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gerrit-public.fairphone.software / platform / external / webrtc / b0f2e0ced4cd7ff23a54d0c494a9264cd62e0b1a / audio
  1. 74dadc1 Ready to support of absolute capture timestamp header extension. by Minyue Li · 4 years, 5 months ago
  2. 8a5776a Only update the current time of a played out frame if a new frame is played out. by Åsa Persson · 4 years, 5 months ago
  3. 9526c55 Refactoring mock_transport to be used separately by Tim Na · 4 years, 5 months ago
  4. 4a14f49 Remove wildcard ownership for build files. by Mirko Bonadei · 4 years, 5 months ago
  5. e52115a Remove inactive OWNERS. by Mirko Bonadei · 4 years, 5 months ago
  6. cad3e0e Replace DataSize and DataRate factories with newer versions by Danil Chapovalov · 4 years, 5 months ago
  7. b506fee Add AbsoluteCaptureTimeReceiver to audio ChannelReceive. by Minyue Li · 4 years, 6 months ago
  8. 74d2b1d Add periodic logging of sync delays. by Åsa Persson · 4 years, 6 months ago
  9. 0c626af Use newer version of TimeDelta and TimeStamp factories in webrtc by Danil Chapovalov · 4 years, 6 months ago
  10. 2fe31a4 Remove ossu@ from audio/ and audio_coding/ OWNERS by Oskar Sundbom · 4 years, 6 months ago
  11. fdbbada Revert "Inlines NullAudioPoller functionality into AudioState class." by Sebastian Jansson · 4 years, 6 months ago
  12. 0e96535 Inlines NullAudioPoller functionality into AudioState class. by Sebastian Jansson · 4 years, 6 months ago
  13. bef818d Default disables legacy overhead calculation. by Sebastian Jansson · 4 years, 6 months ago
  14. c3eb9fd Reland "Reland "Only include overhead if using send side bandwidth estimation."" by Sebastian Jansson · 4 years, 6 months ago
  15. 4356490 Revert "Reland "Only include overhead if using send side bandwidth estimation."" by Mirko Bonadei · 4 years, 6 months ago
  16. 086055d Reland "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 4 years, 6 months ago
  17. c709412 Revert "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 4 years, 6 months ago
  18. 8c79c6e Only include overhead if using send side bandwidth estimation. by Sebastian Jansson · 4 years, 6 months ago
  19. ff0e4db Reland "Send absolute capture time through audio coding module." by Minyue Li · 4 years, 6 months ago
  20. 4175914 Revert "Send absolute capture time through audio coding module." by Minyue Li · 4 years, 6 months ago
  21. 48655cf Send absolute capture time through audio coding module. by Minyue Li · 4 years, 6 months ago
  22. ccbe95f Reformat GN files. by Mirko Bonadei · 4 years, 6 months ago
  23. 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 4 years, 6 months ago
  24. b2b2031 Concatenate string literals at compile time. by Jonas Olsson · 4 years, 6 months ago
  25. b8c775a Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api by Tim Na · 4 years, 7 months ago
  26. fae6400 Add saza@ and peah@ to OWNERS of some audio files by Sam Zackrisson · 4 years, 7 months ago
  27. 4db28b5 Cleanup: Removes redundant includes on message_queue.h by Sebastian Jansson · 4 years, 7 months ago
  28. 1b4e4bf Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing. by Danil Chapovalov · 4 years, 8 months ago
  29. f2c0818 Minor fixes to ChannelSend. by Mirko Bonadei · 4 years, 8 months ago
  30. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 4 years, 8 months ago
  31. 5b82ba3 Adding VoIP specific channel adjustments by Per Åhgren · 4 years, 8 months ago
  32. 662678d Adds injectable trials from peerconnection down to transport controller. by Erik Språng · 4 years, 8 months ago
  33. 39bab5a Add missing assert.h for win no-test build by Jerome Humbert · 4 years, 9 months ago
  34. c3d1f9b Enable injection of a custom NetEqFactory into PeerConnectionFactory. by Ivo Creusen · 4 years, 9 months ago
  35. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 4 years, 9 months ago
  36. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 9 months ago
  37. 85a1000 Use deprecated SingleThreadedTaskQueueForTesting as regular task queue by Danil Chapovalov · 4 years, 9 months ago
  38. 55d19e5 Add gustaf to audio/OWNERS by Gustaf Ullberg · 4 years, 9 months ago
  39. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 4 years, 9 months ago
  40. dabdde6 Avoid running NullAudioPoller without receiving streams by Gustaf Ullberg · 4 years, 9 months ago
  41. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 4 years, 10 months ago
  42. f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 4 years, 10 months ago
  43. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 4 years, 10 months ago
  44. eb90e6f Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest by Danil Chapovalov · 4 years, 10 months ago
  45. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 10 months ago
  46. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 4 years, 10 months ago
  47. 4b64411 NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate by Karl Wiberg · 4 years, 10 months ago
  48. cd0eedb Don't allocate audio if we have no transport sequence number. by Sebastian Jansson · 4 years, 10 months ago
  49. 0a6510d Removes rtp_transport checks in AudioSendStream by Sebastian Jansson · 4 years, 10 months ago
  50. 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 4 years, 10 months ago
  51. ea55b08 Adds support for passing a vector of packets to the paced sender. by Erik Språng · 4 years, 10 months ago
  52. 0429f78 Base overhead calculation for audio priority rate on available data. by Sebastian Jansson · 4 years, 10 months ago
  53. f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 4 years, 10 months ago
  54. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 4 years, 10 months ago
  55. 44db436 Propagate task queue to create test::DirectTransport by TaskQueueBase interface by Danil Chapovalov · 4 years, 10 months ago
  56. 01dd885 Moves contents of bitrate_controller to goog_cc by Sebastian Jansson · 4 years, 10 months ago
  57. 40de3cc Propagating TargetRate struct to BitrateAllocator. by Sebastian Jansson · 4 years, 10 months ago
  58. 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 4 years, 10 months ago
  59. ee5ec9a Replacing local closure classes with C++14 moving capture lambdas. by Sebastian Jansson · 4 years, 10 months ago
  60. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 4 years, 10 months ago
  61. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 4 years, 11 months ago
  62. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 4 years, 11 months ago
  63. 6516f76 Deprecate SingleThreadedTaskQueueForTesting class. by Yves Gerey · 5 years ago
  64. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago
  65. 65024d9 Remove clock drift metric from NetEq. by Jakob Ivarsson · 5 years ago
  66. b6220d9 Delete unused logic for audio RtcpMode::kOff by Niels Möller · 5 years ago
  67. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 5 years ago
  68. b4a6128 Delete unneeded dependencies on libjingle_peerconnection_api by Niels Möller · 5 years ago
  69. 6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 5 years ago
  70. fac7e31 Removes TransportSequenceNumberAllocator by Erik Språng · 5 years ago
  71. 4208a13 Removes deprecated InsertPacket/TimeToSendPacket/TimeToSendPadding by Erik Språng · 5 years ago
  72. d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 5 years ago
  73. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  74. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  75. 54d5d2c Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc by Erik Språng · 5 years ago
  76. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 5 years ago
  77. 58b496b Let StreamStatistician::GetReceiveStreamDataCounters return counters by value by Niels Möller · 5 years ago
  78. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  79. b168678 Add RTC_ prefix to non-standard format specifier macro "PRIdNS" by Oleh Prypin · 5 years ago
  80. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 5 years ago
  81. ed44f54 In ChannelReceive, use AcmReceiver directly, not AudioCodingModule by Niels Möller · 5 years ago
  82. fedd625 Change 2g network pc audio test to more realistic network by Artem Titov · 5 years ago
  83. 054e3bb Reland "Replace the implementation of `GetContributingSources()` on the audio side." by Chen Xing · 5 years ago
  84. da4f093 Reland "Only include payload in bytes sent/received." by Bjorn A Mellem · 5 years ago
  85. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
  86. bcd068d Revert "Only include payload in bytes sent/received." by Bjorn Mellem · 5 years ago
  87. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
  88. 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 5 years ago
  89. 8f319a3 Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" by Alessio Bazzica · 5 years ago
  90. fab3460 Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" by Alessio Bazzica · 5 years ago
  91. 9973933 Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." by Chen Xing · 5 years ago
  92. aa59eca Move RtpPacketSender and merge it with RtpPacketPacer. by Erik Språng · 5 years ago
  93. 74a1b4b Only include payload in bytes sent/received. by Bjorn A Mellem · 5 years ago
  94. cbc91ef Improve low bandwidth audio test instrumentatin, fix PC test by Artem Titov · 5 years ago
  95. 2ab97f6 Migrate WebRTC test infra to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  96. 0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 5 years ago
  97. 4c2c412 Set local ssrc at construction (audio) by Erik Språng · 5 years ago
  98. 24192c2 Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." by Ivo Creusen · 5 years ago
  99. e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 5 years ago
  100. 52e240e Use 16000Hz audio in PC test when specified by Artem Titov · 5 years ago