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webrtc / src.git / fa8f4eee40050e9875a4b8c38d99ea54a112c1ed
  1. fa8f4ee Only combine media transport and ICE states if used for media. by Bjorn A Mellem · 6 years ago
  2. 015c3cb Remove deprecated constructors of RtpSource by Johannes Kron · 6 years ago
  3. 0e1a558 Allowing 40ms audio frame length. by Ying Wang · 6 years ago
  4. 0ee4311 Roll chromium_revision c7f850c75e..d5a13ccb8e (687596:687732) by chromium-webrtc-autoroll · 6 years ago
  5. f5e5d25 BalancedDegradationSettings: add option to configure a min framerate diff. by Åsa Persson · 6 years ago
  6. df625f4 Revert "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Andreassson · 6 years ago
  7. 6094953 Add helper functions to convert between integer milliseconds and fixed-point seconds. by Chen Xing · 6 years ago
  8. 2b9fa09 [GetStats] Expose video codec implementation in standardized metrics. by Henrik Boström · 6 years ago
  9. cc96db6 Simplify stats poller stop in PC level framework by Artem Titov · 6 years ago
  10. 6950b30 Fix thread naming in Call constructor by Erik Språng · 6 years ago
  11. bbeb109 Reporting audio device underrun counter by Alex Narest · 6 years ago
  12. 9b29d69 Make ANA frame length controller more robust to encoder frame lengths. by Minyue Li · 6 years ago
  13. 533c225 Revert "Correct conversion between float and fixed formats" by Henrik Andreassson · 6 years ago
  14. 07a6652 Roll chromium_revision f54998af9c..c7f850c75e (687496:687596) by chromium-webrtc-autoroll · 6 years ago
  15. 98bbd88 Roll chromium_revision 7a2da7b921..f54998af9c (686822:687496) by chromium-webrtc-autoroll · 6 years ago
  16. e5defb1 Sanitize the selected candidate pair in the public API. by Qingsi Wang · 6 years ago
  17. ffc525b Fix a bug/typo in WebRtcSpl_FilterAR which updates the wrong state vector by Jiawei Ou · 6 years ago
  18. 67e43c8 Correct conversion between float and fixed formats by Per Åhgren · 6 years ago
  19. a135127 Remove all AudioBuffer code that is not related to storing audio data by Per Åhgren · 6 years ago
  20. 6e4791f Add check for IsCurrent() for SendTask in SingleThreadedTaskQueueForTesting. by Tommi · 6 years ago
  21. 65feec5 Reenable UlpfecWithNack integration tests by Danil Chapovalov · 6 years ago
  22. 1b247f1 BalancedDegradationSettings: add option to configure min bitrate. by Åsa Persson · 6 years ago
  23. 3aa0d76 Use struct parser for AlrDetector config. by Sebastian Jansson · 6 years ago
  24. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 6 years ago
  25. 09ba219 Roll chromium_revision fa752aeae4..7a2da7b921 (686692:686822) by chromium-webrtc-autoroll · 6 years ago
  26. c759f83 Avoid copying of vectors in RtpPacketInfos. by Minyue Li · 6 years ago
  27. c14b233 Disable the most flaky tests on iOS. by Sami Kalliomäki · 6 years ago
  28. 7daf550 Add new FrameRateEstimator utility class for more precis FPS estimation. by Erik Språng · 6 years ago
  29. 0ee8008 Use struct parser for rate control trial. by Sebastian Jansson · 6 years ago
  30. ad9113f Adds sequence numbers to feedback generator output. by Sebastian Jansson · 6 years ago
  31. 0c38a86 BalancedDegradationSettings: add option to configure no fps limit. by Åsa Persson · 6 years ago
  32. 704c8c4 Re-enable AudioDeviceTest in combination with sanitizers. by Yves Gerey · 6 years ago
  33. 78c56cb Delete deprecated version of ReceiveStatistics::Create by Niels Möller · 6 years ago
  34. 1e04a9b Roll chromium_revision bcb9240637..fa752aeae4 (686583:686692) by chromium-webrtc-autoroll · 6 years ago
  35. fb6edd3 Handle case of empty connection in pair change event by Alex Drake · 6 years ago
  36. bb19942 Roll chromium_revision 6652dd41e1..bcb9240637 (686436:686583) by chromium-webrtc-autoroll · 6 years ago
  37. 68c2a56 Propagating Network Type in Candidate for JNI by Alex Drake · 6 years ago
  38. 608e6ba Add AudioDecoderIsacT::Config to include sampling rate and BWInfo object by Jiawei Ou · 6 years ago
  39. 05497f2 Pull a DataChannelTransportInterface out of MediaTransportInterface. by Bjorn A Mellem · 6 years ago
  40. d419808 Revert "Set the usage pattern bits for adding remote ICE candidates from SDP." by Qingsi Wang · 6 years ago
  41. 7c6f74a Set the usage pattern bits for adding remote ICE candidates from SDP. by Qingsi Wang · 6 years ago
  42. 1a03784 Roll chromium_revision 3ae2445b34..6652dd41e1 (686310:686436) by chromium-webrtc-autoroll · 6 years ago
  43. 9cfdb20 Control PeerConnectionFactory's default min/starting/max bitrates from experiment by Elad Alon · 6 years ago
  44. d781965 Delete StreamDataCountersCallback from ReceiveStatistics by Niels Möller · 6 years ago
  45. 01525f9 Delete method StreamStatistician::GetDataCounters by Niels Möller · 6 years ago
  46. 34aee67 Roll chromium_revision 514a543362..3ae2445b34 (686198:686310) by chromium-webrtc-autoroll · 6 years ago
  47. 43faee0 Implement JNI and objc implementation for Ice Candidate Pair Change event surfacing by Alex Drake · 6 years ago
  48. 519fc44 Roll chromium_revision 01bf391305..514a543362 (686061:686198) by chromium-webrtc-autoroll · 6 years ago
  49. 9809cad Roll chromium_revision f0fd984a31..01bf391305 (685691:686061) by chromium-webrtc-autoroll · 6 years ago
  50. 82d75a6 Use unit types in RoundRobingPacketQueue and PacedSender by Erik Språng · 6 years ago
  51. 4d207a3 Add frames_in_flight metric to catch not delivered frames by Artem Titov · 6 years ago
  52. 110a4de Roll chromium_revision 8f0166a59b..f0fd984a31 (685582:685691) by Yves Gerey · 6 years ago
  53. 40dc98a Print stack trace when a test crash by Danil Chapovalov · 6 years ago
  54. eea605d Make fake network degradation work also for sent audio by Erik Språng · 6 years ago
  55. 58b496b Let StreamStatistician::GetReceiveStreamDataCounters return counters by value by Niels Möller · 6 years ago
  56. 412282a [tsan] Guard audio_device_pulse_linux members from concurrent access. by Yves Gerey · 6 years ago
  57. 1691e88 Remove unused fallback method in PacedSender by Erik Språng · 6 years ago
  58. dc5ed5c Delete NACK-related methods from AudioCodingModule by Niels Möller · 6 years ago
  59. b75d14c audioproc_f: input AEC dump as string, output audio to vector by Sonia-Florina Horchidan · 6 years ago
  60. 81df62b Add field trial to introduce extra delay after target level calculation. by Jakob Ivarsson · 6 years ago
  61. 1544915 Avoid capturing extraneous windows in CroppingWindowCapturerWin by Bryan Ferguson · 6 years ago
  62. e427996 Roll chromium_revision 87ee38fb42..8f0166a59b (685466:685582) by chromium-webrtc-autoroll · 6 years ago
  63. 6b2cec1 Use recommended min bitrate limit provided by encoder. by Sergey Silkin · 6 years ago
  64. 48b48e5 Enable thread check in Call::GetStats(). by Tommi · 6 years ago
  65. e4ba4ee Delete placeholder file rtc_base/function_view.h by Niels Möller · 6 years ago
  66. a52e9bd Use StreamStatistician::BitrateReceived to produce total_bitrate_bps for GetStats. by Niels Möller · 6 years ago
  67. 6685b32 Delete rtc_base/gunit_prod.h by Niels Möller · 6 years ago
  68. e4b4de6 Add missing AppKit dependency by Niels Möller · 6 years ago
  69. 273e263 Delete old placeholder file android_network_monitor_jni.h by Niels Möller · 6 years ago
  70. b90d38a Delete unused Opus-specific methods of AudioCodingModule by Niels Möller · 6 years ago
  71. 45fd69d Roll chromium_revision 6fb8f3c614..87ee38fb42 (685365:685466) by chromium-webrtc-autoroll · 6 years ago
  72. 5297cf3 Delete unused class MockTargetTransferRateObserver by Niels Möller · 6 years ago
  73. 5e4af85 Roll chromium_revision 9230e75a8c..6fb8f3c614 (685264:685365) by chromium-webrtc-autoroll · 6 years ago
  74. 287bff3 Roll chromium_revision 498f5876be..9230e75a8c (685149:685264) by chromium-webrtc-autoroll · 6 years ago
  75. 55251c3 Adds struct parameters parser/encoder. by Sebastian Jansson · 6 years ago
  76. 940c2b5 AEC3: Reduce level of log messages by Gustaf Ullberg · 6 years ago
  77. b6b7d1f Roll chromium_revision 5744654b26..498f5876be (685023:685149) by chromium-webrtc-autoroll · 6 years ago
  78. 78a7138 Remove MediaTransport from Call. by Tommi · 6 years ago
  79. 44327c3 Update test::CreateVideoStreams to use configured scale_resolution_down_by if set. by Åsa Persson · 6 years ago
  80. 383adc0 Delete shim of PRId64 et al. on Windows by Oleh Prypin · 6 years ago
  81. 0d210ee Change return type of of ReceiveStatistics::Create to unique_ptr. by Niels Möller · 6 years ago
  82. c2fe547 Remove unused fallbacks in PacedSender by Erik Språng · 6 years ago
  83. eac47f7 Removing unused fallback variant for the reverb computation by Per Åhgren · 6 years ago
  84. 891d393 Call Call::GetStats() from the correct thread in ProbingEndToEndTest. by Tommi · 6 years ago
  85. aaaf804 Call Call::GetStats() from the correct thread in VideoSendStreamTest. by Tommi · 6 years ago
  86. efffd0a Roll chromium_revision 3d0c04364f..5744654b26 (684897:685023) by chromium-webrtc-autoroll · 6 years ago
  87. 307448f Roll chromium_revision 006302cd2e..3d0c04364f (684781:684897) by chromium-webrtc-autoroll · 6 years ago
  88. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 6 years ago
  89. 2d2bbb1 Filter out duplicate receive codecs in the media engine by Steve Anton · 6 years ago
  90. 3cc2f70 Roll chromium_revision 192da69226..006302cd2e (684664:684781) by chromium-webrtc-autoroll · 6 years ago
  91. b168678 Add RTC_ prefix to non-standard format specifier macro "PRIdNS" by Oleh Prypin · 6 years ago
  92. 12ebfa6 Delete RtcpStatisticsCallback from ReceiveStatistics by Niels Möller · 6 years ago
  93. b668542 Delete unused format specifier macros for NSInteger and NSUInteger by Oleh Prypin · 6 years ago
  94. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 6 years ago
  95. e08648d Add `AbsoluteCaptureTime` to `RtpPacketInfo`. by Chen Xing · 6 years ago
  96. f40a340 Remove deprecated code related to AEC2 by Per Åhgren · 6 years ago
  97. 75caef7 Delete unused ACM members isac_decoder_16k_ and isac_decoder_32k_ by Niels Möller · 6 years ago
  98. d2845f8 Removes unused AudioAllocationSettings from voice engine. by Sebastian Jansson · 6 years ago
  99. d23f67e Call Call::GetStats() from the correct thread in StatsEndToEndTest. by Tommi · 6 years ago
  100. c24a5b1 Fix CallPerfTests to call Call::GetStats() from the right thread. by Tommi · 6 years ago