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gerrit-public.fairphone.software / platform / external / webrtc / 5256d8bc4b67f4ec777a72d0c6cbeb0f182963c2 / audio
  1. f2c0818 Minor fixes to ChannelSend. by Mirko Bonadei · 5 years ago
  2. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  3. 5b82ba3 Adding VoIP specific channel adjustments by Per Åhgren · 5 years ago
  4. 662678d Adds injectable trials from peerconnection down to transport controller. by Erik Språng · 5 years ago
  5. 39bab5a Add missing assert.h for win no-test build by Jerome Humbert · 5 years ago
  6. c3d1f9b Enable injection of a custom NetEqFactory into PeerConnectionFactory. by Ivo Creusen · 5 years ago
  7. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
  8. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  9. 85a1000 Use deprecated SingleThreadedTaskQueueForTesting as regular task queue by Danil Chapovalov · 5 years ago
  10. 55d19e5 Add gustaf to audio/OWNERS by Gustaf Ullberg · 5 years ago
  11. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 5 years ago
  12. dabdde6 Avoid running NullAudioPoller without receiving streams by Gustaf Ullberg · 5 years ago
  13. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
  14. f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 5 years ago
  15. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  16. eb90e6f Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest by Danil Chapovalov · 5 years ago
  17. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  18. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  19. 4b64411 NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate by Karl Wiberg · 5 years ago
  20. cd0eedb Don't allocate audio if we have no transport sequence number. by Sebastian Jansson · 5 years ago
  21. 0a6510d Removes rtp_transport checks in AudioSendStream by Sebastian Jansson · 5 years ago
  22. 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 5 years ago
  23. ea55b08 Adds support for passing a vector of packets to the paced sender. by Erik Språng · 5 years ago
  24. 0429f78 Base overhead calculation for audio priority rate on available data. by Sebastian Jansson · 5 years ago
  25. f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 5 years ago
  26. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
  27. 44db436 Propagate task queue to create test::DirectTransport by TaskQueueBase interface by Danil Chapovalov · 5 years ago
  28. 01dd885 Moves contents of bitrate_controller to goog_cc by Sebastian Jansson · 5 years ago
  29. 40de3cc Propagating TargetRate struct to BitrateAllocator. by Sebastian Jansson · 5 years ago
  30. 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 5 years ago
  31. ee5ec9a Replacing local closure classes with C++14 moving capture lambdas. by Sebastian Jansson · 5 years ago
  32. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  33. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 5 years ago
  34. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
  35. 6516f76 Deprecate SingleThreadedTaskQueueForTesting class. by Yves Gerey · 5 years ago
  36. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago
  37. 65024d9 Remove clock drift metric from NetEq. by Jakob Ivarsson · 5 years ago
  38. b6220d9 Delete unused logic for audio RtcpMode::kOff by Niels Möller · 5 years ago
  39. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 5 years ago
  40. b4a6128 Delete unneeded dependencies on libjingle_peerconnection_api by Niels Möller · 5 years ago
  41. 6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 5 years ago
  42. fac7e31 Removes TransportSequenceNumberAllocator by Erik Språng · 5 years ago
  43. 4208a13 Removes deprecated InsertPacket/TimeToSendPacket/TimeToSendPadding by Erik Språng · 5 years ago
  44. d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 5 years ago
  45. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  46. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  47. 54d5d2c Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc by Erik Språng · 5 years ago
  48. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 5 years ago
  49. 58b496b Let StreamStatistician::GetReceiveStreamDataCounters return counters by value by Niels Möller · 5 years ago
  50. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  51. b168678 Add RTC_ prefix to non-standard format specifier macro "PRIdNS" by Oleh Prypin · 5 years ago
  52. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 5 years ago
  53. ed44f54 In ChannelReceive, use AcmReceiver directly, not AudioCodingModule by Niels Möller · 5 years ago
  54. fedd625 Change 2g network pc audio test to more realistic network by Artem Titov · 5 years ago
  55. 054e3bb Reland "Replace the implementation of `GetContributingSources()` on the audio side." by Chen Xing · 5 years ago
  56. da4f093 Reland "Only include payload in bytes sent/received." by Bjorn A Mellem · 5 years ago
  57. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
  58. bcd068d Revert "Only include payload in bytes sent/received." by Bjorn Mellem · 5 years ago
  59. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
  60. 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 5 years ago
  61. 8f319a3 Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" by Alessio Bazzica · 5 years ago
  62. fab3460 Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" by Alessio Bazzica · 5 years ago
  63. 9973933 Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." by Chen Xing · 5 years ago
  64. aa59eca Move RtpPacketSender and merge it with RtpPacketPacer. by Erik Språng · 5 years ago
  65. 74a1b4b Only include payload in bytes sent/received. by Bjorn A Mellem · 5 years ago
  66. cbc91ef Improve low bandwidth audio test instrumentatin, fix PC test by Artem Titov · 5 years ago
  67. 2ab97f6 Migrate WebRTC test infra to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  68. 0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 5 years ago
  69. 4c2c412 Set local ssrc at construction (audio) by Erik Språng · 5 years ago
  70. 24192c2 Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." by Ivo Creusen · 5 years ago
  71. e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 5 years ago
  72. 52e240e Use 16000Hz audio in PC test when specified by Artem Titov · 5 years ago
  73. b1f2d60 Reland "Fix collection of audio metrics from PC test framework for audio test" by Artem Titov · 5 years ago
  74. 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 5 years ago
  75. 4876cb2 Revert "Fix collection of audio metrics from PC test framework for audio test" by Mirko Bonadei · 5 years ago
  76. d0679bd Enables usage of ChannelMixer in WebRTC's output mixer. by henrika · 5 years ago
  77. 2d0880b Fix collection of audio metrics from PC test framework for audio test by Artem Titov · 5 years ago
  78. 4a126e4 Rename tests to prevent clashing with old audio test by Artem Titov · 5 years ago
  79. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  80. c8263e0 Introduce PC level audio quality test. by Artem Titov · 5 years ago
  81. 2250b05 Adding support for channel mixing between different channel layouts. by henrika · 5 years ago
  82. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  83. 67008df Revert "Replace the implementation of `GetContributingSources()` on the audio side." by Artem Titov · 5 years ago
  84. 8fa7151 Replace the implementation of `GetContributingSources()` on the audio side. by Chen Xing · 5 years ago
  85. 3e8ef94 Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. by Chen Xing · 5 years ago
  86. 225842c Initialize signal processing function pointers statically by Karl Wiberg · 5 years ago
  87. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
  88. f48bca7 Avoid triggering a false error logging when using encryptor and sending DTX. by Minyue Li · 5 years ago
  89. 59b8654 Switch from RtpPacketSender to RtpPacketPacer interface usage. by Erik Språng · 5 years ago
  90. 08fa953 Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" by Danil Chapovalov · 5 years ago
  91. fd5166c Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" by Philip Eliasson · 5 years ago
  92. fc96135 Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory by Danil Chapovalov · 5 years ago
  93. 9ab520e Reland "Avoid encrypting empty audio packet." by Minyue Li · 5 years ago
  94. 6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  95. 87da109 Make ReceiveStatisticsImpl::SetMaxReorderingThreshold apply per ssrc by Niels Möller · 5 years ago
  96. a352248 Add a config flag to disable the audio ALR probing request. by Christoffer Rodbro · 5 years ago
  97. b32f2c7 Publish rtc event log api and default factory for it in api/ by Danil Chapovalov · 5 years ago
  98. b5d9183 Add RTP timestamp to contributing sources by Johannes Kron · 5 years ago
  99. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  100. d703cd0 Revert "Avoid encrypting empty audio packet." by Minyue Li · 5 years ago