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webrtc / src / / e1e8b20444da097c216a1820233ccd4a4900cca0
  1. e1e8b20 Update WebRTC code version (2023-06-10T04:11:03). by webrtc-version-updater · 1 year, 10 months ago
  2. 4133797 Remove expired histograms WebRTC.PeerConnection.SrtpCryptoSuite by Johannes Kron · 1 year, 10 months ago
  3. c7695a5 Document how bitrate probing works from a RTP perspective by Philipp Hancke · 1 year, 10 months ago
  4. 48c44e3 Ensure RtpSenderEgress run on worker queue by Per K · 1 year, 10 months ago
  5. 2b5beb9 Set correct absolute send time on reordered packets by Per K · 1 year, 10 months ago
  6. 682755e Do not support frame tracking id extension in production by Philipp Hancke · 1 year, 10 months ago
  7. 5bcea25 Use version-less CIPD path for android_toolchain by Prashanth Swaminathan · 1 year, 10 months ago
  8. f781ff7 Update WebRTC code version (2023-06-09T04:02:47). by webrtc-version-updater · 1 year, 10 months ago
  9. cde5354 Implement DelayVariationCalculator for events analysis. by Rasmus Brandt · 1 year, 10 months ago
  10. f99e0f4 Remove stale Android NDK [2/2] by Prashanth Swaminathan · 1 year, 10 months ago
  11. 40ad4eb Roll chromium_revision a8db252505..a3756bb36c (1153825:1154916) by Prashanth Swaminathan · 1 year, 10 months ago
  12. 4ee5e5f Disable VideoCaptureTest due to flakyness by Björn Terelius · 1 year, 10 months ago
  13. 37fb647 Disable the roll of 'android_ndk' by Prashanth Swaminathan · 1 year, 10 months ago
  14. 36c945b Update WebRTC code version (2023-06-08T04:11:54). by webrtc-version-updater · 1 year, 10 months ago
  15. 9d9c3f4 [Analysis] Remove old threshold fields by Beining Chen · 1 year, 10 months ago
  16. 89f64b9 Make packet info optional and only set for primary packets in NetEq. by Jakob Ivarsson · 1 year, 10 months ago
  17. 9e639fa Migrate Android NDK to CIPD [1/2] by Prashanth Swaminathan · 1 year, 10 months ago
  18. fc260a18 Add method SetTimestamp in TransformableAudioFrameInterface by Palak Agarwal · 1 year, 10 months ago
  19. 3403acb av1: 8 threads for >720p and tiles config by Jerome Jiang · 1 year, 10 months ago
  20. d615704 Enable frame dropping in libaom AV1 encoder by Sergey Silkin · 1 year, 10 months ago
  21. a458fe5 Update WebRTC code version (2023-06-07T04:12:21). by webrtc-version-updater · 1 year, 10 months ago
  22. 09e0086 Remove ImplForTesting function from MediaChannel by Harald Alvestrand · 1 year, 10 months ago
  23. bd66cfe Roll chromium_revision a5cd053713..a8db252505 (1153688:1153825) by chromium-webrtc-autoroll · 1 year, 10 months ago
  24. 847208e Remove transitional shim classes by Harald Alvestrand · 1 year, 10 months ago
  25. ade07ca Rename current flexfec implementation flexfec_03 by Yosef Twaik · 1 year, 10 months ago
  26. 43df03d Fix spelling mistake ReplaceRemoteDescriptionAndCheckE*r*or by Philipp Hancke · 1 year, 10 months ago
  27. 6d25e96 Roll chromium_revision 404afa6a86..a5cd053713 (1153573:1153688) by chromium-webrtc-autoroll · 1 year, 10 months ago
  28. d3b71c7 Update WebRTC code version (2023-06-06T04:12:09). by webrtc-version-updater · 1 year, 10 months ago
  29. e00a12f Roll chromium_revision 96ad22527d..404afa6a86 (1153423:1153573) by chromium-webrtc-autoroll · 1 year, 10 months ago
  30. 8c4b9ea Remove references to AudioCodec and VideoCodec constructors by Florent Castelli · 1 year, 10 months ago
  31. fd096da Roll chromium_revision 8f3397a259..96ad22527d (1153256:1153423) by chromium-webrtc-autoroll · 1 year, 10 months ago
  32. 77c6230 Add create functions for voice media send and receive channels. by Harald Alvestrand · 1 year, 10 months ago
  33. be316da Ensure that RTCErrorOr<T, E> doesn't require T to be default constructible by Florent Castelli · 1 year, 10 months ago
  34. 0740048 Roll chromium_revision f28b824184..8f3397a259 (1152496:1153256) by chromium-webrtc-autoroll · 1 year, 10 months ago
  35. b0ef5e4 Declare factory functions for video sender and receiver by Harald Alvestrand · 1 year, 10 months ago
  36. 2f0c078 Split WebRtcVoiceChannel into Send and Receive classes by Harald Alvestrand · 1 year, 10 months ago
  37. 1e04d61 Update WebRTC code version (2023-06-05T04:02:35). by webrtc-version-updater · 1 year, 10 months ago
  38. 816f5b1 Create VP9Encoder with a VP9 codec object by Florent Castelli · 1 year, 10 months ago
  39. 968e3c0 rtp_sender: fix typo with spatial_bitmask by Alfred E. Heggestad · 1 year, 10 months ago
  40. 079ce25 Update WebRTC code version (2023-06-04T04:02:33). by webrtc-version-updater · 1 year, 10 months ago
  41. e10f025 Update WebRTC code version (2023-06-03T04:02:02). by webrtc-version-updater · 1 year, 10 months ago
  42. 5278b39 Add H264Encoder::Create() by Florent Castelli · 1 year, 10 months ago
  43. 811e24a Move functionality from AudioCodec and VideoCodec into cricket::Codec by Florent Castelli · 1 year, 10 months ago
  44. b8651de Roll chromium_revision d48b2929db..f28b824184 (1152392:1152496) by chromium-webrtc-autoroll · 1 year, 10 months ago
  45. 54e95bc Propagate time of the last received packet with Timestamp type by Danil Chapovalov · 1 year, 10 months ago
  46. 9a34d80 Apply the "shim" pattern for WebRtcVoiceEngine by Harald Alvestrand · 1 year, 10 months ago
  47. b15a9f0 Fix perf tests. by Jeremy Leconte · 1 year, 10 months ago
  48. 3488726 sdp: reject spec simulcast answers without the rid extension by Philipp Hancke · 1 year, 10 months ago
  49. f785bd4 Split WebRtcVideoMediaChannel into Send and Receive by Harald Alvestrand · 1 year, 10 months ago
  50. 4ad141e Add callback for send codec in audio too by Harald Alvestrand · 1 year, 10 months ago
  51. 371b7af Roll chromium_revision 2478b63fb4..d48b2929db (1151892:1152392) by chromium-webrtc-autoroll · 1 year, 10 months ago
  52. b29ee5b Run the same perf tests on all platforms. by Jeremy Leconte · 1 year, 10 months ago
  53. 267040e Make native VideoTrack pointer public by Jonas Oreland · 1 year, 10 months ago
  54. cfc1a3a Update vpython3 requests by Brian Sheedy · 1 year, 10 months ago
  55. eeacddb Disable flaky PictureIdTests. by Jeremy Leconte · 1 year, 10 months ago
  56. d454815 Use //third_party/cpu_features directly by Prashanth Swaminathan · 1 year, 10 months ago
  57. dab505b Update WebRTC code version (2023-06-02T04:02:59). by webrtc-version-updater · 1 year, 10 months ago
  58. 063b45b Roll chromium_revision faf350b988..2478b63fb4 (1151758:1151892) by chromium-webrtc-autoroll · 1 year, 10 months ago
  59. dba22d3 Move transceiver iteration loop over to the signaling thread. by Tommi · 1 year, 10 months ago
  60. 513ab0c Add a -d option to apply-iwyu by Harald Alvestrand · 1 year, 10 months ago
  61. e24b34c Roll chromium_revision e26eb46a54..faf350b988 (1150524:1151758) by chromium-webrtc-autoroll · 1 year, 10 months ago
  62. b93f69a In VideoCaptureV4L2 create the capture thread last in StartCapture by Andreas Pehrson · 1 year, 10 months ago
  63. e44a155 Add third_party/cpu_features license path. by Jeremy Leconte · 1 year, 10 months ago
  64. 2d59853 Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController. by Ying Wang · 1 year, 10 months ago
  65. 3d6e88e Remove low_bandwidth_audio_test. by Jeremy Leconte · 1 year, 10 months ago
  66. 6110fd9 Update WebRTC code version (2023-06-01T04:12:34). by webrtc-version-updater · 1 year, 10 months ago
  67. cb85143 Fix duplicate 'unix' OS and latest-revision deps by Prashanth Swaminathan · 1 year, 10 months ago
  68. 2197300 Update ReceiveStatistics to use Timestamp/TimeDelta to represent time by Danil Chapovalov · 1 year, 10 months ago
  69. a9bba04 Updating AsyncAudioProcessing API, part 1. by Peter Hanspers · 1 year, 10 months ago
  70. 56d69e2 Add //third_party/cpu_features to DEPS by Prashanth Swaminathan · 1 year, 10 months ago
  71. c18f083 Split MediaChannel concrete functions to MediaChannelUtil by Harald Alvestrand · 1 year, 10 months ago
  72. 94a9d55 Update WebRTC code version (2023-05-31T04:11:01). by webrtc-version-updater · 1 year, 10 months ago
  73. b84fae6 Use sinf instead of std::sinf to improve libstdc++ compatibility by Li-Yu Yu · 1 year, 10 months ago
  74. 9fa5057 Roll chromium_revision da88253915..e26eb46a54 (1150417:1150524) by chromium-webrtc-autoroll · 1 year, 10 months ago
  75. 6acfbb0 Replace std::optional with absl::optional in RtpPacketHistory by Per K · 1 year, 10 months ago
  76. d8098fb Delete struct RTCPReportBlock as no longer used by Danil Chapovalov · 1 year, 10 months ago
  77. d8b88d8 Use the VideoMediaChannelShim for all cases by Harald Alvestrand · 1 year, 10 months ago
  78. 428836d tools: fix small typo in python script by Alfred E. Heggestad · 1 year, 10 months ago
  79. 4bf5238 sdp: reject BUNDLE with RTP header extension id collisions by Philipp Hancke · 1 year, 10 months ago
  80. b184634 Run webrtc_perf_tests on Fuchsia os. by Jeremy Leconte · 1 year, 10 months ago
  81. c73ea4f More systematic null checks before calling native methods by Xavier Lepaul · 1 year, 10 months ago
  82. a3e9c0a Roll chromium_revision c90a8a46d7..da88253915 (1150306:1150417) by chromium-webrtc-autoroll · 1 year, 10 months ago
  83. 97c9623 Make a shim object implementing the VideoMediaChannel interface by Harald Alvestrand · 1 year, 10 months ago
  84. 4c1e959 Change flexfec header reader to parse according to updated RFC. by Yosef Twaik · 1 year, 10 months ago
  85. e4a9a6d Update WebRTC code version (2023-05-30T04:02:06). by webrtc-version-updater · 1 year, 10 months ago
  86. c5e4bcc Roll chromium_revision 599c746c73..c90a8a46d7 (1150194:1150306) by chromium-webrtc-autoroll · 1 year, 10 months ago
  87. 4b14cb7 Roll chromium_revision fa2e063162..599c746c73 (1150086:1150194) by chromium-webrtc-autoroll · 1 year, 10 months ago
  88. 4aaacb4 Update WebRTC code version (2023-05-29T04:03:50). by webrtc-version-updater · 1 year, 10 months ago
  89. e641a97 In RtcpReceiver remove redundand way to represent RTCP report blocks by Danil Chapovalov · 1 year, 10 months ago
  90. b9de471 Update WebRTC code version (2023-05-28T04:11:22). by webrtc-version-updater · 1 year, 10 months ago
  91. 98185b9 Roll chromium_revision 99b12997bf..fa2e063162 (1150050:1150086) by chromium-webrtc-autoroll · 1 year, 10 months ago
  92. a294353 Use type raw for video_codec_perf_tests. by Mirko Bonadei · 1 year, 10 months ago
  93. 01c2efc Roll chromium_revision bddf6cbe18..99b12997bf (1149812:1150050) by chromium-webrtc-autoroll · 1 year, 10 months ago
  94. 9bc8d05 Update WebRTC code version (2023-05-27T04:12:09). by webrtc-version-updater · 1 year, 10 months ago
  95. 9ac543c Roll chromium_revision 1fc947a5da..bddf6cbe18 (1149703:1149812) by chromium-webrtc-autoroll · 1 year, 10 months ago
  96. 87e74f9 Remove unused combined_audio_video_bwe. by Yury Yarashevich · 1 year, 10 months ago
  97. 2bb686d Stop running low_bandwith_audio_tests. by Jeremy Leconte · 1 year, 10 months ago
  98. 6490999 Roll chromium_revision aae661725b..1fc947a5da (1148994:1149703) by chromium-webrtc-autoroll · 1 year, 10 months ago
  99. f0820ff Implement video versions of RTCInboundRtpStreamStats.jitterBuffer{Target,Minimum}Delay by Rasmus Brandt · 1 year, 10 months ago
  100. 9caef2a Use a constant for invalid PipeWire file descriptor by Jan Grulich · 1 year, 10 months ago