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gerrit-public.fairphone.software / platform / external / webrtc / 27f3bf512827b483f9e0c67ce76362d83faa1950
  1. 27f3bf5 Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 7 years ago
  2. 97d5e5b Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 7 years ago
  3. ea8b62a Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 7 years ago
  4. cf76d1c Roll chromium_revision bac605dc3e..ac8224087d (545770:545875) by Autoroller · 7 years ago
  5. 1550292 Adds support for multiple or no media stream ids. by Seth Hampson · 7 years ago
  6. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 7 years ago
  7. 1707168 Add setting to switch between new and legacy AudioDeviceModule in AppRTC by Paulina Hensman · 7 years ago
  8. 6c9a786 Add //base dependency to rtc_base:rtc_event when building with chromium. by John Budorick · 7 years ago
  9. e08a4c0 Delete dead code in RtpReceiverImpl and RTPPayloadRegistry. by Niels Möller · 7 years ago
  10. 250155d Fix histogram logging in InterpolatedGainCurve. by Alex Loiko · 7 years ago
  11. 1b3bd0b Roll chromium_revision 723d8fc2ac..bac605dc3e (545111:545770) by Autoroller · 7 years ago
  12. 947120f Fix of race on access to send side congestion controller. by Sebastian Jansson · 7 years ago
  13. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 7 years ago
  14. 1df1bf8 PeerConnectionInterface::GetStats() with selector argument added. by Henrik Boström · 7 years ago
  15. 0191441 Lowercase Windows headers. by Mirko Bonadei · 7 years ago
  16. 6e091605 Loosen |max_avg_buffer_level_sec| for SimulcastVP8. by Rasmus Brandt · 7 years ago
  17. bfa6b72 Reland "Add linux_internal_compile_lite trybot to CQ" by Oleh Prypin · 7 years ago
  18. 28f2dcd Revert "Add linux_internal_compile_lite trybot to CQ" by Oleh Prypin · 7 years ago
  19. b746c3c Add linux_internal_compile_lite trybot to CQ by Oleh Prypin · 7 years ago
  20. dc4737b Disable FieldTrials/CallOperationEndToEndTest.TransmitsFirstFrame/2 due to flakiness. by Rasmus Brandt · 7 years ago
  21. 04dd176 Reland "Move rtp-specific config out of EncoderSettings." by Niels Möller · 7 years ago
  22. 81de14f Reduce implementations of CallStatsObserver by 2. by Tommi · 7 years ago
  23. 6b28e20 Remove unused 'last_rtt_ms_' variable from vcm::VideoStreamDecoder. by Tommi · 7 years ago
  24. b0ba558 Adding WindowedFilter class for BBR. by Sebastian Jansson · 7 years ago
  25. 0bf0c3d Fix of ReceiveTimeCalculator field trial parsing. by Sebastian Jansson · 7 years ago
  26. 61fce40 Android: Fix compilation of moved aaudio by Magnus Jedvert · 7 years ago
  27. 883f470 Adding DataTransferTracker class for BBR. by Sebastian Jansson · 7 years ago
  28. 2bd41f9 Fix a bug caused by an early return when a TURN port receives a role conflict. by Qingsi Wang · 7 years ago
  29. 2b85792 Move rw_lock_wrapper.h to rtc_base/synchronization/ by Karl Wiberg · 7 years ago
  30. 5e55fe8 Adding flag to enable/disable use of SRTP_AES128_CM_SHA1_32 crypto suite. by Taylor Brandstetter · 7 years ago
  31. 767a2ce Fix for crash when reading from audio file in NetEq. by Ivo Creusen · 7 years ago
  32. bc01047 Simplify RtpVideoStreamReceiver::NotifyReceiverOfFecPacket. by Niels Möller · 7 years ago
  33. 9718711 VideoStreamDecoderImpl implementation, part 1. by philipel · 7 years ago
  34. a9d988d Update VideoSendStreamImpl for configuration transition. by Niels Möller · 7 years ago
  35. 9bd1af1 Adding RttStats class for BBR. by Sebastian Jansson · 7 years ago
  36. 06e363a Annotate AppRTCMobile with @Nulllable. by Sami Kalliomäki · 7 years ago
  37. 6a4d411 Move file_wrapper.h to rtc_base/system/ by Karl Wiberg · 7 years ago
  38. 278d03a Force alignment of JVM called functions. by Yura Yaroshevich · 7 years ago
  39. 66c24fe Update SendStatisticsProxy to use Rtp::payload_name, if available. by Niels Möller · 7 years ago
  40. 8458cff Extend TestAudioDeviceModule API. by Artem Titov · 7 years ago
  41. 866e08d Make rtc::Optional IceConfig parameters interpreted consistently. by Qingsi Wang · 7 years ago
  42. 296a0ce Add MID sending to RTPSender by Steve Anton · 7 years ago
  43. ba2a923 Roll chromium_revision e76515b1f3..723d8fc2ac (545023:545111) by Oleh Prypin · 7 years ago
  44. f7ac09f Changing log levels and logging of the AEC3 render buffer alignment by Per Åhgren · 7 years ago
  45. 36fc5e1 Remove thin_archive config from complete static libraries by Tom Anderson · 7 years ago
  46. 8be087f Update payload_type DCHECKS. by Niels Möller · 7 years ago
  47. 76b7f51 Move timestamp_extrapolator.h to rtc_base/time/ by Karl Wiberg · 7 years ago
  48. 7682c6e Improves in the ERLE estimation for AEC3 by Jesús de Vicente Peña · 7 years ago
  49. 29e7bee Move aligned memory utilities to rtc_base/memory/ by Karl Wiberg · 7 years ago
  50. 12130bb Reporting feedback availability to congestion controller. by Sebastian Jansson · 7 years ago
  51. 1d430f7 Interface for packet feedback availability in SSCC by Sebastian Jansson · 7 years ago
  52. 70899e0 Add support for creating java PCF from native one. by Artem Titov · 7 years ago
  53. e7592d8 Annotate libjingle_peerconnection_java with @Nullable. by Sami Kalliomäki · 7 years ago
  54. 12d6a49 Add payload_name and payload_type to VideoSendStream::Config::Rtp. by Niels Möller · 7 years ago
  55. 9cfb18c Delete obsolete method RtpFeedback::OnInitializeDecoder. by Niels Möller · 7 years ago
  56. 2fee4d6 VideoStreamDecoder skeleton. by philipel · 7 years ago
  57. b3179c7 Remove RTPSender::SetSendPayloadType by Danil Chapovalov · 7 years ago
  58. 76ff622 Roll chromium_revision aa3bb574c4..e76515b1f3 (544920:545023) by Autoroller · 7 years ago
  59. f3e2bf1 Further headset mode robustification based on linear filter convergence by Per Åhgren · 7 years ago
  60. a09993d Skip nullable check when building in Chromium by Oleh Prypin · 7 years ago
  61. 24a842a Add field VideoEncoderConfig::codec_type. by Niels Möller · 7 years ago
  62. 48cca02 Delete ortc methods using cricket::VideoCapturer. by Niels Möller · 7 years ago
  63. 0d44625 Add style guide rules for std::bind and std::function by Karl Wiberg · 7 years ago
  64. 9212097 Match the device config of iOS Debug and Release bots by Oleh Prypin · 7 years ago
  65. 82cea03 Clarify documentation on scoped_refptr::release. by Niels Möller · 7 years ago
  66. 7aabd39 Move asm_defines.h to rtc_base/system/ by Karl Wiberg · 7 years ago
  67. 1aee625 Roll chromium_revision 0187ccfaa8..aa3bb574c4 (544782:544920) by Autoroller · 7 years ago
  68. 5c532d3 Robustification of the echo suppression behavior during headset usage. by Per Åhgren · 7 years ago
  69. 5bdc82a Test for audio codec pair ID assignment by Karl Wiberg · 7 years ago
  70. 77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 7 years ago
  71. ccd1523 Reporting packet feedback availability in VideoSendStream by Sebastian Jansson · 7 years ago
  72. 1b0decc Roll chromium_revision 89bed79700..0187ccfaa8 (544673:544782) by Autoroller · 7 years ago
  73. 2c8773b Avoid DCHECK in P2PTransportChannel::MorePingable. by Taylor Brandstetter · 7 years ago
  74. 32362a6 Android: Simplify moved audio device module by Magnus Jedvert · 7 years ago
  75. 2955d82 Android audio record/track: Remove intermediate JNI manager by Magnus Jedvert · 7 years ago
  76. b34556e Added receive time calculator under field trial. by Sebastian Jansson · 7 years ago
  77. 763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 7 years ago
  78. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 7 years ago
  79. 92be1ca Revert "Move rtp-specific config out of EncoderSettings." by Niels Moller · 7 years ago
  80. 247e0b4 Disabling periodic tasks on SSCC in unit tests. by Sebastian Jansson · 7 years ago
  81. 81e8a43 Roll chromium_revision 3ad75d0441..89bed79700 (544573:544673) by Autoroller · 7 years ago
  82. 29b204e Tracking packet feedback availability in BitrateAllocator. by Sebastian Jansson · 7 years ago
  83. fe617a3 Adding has_packet_feedback to LimitObserver callback. by Sebastian Jansson · 7 years ago
  84. bc900cb Move rtp-specific config out of EncoderSettings. by Niels Möller · 7 years ago
  85. c3d1e09 Make sure RTCMTLVideoView.h ends up in framework headers. by Anders Carlsson · 7 years ago
  86. 08006d4 Android AppRTCMobile: Use new audio device code by Magnus Jedvert · 7 years ago
  87. 82fad3d Remove TemporalLayersFactory and associated classes by Erik Språng · 7 years ago
  88. 8fc7948 Android: Generate audio JNI code by Magnus Jedvert · 7 years ago
  89. 37e3602 Android: Add henrika@ as owner of audio code by Magnus Jedvert · 7 years ago
  90. e7fac68 Introduce Nullable annotation. by Sami Kalliomäki · 7 years ago
  91. 7531a76 Delete unused header media/base/test/mock_mediachannel.h. by Niels Möller · 7 years ago
  92. 9611442 Remove unneeded migration helper. by Sami Kalliomäki · 7 years ago
  93. e61631d Roll chromium_revision 2f0e6b63b5..3ad75d0441 (544446:544573) by Autoroller · 7 years ago
  94. b88bfc9 Roll chromium_revision 8c0344a12e..2f0e6b63b5 (544337:544446) by Autoroller · 7 years ago
  95. db67ba1 Report SRTP error codes to UMA by Steve Anton · 7 years ago
  96. e9d2e4d Provide the option of injecting rtc::TaskQueue when creating RtcEventLogImpl via factory methods. by Dino Radaković · 7 years ago
  97. fe48ee9 Fixing zlib license generation. by Mirko Bonadei · 7 years ago
  98. b38b05b Adding srte as owner in modules/congestion_controller. by Sebastian Jansson · 7 years ago
  99. 89dd7bf Move android audio device code into sdk/android by Paulina Hensman · 7 years ago
  100. 4d22a6d Delete unneeded includes of wav_file.h and file_wrapper.h. by Niels Möller · 7 years ago