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webrtc / src.git / e7592d8d5fed88abc1bf8f4ee2492aa98cd63bb9
  1. e7592d8 Annotate libjingle_peerconnection_java with @Nullable. by Sami Kalliomäki · 7 years ago
  2. 12d6a49 Add payload_name and payload_type to VideoSendStream::Config::Rtp. by Niels Möller · 7 years ago
  3. 9cfb18c Delete obsolete method RtpFeedback::OnInitializeDecoder. by Niels Möller · 7 years ago
  4. 2fee4d6 VideoStreamDecoder skeleton. by philipel · 7 years ago
  5. b3179c7 Remove RTPSender::SetSendPayloadType by Danil Chapovalov · 7 years ago
  6. 76ff622 Roll chromium_revision aa3bb574c4..e76515b1f3 (544920:545023) by Autoroller · 7 years ago
  7. f3e2bf1 Further headset mode robustification based on linear filter convergence by Per Åhgren · 7 years ago
  8. a09993d Skip nullable check when building in Chromium by Oleh Prypin · 7 years ago
  9. 24a842a Add field VideoEncoderConfig::codec_type. by Niels Möller · 7 years ago
  10. 48cca02 Delete ortc methods using cricket::VideoCapturer. by Niels Möller · 7 years ago
  11. 0d44625 Add style guide rules for std::bind and std::function by Karl Wiberg · 7 years ago
  12. 9212097 Match the device config of iOS Debug and Release bots by Oleh Prypin · 7 years ago
  13. 82cea03 Clarify documentation on scoped_refptr::release. by Niels Möller · 7 years ago
  14. 7aabd39 Move asm_defines.h to rtc_base/system/ by Karl Wiberg · 7 years ago
  15. 1aee625 Roll chromium_revision 0187ccfaa8..aa3bb574c4 (544782:544920) by Autoroller · 7 years ago
  16. 5c532d3 Robustification of the echo suppression behavior during headset usage. by Per Åhgren · 7 years ago
  17. 5bdc82a Test for audio codec pair ID assignment by Karl Wiberg · 7 years ago
  18. 77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 7 years ago
  19. ccd1523 Reporting packet feedback availability in VideoSendStream by Sebastian Jansson · 7 years ago
  20. 1b0decc Roll chromium_revision 89bed79700..0187ccfaa8 (544673:544782) by Autoroller · 7 years ago
  21. 2c8773b Avoid DCHECK in P2PTransportChannel::MorePingable. by Taylor Brandstetter · 7 years ago
  22. 32362a6 Android: Simplify moved audio device module by Magnus Jedvert · 7 years ago
  23. 2955d82 Android audio record/track: Remove intermediate JNI manager by Magnus Jedvert · 7 years ago
  24. b34556e Added receive time calculator under field trial. by Sebastian Jansson · 7 years ago
  25. 763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 7 years ago
  26. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 7 years ago
  27. 92be1ca Revert "Move rtp-specific config out of EncoderSettings." by Niels Moller · 7 years ago
  28. 247e0b4 Disabling periodic tasks on SSCC in unit tests. by Sebastian Jansson · 7 years ago
  29. 81e8a43 Roll chromium_revision 3ad75d0441..89bed79700 (544573:544673) by Autoroller · 7 years ago
  30. 29b204e Tracking packet feedback availability in BitrateAllocator. by Sebastian Jansson · 7 years ago
  31. fe617a3 Adding has_packet_feedback to LimitObserver callback. by Sebastian Jansson · 7 years ago
  32. bc900cb Move rtp-specific config out of EncoderSettings. by Niels Möller · 7 years ago
  33. c3d1e09 Make sure RTCMTLVideoView.h ends up in framework headers. by Anders Carlsson · 7 years ago
  34. 08006d4 Android AppRTCMobile: Use new audio device code by Magnus Jedvert · 7 years ago
  35. 82fad3d Remove TemporalLayersFactory and associated classes by Erik Språng · 7 years ago
  36. 8fc7948 Android: Generate audio JNI code by Magnus Jedvert · 7 years ago
  37. 37e3602 Android: Add henrika@ as owner of audio code by Magnus Jedvert · 7 years ago
  38. e7fac68 Introduce Nullable annotation. by Sami Kalliomäki · 7 years ago
  39. 7531a76 Delete unused header media/base/test/mock_mediachannel.h. by Niels Möller · 7 years ago
  40. 9611442 Remove unneeded migration helper. by Sami Kalliomäki · 7 years ago
  41. e61631d Roll chromium_revision 2f0e6b63b5..3ad75d0441 (544446:544573) by Autoroller · 7 years ago
  42. b88bfc9 Roll chromium_revision 8c0344a12e..2f0e6b63b5 (544337:544446) by Autoroller · 7 years ago
  43. db67ba1 Report SRTP error codes to UMA by Steve Anton · 7 years ago
  44. e9d2e4d Provide the option of injecting rtc::TaskQueue when creating RtcEventLogImpl via factory methods. by Dino Radaković · 7 years ago
  45. fe48ee9 Fixing zlib license generation. by Mirko Bonadei · 7 years ago
  46. b38b05b Adding srte as owner in modules/congestion_controller. by Sebastian Jansson · 7 years ago
  47. 89dd7bf Move android audio device code into sdk/android by Paulina Hensman · 7 years ago
  48. 4d22a6d Delete unneeded includes of wav_file.h and file_wrapper.h. by Niels Möller · 7 years ago
  49. 8ef59a4 Added data member access methods to FakeNetworkPipe. by Christoffer Rodbro · 7 years ago
  50. 3dc0125c Moving ConfigureEncoderTask to the calling scope. by Sebastian Jansson · 7 years ago
  51. 68a7168 Roll chromium_revision ee966518c2..8c0344a12e (544233:544337) by Autoroller · 7 years ago
  52. 2e0da5a Remove EncodedFrame picture_id/spatial_layer references. by philipel · 7 years ago
  53. f18072e Enable SVC based on number of SSRCs. by Sergey Silkin · 7 years ago
  54. eb98c72 Minor improvements in ADM unittest for Windows. by henrika · 7 years ago
  55. faed538 Delete obsolete alias RateLimiter and rtc_base/ratelimiter.h by Niels Möller · 7 years ago
  56. 9047dac Disable flaky test SendSideCongestionControllerTest/PacerQueueEncodeRatePushback. by Rasmus Brandt · 7 years ago
  57. d328ec3 Change valueOf -> parseBoolean to avoid unneeded boxing by Oleh Prypin · 7 years ago
  58. a98bb2d Roll chromium_revision 02de71e34e..ee966518c2 (544129:544233) by Autoroller · 7 years ago
  59. 114155b Roll chromium_revision 163641576c..02de71e34e (544029:544129) by Autoroller · 7 years ago
  60. 01cb5f2 Fix issue where sockets bound to temporary IPv6 addresses are discarded. by Taylor Brandstetter · 7 years ago
  61. 3d976f6 Discard link to media channel when audio sender stopped. by Harald Alvestrand · 7 years ago
  62. bb60a3a Refactor VP8 TemporalLayers by Erik Språng · 7 years ago
  63. d757356 Fixing -Wstrict-prototypes warnings. by Mirko Bonadei · 7 years ago
  64. def1ef5 New equality operators, for structs related to webrtc::VideoCodec. by Niels Möller · 7 years ago
  65. 5b3541f RTCStatsCollector::GetStatsReport() with optional selector argument. by Henrik Boström · 7 years ago
  66. 4a73cd4 Adding tests of TaskQueueCongestionControl field trial. by Sebastian Jansson · 7 years ago
  67. e62f600 Extend WavReader and WavWriter API. by Artem Titov · 7 years ago
  68. 451dfdf Roll chromium_revision 7a8a322ad7..163641576c (543921:544029) by Autoroller · 7 years ago
  69. 0fa82a6 Moved FrameKey to api/video/encoded_frame.h and renamed it to VideoLayerFrameId. by philipel · 7 years ago
  70. 9c1ee36 Fix low_bandwidth_audio_perf_test resource dependency on Android by Oleh Prypin · 7 years ago
  71. b9a02e5 Change place of UMA logging in AudioMixer. by Alex Loiko · 7 years ago
  72. 5370124 Replacing unique pointer with raw pointer in SSCC checks. by Sebastian Jansson · 7 years ago
  73. 04d4950 Revert "Using safe casts of allocation limits in Call." by Oleh Prypin · 7 years ago
  74. 4a9b4d6 Using safe casts of allocation limits in Call. by Sebastian Jansson · 7 years ago
  75. 8d8cb56 Delete obsolete methods from MockRtpTransportControllerSend by Sebastian Jansson · 7 years ago
  76. b708e93 Bring mb up to date with Chromium's changes by Oleh Prypin · 7 years ago
  77. 8d2c5a8 Detangling target dependencies in rtc_base_approved. by Tommi · 7 years ago
  78. 7b2676f Fix low_bandwidth_audio_perf_test binary dependency on Windows by Oleh Prypin · 7 years ago
  79. d2c8332 Revert "Relaxing no-streams presubmit check (streams are allowed in tests)." by Mirko Bonadei · 7 years ago
  80. 7311918 Add an example app for iOS native API. by Anders Carlsson · 7 years ago
  81. 8cf0a87 Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
  82. 73ac908 Relaxing no-streams presubmit check (streams are allowed in tests). by Mirko Bonadei · 7 years ago
  83. 9fa35e5 Fix path to proto in py_event_log_analyzer/pb_parse.py by Oleh Prypin · 7 years ago
  84. 8a1b20a Roll chromium_revision 33aa22e76e..7a8a322ad7 (543816:543921) by Autoroller · 7 years ago
  85. cdd2a97 Roll chromium_revision ce851e47bd..33aa22e76e (543685:543816) by Autoroller · 7 years ago
  86. 317a522 Fixes to posting delayed process tasks in SSCC. by Sebastian Jansson · 7 years ago
  87. 4ccc1c4 Don't destroy a receive stream's sink before reassigning it. by Oskar Sundbom · 7 years ago
  88. 3bb1194 Revert "Add 'is_chrome_branded' guard to the default of 'rtc_use_h264'" by Patrik Höglund · 7 years ago
  89. 35468356 Roll chromium_revision e0e02de5a7..ce851e47bd (543578:543685) by Autoroller · 7 years ago
  90. bf3dbb4 Delete payload_type from VCMEncoderDatabase and vcm::VideoSender. by Niels Möller · 7 years ago
  91. 5bf8ccd Delete encoder caching in WebRtcVideoSendStream. by Niels Möller · 7 years ago
  92. 677f42c Enable ContinuousAfterStreamCountChangeSimulcastEncoderAdapter picture id tests. by Åsa Persson · 7 years ago
  93. d132ce1 Remove unnecessary copies from AsyncInvoke by Cameron Pickett · 7 years ago
  94. 465a5d9 Refactor payload types constants in CallTest by Ilya Nikolaevskiy · 7 years ago
  95. af9e87b Delete unused methods from vcm::VideoCodingModule. by Niels Möller · 7 years ago
  96. eef09fc Fix race in DegradedCall::DestroyVideoSendStream by Erik Språng · 7 years ago
  97. 883d00f Add support of AAudio in native WebRTC on Android O and above by henrika · 7 years ago
  98. 815f3b6 Fix podspec iOS version. by Kári Tristan Helgason · 7 years ago
  99. 7696bef Remove the public_deps to fileutils from test_support. by Patrik Höglund · 7 years ago
  100. 4de9eb2 Roll chromium_revision bc2c5b551b..e0e02de5a7 (543473:543578) by Autoroller · 7 years ago