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gerrit-public.fairphone.software / platform / external / webrtc / 7940da0f2ef535ef2684606d816c763c2df46981
  1. 7940da0 Integration of media_transport in JSepTransportController by Anton Sukhanov · 6 years ago
  2. 6cc9cca Don't reset streams for the FrameEncryptor / FrameDecryptor unless they changed. by Benjamin Wright · 6 years ago
  3. da67c16 Roll chromium_revision 8a25f94ac2..0d09089dd5 (598237:598349) by chromium-webrtc-autoroll · 6 years ago
  4. ca27091 Remove rtc_base:rtc_base_approved_generic. by Mirko Bonadei · 6 years ago
  5. ede8796 Print per-frame VMAF score instead of average. by Paulina Hensman · 6 years ago
  6. b3b0179 Fix backwards logic in rtc::Buffer::OnMovedFrom() by Karl Wiberg · 6 years ago
  7. 0213786 Add certificate gen/set functionality to bring Android closer to JS API by Michael Iedema · 6 years ago
  8. dcc0238 Don't increment timestamp on drop/reencode in LibvpxVp8Encoder. by Erik Språng · 6 years ago
  9. 5526e45 vp9: change x-google-profile-id to profile-id by Philipp Hancke · 6 years ago
  10. 028248c Add `rtc_enable_symbol_export` to incrementally create a WebRTC component. by Mirko Bonadei · 6 years ago
  11. b686396 Makes AudioSendStream signal that it's part of allocation. by Sebastian Jansson · 6 years ago
  12. 99a70a2 Remove rtc_base_approved_objc and introduce rtc_base:logging_mac. by Mirko Bonadei · 6 years ago
  13. edc49c1 [Cleanup] Remove unused swap function. by Yves Gerey · 6 years ago
  14. a4c8514 Add JSON parsing and corresponding ToString to EchoCanceller3Config by Sam Zackrisson · 6 years ago
  15. 2558c4e Remove ortc folder. by Mirko Bonadei · 6 years ago
  16. 88b68ac Create field trial for setting a minimum value for Opus encoder packet loss rate by Jakob Ivarsson · 6 years ago
  17. f08dd9d Disable flaky tests on mac perf bot by Ilya Nikolaevskiy · 6 years ago
  18. 1bca65b Makes RtpSender indicate allocation and feedback status on packets. by Sebastian Jansson · 6 years ago
  19. 81125f0 Implement (mostly) standards-compliant RTCIceTransportState. by Jonas Olsson · 6 years ago
  20. 5f35e96 Roll chromium_revision 476ae6d661..8a25f94ac2 (598136:598237) by chromium-webrtc-autoroll · 6 years ago
  21. c87b8c1 Moves GoogCC factory to API. by Sebastian Jansson · 6 years ago
  22. 0d8c100 AEC3: Decrease the suppression during the echo-only case by Per Åhgren · 6 years ago
  23. 463c764 Roll chromium_revision cfe6e706d0..476ae6d661 (598018:598136) by chromium-webrtc-autoroll · 6 years ago
  24. aabf204 Remove container typedefs from RelayServer by Steve Anton · 6 years ago
  25. 11358fe Use unique_ptr in port_unittest by Steve Anton · 6 years ago
  26. 13d392d AEC3: Utilize dominant nearend functionality to increase transparency by Per Åhgren · 6 years ago
  27. 3a3f027 Roll chromium_revision 0cf8926390..cfe6e706d0 (597915:598018) by chromium-webrtc-autoroll · 6 years ago
  28. 0378997 Adds flags indicating presence in allocation and feedback per packet. by Sebastian Jansson · 6 years ago
  29. 30e2d6e Moves locking outside function in RtpSender. by Sebastian Jansson · 6 years ago
  30. 789f459 Adds fields for unacknowledged data to transport feedback. by Sebastian Jansson · 6 years ago
  31. 20a49f3 Don't try to use CN if voice codec isn't mono by Karl Wiberg · 6 years ago
  32. 5fcc4de Roll chromium_revision f362b3e857..0cf8926390 (597811:597915) by chromium-webrtc-autoroll · 6 years ago
  33. 759f959 Refactor tests with ConfigurableFrameSizeEncoder by Niels Möller · 6 years ago
  34. 040f87f AEC3: Allow a more stable filter during double-talk by Gustaf Ullberg · 6 years ago
  35. 7730193 Remove SetExecutablePath, simplify ResourcePath by Patrik Höglund · 6 years ago
  36. 7004571 AEC3: Decrease the modelling of the reverb by Per Åhgren · 6 years ago
  37. d76a0fc Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport by erikvarga@webrtc.org · 6 years ago
  38. b674cd1 Enable multithreading in libvpx VP9 decoder. by Sergey Silkin · 6 years ago
  39. d0bc462 Check if __IPHONE_OS_VERSION_MAX_ALLOWED is defined before reference by Joel Sutherland · 6 years ago
  40. 0414040 Fix race condition for SupportsFlexfecWithMultithreadedH264/0 test. by Yves Gerey · 6 years ago
  41. bf47198 Roll chromium_revision ba2e073e2c..f362b3e857 (597606:597811) by chromium-webrtc-autoroll · 6 years ago
  42. 4ff7214 Using TaskQueue for congestion controller by default. by Sebastian Jansson · 6 years ago
  43. 4b14416 Roll chromium_revision 0cdd2e3eab..ba2e073e2c (597498:597606) by chromium-webrtc-autoroll · 6 years ago
  44. e0c2e97 Pass MediaTransportFactory to PeerConnectionFactory. by Piotr (Peter) Slatala · 6 years ago
  45. 1e05486 Added the new generic descriptor extension to WebRtcVideoEngine::GetCapabilities. by philipel · 6 years ago
  46. ab09039 Add comment that xcode version needs to be updated in two places by Oleh Prypin · 6 years ago
  47. 16fe3f2 Revert "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  48. 99eea42 Reland "Reland "Export symbols needed by the Chromium component build (part 1)."" by Mirko Bonadei · 6 years ago
  49. 2e068e8 Adds RTT based backoff trial to SendSideBandwidthEstimation. by Sebastian Jansson · 6 years ago
  50. d2fb1bf Generate module.modulemap file when building Mac Framework by Joel Sutherland · 6 years ago
  51. e6708f3 Notify a rotation about autoroll CLs by Oleh Prypin · 6 years ago
  52. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
  53. 3a74239 Fix compilation issues on media_transport_interface.h by Niels Möller · 6 years ago
  54. 788c51c Pass HeaderExtensionMap by reference in rtc_event_log2rtp_dump. by Bjorn Terelius · 6 years ago
  55. b6a8942 Fix race condition for GetContributingSources test. by Yves Gerey · 6 years ago
  56. 666fb32 Rename DefaultNetworkSimulationConfig into BuiltInNetworkBehaviorConfig. by Artem Titov · 6 years ago
  57. 6a8327f Roll chromium_revision ccb83d4a55..0cdd2e3eab (597330:597498) by chromium-webrtc-autoroll · 6 years ago
  58. 7c1744d Reland "Reland "Using units in SendSideBandwidthEstimation."" by Sebastian Jansson · 6 years ago
  59. 841c912 Changed FakeVp8Encoder to write dimensions in payload. by Per Kjellander · 6 years ago
  60. a4de9c8 Revert "Reland "Using units in SendSideBandwidthEstimation."" by Sebastian Jansson · 6 years ago
  61. e2cb26c Reland "Using units in SendSideBandwidthEstimation." by Sebastian Jansson · 6 years ago
  62. 917e596 Revert "Using units in SendSideBandwidthEstimation." by Oleh Prypin · 6 years ago
  63. 2e00abc Reland "[cleanup] Remove useless includes." by Yves Gerey · 6 years ago
  64. 4dc66c5 Move EncodedImage class to api/video/ by Niels Möller · 6 years ago
  65. 38537ed Fix visibility of api/units build targets. by Mirko Bonadei · 6 years ago
  66. 343f414 Allows copy and assignment of field trial parameters. by Sebastian Jansson · 6 years ago
  67. e53341c Roll chromium_revision 7099444bc9..ccb83d4a55 (597172:597330) by chromium-webrtc-autoroll · 6 years ago
  68. 35b5e5f Using units in SendSideBandwidthEstimation. by Sebastian Jansson · 6 years ago
  69. 9f80b97 Fix fuzzer build failures on Windows by Jonathan Metzman · 6 years ago
  70. 4d6f605 Roll chromium_revision d62b62d830..7099444bc9 (597059:597172) by chromium-webrtc-autoroll · 6 years ago
  71. ef8a3eb Include NTP value in playout path. by Niklas Enbom · 6 years ago
  72. a23dc78 Removes initial window field trial. by Sebastian Jansson · 6 years ago
  73. a6471eb Reland "Tidy up and increase exception handling in compare_videos" by Paulina Hensman · 6 years ago
  74. 6c19dec Adds Clamping functions for DataRate. by Sebastian Jansson · 6 years ago
  75. b88fe02 Removes logging spam from congestion window. by Sebastian Jansson · 6 years ago
  76. f2637a8 Reland of 'Bug in histogram metric reporting.' by Alex Loiko · 6 years ago
  77. e28dedf Remove old data files. by Rasmus Brandt · 6 years ago
  78. 64be7fa Move FecController to RtpVideoSender. by Stefan Holmer · 6 years ago
  79. 8e87852 Remove old video_bitrate_allocator.h by Rasmus Brandt · 6 years ago
  80. 8434aeb Use Chromium's code for locating the src dir. by Patrik Höglund · 6 years ago
  81. 0a5792e Add UMA metric and logging of frames dropped in the render queue. by Stefan Holmer · 6 years ago
  82. 96a0f61 Revert "[cleanup] Remove useless includes." by Oleh Prypin · 6 years ago
  83. 1cd7391 Turning off a stream should results in target bitrate 0 signal by Erik Språng · 6 years ago
  84. be8b534 [cleanup] Remove useless includes. by Yves Gerey · 6 years ago
  85. a113450 Roll chromium_revision 618ddbcb7f..d62b62d830 (596951:597059) by chromium-webrtc-autoroll · 6 years ago
  86. 8ea1e9d Switch webrtc from deprecated usages of NetworkSimulationInterface by Artem Titov · 6 years ago
  87. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
  88. 6c966ea Remove @SuppressLint(NewApi) and guard @TargetApi methods by Paulina Hensman · 6 years ago
  89. 97c65b7 Make modules/audio_mixer:audio_mixer_impl publicly visible. by Mirko Bonadei · 6 years ago
  90. 44b384d Delete support for VoIP metrics (RFC 3611 4.7) by Niels Möller · 6 years ago
  91. 4bb1e4a Lower gain parameters for AGC2. by Alex Loiko · 6 years ago
  92. 5fbc0e0 Hide libvpx vp8 encoder behind an interface and add mock for testing. by Erik Språng · 6 years ago
  93. 78cdde3 Add support for sending RTP two-byte header extensions. by Johannes Kron · 6 years ago
  94. 7880be1 Don't include <memory.h> in aligned_malloc.cc. by Mirko Bonadei · 6 years ago
  95. 0a74e09 Roll chromium_revision 0af97cea37..618ddbcb7f (596847:596951) by chromium-webrtc-autoroll · 6 years ago
  96. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  97. aa43b7b Roll chromium_revision c5c13a1e38..0af97cea37 (596716:596847) by chromium-webrtc-autoroll · 6 years ago
  98. 264079a Roll chromium_revision 70554c2519..c5c13a1e38 (596607:596716) by chromium-webrtc-autoroll · 6 years ago
  99. f638bbc Set the generic_descriptor flag in the parameterized fullstack tests to actually use the generic descriptor. by philipel · 6 years ago
  100. 8782a58 Send rtcp target bitrate immediately on new bitrate allocation structure by Erik Språng · 6 years ago