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webrtc / src / / f0895f77336696995b173a89fea5815d17fd9b47
  1. f0895f7 Roll chromium_revision b77c10eec3..553b5dd870 (1256869:1257025) by chromium-webrtc-autoroll · 1 year, 1 month ago
  2. 0a92fe8 Roll chromium_revision 4433ef31eb..b77c10eec3 (1256671:1256869) by chromium-webrtc-autoroll · 1 year, 1 month ago
  3. 8bb54c1 Penultimate split-up of rtc_p2p build target by Harald Alvestrand · 1 year, 1 month ago
  4. 310c9d4 Roll chromium_revision 9983c30299..4433ef31eb (1256551:1256671) by chromium-webrtc-autoroll · 1 year, 1 month ago
  5. 26540f6 Update WebRTC code version (2024-02-06T04:02:15). by webrtc-version-updater · 1 year, 1 month ago
  6. 1188d08 Roll chromium_revision 224f3cf6ba..9983c30299 (1256436:1256551) by chromium-webrtc-autoroll · 1 year, 1 month ago
  7. 559e854 Roll chromium_revision 0627ca51a7..224f3cf6ba (1256272:1256436) by chromium-webrtc-autoroll · 1 year, 1 month ago
  8. 57a1232 Remove WebRtcVideoSendChannel::kDefaultQpMax by Sergey Silkin · 1 year, 1 month ago
  9. 1df2690 Roll chromium_revision 82f331232d..0627ca51a7 (1256154:1256272) by chromium-webrtc-autoroll · 1 year, 1 month ago
  10. 5ff04d1 Avoid zero duration packets in NetEq test with replacement audio. by Jakob Ivarsson · 1 year, 1 month ago
  11. 5b90b96 Provide Environment for VideoDecoder in video_coding/ tests by Danil Chapovalov · 1 year, 1 month ago
  12. 0358a2c Roll chromium_revision 844caa73fd..82f331232d (1252127:1256154) by chromium-webrtc-autoroll · 1 year, 1 month ago
  13. ff61626 Fix Chromium roll failures due to classpath format changes in jni_zero by Byoungchan Lee · 1 year, 1 month ago
  14. f6ae657 Adapt NetEq delay to received FEC (both RED and codec inband). by Jakob Ivarsson · 1 year, 1 month ago
  15. 35fe958 Add RTC_EXPORT for rtc::IPIsLinkLocal and rtc::IPIsLoopback by Artem Titov · 1 year, 1 month ago
  16. bda5cc6 Clean up use of WebRTC-UseStandardBytesStats trial in tests by Philipp Hancke · 1 year, 1 month ago
  17. 2212f86 Update WebRTC code version (2024-02-04T04:03:09). by webrtc-version-updater · 1 year, 1 month ago
  18. 707296a Update WebRTC code version (2024-02-03T04:02:09). by webrtc-version-updater · 1 year, 1 month ago
  19. 8c371f2 Reland "Take out Fuchsia-only SDES-enabling parameters" by Harald Alvestrand · 1 year, 1 month ago
  20. f19c7ca Fix crash when rolling libaom. by Jeremy Leconte · 1 year, 1 month ago
  21. 9d65366 Update WebRTC code version (2024-02-02T04:03:16). by webrtc-version-updater · 1 year, 1 month ago
  22. 687ef0a Revert "Remove post-decode VAD" by Jeremy Leconte · 1 year, 1 month ago
  23. 53e41a2 Ignore old, duplicate and overlapping packets in packet arrival history. by Jakob Ivarsson · 1 year, 1 month ago
  24. b5f2b17 Remove SOCKS5 support by Harald Alvestrand · 1 year, 1 month ago
  25. c1cc6a3 sdp: backfill default codec parameters for AV1 by Philipp Hancke · 1 year, 1 month ago
  26. 89cf26f Remove post-decode VAD by Tomas Lundqvist · 1 year, 1 month ago
  27. 2c169ae Rename kLocal to kHost and kStun to kSrflx by Tommi · 1 year, 1 month ago
  28. d071dc1 Bypass global field trial string in RtcEventLog unittests by Danil Chapovalov · 1 year, 1 month ago
  29. 657b65f Remove special-casing of TWCCv2 negotiation by Philipp Hancke · 1 year, 1 month ago
  30. 27a452d Rewrite webrtc_sdp unittest to use DTLS not SDES by Harald Alvestrand · 1 year, 1 month ago
  31. 4860148 Add WebRTC-LibaomAv1Encoder-MaxConsecFrameDrop parameter to explicitly limit the maximum consecutive frame drop by Dan Tan · 1 year, 1 month ago
  32. 1d3e286 Fix a fuzzer-found issue in G.722 decoder by Henrik Lundin · 1 year, 1 month ago
  33. 26ad5b8 Fix a fuzzer-found issue in PCM/G.711 decoder by Henrik Lundin · 1 year, 1 month ago
  34. 9b7f364 Fix a fuzzer-found issue in PCM16 decoder by Henrik Lundin · 1 year, 1 month ago
  35. 14b016f In RtcEventLogEncoderNewFormat use propagated instead of global field trials by Danil Chapovalov · 1 year, 1 month ago
  36. c0741e9 Revert "Take out Fuchsia-only SDES-enabling parameters" by Olga Sharonova · 1 year, 1 month ago
  37. 7cb4ce0 Remove IceCandidateType::kNumValues by Tommi · 1 year, 1 month ago
  38. 3cbe63e Do not register receiver for REMB until it starts receiving by Philipp Hancke · 1 year, 1 month ago
  39. c4dd03d Remove kUnknown as a possible value for IceCandidateType. by Tommi · 1 year, 1 month ago
  40. 958c9ac Allow VideoCaptureModulePipeWire to be shared with more consumers by Jan Grulich · 1 year, 1 month ago
  41. 365cf14 Revert "Test new tree." by Mirko Bonadei · 1 year, 1 month ago
  42. c6675ed Test new tree. by Mirko Bonadei · 1 year, 1 month ago
  43. cc83e32 Fix H.265 bitstream parser incorrect PPS reference. by Qiu Jianlin · 1 year, 1 month ago
  44. 056782c Implement Socket::RecvFrom(ReceiveBuffer& buffer) in PhysicalSocketServer by Per K · 1 year, 1 month ago
  45. 59f3b35 Take out Fuchsia-only SDES-enabling parameters by Harald Alvestrand · 1 year, 1 month ago
  46. 765024e test: fix fuzzers line-endings by Alfred E. Heggestad · 1 year, 1 month ago
  47. 05a6f3b Update WebRTC code version (2024-01-30T04:07:38). by webrtc-version-updater · 1 year, 1 month ago
  48. 68e85b8 Adds WebRTC-Audio-PriorityBitrate for controlling audio/video rate allocation by Dan Tan · 1 year, 1 month ago
  49. 62cee88 Propagate Environment through QualityAnalyzingVideoDecoderFactory by Danil Chapovalov · 1 year, 1 month ago
  50. 7aa7972 Propagate sequence number to cloned encoded audio frames by Tony Herre · 1 year, 1 month ago
  51. f43e8eb Add RTP depacketizer for H265 by qwu16 · 1 year, 1 month ago
  52. 6adf224 Compute scaling factors for not-explicitly configured layers in VP9 encoder by Ilya Nikolaevskiy · 1 year, 1 month ago
  53. 98db63c Introduce RtpTransportConfig:allow_bandwidht_estimation_probe_without_media by Per K · 1 year, 1 month ago
  54. 89db1c5 Update WebRTC code version (2024-01-29T04:16:27). by webrtc-version-updater · 1 year, 1 month ago
  55. 0c4165e Update WebRTC code version (2024-01-27T04:11:16). by webrtc-version-updater · 1 year, 2 months ago
  56. 698b4e7 Update more Candidate type checkers to use Candidate::is_* by Tommi · 1 year, 2 months ago
  57. 9c687460 Consolidate encoded transform mocks into api/test/ by Tony Herre · 1 year, 2 months ago
  58. 1dccfeb Set InterLayerPredMode based on scalability mode for VP9. by Åsa Persson · 1 year, 2 months ago
  59. 979b6d6 Refactor RtpVideoSender::SetActiveModules. by Per K · 1 year, 2 months ago
  60. 9c166e0 Remove VideoSendStream::StartPerRtpStream by Per K · 1 year, 2 months ago
  61. 9a953b2 Detangle p2p/connection.cc and port.cc by Harald Alvestrand · 1 year, 2 months ago
  62. d213dd5 Pass Environment to VideoDecoders through VideoCodecTester by Danil Chapovalov · 1 year, 2 months ago
  63. 523eff6 [Stats] Delete unused RTCStatsMember type alias. by Henrik Boström · 1 year, 2 months ago
  64. 6a3bbef Reland "Enable DD and VLA header extensions by default for Simulcast/SVC" by Philipp Hancke · 1 year, 2 months ago
  65. 7f8470a Update WebRTC code version (2024-01-26T04:12:47). by webrtc-version-updater · 1 year, 2 months ago
  66. 0817380 Pass Environment when creating VideoDecoder in VideoReceiveStream2 by Danil Chapovalov · 1 year, 2 months ago
  67. ac58a33 [Stats] Migrate from the RTCStatsMember type alias to absl::optional. by Henrik Boström · 1 year, 2 months ago
  68. a310d78 Refactor a lot of the p2p:rtc_p2p target by Harald Alvestrand · 1 year, 2 months ago
  69. e3a4bdb Roll chromium_revision 712952759e..844caa73fd (1251936:1252127) by chromium-webrtc-autoroll · 1 year, 2 months ago
  70. cc70a6d Guard GenerateUniqueId aginst concurrent access. by Mirko Bonadei · 1 year, 2 months ago
  71. c3624d0 Add field trial that enables Opus PLC. by Jakob Ivarsson · 1 year, 2 months ago
  72. de3c726 Update to vpython 3.11 and remove .vpython (v2.x) by Christoffer Dewerin · 1 year, 2 months ago
  73. c13a7f9 Change string constant to constexpr char[] to unblock roll. by Henrik Boström · 1 year, 2 months ago
  74. b7e8a10 Roll chromium_revision cf886b3ada..712952759e (1250272:1251936) by chromium-webrtc-autoroll · 1 year, 2 months ago
  75. a8375bb iOS: Fix building tests on real devices by Florent Castelli · 1 year, 2 months ago
  76. 5372bce Reland "[Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>." by Henrik Boström · 1 year, 2 months ago
  77. 44e4453 Fix Port test and supply a legal value for the port type. by Tommi · 1 year, 2 months ago
  78. 1b61c71 Expose setCodecPreferences/getCapabilities for iOS by Karim H · 1 year, 2 months ago
  79. c708c00 Add VideoDecoderFactory function to pass Environment for VideoDecoder construction by Danil Chapovalov · 1 year, 2 months ago
  80. 340d6c0 Remove packet overhead lock and cached bitrate constraints. by Jakob Ivarsson · 1 year, 2 months ago
  81. eb76f19 Implement Newline Check in the Presubmit by Byoungchan Lee · 1 year, 2 months ago
  82. 25b2982 Roll chromium_revision e1fb84c37d..cf886b3ada (1250109:1250272) by chromium-webrtc-autoroll · 1 year, 2 months ago
  83. be2786c Move candidate type preference defaults to the Candidate class by Tommi · 1 year, 2 months ago
  84. fd54a61 Revert "[Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>." by Henrik Boström · 1 year, 2 months ago
  85. 6fa743f Roll chromium_revision 336689d906..e1fb84c37d (1249985:1250109) by chromium-webrtc-autoroll · 1 year, 2 months ago
  86. 79ac694 [Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>. by Henrik Boström · 1 year, 2 months ago
  87. f1fc6ab Remove usage of the rtc::TaskQueue in video/ by Danil Chapovalov · 1 year, 2 months ago
  88. 3484381 [Stats] Delete ValueToString/ToJson, ToString does the job. by Henrik Boström · 1 year, 2 months ago
  89. 9074f0b Roll chromium_revision eb15a443b5..336689d906 (1249859:1249985) by chromium-webrtc-autoroll · 1 year, 2 months ago
  90. 1768705 Revert^4 "Delete pc/peerconnection build target" by Harald Alvestrand · 1 year, 2 months ago
  91. 11f87b2 Update WebRTC code version (2024-01-21T04:12:31). by webrtc-version-updater · 1 year, 2 months ago
  92. 7a1f85f Roll chromium_revision fc694f0482..eb15a443b5 (1249756:1249859) by chromium-webrtc-autoroll · 1 year, 2 months ago
  93. 798e451 Adds WebRTC-AV1-OverridePriorityBitrate to change bit rate allocation between audio and video by Dan Tan · 1 year, 2 months ago
  94. 5dc6c14 Update WebRTC code version (2024-01-20T04:12:27). by webrtc-version-updater · 1 year, 2 months ago
  95. 1c1c260 Roll chromium_revision fbebadaea2..fc694f0482 (1249624:1249756) by chromium-webrtc-autoroll · 1 year, 2 months ago
  96. 7ee67e1 Roll chromium_revision 4b479b78e7..fbebadaea2 (1247241:1249624) by chromium-webrtc-autoroll · 1 year, 2 months ago
  97. 25be2f8 Remove legacy, unused, preference() property from Candidate by Tommi · 1 year, 2 months ago
  98. 67ea392 Set visibility for RTC event log events BUILD targets by Björn Terelius · 1 year, 2 months ago
  99. 6a99212 Tighten som DCHECKs to CHECKs in VP9 packetization. by Erik Språng · 1 year, 2 months ago
  100. a9ef127 Fix fuzzing issue for h265 bitstream parser by qwu16 · 1 year, 2 months ago