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gerrit-public.fairphone.software / platform / external / webrtc / 5297bd21b1f81f35af3df8173fea26c5d8952f87
  1. 5297bd2 Fixed crash when PCF is destroyed before PC in ObjC by Yura Yaroshevich · 6 years ago
  2. f04148c Enable any address ports by default. by Qingsi Wang · 6 years ago
  3. 188301c Roll chromium_revision 6e14efc13e..20579735a6 (568572:568689) by Autoroller · 6 years ago
  4. 6109d03 Mark unused/deprecated DTMF methods for removal by Steve Anton · 6 years ago
  5. 1d4a76d Fixing flakiness in PeerConnectionIntegrationTest. by Seth Hampson · 6 years ago
  6. 66cadcc Replace rtc::Optional with absl::optional in pc by Danil Chapovalov · 6 years ago
  7. 751a817 Roll chromium_revision c27ef6f9f6..6e14efc13e (568443:568572) by Autoroller · 6 years ago
  8. a465344 Return SSRC stats with the old stats API when SSRCs are unsignaled. by Taylor Brandstetter · 6 years ago
  9. 0a5fe77 Clean up in module_common_types.h by removing the unused struct RTPAudioHeader. by philipel · 6 years ago
  10. 7e9a619 Add setter method EncodedFrame::SetTimestamp. by Niels Möller · 6 years ago
  11. acef18d Roll chromium_revision 9d565db4c0..c27ef6f9f6 (568343:568443) by Autoroller · 6 years ago
  12. fa2b8d7 Add separate native library for instrumentationtests by Paulina Hensman · 6 years ago
  13. faf2827 Add Parsing/Building generic frame descriptor extension by Danil Chapovalov · 6 years ago
  14. 709c822 Add nisse@ as owner of api/video/ by Niels Möller · 6 years ago
  15. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  16. b602123 Replace rtc::Optional with absl::optional in modules/audio_coding by Danil Chapovalov · 6 years ago
  17. bbfcc70 AEC3: Unittests for MovingAverage by Gustaf Ullberg · 6 years ago
  18. 1998e9d Drop tools/gyp from dependencies by Niels Möller · 6 years ago
  19. f344dbb Cover AecDump calls in APM fuzzer. by Alex Loiko · 6 years ago
  20. 8406c43 AEC3: Average the spectrum of multiple nearend frames in the suppressor. by Gustaf Ullberg · 6 years ago
  21. 5d848f3 Delete picture id and tl0 index from CodecSpecificInfo. by Niels Möller · 6 years ago
  22. db9f7ab Replace rtc::Optional with absl::optional in modules/audio processing by Danil Chapovalov · 6 years ago
  23. c66613d Android: Simlify createOesTextureBuffer() in VideoFrameBufferTest by Magnus Jedvert · 6 years ago
  24. 8a5edb2 Always enable 'delay-agnostic' in APM fuzzer. by Alex Loiko · 6 years ago
  25. 790da37 Fuzz AEC field trial killswitches by Sam Zackrisson · 6 years ago
  26. af998e2 Remove non-API beamformer references by Sam Zackrisson · 6 years ago
  27. 024eeff Roll chromium_revision 9df92afb16..9d565db4c0 (566630:568343) by Autoroller · 6 years ago
  28. aac7dee [desktopCapture Mac]reorder execution order in start/release processing by braveyao · 6 years ago
  29. 15ac521 Removing unused cricket::Port constructor. by Taylor Brandstetter · 6 years ago
  30. 6bbeb08 Extract rtc_base/base64.h and rtc_base/base64.cc into separate target. by Artem Titov · 6 years ago
  31. 6250fdd Delete FakeWebRtcVcmFactory::OnDestroyed method. by Niels Möller · 6 years ago
  32. 9394f6f Stop using the beamformer inside APM by Sam Zackrisson · 6 years ago
  33. 431abd9 Replace rtc::Optional with absl::optional in test and rtc_tools by Danil Chapovalov · 6 years ago
  34. 9bf3158 Pass buffer with size when writing rtp header extension by Danil Chapovalov · 6 years ago
  35. 0040b66 Replace rtc::Optional with absl::optional by Danil Chapovalov · 6 years ago
  36. ae18886 Disable new RTC_CHECK unittest by Jonas Olsson · 6 years ago
  37. 00c7183 Replace rtc::Optional with absl::optional in media, ortc, p2p by Danil Chapovalov · 6 years ago
  38. 5a9ba68 Add base64 webrtc owned third_party dep by Artem Titov · 6 years ago
  39. 5adf07d Make instructions for checkin_chrome_dep a bit clearer. by Patrik Höglund · 6 years ago
  40. ce4829a Adds trial to ignore video pacing for audio packets. by Sebastian Jansson · 6 years ago
  41. f8e5c11 Refactor checks to use a copy of the new logging backend. by Jonas Olsson · 6 years ago
  42. 6a9bd74 Fix a downstream test failure. by Ying Wang · 6 years ago
  43. c235a8d Adds trial to always send padding packets when not sending video. by Sebastian Jansson · 6 years ago
  44. fc50110 Remove stringstreams from modules/video_coding/ by Jonas Olsson · 6 years ago
  45. 5c43150 Makes BBR congestion window more similar to QUIC. by Sebastian Jansson · 6 years ago
  46. fb4d66b Improves buffer time calculation in network control tester. by Sebastian Jansson · 6 years ago
  47. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  48. e61d72b Disables congestion window in pacer when CongestionWindowPushback is enabled. by Ying Wang · 6 years ago
  49. 92b24f0 Delete an unneeded include of pathutils.h. by Niels Möller · 6 years ago
  50. fc9dcb6 Remove wire-up for cancelled experement on VAAPI VP8 encoding by Ilya Nikolaevskiy · 6 years ago
  51. d264df5 Replace rtc::Optional with absl::optional in modules/rtp_rtcp by Danil Chapovalov · 6 years ago
  52. 394b4eb Delete unused methods on rtc::Pathname. by Niels Möller · 6 years ago
  53. 65c61dc Android: Add helper class for generating OpenGL shaders by Magnus Jedvert · 6 years ago
  54. 8643b78 Moved NackModule and VCMPacket to their own targets by Ilya Nikolaevskiy · 6 years ago
  55. 88aee28 Remove support for old test modes in EncodeDecodeTest by Karl Wiberg · 6 years ago
  56. d477129 Remove dead RED code in TestRedFec by Karl Wiberg · 6 years ago
  57. 8fbe4f1 Remove executable insert_packet_with_timing by Karl Wiberg · 6 years ago
  58. 0f173bd Revert "Drop tools/gyp from dependencies." by Artem Titov · 6 years ago
  59. 0a5fdbb Use RTC_HISTOGRAM_ENUMERATION to report SRTP/SRTCP unprotect error. by Zhi Huang · 6 years ago
  60. 9eb3886 Adds field trial parser. by Sebastian Jansson · 6 years ago
  61. 7c32c86 Metal view: Update drawable size when rotating. by Peter Hanspers · 6 years ago
  62. 724a97d Drop tools/gyp from dependencies. by Niels Möller · 6 years ago
  63. a6fc636 Add ivoc@ and saza@ to audio_processing OWNERS by Sam Zackrisson · 6 years ago
  64. 6507054 Android: Add tests for VideoFrame.Buffer.toI420() and cropAndScale() by Magnus Jedvert · 6 years ago
  65. d1f970d Change echo detector to scoped_refptr by Ivo Creusen · 6 years ago
  66. 4e952a3 Remove unused WavFile::FormatAsString method. by Jonas Olsson · 6 years ago
  67. 671cae2 Handle FileRotatingStreams with long file names by Jonas Olsson · 6 years ago
  68. 1b36894 Reland "Refactor the regathering of candidates in P2PTransportChannel." by Qingsi Wang · 6 years ago
  69. 5d16d7f Add a DCHECK for null port in FakePortAllocator. by Qingsi Wang · 6 years ago
  70. 60b6c1d [Unified Plan] Clear RtpSender "SSRC" when the SDP has no send streams by Steve Anton · 6 years ago
  71. f7d7e90 Replace std:remove on vector::erase in streamparams_unittest.cc by Artem Titov · 6 years ago
  72. aeb0a64 AEC3: Increase the range of reported echo path delay metrics by Per Åhgren · 6 years ago
  73. b7700d3 Android: Fix VideoTrack behavior for adding/removing VideoSinks by Magnus Jedvert · 6 years ago
  74. 493c78a Replace all use of rtc::Pathname in generator_unittest.cc. by Niels Möller · 6 years ago
  75. fabb12e Introduce list of fields to put into codec agnostic descriptor by Danil Chapovalov · 6 years ago
  76. 075cb2b AEC3: Changes to how the reverberation decay is applied. by Jesús de Vicente Peña · 6 years ago
  77. 9633cff Remove "webrtc_rtp" traces. by Jonas Olsson · 6 years ago
  78. 29c36b2 Add ow2_asm license by Yura Yaroshevich · 6 years ago
  79. 789221f Adding WebRTC-Audio-ForceNoTWCC field trial by Alex Narest · 6 years ago
  80. e3cf3d0 Use enum class for VideoCodecMode and VideoCodecComplexity. by Niels Möller · 6 years ago
  81. 037b37a Add implementation of EncodedFrame::Timestamp. by Niels Möller · 6 years ago
  82. e4a17c5 Moved timing related logic into its own function in webrtc::PayloadRouter. by philipel · 6 years ago
  83. 1f4d7a2 Revert "Refactor the regathering of candidates in P2PTransportChannel." by Qingsi Wang · 6 years ago
  84. 241d0c1 Remove ContinualGatheringPolicy::GATHER_CONTINUALLY_AND_RECOVER. by Qingsi Wang · 6 years ago
  85. aed7164 Updated PeerConnection integration test to fix race condition. by Seth Hampson · 6 years ago
  86. 2cf61e3 Roll chromium_revision ef538e3112..9df92afb16 (566490:566630) by Autoroller · 6 years ago
  87. 14f8aba Refactor the regathering of candidates in P2PTransportChannel. by Qingsi Wang · 6 years ago
  88. b57e169 Add a flag to actively reset the SRTP parameters by Zhi Huang · 6 years ago
  89. 9dce71b Reland "Use absl::optional instead or rtc::Optional" by Danil Chapovalov · 6 years ago
  90. 57dc9e3 Roll chromium_revision ffaf1e2ba6..ef538e3112 (565764:566490) by Autoroller · 6 years ago
  91. abe301f Add HeaderExtensions to RtpParameters by Florent Castelli · 6 years ago
  92. 867e510 Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled. by Alex Narest · 6 years ago
  93. 910540d Explicitly setting use_lld=false on MSVC bots. by Mirko Bonadei · 6 years ago
  94. 7af087a Metal renderer does not handle i420 frames correctly. by Peter Hanspers · 6 years ago
  95. dd3e0ab Make rtc_software_fallback_wrappers target visible. by Anders Carlsson · 6 years ago
  96. cf15eb5 Release ADM after passing it to PCF in AppRTC by Paulina Hensman · 6 years ago
  97. d294c85 LogMessage::UpdateMinLogSeverity: Don't ignore all but the last stream by Karl Wiberg · 6 years ago
  98. 798b282 Don't update internal state of the FrameBuffer2 when an undecodable frame is inserted. by philipel · 6 years ago
  99. dadaaee Remove stringstreams from p2p/ by Jonas Olsson · 6 years ago
  100. 39da65b remove unused UNSHIPPED trace macros by Jonas Olsson · 6 years ago