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gerrit-public.fairphone.software / platform / external / webrtc / f0db2e2ef5d286f8ebecabc73c2bc18e18083c6c
  1. f0db2e2 nit: Missing space in build_overrides/build.gni by Elad Alon · 6 years ago
  2. dd886082 AGC2 flags: remove deprecated fields. by Alessio Bazzica · 6 years ago
  3. a06bf85 Add a presubmit check for absl/memory/memory.h inclusion by tzik · 6 years ago
  4. 9514071 Android: Support externally aligned timestamps by Magnus Jedvert · 6 years ago
  5. 2277ac6 Adds OWNERS to rtc_base/experiments. by Sebastian Jansson · 6 years ago
  6. f01d8c8 Add android bindings for PeerConnectionState. by Jonas Olsson · 6 years ago
  7. 586725d Add ios bindings for PeerConnectionState. by Jonas Olsson · 6 years ago
  8. 58376f3 Make member internal::SynchronousMethodCall::e_ a non-pointer. by Niels Möller · 6 years ago
  9. a859d41 Increasing visibility of api/transport build targets. by Mirko Bonadei · 6 years ago
  10. dbb988b Change ReceiveStatistics to implement RtpPacketSinkInterface, part 2. by Niels Möller · 6 years ago
  11. 7af4ac8 Roll chromium_revision 4ffd688e44..92e84c81c1 (608180:608282) by chromium-webrtc-autoroll · 6 years ago
  12. c25d234 Adds OWNERS to api/transport. by Sebastian Jansson · 6 years ago
  13. d575a2d Roll chromium_revision 3d76a59d7d..4ffd688e44 (608069:608180) by chromium-webrtc-autoroll · 6 years ago
  14. 30599b0 Roll chromium_revision fbed28d429..3d76a59d7d (607938:608069) by chromium-webrtc-autoroll · 6 years ago
  15. b1e4775 Exposing rtcp report interval setting in objc api by Jiawei Ou · 6 years ago
  16. 83aa5ac Adding Microsoft Corporation (*@microsoft.com) to WebRTC AUTHORS by James Cadd · 6 years ago
  17. dd9390c Prevent channels being set on stopped transceiver. by Amit Hilbuch · 6 years ago
  18. 1724a80 AEC3: Turn off the specific suppressor mode for stationary render by Per Åhgren · 6 years ago
  19. cc55032 Adding shampson to media/OWNERS. by Seth Hampson · 6 years ago
  20. 2464348 Don't reset RTT Backoff timeout on route change. by Sebastian Jansson · 6 years ago
  21. fdc635d Remove deprecated APIs from RTC event log parser. by Bjorn Terelius · 6 years ago
  22. 3bc696f Android EglRenderer: Replace unicoce character with ascii character by Magnus Jedvert · 6 years ago
  23. 76f9954 Remove the old RTC event log parser. by Bjorn Terelius · 6 years ago
  24. 38578ca Roll chromium_revision db720b4ab9..fbed28d429 (606025:607938) by chromium-webrtc-autoroll · 6 years ago
  25. a038e71 Less strict audio codec tests to accomodate opus switch to SSE. by Yves Gerey · 6 years ago
  26. fb6fd4b Fix lint errors for android manifests. by Yves Gerey · 6 years ago
  27. 6ef89e7 Rectify comment about 'build_with_chromium'. by Mirko Bonadei · 6 years ago
  28. c58c8a5 Adding mbonadei@ to build_overrides/OWNERS. by Mirko Bonadei · 6 years ago
  29. 42b715a Add visibility to ana config proto by Piotr (Peter) Slatala · 6 years ago
  30. 6dbf0e4 Remove all aliases to rtc::Thread by Danil Chapovalov · 6 years ago
  31. 428a160 Remove rtc_event_log2text by Bjorn Terelius · 6 years ago
  32. 95ca6e1 AudioSource allows implementations to return settings by Piotr (Peter) Slatala · 6 years ago
  33. bc4cf89 Run some peer connection end-to-end tests with an empty audio encoder factory by Karl Wiberg · 6 years ago
  34. de8e6e6 Refactor bitrate configuration in CallTest by Niels Möller · 6 years ago
  35. c7e3af1 Remove rtc_event_log2stats. by Bjorn Terelius · 6 years ago
  36. 8544799 Introduce DLOG to video and voiceengine. by Jonas Olsson · 6 years ago
  37. 318da51 Reland "Add support for screen sharing with PipeWire on Wayland" by Tomas Popela · 6 years ago
  38. 1e2542f AGC2: adding level estimation option (RMS or peak-based). by Alessio Bazzica · 6 years ago
  39. 44ca9a3 Allow usage of stringstream under examples/. by Mirko Bonadei · 6 years ago
  40. 105edca Remove some unused RentACodec static methods by Karl Wiberg · 6 years ago
  41. a33c895 AEC3: Corrected erroneous if-statement that always returned true by Per Åhgren · 6 years ago
  42. b739666 Add missing include of unistd.h by Niels Möller · 6 years ago
  43. 90e6745 Delete deprecated class WrappedI420Buffer by Niels Möller · 6 years ago
  44. f4ce0e4 Configs to run slow_tests. by Mirko Bonadei · 6 years ago
  45. 8fb5746 Delete obsolete interface class RtpData by Niels Möller · 6 years ago
  46. fd20171 Adds setup of RTP Extensions in Scenario tests. by Sebastian Jansson · 6 years ago
  47. cb7eddb Add tests for cpu overuse scaling. by Åsa Persson · 6 years ago
  48. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  49. 4aeb35b Explicitly retain self in objc blocks to avoid compiler warning. by Jiawei Ou · 6 years ago
  50. 0c32e33 Allows change of fake encoder max rate in scenarios tests. by Sebastian Jansson · 6 years ago
  51. 985ee68 Add support for screenshare content type in scenario tests. by Sebastian Jansson · 6 years ago
  52. 2b101d2 Simplifies audio priority rate config in scenario tests. by Sebastian Jansson · 6 years ago
  53. aee8380 Remove obsolete comment (WebRtcSessionDescriptionFactory ctor) by Elad Alon · 6 years ago
  54. 6b64c43 Using early acknowledged rate for safe reset in GoogCC. by Sebastian Jansson · 6 years ago
  55. f1cc3a2 In RTP to NTP estimator use linear regression instead of ad hoc filter by Ilya Nikolaevskiy · 6 years ago
  56. c42d624 Event log - Use ToUnsigned() and ToSigned() on timestamp_ms by Elad Alon · 6 years ago
  57. 19084f8 Event logs - encode N channels as N-1 by Elad Alon · 6 years ago
  58. 49c33ce AudioCodingModule: Remove support for creating encoders by Karl Wiberg · 6 years ago
  59. 80c6762 Tweak ChannelReceive interface, to make it closer to ChannelReceiveProxy by Niels Möller · 6 years ago
  60. 140b1d9 Eliminate use of EventWrapper from android audio device tests by Niels Möller · 6 years ago
  61. f4a3f9c Add RtcEvent::timestamp_ms() by Elad Alon · 6 years ago
  62. 89f874e Add offer_extmap_allow_mixed to RTCConfiguration by Johannes Kron · 6 years ago
  63. 5ae3a02 Revert "Run robolectric tests for Android on several Android API versions" by Danil Chapovalov · 6 years ago
  64. 20f60f0 Fuzzer crash in AGC2. by Alex Loiko · 6 years ago
  65. cfe3b6a Remove most of api/ortc/. by Jonas Olsson · 6 years ago
  66. 8584667 Fix overflow for high bitrates in BitrateProber by Johannes Kron · 6 years ago
  67. 09102a0 Revert "Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus."" by Yves Gerey · 6 years ago
  68. 0b1b5c1 Hide RtcEvent members behind accessors by Elad Alon · 6 years ago
  69. eb809f3 Event logs - separate audio_level and voice_activity by Elad Alon · 6 years ago
  70. 466620b Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus." by Yves Gerey · 6 years ago
  71. 56a4b32 Rename fields in rtc_event_log2.proto by Elad Alon · 6 years ago
  72. a2eb0a7 Fix up an outdated comment in peerconnection_integrationtest.cc. by Bjorn Mellem · 6 years ago
  73. 7127f34 Signal Network route change in fake ice. by Piotr (Peter) Slatala · 6 years ago
  74. d95b0a2 Use delta-encoding in new WebRTC event logs by Elad Alon · 6 years ago
  75. 7246720 Clean up root OWNERS. by Patrik Höglund · 6 years ago
  76. e598e6b Run robolectric tests for Android on several Android API versions by Artem Titarenko · 6 years ago
  77. 9973fa8 Pass HdrMetadata between VideoFrame and EncodedImage for VP9 by Johannes Kron · 6 years ago
  78. 6c373cc Add support for audio in latency visualization. by Bjorn Terelius · 6 years ago
  79. d8aa9f9 Fix flaky JsepTransportControllerTests. by Jonas Olsson · 6 years ago
  80. ad1d9f0 Add RTP header extension for HDR metadata by Johannes Kron · 6 years ago
  81. ee45f90 In RTP to NTP estimator do not allow huge jumps in NTP timestamps by Ilya Nikolaevskiy · 6 years ago
  82. 06f6bc9 Reintroduce missing dependencies in libwebrtc.a library. by Yves Gerey · 6 years ago
  83. 175aa2e Implement data channels over media transport. by Bjorn Mellem · 6 years ago
  84. c2ebe21 Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" by Jiawei Ou · 6 years ago
  85. 0393c64 [Win/boringSSL] Add nasm as part of required dependencies. by Yves Gerey · 6 years ago
  86. ada077f Callback changes to media transport interface: by Piotr (Peter) Slatala · 6 years ago
  87. 87e1619 Add owners for media_transport_interface by Piotr (Peter) Slatala · 6 years ago
  88. d3438aa Add ability to specify if rate controller of video encoder is trusted. by Erik Språng · 6 years ago
  89. 6528d8a In Android encoders, cache EncoderInfo in InitEncode. by Erik Språng · 6 years ago
  90. 260770c Delete rtc::Filesystem. Move needed functions to filerotatingstream.cc. by Niels Möller · 6 years ago
  91. b0550bd Eliminate use of EventWrapper from mac audio device by Niels Möller · 6 years ago
  92. c94b22e Add magjed/nisse/sprang/brandtr as api/video_codecs owners by Erik Språng · 6 years ago
  93. c5dd300 Introduce RtpPacket::GetExtension accessor that return result by Danil Chapovalov · 6 years ago
  94. 357f596 Split a separate codecs target off of :video_jni by Jonathan Yu · 6 years ago
  95. 5bb1ed6 Eliminate use of EventWrapper from ios audio device tests by Niels Möller · 6 years ago
  96. a33c7af Tolerate optional chunks in WAV files by Alessio Bazzica · 6 years ago
  97. c496d58 Add flag for fast jitter buffer playout in neteq simulation by Sam Zackrisson · 6 years ago
  98. e6c2c08 MsanUninitialized: restric type check to msan case. by Alessio Bazzica · 6 years ago
  99. c4e9825 Delete classes EventFactory and EventFactoryImpl. by Niels Möller · 6 years ago
  100. 2a74263 Make the bitrate_allocator param optional to prepare for its removal by Oleh Prypin · 6 years ago