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gerrit-public.fairphone.software / platform / external / webrtc / d4262dffa01f2fb71200d9d2057463d870f5b14f / audio
  1. 71c9a18 Switch low bw audio test to histograms. by Patrik Höglund · 5 years ago
  2. 1b20c41 Greatly simplify flags for test binaries. by Patrik Höglund · 5 years ago
  3. 7eab0a8 Split RMS level measurement utility from APM by Sam Zackrisson · 5 years ago
  4. dc5522b APM: Removing the redundant VAD output from the integer API by Per Åhgren · 5 years ago
  5. 71652f4 APM: Localize/abstract the usage of AudioFrame by Per Åhgren · 5 years ago
  6. 0c96449 Clamp stable target bitrate to min/max allocated bitrate. by Jakob Ivarsson · 5 years ago
  7. b8e69ef Write protos as binary. by Patrik Höglund · 5 years ago
  8. e618cc9 Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API by Artem Titov · 5 years ago
  9. c771084 Remove RTC_NOTREACHED from audio_send_stream when ANA didn't work by Alejandro Luebs · 5 years ago
  10. 01ab084 Add minimum overhead to configured priorty bitrate instead of maximum. by Jakob Ivarsson · 5 years ago
  11. d14525e Make sure that the audio stream is allocated with the correct overhead. by Jakob Ivarsson · 5 years ago
  12. 74dadc1 Ready to support of absolute capture timestamp header extension. by Minyue Li · 5 years ago
  13. 8a5776a Only update the current time of a played out frame if a new frame is played out. by Åsa Persson · 5 years ago
  14. 9526c55 Refactoring mock_transport to be used separately by Tim Na · 5 years ago
  15. 4a14f49 Remove wildcard ownership for build files. by Mirko Bonadei · 5 years ago
  16. e52115a Remove inactive OWNERS. by Mirko Bonadei · 5 years ago
  17. cad3e0e Replace DataSize and DataRate factories with newer versions by Danil Chapovalov · 5 years ago
  18. b506fee Add AbsoluteCaptureTimeReceiver to audio ChannelReceive. by Minyue Li · 5 years ago
  19. 74d2b1d Add periodic logging of sync delays. by Åsa Persson · 5 years ago
  20. 0c626af Use newer version of TimeDelta and TimeStamp factories in webrtc by Danil Chapovalov · 5 years ago
  21. 2fe31a4 Remove ossu@ from audio/ and audio_coding/ OWNERS by Oskar Sundbom · 5 years ago
  22. fdbbada Revert "Inlines NullAudioPoller functionality into AudioState class." by Sebastian Jansson · 5 years ago
  23. 0e96535 Inlines NullAudioPoller functionality into AudioState class. by Sebastian Jansson · 5 years ago
  24. bef818d Default disables legacy overhead calculation. by Sebastian Jansson · 5 years ago
  25. c3eb9fd Reland "Reland "Only include overhead if using send side bandwidth estimation."" by Sebastian Jansson · 5 years ago
  26. 4356490 Revert "Reland "Only include overhead if using send side bandwidth estimation."" by Mirko Bonadei · 5 years ago
  27. 086055d Reland "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 5 years ago
  28. c709412 Revert "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 5 years ago
  29. 8c79c6e Only include overhead if using send side bandwidth estimation. by Sebastian Jansson · 5 years ago
  30. ff0e4db Reland "Send absolute capture time through audio coding module." by Minyue Li · 5 years ago
  31. 4175914 Revert "Send absolute capture time through audio coding module." by Minyue Li · 5 years ago
  32. 48655cf Send absolute capture time through audio coding module. by Minyue Li · 5 years ago
  33. ccbe95f Reformat GN files. by Mirko Bonadei · 5 years ago
  34. 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 5 years ago
  35. b2b2031 Concatenate string literals at compile time. by Jonas Olsson · 5 years ago
  36. b8c775a Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api by Tim Na · 5 years ago
  37. fae6400 Add saza@ and peah@ to OWNERS of some audio files by Sam Zackrisson · 5 years ago
  38. 4db28b5 Cleanup: Removes redundant includes on message_queue.h by Sebastian Jansson · 5 years ago
  39. 1b4e4bf Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing. by Danil Chapovalov · 5 years ago
  40. f2c0818 Minor fixes to ChannelSend. by Mirko Bonadei · 5 years ago
  41. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  42. 5b82ba3 Adding VoIP specific channel adjustments by Per Åhgren · 5 years ago
  43. 662678d Adds injectable trials from peerconnection down to transport controller. by Erik Språng · 5 years ago
  44. 39bab5a Add missing assert.h for win no-test build by Jerome Humbert · 5 years ago
  45. c3d1f9b Enable injection of a custom NetEqFactory into PeerConnectionFactory. by Ivo Creusen · 5 years ago
  46. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
  47. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  48. 85a1000 Use deprecated SingleThreadedTaskQueueForTesting as regular task queue by Danil Chapovalov · 5 years ago
  49. 55d19e5 Add gustaf to audio/OWNERS by Gustaf Ullberg · 5 years ago
  50. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 5 years ago
  51. dabdde6 Avoid running NullAudioPoller without receiving streams by Gustaf Ullberg · 5 years ago
  52. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
  53. f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 5 years ago
  54. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  55. eb90e6f Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest by Danil Chapovalov · 5 years ago
  56. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  57. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  58. 4b64411 NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate by Karl Wiberg · 5 years ago
  59. cd0eedb Don't allocate audio if we have no transport sequence number. by Sebastian Jansson · 5 years ago
  60. 0a6510d Removes rtp_transport checks in AudioSendStream by Sebastian Jansson · 5 years ago
  61. 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 5 years ago
  62. ea55b08 Adds support for passing a vector of packets to the paced sender. by Erik Språng · 5 years ago
  63. 0429f78 Base overhead calculation for audio priority rate on available data. by Sebastian Jansson · 5 years ago
  64. f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 5 years ago
  65. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
  66. 44db436 Propagate task queue to create test::DirectTransport by TaskQueueBase interface by Danil Chapovalov · 5 years ago
  67. 01dd885 Moves contents of bitrate_controller to goog_cc by Sebastian Jansson · 5 years ago
  68. 40de3cc Propagating TargetRate struct to BitrateAllocator. by Sebastian Jansson · 5 years ago
  69. 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 5 years ago
  70. ee5ec9a Replacing local closure classes with C++14 moving capture lambdas. by Sebastian Jansson · 5 years ago
  71. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  72. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 5 years ago
  73. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
  74. 6516f76 Deprecate SingleThreadedTaskQueueForTesting class. by Yves Gerey · 5 years ago
  75. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago
  76. 65024d9 Remove clock drift metric from NetEq. by Jakob Ivarsson · 5 years ago
  77. b6220d9 Delete unused logic for audio RtcpMode::kOff by Niels Möller · 5 years ago
  78. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 5 years ago
  79. b4a6128 Delete unneeded dependencies on libjingle_peerconnection_api by Niels Möller · 5 years ago
  80. 6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 5 years ago
  81. fac7e31 Removes TransportSequenceNumberAllocator by Erik Språng · 5 years ago
  82. 4208a13 Removes deprecated InsertPacket/TimeToSendPacket/TimeToSendPadding by Erik Språng · 5 years ago
  83. d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 6 years ago
  84. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 6 years ago
  85. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 6 years ago
  86. 54d5d2c Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc by Erik Språng · 6 years ago
  87. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 6 years ago
  88. 58b496b Let StreamStatistician::GetReceiveStreamDataCounters return counters by value by Niels Möller · 6 years ago
  89. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 6 years ago
  90. b168678 Add RTC_ prefix to non-standard format specifier macro "PRIdNS" by Oleh Prypin · 6 years ago
  91. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 6 years ago
  92. ed44f54 In ChannelReceive, use AcmReceiver directly, not AudioCodingModule by Niels Möller · 6 years ago
  93. fedd625 Change 2g network pc audio test to more realistic network by Artem Titov · 6 years ago
  94. 054e3bb Reland "Replace the implementation of `GetContributingSources()` on the audio side." by Chen Xing · 6 years ago
  95. da4f093 Reland "Only include payload in bytes sent/received." by Bjorn A Mellem · 6 years ago
  96. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 6 years ago
  97. bcd068d Revert "Only include payload in bytes sent/received." by Bjorn Mellem · 6 years ago
  98. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 6 years ago
  99. 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 6 years ago
  100. 8f319a3 Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" by Alessio Bazzica · 6 years ago