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gerrit-public.fairphone.software / platform / external / webrtc / 758d94610672bc99bc23e3571871439b4eadd37f
  1. 758d946 Add origin trial ids to non-standard stats members. by Jakob Ivarsson · 6 years ago
  2. edd2054 Minor fixes and refactoring for RtpTransport until the Demux. by Amit Hilbuch · 6 years ago
  3. 342989d Reland "Add winmm.lib as a Windows dep for timeutils." by Noah Richards · 6 years ago
  4. 82b7ff5 Don't store last rendered frame in DefaultVideoQualityAnalyzer by Artem Titov · 6 years ago
  5. ded1e4f Disable flaky call_perf tests for iOS devices by Artem Titarenko · 6 years ago
  6. 4fa9ede Refactor DefaultEncodedImageDataInjector to let EncodedImage own the data. by Niels Möller · 6 years ago
  7. 80cfd81 Move PeerConnectionComponents when creating PeerConnectionDependencies. by Artem Titov · 6 years ago
  8. 276cdfc Rename resolution_of_encoded_image into resolution_of_rendered_frame. by Artem Titov · 6 years ago
  9. 608d801 Use deque instead of list in DefaultVideoQualityAnalyzer. by Artem Titov · 6 years ago
  10. 2809cbb Add 'src/third_party/android_ndk' and '...sdk' dependencies to adapt to changed Android SDK path by Oleksandr Iakovenko · 6 years ago
  11. 5165543 DecodeFrameHistory can now deal with negative picture IDs. by philipel · 6 years ago
  12. c1f3f07 Revert "Add winmm.lib as a Windows dep for timeutils." by Sam Zackrisson · 6 years ago
  13. f0b8dee Qualify cmath functions. by Mirko Bonadei · 6 years ago
  14. 93e2120 Qualify cmath functions. by Mirko Bonadei · 6 years ago
  15. 9e06ce0 Add winmm.lib as a Windows dep for timeutils. by Noah Richards · 6 years ago
  16. db4def9 Update parsing of stun and turn urls for RFC 7064-7065 by Niels Möller · 6 years ago
  17. d360263 Roll chromium_revision b8ead26ca6..4c1c5d8822 (641562:641685) by chromium-webrtc-autoroll · 6 years ago
  18. 7fbfaa4 PeerConnection::SetBitrate now also configures media transport. by Piotr (Peter) Slatala · 6 years ago
  19. ae88f39 Revert "Adding support for enum class in RTC_CHECK and RTC_LOG." by Steve Anton · 6 years ago
  20. 946b968 Add support for target rate constraints by Piotr (Peter) Slatala · 6 years ago
  21. 6b6f537 Adding support for enum class in RTC_CHECK and RTC_LOG. by Amit Hilbuch · 6 years ago
  22. cb8284e Add ownership to fake_media_transport by Piotr (Peter) Slatala · 6 years ago
  23. 37b5662 Remove zero lower bound of estimated inter-arrival time. by Jakob Ivarsson · 6 years ago
  24. 59c8569 Remove spammy log message from RtpSenderVideo::AddRtpHeaderExtensions. by philipel · 6 years ago
  25. 7edc49c Mark neteq_rtpplay as publicly visible. by Mirko Bonadei · 6 years ago
  26. 2e6552d Roll chromium_revision 6abc3675fb..b8ead26ca6 (641307:641562) by chromium-webrtc-autoroll · 6 years ago
  27. 7dbc0eb Makes loss based controller test more robust. by Sebastian Jansson · 6 years ago
  28. 6d83592 Improve handling of packets with unknown ssrc. by Jonas Oreland · 6 years ago
  29. 0611a15 Make the stacktrace unit test more robust by Karl Wiberg · 6 years ago
  30. 2236bb9 Reduce smoke test video resolution. by Artem Titov · 6 years ago
  31. 02ba0ac [build] Port: Use CIPD packages for GN instead of GCS by Oleh Prypin · 6 years ago
  32. df644be webrtc: Remove use_drfuzz. by Nico Weber · 6 years ago
  33. 7583467 Roll chromium_revision cf85bf419e..6abc3675fb (641142:641307) by chromium-webrtc-autoroll · 6 years ago
  34. ba82e00 Add API to schedule environment changing actions during test in PC E2E framework by Artem Titov · 6 years ago
  35. 6cac21d Remove dependency on winsdk_samples. by Mirko Bonadei · 6 years ago
  36. 47dbcab Fuzzing support for RTPDump VP8 and VP9 Streams. by Benjamin Wright · 6 years ago
  37. e07d3b4 Remove crbug.com/904400 workaround. by Mirko Bonadei · 6 years ago
  38. 154d839 Fix misaligned read in StunMessage::Read by Andrew Royes · 6 years ago
  39. 2f5f061 Remove unused variable DefaultTemporalLayers::kKeyframeBuffer. by philipel · 6 years ago
  40. ad31c98 Don't use the Process method of vcm::VideoReceiver by Niels Möller · 6 years ago
  41. 7bf8c7f Add public API for NetworkEmulationManager by Artem Titov · 6 years ago
  42. 69008a8 Avoid div-by-zero in VideoCodecTest stats calculation. by Rasmus Brandt · 6 years ago
  43. 35816cc Revert "Log an error if the RTT is negative" by Magnus Jedvert · 6 years ago
  44. 1e08724 Roll chromium_revision 31e0a71127..cf85bf419e (641033:641142) by chromium-webrtc-autoroll · 6 years ago
  45. 647d5e6 Increase the default maximum jitter buffer size to 200 packets. by Jakob Ivarsson · 6 years ago
  46. dbce090 Qualify cmath functions. by Mirko Bonadei · 6 years ago
  47. bfe4948 Roll chromium_revision 5cef02b5fd..31e0a71127 (640862:641033) by chromium-webrtc-autoroll · 6 years ago
  48. 17b050f Fixes ClangTidy errors in audio/ by Benjamin Wright · 6 years ago
  49. 8965fbc ClangTidy fixes for common_audio/ by Benjamin Wright · 6 years ago
  50. c6fa6d9 ClangTidy fixes for examples/ by Benjamin Wright · 6 years ago
  51. 65cccca Roll chromium_revision b2075e83fd..5cef02b5fd (640732:640862) by chromium-webrtc-autoroll · 6 years ago
  52. b5207b4 Revert "SeqNumUnwrapper::Unwrap now returns int64_t instead of uint64_t." by Philip Eliasson · 6 years ago
  53. 38e6c66 CNAME is missing in simulcast layers. by Amit Hilbuch · 6 years ago
  54. f1c9e21 ClangTidy fixes for logging/ by Benjamin Wright · 6 years ago
  55. 2789766 Roll chromium_revision fc637deb51..b2075e83fd (640618:640732) by chromium-webrtc-autoroll · 6 years ago
  56. 10db597 Support different capture resolutions in new video_loopback. by Kári Tristan Helgason · 6 years ago
  57. 1ddc763 Qualify cmath functions. by Mirko Bonadei · 6 years ago
  58. b0f968a SeqNumUnwrapper::Unwrap now returns int64_t instead of uint64_t. by philipel · 6 years ago
  59. e8efbbd AEC3: Removing unused parameters by Per Åhgren · 6 years ago
  60. ab03638 Let threads opt in to having their stack traces printed by Karl Wiberg · 6 years ago
  61. 9249fbf AEC3: Redesign delay headroom by Gustaf Ullberg · 6 years ago
  62. 41f9f2c ClangTidy fixes for call/ by Benjamin Wright · 6 years ago
  63. 009ab3c Delete EncodedImage::GetBufferPaddingBytes by Niels Möller · 6 years ago
  64. 1f4173e Fix ClangTidy issues in video/ by Benjamin Wright · 6 years ago
  65. a594ef0 Log an error if the RTT is negative by Evan Shrubsole · 6 years ago
  66. d841ea6 Skip return value check for stopped repeating tasks. by Sebastian Jansson · 6 years ago
  67. 5d7e439 Roll chromium_revision acb568e104..fc637deb51 (640514:640618) by chromium-webrtc-autoroll · 6 years ago
  68. 3cc45d4 Add a test that all //api/test headers are compilable. by Harald Alvestrand · 6 years ago
  69. d5e1c37 SSLCertificate basic fuzzer. by Benjamin Wright · 6 years ago
  70. 3aa584f Fixes ClangTidy issues in api/ by Benjamin Wright · 6 years ago
  71. ce66bb4 Adding simulcast examples to the fuzzing corpus. by Amit Hilbuch · 6 years ago
  72. 1295b0d Add basic fuzzing for rtp_header_parser.h/cc. by Benjamin Wright · 6 years ago
  73. ec4cdba Roll chromium_revision a9ac2956aa..acb568e104 (640406:640514) by chromium-webrtc-autoroll · 6 years ago
  74. 7f3687c Integrate parsing of SCTP messages into WebRTC Fuzzers. by Benjamin Wright · 6 years ago
  75. 7a7cf94 Roll chromium_revision c0acb51236..a9ac2956aa (640306:640406) by chromium-webrtc-autoroll · 6 years ago
  76. 45a2cd2 Fixing documentation for CopyOnWriteBuffer. by Amit Hilbuch · 6 years ago
  77. d6c4b80 Add Fuzzing support for ParseRtcpPacketSenderSsrc. by Benjamin Wright · 6 years ago
  78. baf271f DefaultVideoQualityAnalyzer cleanup. by Artem Titov · 6 years ago
  79. 982dc79 Preserve legacy behavior for old OveruseFrameDetector by Erik Språng · 6 years ago
  80. 5ce38ff Making UpdatesTargetRateBasedOnLinkCapacity more robust. by Sebastian Jansson · 6 years ago
  81. 5ad789c Reland "NetEQ RTP Play: Optionally write output audio file" by Alessio Bazzica · 6 years ago
  82. 123f345 Cleanup of scenario test framework. by Sebastian Jansson · 6 years ago
  83. 9a66d5e Add support to audioproc_f to generate a custom call order file. by Ivo Creusen · 6 years ago
  84. f84b95d Rename network_manager -> emulation. by Artem Titov · 6 years ago
  85. 3c589be Reland "Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video." by Rasmus Brandt · 6 years ago
  86. c032109 Improve bitstream dumping logic in VideoReceiveStream by Ilya Nikolaevskiy · 6 years ago
  87. 133b307 Delete method VideoStreamDecoder::UpdateRtt by Niels Möller · 6 years ago
  88. ecc11eb Roll chromium_revision 48038209dc..c0acb51236 (640007:640306) by chromium-webrtc-autoroll · 6 years ago
  89. cc35032 Replace abs with std::abs in audio_coding/neteq/histogram.cc by Piasy · 6 years ago
  90. 2086347 Move creation of rtc::NetworkManager into network emulation layer by Artem Titov · 6 years ago
  91. 9699f09 Add new webrtc_perf_test for lower stream of vp8 simulcast screenshare by Ilya Nikolaevskiy · 6 years ago
  92. 1175ae0 Add log based GoogCC simulation to visualizer. by Sebastian Jansson · 6 years ago
  93. 7ae8d64 Restore VideoCodecInitializer to use only the 1st stream maxFramerate by Ilya Nikolaevskiy · 6 years ago
  94. 77efcd8 Reland "Replacing rtc::Thread with task queue for TestAudioDeviceModule." by Sebastian Jansson · 6 years ago
  95. 793597b Removes TaskQueueBase::Current call in repeating task. by Sebastian Jansson · 6 years ago
  96. cda86dd Removes usages of repeating task without task queue argument. by Sebastian Jansson · 6 years ago
  97. e7a5f7b Modifying MediaChannel to accept CopyOnWriteBuffer by value. by Amit Hilbuch · 6 years ago
  98. dfaea9d Fuzz rtc::StringToNumber. by Benjamin Wright · 6 years ago
  99. 6a5e976 Add generic depacketizer fuzzer to WebRTC. by Benjamin Wright · 6 years ago
  100. ade5cb8 Field trial fuzzer. by Benjamin Wright · 6 years ago