Google Git
Sign in
webrtc / src.git / e71b55fb276a95589e1729930a45c3bf869871ae
  1. e71b55f build: merge media_constants and engine_constants by Philipp Hancke · 4 years, 2 months ago
  2. f109193 Remove VideoLayerFrameId::spatial_layer, use EncodedImage::SpatialIndex instead. by philipel · 4 years, 2 months ago
  3. cae4656 Split peer_connection_integrationtest.cc into pieces by Harald Alvestrand · 4 years, 2 months ago
  4. aee2c6a In android video encoder wrapper fill codec-agnostic frame dependencies by Danil Chapovalov · 4 years, 2 months ago
  5. 590b1ba Add lock annotations to DxgiDuplicatorController by Niels Möller · 4 years, 2 months ago
  6. 8f8bd4d Roll chromium_revision 2f598fb8a8..b5988d40c8 (853215:853388) by chromium-webrtc-autoroll · 4 years, 2 months ago
  7. 80d661c Use monotonic clock for legacy stats caching. by Taylor Brandstetter · 4 years, 2 months ago
  8. 95c8f0d Roll chromium_revision 11a94c709b..2f598fb8a8 (853086:853215) by chromium-webrtc-autoroll · 4 years, 2 months ago
  9. 5a58595 Update AGC2 tuning by Alessio Bazzica · 4 years, 2 months ago
  10. 45d2234 Test and fix unscalable video structure. by Danil Chapovalov · 4 years, 2 months ago
  11. dea374a Deliver packet info to source listeners when audio playout is disabled. by Ranveer Aggarwal · 4 years, 2 months ago
  12. 0710b40 Migrate perf tests result writing API to absl::string_view by Artem Titov · 4 years, 2 months ago
  13. 7666d89 Roll chromium_revision 897c000738..11a94c709b (852985:853086) by chromium-webrtc-autoroll · 4 years, 2 months ago
  14. ec23d6d Remove blanket disabling of TSAN for peer_connection_integrationtests by Harald Alvestrand · 4 years, 2 months ago
  15. cc47437 [Battery]: Delay start of CallStats. by Etienne Pierre-Doray · 4 years, 2 months ago
  16. 4eb47f2 Delete unused class SignalThread by Niels Möller · 4 years, 2 months ago
  17. 8cb9706 AddRemoteCandidate on the network thread by Tomas Gunnarsson · 4 years, 2 months ago
  18. ff0e01f Implement audio_interruption metric for kCodecPlc by Pablo Barrera González · 4 years, 2 months ago
  19. 983627c Update VirtualSocketServer to use more std::unique_ptr by Niels Möller · 4 years, 2 months ago
  20. ed8abad Convert third_party/android_deps:androidx refs to third_party/androidx by Peter Kotwicz · 4 years, 2 months ago
  21. de7ee3a Reland "AV1: change update freq and disable denoiser explicitly." by Jerome Jiang · 4 years, 2 months ago
  22. c3257d0 Reland "Remove thread hops from events provided by JsepTransportController." by Tommi · 4 years, 2 months ago
  23. f28d7e8 Roll chromium_revision 369c665c3d..897c000738 (852709:852985) by chromium-webrtc-autoroll · 4 years, 2 months ago
  24. 8bf61a3 Add tests for datachannel continuity under network outages. by Harald Alvestrand · 4 years, 2 months ago
  25. c5dbe94 Roll chromium_revision fa2088e13f..369c665c3d (852593:852709) by chromium-webrtc-autoroll · 4 years, 2 months ago
  26. 4b1c72c Reland "Replace field trials with WebRtcKeyValueConfig in LossBasedBandwidthEstimation" by Per Kjellander · 4 years, 2 months ago
  27. d15a575 Use SequenceChecker from public API by Artem Titov · 4 years, 2 months ago
  28. 92eebef Reland "Fix unsynchronized access to mid_to_transport_ in JsepTransportController" by Tomas Gunnarsson · 4 years, 2 months ago
  29. ba69442 Configure threads with their own warning deadlines. by Harald Alvestrand · 4 years, 2 months ago
  30. 9e1f08a Roll chromium_revision 1e9271a080..fa2088e13f (852360:852593) by chromium-webrtc-autoroll · 4 years, 2 months ago
  31. 06159aa Remove deprecated thread checker by Artem Titov · 4 years, 2 months ago
  32. e44f24e Handle longer AudioSendStream::Config strings by Evan Shrubsole · 4 years, 2 months ago
  33. 7bad75b3 Revert "Replace field trials with WebRtcKeyValueConfig in LossBasedBandwidthEstimation" by Mirko Bonadei · 4 years, 2 months ago
  34. fa5ad8c Revert "AV1: change update freq and disable denoiser explicitly." by Mirko Bonadei · 4 years, 2 months ago
  35. ad8a00d Replace casted uses of [OCMArg anyPointer] with [OCMArg anyObjectRef]. by Mirko Bonadei · 4 years, 2 months ago
  36. 028d5be Roll chromium_revision b86bc4feb4..1e9271a080 (852112:852360) by chromium-webrtc-autoroll · 4 years, 2 months ago
  37. 51f8e09 Replace field trials with WebRtcKeyValueConfig in LossBasedBandwidthEstimation by Per Kjellander · 4 years, 2 months ago
  38. bef0850 PRESUBMIT.py: Use input_api.owners_client to check DEPS includes owners. by Edward Lesmes · 4 years, 2 months ago
  39. abf5701 AV1: change update freq and disable denoiser explicitly. by Jerome Jiang · 4 years, 2 months ago
  40. bb8f32f peerconnection: measure bundle policy usage by Philipp Hancke · 4 years, 2 months ago
  41. 6d17602 Replace some RecursiveCriticalSection with Mutex, in PhysicalSocketServer. by Niels Möller · 4 years, 2 months ago
  42. c3a486c Delete unused sigslot variables. by Lahiru Ginnaliya Gamathige · 4 years, 2 months ago
  43. 6b143c1 Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController" by Guido Urdaneta · 4 years, 2 months ago
  44. 6e4fcac Revert "Remove thread hops from events provided by JsepTransportController." by Guido Urdaneta · 4 years, 2 months ago
  45. 82ce7e5 Fix PacedSender class to use plain mutex, rather than RecursiveCriticalSection by Niels Möller · 4 years, 2 months ago
  46. 0a144a7 Adding initial support for lock-less informing of muting by Per Åhgren · 4 years, 2 months ago
  47. 7e225cf Roll chromium_revision 4231e93dbb..b86bc4feb4 (850549:852112) by Mirko Bonadei · 4 years, 3 months ago
  48. b609718 Update WebRTC code version (2021-02-09T04:03:22). by webrtc-version-updater · 4 years, 3 months ago
  49. 7ef97f6 Relax limit on audio samples watching even more by Harald Alvestrand · 4 years, 3 months ago
  50. 203b549 Using nullopt when connection is nullptr. by Tim Na · 4 years, 3 months ago
  51. ccdf50b Move metadata in OWNERS files to DIR_METADATA files. by Josip Sokcevic · 4 years, 3 months ago
  52. f554b3c Remove thread hops from events provided by JsepTransportController. by Tomas Gunnarsson · 4 years, 3 months ago
  53. f4fa763 Update PsnrIsLowWhenNetworkIsBad test jitter_buffer mean value, as the congestion window default config changed. by Ying Wang · 4 years, 3 months ago
  54. 1142e0b Avoid crashing on error code 6450 in isac. by Ivo Creusen · 4 years, 3 months ago
  55. 68c225d Make 48 kHz maximum rate default for all devices by Per Åhgren · 4 years, 3 months ago
  56. 6cd4058 Fix unsynchronized access to mid_to_transport_ in JsepTransportController by Tomas Gunnarsson · 4 years, 3 months ago
  57. 54ea85c video frame: fix grammar by Philipp Hancke · 4 years, 3 months ago
  58. b586d82 Move SequenceChecker header to API: step 1, move header only by Artem Titov · 4 years, 3 months ago
  59. 4e57d4b Fix recursive locking in SignalThread. by Niels Möller · 4 years, 3 months ago
  60. a0848dd Correct SpatialLayer in VP9 unittest. by Hua, Chunbo · 4 years, 3 months ago
  61. 69c0118 Update WebRTC code version (2021-02-08T04:03:13). by webrtc-version-updater · 4 years, 3 months ago
  62. c3c63c8 Update WebRTC code version (2021-02-07T04:04:50). by webrtc-version-updater · 4 years, 3 months ago
  63. 9554a7b Account for extra capacity rtx packet might need by Danil Chapovalov · 4 years, 3 months ago
  64. d42413a fix RTP_DUMP timestamps by Philipp Hancke · 4 years, 3 months ago
  65. 129caca Update WebRTC code version (2021-02-06T04:03:11). by webrtc-version-updater · 4 years, 3 months ago
  66. 879d33b Add more refined control over dumping of data and the aecdump content by Per Åhgren · 4 years, 3 months ago
  67. d4ad2ef Remove accessor_lock_ in jsep_transport by Harald Alvestrand · 4 years, 3 months ago
  68. 32edd54 Add androidx dependency to webrtc by Andrey Logvin · 4 years, 3 months ago
  69. 6a48a1d Delete most use of `accessor_lock_` in JsepTransport. by Niels Möller · 4 years, 3 months ago
  70. 4593047 Make congestion window pushback drop frame experiment config default. by Ying Wang · 4 years, 3 months ago
  71. 1184b55 Fixed missing define ENETUNREACH. Every 2 sec logmessage Connect failed with 10051 by Berthold Herrmann · 4 years, 3 months ago
  72. ab9d6e1 Delete null JsepTransport constructor argument datagram_rtp_transport. by Niels Möller · 4 years, 3 months ago
  73. 3b9abd8 Avoiding the noise pumping during DTX regions by just forwarding the refresh DTX packets that decrease the comfort noise level at the decoder. by Jesús de Vicente Peña · 4 years, 3 months ago
  74. 483b31c Reland "Enable Video-QualityScaling experiment by default" by Ilya Nikolaevskiy · 4 years, 3 months ago
  75. aab91c7 Remove temporal layer restriction for forced resolution based fallback. by Åsa Persson · 4 years, 3 months ago
  76. b6d87dd Update WebRTC code version (2021-02-05T04:04:12). by webrtc-version-updater · 4 years, 3 months ago
  77. 8db9534 Support event log visualization in python3 by Björn Terelius · 4 years, 3 months ago
  78. 65b901b Clean up previously deleted RTCP VOIP metrics block. by Björn Terelius · 4 years, 3 months ago
  79. 7f354f8 Use bandwidth allocation in DropDueToSize when incoming resolution increases. by Åsa Persson · 4 years, 3 months ago
  80. 3ba7beb Use callback_list for port destroy operation. by Lahiru Ginnaliya Gamathige · 4 years, 3 months ago
  81. db82165 Add missing compile-time thread annotations for BaseChannel methods. by Niels Möller · 4 years, 3 months ago
  82. c5d4810 Const-declare some JsepTransport members, and delete always-null members. by Niels Möller · 4 years, 3 months ago
  83. 653d534 Roll chromium_revision 343080f0c8..4231e93dbb (850400:850549) by chromium-webrtc-autoroll · 4 years, 3 months ago
  84. ad99c81 Just adding my message in whitespace. by Alessio Bazzica · 4 years, 3 months ago
  85. a208861 Reland "Fix data race for config_ in AudioSendStream" by Artem Titov · 4 years, 3 months ago
  86. 4ef5638 Parse and plot RTCP BYE in RTC event log. by Björn Terelius · 4 years, 3 months ago
  87. 14b036d Make PipeWire 0.3 default version by Jan Grulich · 4 years, 3 months ago
  88. 20f7456 Fix unsynchronized access to jsep_transports_by_name_. by Tomas Gunnarsson · 4 years, 3 months ago
  89. 426b6e4 changed src\modules\audio_device\win\audio_device_core_win.cc , and it is working by Berthold Herrmann · 4 years, 3 months ago
  90. c1e7878 Remove deprecated ExpectationToString in SequenceChecker by Artem Titov · 4 years, 3 months ago
  91. 8ddbc04 Roll chromium_revision d32cb9c63c..343080f0c8 (850300:850400) by chromium-webrtc-autoroll · 4 years, 3 months ago
  92. 1d71fd9 Roll chromium_revision e4e9ee4776..d32cb9c63c (850184:850300) by chromium-webrtc-autoroll · 4 years, 3 months ago
  93. 1651c6c Roll chromium_revision 415eaa7c56..e4e9ee4776 (850009:850184) by chromium-webrtc-autoroll · 4 years, 3 months ago
  94. ad32586 Reland "Prepare to avoid hops to worker for network events." by Tomas Gunnarsson · 4 years, 3 months ago
  95. 3f7990d Split sequence checker on two headers before moving to API by Artem Titov · 4 years, 3 months ago
  96. 7301253 Add IncludeBlocks to clang-format. by Mirko Bonadei · 4 years, 3 months ago
  97. 14cad9f Fix clang-tidy: performance-inefficient-vector-operation. by Mirko Bonadei · 4 years, 3 months ago
  98. d582702 Add hta@ to WebRTC's root OWNERS. by Mirko Bonadei · 4 years, 3 months ago
  99. b582305 In VP9 encoder avoid crashing when encoder produce an unexpected frame by Danil Chapovalov · 4 years, 3 months ago
  100. 5361022 Delete unused function webrtc::AudioProcessing::MutateConfig by Sam Zackrisson · 4 years, 3 months ago