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gerrit-public.fairphone.software / platform / external / webrtc / 196100efa62583dca81c52e8abc06a29b086764a
  1. 196100e Replace rtc::Optional with absl::optional by Danil Chapovalov · 6 years ago
  2. ae810c1 Create a peer connection factory builder by Jiawei Ou · 6 years ago
  3. f1e3cb4 Add OnLogMessage(msg, sev, tag) to logsinks by Paulina Hensman · 6 years ago
  4. db38972 Remove nonlinear beamformer API from APM by Sam Zackrisson · 6 years ago
  5. 7b55c73 Add RTPVideoHeader accessor. by philipel · 6 years ago
  6. fd2457b Roll chromium_revision e1ef7d4b6b..b47e7752c6 (568794:569173) by Autoroller · 6 years ago
  7. 5b8dd4d Fix a dangling-pointer bug in P2PTransportChannel unit tests. by Qingsi Wang · 6 years ago
  8. b983bae Remove unused/deprecated DTMF methods by Steve Anton · 6 years ago
  9. 3d8b171 Removing some TSan suppressions around Thread class. by Taylor Brandstetter · 6 years ago
  10. db6af36 Add RNN-VAD to AGC2. by Alex Loiko · 6 years ago
  11. 87a9353 Add format check to `git cl presubmit` by Yves Gerey · 6 years ago
  12. beb2d98 Removing usage of //build/config/compiler:no_size_t_to_int_warning. by Mirko Bonadei · 6 years ago
  13. de212ca Removing some MSVC warning suppression flags. by Mirko Bonadei · 6 years ago
  14. b23db02 Add mbonadei@ to build configs OWNERS. by Mirko Bonadei · 6 years ago
  15. bc5c934 Adding mbonadei@ to build_files WATCHLIST. by Mirko Bonadei · 6 years ago
  16. a97c931 Fix a bug where TestAudioDeviceModule crashes if destroyed uninitialized. by Sami Kalliomäki · 6 years ago
  17. aaa483b Revert "Remove deprecated mac capture code." by Kári Helgason · 6 years ago
  18. caca556 Roll chromium_revision 20579735a6..e1ef7d4b6b (568689:568794) by Autoroller · 6 years ago
  19. 88703e8 Let git-hyper-blame ignore format commit. by Yves Gerey · 6 years ago
  20. 056a68d Revert "Enable any address ports by default." by Mirko Bonadei · 6 years ago
  21. 80c0f06 Init GainControlImpl with correct lock. by Alex Loiko · 6 years ago
  22. 5565981 Add functionality to set min/max bitrate per simulcast layer through RtpEncodingParameters. by Åsa Persson · 6 years ago
  23. f88a22c Delete pre_decode_callback. by Niels Möller · 6 years ago
  24. 5297bd2 Fixed crash when PCF is destroyed before PC in ObjC by Yura Yaroshevich · 6 years ago
  25. f04148c Enable any address ports by default. by Qingsi Wang · 6 years ago
  26. 188301c Roll chromium_revision 6e14efc13e..20579735a6 (568572:568689) by Autoroller · 6 years ago
  27. 6109d03 Mark unused/deprecated DTMF methods for removal by Steve Anton · 6 years ago
  28. 1d4a76d Fixing flakiness in PeerConnectionIntegrationTest. by Seth Hampson · 6 years ago
  29. 66cadcc Replace rtc::Optional with absl::optional in pc by Danil Chapovalov · 6 years ago
  30. 751a817 Roll chromium_revision c27ef6f9f6..6e14efc13e (568443:568572) by Autoroller · 6 years ago
  31. a465344 Return SSRC stats with the old stats API when SSRCs are unsignaled. by Taylor Brandstetter · 6 years ago
  32. 0a5fe77 Clean up in module_common_types.h by removing the unused struct RTPAudioHeader. by philipel · 6 years ago
  33. 7e9a619 Add setter method EncodedFrame::SetTimestamp. by Niels Möller · 6 years ago
  34. acef18d Roll chromium_revision 9d565db4c0..c27ef6f9f6 (568343:568443) by Autoroller · 6 years ago
  35. fa2b8d7 Add separate native library for instrumentationtests by Paulina Hensman · 6 years ago
  36. faf2827 Add Parsing/Building generic frame descriptor extension by Danil Chapovalov · 6 years ago
  37. 709c822 Add nisse@ as owner of api/video/ by Niels Möller · 6 years ago
  38. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  39. b602123 Replace rtc::Optional with absl::optional in modules/audio_coding by Danil Chapovalov · 6 years ago
  40. bbfcc70 AEC3: Unittests for MovingAverage by Gustaf Ullberg · 6 years ago
  41. 1998e9d Drop tools/gyp from dependencies by Niels Möller · 6 years ago
  42. f344dbb Cover AecDump calls in APM fuzzer. by Alex Loiko · 6 years ago
  43. 8406c43 AEC3: Average the spectrum of multiple nearend frames in the suppressor. by Gustaf Ullberg · 6 years ago
  44. 5d848f3 Delete picture id and tl0 index from CodecSpecificInfo. by Niels Möller · 6 years ago
  45. db9f7ab Replace rtc::Optional with absl::optional in modules/audio processing by Danil Chapovalov · 6 years ago
  46. c66613d Android: Simlify createOesTextureBuffer() in VideoFrameBufferTest by Magnus Jedvert · 6 years ago
  47. 8a5edb2 Always enable 'delay-agnostic' in APM fuzzer. by Alex Loiko · 6 years ago
  48. 790da37 Fuzz AEC field trial killswitches by Sam Zackrisson · 6 years ago
  49. af998e2 Remove non-API beamformer references by Sam Zackrisson · 6 years ago
  50. 024eeff Roll chromium_revision 9df92afb16..9d565db4c0 (566630:568343) by Autoroller · 6 years ago
  51. aac7dee [desktopCapture Mac]reorder execution order in start/release processing by braveyao · 6 years ago
  52. 15ac521 Removing unused cricket::Port constructor. by Taylor Brandstetter · 6 years ago
  53. 6bbeb08 Extract rtc_base/base64.h and rtc_base/base64.cc into separate target. by Artem Titov · 6 years ago
  54. 6250fdd Delete FakeWebRtcVcmFactory::OnDestroyed method. by Niels Möller · 6 years ago
  55. 9394f6f Stop using the beamformer inside APM by Sam Zackrisson · 6 years ago
  56. 431abd9 Replace rtc::Optional with absl::optional in test and rtc_tools by Danil Chapovalov · 6 years ago
  57. 9bf3158 Pass buffer with size when writing rtp header extension by Danil Chapovalov · 6 years ago
  58. 0040b66 Replace rtc::Optional with absl::optional by Danil Chapovalov · 6 years ago
  59. ae18886 Disable new RTC_CHECK unittest by Jonas Olsson · 6 years ago
  60. 00c7183 Replace rtc::Optional with absl::optional in media, ortc, p2p by Danil Chapovalov · 6 years ago
  61. 5a9ba68 Add base64 webrtc owned third_party dep by Artem Titov · 6 years ago
  62. 5adf07d Make instructions for checkin_chrome_dep a bit clearer. by Patrik Höglund · 6 years ago
  63. ce4829a Adds trial to ignore video pacing for audio packets. by Sebastian Jansson · 6 years ago
  64. f8e5c11 Refactor checks to use a copy of the new logging backend. by Jonas Olsson · 6 years ago
  65. 6a9bd74 Fix a downstream test failure. by Ying Wang · 6 years ago
  66. c235a8d Adds trial to always send padding packets when not sending video. by Sebastian Jansson · 6 years ago
  67. fc50110 Remove stringstreams from modules/video_coding/ by Jonas Olsson · 6 years ago
  68. 5c43150 Makes BBR congestion window more similar to QUIC. by Sebastian Jansson · 6 years ago
  69. fb4d66b Improves buffer time calculation in network control tester. by Sebastian Jansson · 6 years ago
  70. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  71. e61d72b Disables congestion window in pacer when CongestionWindowPushback is enabled. by Ying Wang · 6 years ago
  72. 92b24f0 Delete an unneeded include of pathutils.h. by Niels Möller · 6 years ago
  73. fc9dcb6 Remove wire-up for cancelled experement on VAAPI VP8 encoding by Ilya Nikolaevskiy · 6 years ago
  74. d264df5 Replace rtc::Optional with absl::optional in modules/rtp_rtcp by Danil Chapovalov · 6 years ago
  75. 394b4eb Delete unused methods on rtc::Pathname. by Niels Möller · 6 years ago
  76. 65c61dc Android: Add helper class for generating OpenGL shaders by Magnus Jedvert · 6 years ago
  77. 8643b78 Moved NackModule and VCMPacket to their own targets by Ilya Nikolaevskiy · 6 years ago
  78. 88aee28 Remove support for old test modes in EncodeDecodeTest by Karl Wiberg · 6 years ago
  79. d477129 Remove dead RED code in TestRedFec by Karl Wiberg · 6 years ago
  80. 8fbe4f1 Remove executable insert_packet_with_timing by Karl Wiberg · 6 years ago
  81. 0f173bd Revert "Drop tools/gyp from dependencies." by Artem Titov · 6 years ago
  82. 0a5fdbb Use RTC_HISTOGRAM_ENUMERATION to report SRTP/SRTCP unprotect error. by Zhi Huang · 6 years ago
  83. 9eb3886 Adds field trial parser. by Sebastian Jansson · 6 years ago
  84. 7c32c86 Metal view: Update drawable size when rotating. by Peter Hanspers · 6 years ago
  85. 724a97d Drop tools/gyp from dependencies. by Niels Möller · 6 years ago
  86. a6fc636 Add ivoc@ and saza@ to audio_processing OWNERS by Sam Zackrisson · 6 years ago
  87. 6507054 Android: Add tests for VideoFrame.Buffer.toI420() and cropAndScale() by Magnus Jedvert · 6 years ago
  88. d1f970d Change echo detector to scoped_refptr by Ivo Creusen · 6 years ago
  89. 4e952a3 Remove unused WavFile::FormatAsString method. by Jonas Olsson · 6 years ago
  90. 671cae2 Handle FileRotatingStreams with long file names by Jonas Olsson · 6 years ago
  91. 1b36894 Reland "Refactor the regathering of candidates in P2PTransportChannel." by Qingsi Wang · 6 years ago
  92. 5d16d7f Add a DCHECK for null port in FakePortAllocator. by Qingsi Wang · 6 years ago
  93. 60b6c1d [Unified Plan] Clear RtpSender "SSRC" when the SDP has no send streams by Steve Anton · 6 years ago
  94. f7d7e90 Replace std:remove on vector::erase in streamparams_unittest.cc by Artem Titov · 6 years ago
  95. aeb0a64 AEC3: Increase the range of reported echo path delay metrics by Per Åhgren · 6 years ago
  96. b7700d3 Android: Fix VideoTrack behavior for adding/removing VideoSinks by Magnus Jedvert · 6 years ago
  97. 493c78a Replace all use of rtc::Pathname in generator_unittest.cc. by Niels Möller · 6 years ago
  98. fabb12e Introduce list of fields to put into codec agnostic descriptor by Danil Chapovalov · 6 years ago
  99. 075cb2b AEC3: Changes to how the reverberation decay is applied. by Jesús de Vicente Peña · 6 years ago
  100. 9633cff Remove "webrtc_rtp" traces. by Jonas Olsson · 6 years ago