand is_linux to is_linux || is_chromeos. by Hidehiko Abe · 4 years, 6 months ago
  12. bef7b05 Make AV sync robust to failures to set a desired minimum delay by Ivo Creusen · 4 years, 7 months ago
  13. 9e9c8b7 Delete obsolete method AudioReceiveStream::OnRtpPacket by Niels Möller · 4 years, 7 months ago
  14. abdb470 Make MessageHandler cleanup optional. by Tomas Gunnarsson · 4 years, 7 months ago
  15. a534729 DTMF Event Sub-API on VoIP API by Jason Long · 4 years, 7 months ago
  16. fde2b24 Reland "Call OnReceivedOverhead after audio network adaptor is created." by Jakob Ivarsson · 4 years, 7 months ago
  17. c8ac358 Revert "Call OnReceivedOverhead after audio network adaptor is created." by Erik Språng · 4 years, 7 months ago
  18. a135557 Call OnReceivedOverhead after audio network adaptor is created. by Jakob Ivarsson · 4 years, 7 months ago
  19. ed97116 Log audio network adaptor and DSCP in AudioSendStream. by Jakob Ivarsson · 4 years, 7 months ago
  20. dba1f94 Added Error Checking in Ingress/Egress and Extra Unit Tests by Jason Long · 4 years, 8 months ago
  21. 3d22108 Remove unused critical section includes. by Markus Handell · 4 years, 8 months ago
  22. a166a35 webrtc::AudioSendStream: Add lock annotation to audio_level_ by Sam Zackrisson · 4 years, 9 months ago
  23. 6287280 Migrate audio/ to use webrtc::Mutex by Markus Handell · 4 years, 9 months ago
  24. 1ff3c58 Add TimeController to the CreatePeerConnectionE2EQualityTestFixture API by Artem Titov · 4 years, 9 months ago
  25. 2b4d2f3 Removes locking in TransportFeedbackProxy. by Erik Språng · 4 years, 9 months ago
  26. edcd966 negotiate RED codec for audio by Philipp Hancke · 4 years, 9 months ago
  27. 08ce986 Switch to absl single target when building with Chromium. by Mirko Bonadei · 4 years, 10 months ago
  28. 2dcf348 Use absl_deps in order to preapre to the Abseil component build release. by Mirko Bonadei · 4 years, 10 months ago
  29. 506d4eb Add missing headers to fix chromium roll by Artem Titov · 4 years, 10 months ago
  30. f25761d Remove dependency from RtpRtcp on the Module interface. by Tomas Gunnarsson · 4 years, 10 months ago
  31. fae0562 Deprecate the static RtpRtcp::Create() method. by Tomas Gunnarsson · 4 years, 10 months ago
  32. 1a49756 fix typos in comments by Philipp Hancke · 4 years, 10 months ago
  33. 909f3a5 Rename several more tests that use EXPECT_DEATH to *DeathTest. by Tommi · 4 years, 10 months ago
  34. cf6544a Avoids unnecessary calls to audio encoder. by Erik Språng · 4 years, 10 months ago
  35. f9c6b68 In audio/ replace mock macros with unified MOCK_METHOD macro by Danil Chapovalov · 4 years, 10 months ago
  36. 04e1bab Replaces OverheadObserver with simple getter. by Erik Språng · 4 years, 11 months ago
  37. c0df5fc VoIP API implementation on top of AudioIngress/Egress by Tim Na · 4 years, 11 months ago
  38. d719708 Add unit tests for audio channel send frame transformer delegate. by Marina Ciocea · 4 years, 11 months ago
  39. 701ccf9 Add unit tests for audio receive channel frame transformer delegate. by Marina Ciocea · 4 years, 11 months ago
  40. 9abc6bd Reduce audiosendstream dependency on rttstats (dead code). by Tommi · 5 years ago
  41. cc73ed3 APM: Add build flag to allow building WebRTC without APM by Per Åhgren · 5 years ago
  42. f4ee036 [InsertableStreams] Clear callback to audio receive channel in delegate. by Marina Ciocea · 5 years ago
  43. 11f92bc Audio ingress implementation for voip api. by Tim Na · 5 years ago
  44. b9d4685 insertable streams: include rtp_timestamp offset for audio by Philipp Hancke · 5 years ago
  45. fc23cc0 [InsertableStreams] Don't include the header in the transformable payload. by Marina Ciocea · 5 years ago
  46. 4862320 Transform received audio frames in ChannelReceive. by Marina Ciocea · 5 years ago
  47. 3e9af7f Insert audio frame transformer between depacketizer and decoder. by Marina Ciocea · 5 years ago
  48. 65674d8 Transform encoded frames in ChannelSend. by Marina Ciocea · 5 years ago
  49. d2aa8f9 Insert audio frame transformer between encoder and packetizer. by Marina Ciocea · 5 years ago
  50. a7a0173 Remove the histogram flag and all Chart JSON code. by Patrik Höglund · 5 years ago
  51. a298fd5 Don't double import protobuf code. by Patrik Höglund · 5 years ago
  52. 8ab3c77 Audio egress implementation for initial voip api in api/voip. by Tim Na · 5 years ago
  53. a782299 Add PYTHONPATH entry for proto internal. by Patrik Höglund · 5 years ago
  54. e4ae1ce Reland: Add in missing protobuf code. by Patrik Höglund · 5 years ago
  55. 749dff1 Revert "Add in missing protobuf code." by Patrik Höglund · 5 years ago
  56. 0a663bf Add in missing protobuf code. by Patrik Höglund · 5 years ago
  57. 28b8a0b Partial revert of flag simplification. by Patrik Höglund · 5 years ago
  58. 7ac010c Fix low bw test. by Patrik Höglund · 5 years ago
  59. 3246466 Add debug info to low bw test. by Patrik Höglund · 5 years ago
  60. 3b4bbf5 Reland "Switch low bw audio test to histograms." by Patrik Höglund · 5 years ago
  61. ad502fa Revert "Switch low bw audio test to histograms." by Patrik Höglund · 5 years ago
  62. 71c9a18 Switch low bw audio test to histograms. by Patrik Höglund · 5 years ago
  63. 1b20c41 Greatly simplify flags for test binaries. by Patrik Höglund · 5 years ago
  64. 7eab0a8 Split RMS level measurement utility from APM by Sam Zackrisson · 5 years ago
  65. dc5522b APM: Removing the redundant VAD output from the integer API by Per Åhgren · 5 years ago
  66. 71652f4 APM: Localize/abstract the usage of AudioFrame by Per Åhgren · 5 years ago
  67. 0c96449 Clamp stable target bitrate to min/max allocated bitrate. by Jakob Ivarsson · 5 years ago
  68. b8e69ef Write protos as binary. by Patrik Höglund · 5 years ago
  69. e618cc9 Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API by Artem Titov · 5 years ago
  70. c771084 Remove RTC_NOTREACHED from audio_send_stream when ANA didn't work by Alejandro Luebs · 5 years ago
  71. 01ab084 Add minimum overhead to configured priorty bitrate instead of maximum. by Jakob Ivarsson · 5 years ago
  72. d14525e Make sure that the audio stream is allocated with the correct overhead. by Jakob Ivarsson · 5 years ago
  73. 74dadc1 Ready to support of absolute capture timestamp header extension. by Minyue Li · 5 years ago
  74. 8a5776a Only update the current time of a played out frame if a new frame is played out. by Åsa Persson · 5 years ago
  75. 9526c55 Refactoring mock_transport to be used separately by Tim Na · 5 years ago
  76. 4a14f49 Remove wildcard ownership for build files. by Mirko Bonadei · 5 years ago
  77. e52115a Remove inactive OWNERS. by Mirko Bonadei · 5 years ago
  78. cad3e0e Replace DataSize and DataRate factories with newer versions by Danil Chapovalov · 5 years ago
  79. b506fee Add AbsoluteCaptureTimeReceiver to audio ChannelReceive. by Minyue Li · 5 years ago
  80. 74d2b1d Add periodic logging of sync delays. by Åsa Persson · 5 years ago
  81. 0c626af Use newer version of TimeDelta and TimeStamp factories in webrtc by Danil Chapovalov · 5 years ago
  82. 2fe31a4 Remove ossu@ from audio/ and audio_coding/ OWNERS by Oskar Sundbom · 5 years ago
  83. fdbbada Revert "Inlines NullAudioPoller functionality into AudioState class." by Sebastian Jansson · 5 years ago
  84. 0e96535 Inlines NullAudioPoller functionality into AudioState class. by Sebastian Jansson · 5 years ago
  85. bef818d Default disables legacy overhead calculation. by Sebastian Jansson · 5 years ago
  86. c3eb9fd Reland "Reland "Only include overhead if using send side bandwidth estimation."" by Sebastian Jansson · 5 years ago
  87. 4356490 Revert "Reland "Only include overhead if using send side bandwidth estimation."" by Mirko Bonadei · 5 years ago
  88. 086055d Reland "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 5 years ago
  89. c709412 Revert "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 5 years ago
  90. 8c79c6e Only include overhead if using send side bandwidth estimation. by Sebastian Jansson · 5 years ago
  91. ff0e4db Reland "Send absolute capture time through audio coding module." by Minyue Li · 5 years ago
  92. 4175914 Revert "Send absolute capture time through audio coding module." by Minyue Li · 5 years ago
  93. 48655cf Send absolute capture time through audio coding module. by Minyue Li · 5 years ago
  94. ccbe95f Reformat GN files. by Mirko Bonadei · 5 years ago
  95. 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 5 years ago
  96. b2b2031 Concatenate string literals at compile time. by Jonas Olsson · 5 years ago
  97. b8c775a Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api by Tim Na · 5 years ago
  98. fae6400 Add saza@ and peah@ to OWNERS of some audio files by Sam Zackrisson · 5 years ago
  99. 4db28b5 Cleanup: Removes redundant includes on message_queue.h by Sebastian Jansson · 5 years ago
  100. 1b4e4bf Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing. by Danil Chapovalov · 5 years ago