Google Git
Sign in
webrtc / src.git / 83aa5ace99a52bbe85055d89ed6837930f8d1f2c
  1. 83aa5ac Adding Microsoft Corporation (*@microsoft.com) to WebRTC AUTHORS by James Cadd · 6 years ago
  2. dd9390c Prevent channels being set on stopped transceiver. by Amit Hilbuch · 6 years ago
  3. 1724a80 AEC3: Turn off the specific suppressor mode for stationary render by Per Åhgren · 6 years ago
  4. cc550320 Adding shampson to media/OWNERS. by Seth Hampson · 6 years ago
  5. 2464348 Don't reset RTT Backoff timeout on route change. by Sebastian Jansson · 6 years ago
  6. fdc635d Remove deprecated APIs from RTC event log parser. by Bjorn Terelius · 6 years ago
  7. 3bc696f Android EglRenderer: Replace unicoce character with ascii character by Magnus Jedvert · 6 years ago
  8. 76f9954 Remove the old RTC event log parser. by Bjorn Terelius · 6 years ago
  9. 38578ca Roll chromium_revision db720b4ab9..fbed28d429 (606025:607938) by chromium-webrtc-autoroll · 6 years ago
  10. a038e71 Less strict audio codec tests to accomodate opus switch to SSE. by Yves Gerey · 6 years ago
  11. fb6fd4b Fix lint errors for android manifests. by Yves Gerey · 6 years ago
  12. 6ef89e7 Rectify comment about 'build_with_chromium'. by Mirko Bonadei · 6 years ago
  13. c58c8a5 Adding mbonadei@ to build_overrides/OWNERS. by Mirko Bonadei · 6 years ago
  14. 42b715a Add visibility to ana config proto by Piotr (Peter) Slatala · 6 years ago
  15. 6dbf0e4 Remove all aliases to rtc::Thread by Danil Chapovalov · 6 years ago
  16. 428a160 Remove rtc_event_log2text by Bjorn Terelius · 6 years ago
  17. 95ca6e1 AudioSource allows implementations to return settings by Piotr (Peter) Slatala · 6 years ago
  18. bc4cf89 Run some peer connection end-to-end tests with an empty audio encoder factory by Karl Wiberg · 6 years ago
  19. de8e6e6 Refactor bitrate configuration in CallTest by Niels Möller · 6 years ago
  20. c7e3af1 Remove rtc_event_log2stats. by Bjorn Terelius · 6 years ago
  21. 8544799 Introduce DLOG to video and voiceengine. by Jonas Olsson · 6 years ago
  22. 318da51 Reland "Add support for screen sharing with PipeWire on Wayland" by Tomas Popela · 6 years ago
  23. 1e2542f AGC2: adding level estimation option (RMS or peak-based). by Alessio Bazzica · 6 years ago
  24. 44ca9a3 Allow usage of stringstream under examples/. by Mirko Bonadei · 6 years ago
  25. 105edca Remove some unused RentACodec static methods by Karl Wiberg · 6 years ago
  26. a33c895 AEC3: Corrected erroneous if-statement that always returned true by Per Åhgren · 6 years ago
  27. b739666 Add missing include of unistd.h by Niels Möller · 6 years ago
  28. 90e6745 Delete deprecated class WrappedI420Buffer by Niels Möller · 6 years ago
  29. f4ce0e4 Configs to run slow_tests. by Mirko Bonadei · 6 years ago
  30. 8fb5746 Delete obsolete interface class RtpData by Niels Möller · 6 years ago
  31. fd20171 Adds setup of RTP Extensions in Scenario tests. by Sebastian Jansson · 6 years ago
  32. cb7eddb Add tests for cpu overuse scaling. by Åsa Persson · 6 years ago
  33. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  34. 4aeb35b Explicitly retain self in objc blocks to avoid compiler warning. by Jiawei Ou · 6 years ago
  35. 0c32e33 Allows change of fake encoder max rate in scenarios tests. by Sebastian Jansson · 6 years ago
  36. 985ee68 Add support for screenshare content type in scenario tests. by Sebastian Jansson · 6 years ago
  37. 2b101d2 Simplifies audio priority rate config in scenario tests. by Sebastian Jansson · 6 years ago
  38. aee8380 Remove obsolete comment (WebRtcSessionDescriptionFactory ctor) by Elad Alon · 6 years ago
  39. 6b64c43 Using early acknowledged rate for safe reset in GoogCC. by Sebastian Jansson · 6 years ago
  40. f1cc3a2 In RTP to NTP estimator use linear regression instead of ad hoc filter by Ilya Nikolaevskiy · 6 years ago
  41. c42d624 Event log - Use ToUnsigned() and ToSigned() on timestamp_ms by Elad Alon · 6 years ago
  42. 19084f8 Event logs - encode N channels as N-1 by Elad Alon · 6 years ago
  43. 49c33ce AudioCodingModule: Remove support for creating encoders by Karl Wiberg · 6 years ago
  44. 80c6762 Tweak ChannelReceive interface, to make it closer to ChannelReceiveProxy by Niels Möller · 6 years ago
  45. 140b1d9 Eliminate use of EventWrapper from android audio device tests by Niels Möller · 6 years ago
  46. f4a3f9c Add RtcEvent::timestamp_ms() by Elad Alon · 6 years ago
  47. 89f874e Add offer_extmap_allow_mixed to RTCConfiguration by Johannes Kron · 6 years ago
  48. 5ae3a02 Revert "Run robolectric tests for Android on several Android API versions" by Danil Chapovalov · 6 years ago
  49. 20f60f0 Fuzzer crash in AGC2. by Alex Loiko · 6 years ago
  50. cfe3b6a Remove most of api/ortc/. by Jonas Olsson · 6 years ago
  51. 8584667 Fix overflow for high bitrates in BitrateProber by Johannes Kron · 6 years ago
  52. 09102a0 Revert "Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus."" by Yves Gerey · 6 years ago
  53. 0b1b5c1 Hide RtcEvent members behind accessors by Elad Alon · 6 years ago
  54. eb809f30d Event logs - separate audio_level and voice_activity by Elad Alon · 6 years ago
  55. 466620b Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus." by Yves Gerey · 6 years ago
  56. 56a4b32 Rename fields in rtc_event_log2.proto by Elad Alon · 6 years ago
  57. a2eb0a7 Fix up an outdated comment in peerconnection_integrationtest.cc. by Bjorn Mellem · 6 years ago
  58. 7127f34 Signal Network route change in fake ice. by Piotr (Peter) Slatala · 6 years ago
  59. d95b0a2 Use delta-encoding in new WebRTC event logs by Elad Alon · 6 years ago
  60. 7246720 Clean up root OWNERS. by Patrik Höglund · 6 years ago
  61. e598e6b Run robolectric tests for Android on several Android API versions by Artem Titarenko · 6 years ago
  62. 9973fa8 Pass HdrMetadata between VideoFrame and EncodedImage for VP9 by Johannes Kron · 6 years ago
  63. 6c373cc Add support for audio in latency visualization. by Bjorn Terelius · 6 years ago
  64. d8aa9f9 Fix flaky JsepTransportControllerTests. by Jonas Olsson · 6 years ago
  65. ad1d9f0 Add RTP header extension for HDR metadata by Johannes Kron · 6 years ago
  66. ee45f90 In RTP to NTP estimator do not allow huge jumps in NTP timestamps by Ilya Nikolaevskiy · 6 years ago
  67. 06f6bc9 Reintroduce missing dependencies in libwebrtc.a library. by Yves Gerey · 6 years ago
  68. 175aa2e Implement data channels over media transport. by Bjorn Mellem · 6 years ago
  69. c2ebe21 Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" by Jiawei Ou · 6 years ago
  70. 0393c64 [Win/boringSSL] Add nasm as part of required dependencies. by Yves Gerey · 6 years ago
  71. ada077f Callback changes to media transport interface: by Piotr (Peter) Slatala · 6 years ago
  72. 87e1619 Add owners for media_transport_interface by Piotr (Peter) Slatala · 6 years ago
  73. d3438aa Add ability to specify if rate controller of video encoder is trusted. by Erik Språng · 6 years ago
  74. 6528d8a In Android encoders, cache EncoderInfo in InitEncode. by Erik Språng · 6 years ago
  75. 260770c Delete rtc::Filesystem. Move needed functions to filerotatingstream.cc. by Niels Möller · 6 years ago
  76. b0550bd Eliminate use of EventWrapper from mac audio device by Niels Möller · 6 years ago
  77. c94b22e Add magjed/nisse/sprang/brandtr as api/video_codecs owners by Erik Språng · 6 years ago
  78. c5dd300 Introduce RtpPacket::GetExtension accessor that return result by Danil Chapovalov · 6 years ago
  79. 357f596 Split a separate codecs target off of :video_jni by Jonathan Yu · 6 years ago
  80. 5bb1ed6 Eliminate use of EventWrapper from ios audio device tests by Niels Möller · 6 years ago
  81. a33c7af Tolerate optional chunks in WAV files by Alessio Bazzica · 6 years ago
  82. c496d58 Add flag for fast jitter buffer playout in neteq simulation by Sam Zackrisson · 6 years ago
  83. e6c2c08 MsanUninitialized: restric type check to msan case. by Alessio Bazzica · 6 years ago
  84. c4e9825 Delete classes EventFactory and EventFactoryImpl. by Niels Möller · 6 years ago
  85. 2a74263 Make the bitrate_allocator param optional to prepare for its removal by Oleh Prypin · 6 years ago
  86. cd2e105 Reenable test RampUpTest.AudioTransportSequenceNumber by Niels Möller · 6 years ago
  87. 694ed17 Add a style rule about not using const optional<T>& arguments by Karl Wiberg · 6 years ago
  88. f0e7440 Add missing conditional defines to neteq test and tools targets by Sam Zackrisson · 6 years ago
  89. 689983f Deprecate EventFactory and delete all usage. by Niels Möller · 6 years ago
  90. 54b4924 Update H264 encoder to use GetEncoderInfo by Erik Språng · 6 years ago
  91. 1060870 Update LibVpxVp8Encoder to use GetEncoderInfo by Erik Språng · 6 years ago
  92. 727d164 Update VP9 encoder to use GetEncoderInfo by Erik Språng · 6 years ago
  93. 5473a45 Remove multiple RTX codec entries in GetRtpReceiver/SenderCapabilities by Florent Castelli · 6 years ago
  94. 75de46a Update SimulcastEncoderAdapter merging of EncoderInfo by Erik Språng · 6 years ago
  95. e6a2d94 Clear FrameBuffer if there were no frames received for 10 minutes by Ilya Nikolaevskiy · 6 years ago
  96. b768e88 Reland "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  97. bdc6c40 Add field trial for target bitrate RTCP XR message. by Rasmus Brandt · 6 years ago
  98. d565918 Delete NullEventFactory by Niels Möller · 6 years ago
  99. e769ed9 Roll chromium_revision 38dcb5ed01..db720b4ab9 (605924:606025) by chromium-webrtc-autoroll · 6 years ago
  100. 50f60cb Rename software codec classes and move them into api/ by Jonathan Yu · 6 years ago