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gerrit-public.fairphone.software / platform / external / webrtc / 79aab3ff51105d7ef38152e27cfcdb513313a6db
  1. 79aab3f VideoStreamDecoderImpl implementation, part 2. by philipel · 6 years ago
  2. ef99888 Delete OnIncomingCSRCChanged and related code. by Niels Möller · 6 years ago
  3. 150dcb0 Delete logic to set picture id and tl0 pic idx in encoders. by Niels Möller · 6 years ago
  4. d45e078 Initialize variables to prevent usage of trash values, for exmaple, if NeedMorePlayData won't set the right value by Artem Titov · 6 years ago
  5. 959a1b9 Disable FieldTrials/CallOperationEndToEndTest.ObserversEncodedFrames due to flakiness. by Ying Wang · 6 years ago
  6. 6c95bdc Adding network control test helper. by Sebastian Jansson · 6 years ago
  7. ca09125 Fixing the change in number of temporal layers in screen sharing mode. by Sergey Silkin · 7 years ago
  8. 370228c Move ScreenCapturerMac to mac sub directory by Julien Isorce · 6 years ago
  9. 16ccef7 Allow implementation defined STUN attributes in 0x4000-0x7FFF range by Jonas Oreland · 6 years ago
  10. be682d4 Fix chromium warnings for SdpVideoFormat. by Niels Möller · 6 years ago
  11. 27153ba Roll chromium_revision ac8224087d..3472e079dc (545875:546039) by Autoroller · 6 years ago
  12. 7149136 Lowercase windows includes in win audio device. by Noah Richards · 6 years ago
  13. 27f3bf5 Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 6 years ago
  14. 97d5e5b Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 6 years ago
  15. ea8b62a Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 6 years ago
  16. cf76d1c Roll chromium_revision bac605dc3e..ac8224087d (545770:545875) by Autoroller · 6 years ago
  17. 1550292 Adds support for multiple or no media stream ids. by Seth Hampson · 6 years ago
  18. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 6 years ago
  19. 1707168 Add setting to switch between new and legacy AudioDeviceModule in AppRTC by Paulina Hensman · 6 years ago
  20. 6c9a786 Add //base dependency to rtc_base:rtc_event when building with chromium. by John Budorick · 6 years ago
  21. e08a4c0 Delete dead code in RtpReceiverImpl and RTPPayloadRegistry. by Niels Möller · 6 years ago
  22. 250155d Fix histogram logging in InterpolatedGainCurve. by Alex Loiko · 6 years ago
  23. 1b3bd0b Roll chromium_revision 723d8fc2ac..bac605dc3e (545111:545770) by Autoroller · 6 years ago
  24. 947120f Fix of race on access to send side congestion controller. by Sebastian Jansson · 6 years ago
  25. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 6 years ago
  26. 1df1bf8 PeerConnectionInterface::GetStats() with selector argument added. by Henrik Boström · 6 years ago
  27. 0191441 Lowercase Windows headers. by Mirko Bonadei · 6 years ago
  28. 6e091605 Loosen |max_avg_buffer_level_sec| for SimulcastVP8. by Rasmus Brandt · 6 years ago
  29. bfa6b72 Reland "Add linux_internal_compile_lite trybot to CQ" by Oleh Prypin · 6 years ago
  30. 28f2dcd Revert "Add linux_internal_compile_lite trybot to CQ" by Oleh Prypin · 6 years ago
  31. b746c3c Add linux_internal_compile_lite trybot to CQ by Oleh Prypin · 6 years ago
  32. dc4737b Disable FieldTrials/CallOperationEndToEndTest.TransmitsFirstFrame/2 due to flakiness. by Rasmus Brandt · 6 years ago
  33. 04dd176 Reland "Move rtp-specific config out of EncoderSettings." by Niels Möller · 6 years ago
  34. 81de14f Reduce implementations of CallStatsObserver by 2. by Tommi · 6 years ago
  35. 6b28e20 Remove unused 'last_rtt_ms_' variable from vcm::VideoStreamDecoder. by Tommi · 6 years ago
  36. b0ba558 Adding WindowedFilter class for BBR. by Sebastian Jansson · 6 years ago
  37. 0bf0c3d Fix of ReceiveTimeCalculator field trial parsing. by Sebastian Jansson · 6 years ago
  38. 61fce40 Android: Fix compilation of moved aaudio by Magnus Jedvert · 6 years ago
  39. 883f470 Adding DataTransferTracker class for BBR. by Sebastian Jansson · 6 years ago
  40. 2bd41f9 Fix a bug caused by an early return when a TURN port receives a role conflict. by Qingsi Wang · 6 years ago
  41. 2b85792 Move rw_lock_wrapper.h to rtc_base/synchronization/ by Karl Wiberg · 6 years ago
  42. 5e55fe8 Adding flag to enable/disable use of SRTP_AES128_CM_SHA1_32 crypto suite. by Taylor Brandstetter · 6 years ago
  43. 767a2ce Fix for crash when reading from audio file in NetEq. by Ivo Creusen · 6 years ago
  44. bc01047 Simplify RtpVideoStreamReceiver::NotifyReceiverOfFecPacket. by Niels Möller · 6 years ago
  45. 9718711 VideoStreamDecoderImpl implementation, part 1. by philipel · 6 years ago
  46. a9d988d Update VideoSendStreamImpl for configuration transition. by Niels Möller · 6 years ago
  47. 9bd1af1 Adding RttStats class for BBR. by Sebastian Jansson · 6 years ago
  48. 06e363a Annotate AppRTCMobile with @Nulllable. by Sami Kalliomäki · 6 years ago
  49. 6a4d411 Move file_wrapper.h to rtc_base/system/ by Karl Wiberg · 6 years ago
  50. 278d03a Force alignment of JVM called functions. by Yura Yaroshevich · 6 years ago
  51. 66c24fe Update SendStatisticsProxy to use Rtp::payload_name, if available. by Niels Möller · 6 years ago
  52. 8458cff Extend TestAudioDeviceModule API. by Artem Titov · 6 years ago
  53. 866e08d Make rtc::Optional IceConfig parameters interpreted consistently. by Qingsi Wang · 6 years ago
  54. 296a0ce Add MID sending to RTPSender by Steve Anton · 6 years ago
  55. ba2a923 Roll chromium_revision e76515b1f3..723d8fc2ac (545023:545111) by Oleh Prypin · 6 years ago
  56. f7ac09f Changing log levels and logging of the AEC3 render buffer alignment by Per Åhgren · 6 years ago
  57. 36fc5e1 Remove thin_archive config from complete static libraries by Tom Anderson · 6 years ago
  58. 8be087f Update payload_type DCHECKS. by Niels Möller · 6 years ago
  59. 76b7f51 Move timestamp_extrapolator.h to rtc_base/time/ by Karl Wiberg · 6 years ago
  60. 7682c6e Improves in the ERLE estimation for AEC3 by Jesús de Vicente Peña · 6 years ago
  61. 29e7bee Move aligned memory utilities to rtc_base/memory/ by Karl Wiberg · 6 years ago
  62. 12130bb Reporting feedback availability to congestion controller. by Sebastian Jansson · 6 years ago
  63. 1d430f7 Interface for packet feedback availability in SSCC by Sebastian Jansson · 6 years ago
  64. 70899e0 Add support for creating java PCF from native one. by Artem Titov · 6 years ago
  65. e7592d8 Annotate libjingle_peerconnection_java with @Nullable. by Sami Kalliomäki · 6 years ago
  66. 12d6a49 Add payload_name and payload_type to VideoSendStream::Config::Rtp. by Niels Möller · 6 years ago
  67. 9cfb18c Delete obsolete method RtpFeedback::OnInitializeDecoder. by Niels Möller · 6 years ago
  68. 2fee4d6 VideoStreamDecoder skeleton. by philipel · 6 years ago
  69. b3179c7 Remove RTPSender::SetSendPayloadType by Danil Chapovalov · 6 years ago
  70. 76ff622 Roll chromium_revision aa3bb574c4..e76515b1f3 (544920:545023) by Autoroller · 6 years ago
  71. f3e2bf1 Further headset mode robustification based on linear filter convergence by Per Åhgren · 6 years ago
  72. a09993d Skip nullable check when building in Chromium by Oleh Prypin · 6 years ago
  73. 24a842a Add field VideoEncoderConfig::codec_type. by Niels Möller · 6 years ago
  74. 48cca02 Delete ortc methods using cricket::VideoCapturer. by Niels Möller · 6 years ago
  75. 0d44625 Add style guide rules for std::bind and std::function by Karl Wiberg · 6 years ago
  76. 9212097 Match the device config of iOS Debug and Release bots by Oleh Prypin · 6 years ago
  77. 82cea03 Clarify documentation on scoped_refptr::release. by Niels Möller · 6 years ago
  78. 7aabd39 Move asm_defines.h to rtc_base/system/ by Karl Wiberg · 6 years ago
  79. 1aee625 Roll chromium_revision 0187ccfaa8..aa3bb574c4 (544782:544920) by Autoroller · 6 years ago
  80. 5c532d3 Robustification of the echo suppression behavior during headset usage. by Per Åhgren · 6 years ago
  81. 5bdc82a Test for audio codec pair ID assignment by Karl Wiberg · 6 years ago
  82. 77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 6 years ago
  83. ccd1523 Reporting packet feedback availability in VideoSendStream by Sebastian Jansson · 6 years ago
  84. 1b0decc Roll chromium_revision 89bed79700..0187ccfaa8 (544673:544782) by Autoroller · 6 years ago
  85. 2c8773b Avoid DCHECK in P2PTransportChannel::MorePingable. by Taylor Brandstetter · 6 years ago
  86. 32362a6 Android: Simplify moved audio device module by Magnus Jedvert · 6 years ago
  87. 2955d82 Android audio record/track: Remove intermediate JNI manager by Magnus Jedvert · 6 years ago
  88. b34556e Added receive time calculator under field trial. by Sebastian Jansson · 6 years ago
  89. 763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 6 years ago
  90. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 6 years ago
  91. 92be1ca Revert "Move rtp-specific config out of EncoderSettings." by Niels Moller · 6 years ago
  92. 247e0b4 Disabling periodic tasks on SSCC in unit tests. by Sebastian Jansson · 6 years ago
  93. 81e8a43 Roll chromium_revision 3ad75d0441..89bed79700 (544573:544673) by Autoroller · 6 years ago
  94. 29b204e Tracking packet feedback availability in BitrateAllocator. by Sebastian Jansson · 6 years ago
  95. fe617a3 Adding has_packet_feedback to LimitObserver callback. by Sebastian Jansson · 6 years ago
  96. bc900cb Move rtp-specific config out of EncoderSettings. by Niels Möller · 6 years ago
  97. c3d1e09 Make sure RTCMTLVideoView.h ends up in framework headers. by Anders Carlsson · 6 years ago
  98. 08006d4 Android AppRTCMobile: Use new audio device code by Magnus Jedvert · 6 years ago
  99. 82fad3d Remove TemporalLayersFactory and associated classes by Erik Språng · 6 years ago
  100. 8fc7948 Android: Generate audio JNI code by Magnus Jedvert · 6 years ago