Google Git
Sign in
webrtc / src / / 1cb54bee7ac24eb67ea8d751b1f87a5632abbe7b
  1. 1cb54be Delete unused killswitch flag related to scalability mode. by Henrik Boström · 1 year, 9 months ago
  2. 49ace8b Merge the codec types by Florent Castelli · 1 year, 9 months ago
  3. 0451baa Roll chromium_revision 13b44452d4..bb8855a075 (1157265:1157397) by chromium-webrtc-autoroll · 1 year, 9 months ago
  4. 3cf60be Update WebRTC code version (2023-06-14T04:02:08). by webrtc-version-updater · 1 year, 9 months ago
  5. 7d77e24 Roll chromium_revision 8d5bd97af2..13b44452d4 (1157142:1157265) by chromium-webrtc-autoroll · 1 year, 9 months ago
  6. eb76ed9 Roll chromium_revision 6f33bc2255..8d5bd97af2 (1156959:1157142) by chromium-webrtc-autoroll · 1 year, 9 months ago
  7. 1f235d6 Roll chromium_revision 2710d66105..6f33bc2255 (1156780:1156959) by chromium-webrtc-autoroll · 1 year, 9 months ago
  8. 61deed5 Change flexfec header writer to finalize header according to updated RFC by Yosef Twaik · 1 year, 9 months ago
  9. bde7c6a Change FinalizeFecHeader to receive list of streams by Yosef Twaik · 1 year, 9 months ago
  10. cfa0f81 Fix DEPS path for clang-format scripts by Florent Castelli · 1 year, 9 months ago
  11. 18f66fc [rct_tools/video_encoder] Add video encoder tool by Jianhui Dai · 1 year, 9 months ago
  12. febf558 Revert "Adopt EglThread in EglRenderer" by Linus Nilsson · 1 year, 9 months ago
  13. c8b217a Roll chromium_revision 0e945e28ee..2710d66105 (1156642:1156780) by chromium-webrtc-autoroll · 1 year, 9 months ago
  14. 9bb7f81 Update WebRTC code version (2023-06-13T04:07:18). by webrtc-version-updater · 1 year, 9 months ago
  15. 491d1d6 Roll chromium_revision a3f4dda807..0e945e28ee (1156488:1156642) by chromium-webrtc-autoroll · 1 year, 9 months ago
  16. e370f2f Roll chromium_revision 6f1f457b3d..a3f4dda807 (1156303:1156488) by chromium-webrtc-autoroll · 1 year, 9 months ago
  17. f6f642d Roll chromium_revision dd02f6b781..6f1f457b3d (1156166:1156303) by chromium-webrtc-autoroll · 1 year, 9 months ago
  18. c0e2418 Sort WebRtcAudio{Send,Receive}Channel implementation by Harald Alvestrand · 1 year, 9 months ago
  19. 4f6783c Roll chromium_revision a3756bb36c..dd02f6b781 (1154916:1156166) by Mirko Bonadei · 1 year, 9 months ago
  20. de92338 Update parameters' type from NSString to AVAudioSession*. by Abby Yeh · 1 year, 9 months ago
  21. ee58849 Make SetRTPTimestamp pure virtual in TransformableAudioFrameInterface by Palak Agarwal · 1 year, 9 months ago
  22. 47bdcc1 When updating audio session, update category, mode, options at once. by Abby Yeh · 1 year, 9 months ago
  23. e1e8b20 Update WebRTC code version (2023-06-10T04:11:03). by webrtc-version-updater · 1 year, 10 months ago
  24. 4133797 Remove expired histograms WebRTC.PeerConnection.SrtpCryptoSuite by Johannes Kron · 1 year, 10 months ago
  25. c7695a5 Document how bitrate probing works from a RTP perspective by Philipp Hancke · 1 year, 10 months ago
  26. 48c44e3 Ensure RtpSenderEgress run on worker queue by Per K · 1 year, 10 months ago
  27. 2b5beb9 Set correct absolute send time on reordered packets by Per K · 1 year, 10 months ago
  28. 682755e Do not support frame tracking id extension in production by Philipp Hancke · 1 year, 10 months ago
  29. 5bcea25 Use version-less CIPD path for android_toolchain by Prashanth Swaminathan · 1 year, 10 months ago
  30. f781ff7 Update WebRTC code version (2023-06-09T04:02:47). by webrtc-version-updater · 1 year, 10 months ago
  31. cde5354 Implement DelayVariationCalculator for events analysis. by Rasmus Brandt · 1 year, 10 months ago
  32. f99e0f4 Remove stale Android NDK [2/2] by Prashanth Swaminathan · 1 year, 10 months ago
  33. 40ad4eb Roll chromium_revision a8db252505..a3756bb36c (1153825:1154916) by Prashanth Swaminathan · 1 year, 10 months ago
  34. 4ee5e5f Disable VideoCaptureTest due to flakyness by Björn Terelius · 1 year, 10 months ago
  35. 37fb647 Disable the roll of 'android_ndk' by Prashanth Swaminathan · 1 year, 10 months ago
  36. 36c945b Update WebRTC code version (2023-06-08T04:11:54). by webrtc-version-updater · 1 year, 10 months ago
  37. 9d9c3f4 [Analysis] Remove old threshold fields by Beining Chen · 1 year, 10 months ago
  38. 89f64b9 Make packet info optional and only set for primary packets in NetEq. by Jakob Ivarsson · 1 year, 10 months ago
  39. 9e639fa Migrate Android NDK to CIPD [1/2] by Prashanth Swaminathan · 1 year, 10 months ago
  40. fc260a18 Add method SetTimestamp in TransformableAudioFrameInterface by Palak Agarwal · 1 year, 10 months ago
  41. 3403acb av1: 8 threads for >720p and tiles config by Jerome Jiang · 1 year, 10 months ago
  42. d615704 Enable frame dropping in libaom AV1 encoder by Sergey Silkin · 1 year, 10 months ago
  43. a458fe5 Update WebRTC code version (2023-06-07T04:12:21). by webrtc-version-updater · 1 year, 10 months ago
  44. 09e0086 Remove ImplForTesting function from MediaChannel by Harald Alvestrand · 1 year, 10 months ago
  45. bd66cfe Roll chromium_revision a5cd053713..a8db252505 (1153688:1153825) by chromium-webrtc-autoroll · 1 year, 10 months ago
  46. 847208e Remove transitional shim classes by Harald Alvestrand · 1 year, 10 months ago
  47. ade07ca Rename current flexfec implementation flexfec_03 by Yosef Twaik · 1 year, 10 months ago
  48. 43df03d Fix spelling mistake ReplaceRemoteDescriptionAndCheckE*r*or by Philipp Hancke · 1 year, 10 months ago
  49. 6d25e96 Roll chromium_revision 404afa6a86..a5cd053713 (1153573:1153688) by chromium-webrtc-autoroll · 1 year, 10 months ago
  50. d3b71c7 Update WebRTC code version (2023-06-06T04:12:09). by webrtc-version-updater · 1 year, 10 months ago
  51. e00a12f Roll chromium_revision 96ad22527d..404afa6a86 (1153423:1153573) by chromium-webrtc-autoroll · 1 year, 10 months ago
  52. 8c4b9ea Remove references to AudioCodec and VideoCodec constructors by Florent Castelli · 1 year, 10 months ago
  53. fd096da Roll chromium_revision 8f3397a259..96ad22527d (1153256:1153423) by chromium-webrtc-autoroll · 1 year, 10 months ago
  54. 77c6230 Add create functions for voice media send and receive channels. by Harald Alvestrand · 1 year, 10 months ago
  55. be316da Ensure that RTCErrorOr<T, E> doesn't require T to be default constructible by Florent Castelli · 1 year, 10 months ago
  56. 0740048 Roll chromium_revision f28b824184..8f3397a259 (1152496:1153256) by chromium-webrtc-autoroll · 1 year, 10 months ago
  57. b0ef5e4 Declare factory functions for video sender and receiver by Harald Alvestrand · 1 year, 10 months ago
  58. 2f0c078 Split WebRtcVoiceChannel into Send and Receive classes by Harald Alvestrand · 1 year, 10 months ago
  59. 1e04d61 Update WebRTC code version (2023-06-05T04:02:35). by webrtc-version-updater · 1 year, 10 months ago
  60. 816f5b1 Create VP9Encoder with a VP9 codec object by Florent Castelli · 1 year, 10 months ago
  61. 968e3c0 rtp_sender: fix typo with spatial_bitmask by Alfred E. Heggestad · 1 year, 10 months ago
  62. 079ce25 Update WebRTC code version (2023-06-04T04:02:33). by webrtc-version-updater · 1 year, 10 months ago
  63. e10f025 Update WebRTC code version (2023-06-03T04:02:02). by webrtc-version-updater · 1 year, 10 months ago
  64. 5278b39 Add H264Encoder::Create() by Florent Castelli · 1 year, 10 months ago
  65. 811e24a Move functionality from AudioCodec and VideoCodec into cricket::Codec by Florent Castelli · 1 year, 10 months ago
  66. b8651de Roll chromium_revision d48b2929db..f28b824184 (1152392:1152496) by chromium-webrtc-autoroll · 1 year, 10 months ago
  67. 54e95bc Propagate time of the last received packet with Timestamp type by Danil Chapovalov · 1 year, 10 months ago
  68. 9a34d80 Apply the "shim" pattern for WebRtcVoiceEngine by Harald Alvestrand · 1 year, 10 months ago
  69. b15a9f0 Fix perf tests. by Jeremy Leconte · 1 year, 10 months ago
  70. 3488726 sdp: reject spec simulcast answers without the rid extension by Philipp Hancke · 1 year, 10 months ago
  71. f785bd4 Split WebRtcVideoMediaChannel into Send and Receive by Harald Alvestrand · 1 year, 10 months ago
  72. 4ad141e Add callback for send codec in audio too by Harald Alvestrand · 1 year, 10 months ago
  73. 371b7af Roll chromium_revision 2478b63fb4..d48b2929db (1151892:1152392) by chromium-webrtc-autoroll · 1 year, 10 months ago
  74. b29ee5b Run the same perf tests on all platforms. by Jeremy Leconte · 1 year, 10 months ago
  75. 267040e Make native VideoTrack pointer public by Jonas Oreland · 1 year, 10 months ago
  76. cfc1a3a Update vpython3 requests by Brian Sheedy · 1 year, 10 months ago
  77. eeacddb Disable flaky PictureIdTests. by Jeremy Leconte · 1 year, 10 months ago
  78. d454815 Use //third_party/cpu_features directly by Prashanth Swaminathan · 1 year, 10 months ago
  79. dab505b Update WebRTC code version (2023-06-02T04:02:59). by webrtc-version-updater · 1 year, 10 months ago
  80. 063b45b Roll chromium_revision faf350b988..2478b63fb4 (1151758:1151892) by chromium-webrtc-autoroll · 1 year, 10 months ago
  81. dba22d3 Move transceiver iteration loop over to the signaling thread. by Tommi · 1 year, 10 months ago
  82. 513ab0c Add a -d option to apply-iwyu by Harald Alvestrand · 1 year, 10 months ago
  83. e24b34c Roll chromium_revision e26eb46a54..faf350b988 (1150524:1151758) by chromium-webrtc-autoroll · 1 year, 10 months ago
  84. b93f69a In VideoCaptureV4L2 create the capture thread last in StartCapture by Andreas Pehrson · 1 year, 10 months ago
  85. e44a155 Add third_party/cpu_features license path. by Jeremy Leconte · 1 year, 10 months ago
  86. 2d59853 Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController. by Ying Wang · 1 year, 10 months ago
  87. 3d6e88e Remove low_bandwidth_audio_test. by Jeremy Leconte · 1 year, 10 months ago
  88. 6110fd9 Update WebRTC code version (2023-06-01T04:12:34). by webrtc-version-updater · 1 year, 10 months ago
  89. cb85143 Fix duplicate 'unix' OS and latest-revision deps by Prashanth Swaminathan · 1 year, 10 months ago
  90. 2197300 Update ReceiveStatistics to use Timestamp/TimeDelta to represent time by Danil Chapovalov · 1 year, 10 months ago
  91. a9bba04 Updating AsyncAudioProcessing API, part 1. by Peter Hanspers · 1 year, 10 months ago
  92. 56d69e2 Add //third_party/cpu_features to DEPS by Prashanth Swaminathan · 1 year, 10 months ago
  93. c18f083 Split MediaChannel concrete functions to MediaChannelUtil by Harald Alvestrand · 1 year, 10 months ago
  94. 94a9d55 Update WebRTC code version (2023-05-31T04:11:01). by webrtc-version-updater · 1 year, 10 months ago
  95. b84fae6 Use sinf instead of std::sinf to improve libstdc++ compatibility by Li-Yu Yu · 1 year, 10 months ago
  96. 9fa5057 Roll chromium_revision da88253915..e26eb46a54 (1150417:1150524) by chromium-webrtc-autoroll · 1 year, 10 months ago
  97. 6acfbb0 Replace std::optional with absl::optional in RtpPacketHistory by Per K · 1 year, 10 months ago
  98. d8098fb Delete struct RTCPReportBlock as no longer used by Danil Chapovalov · 1 year, 10 months ago
  99. d8b88d8 Use the VideoMediaChannelShim for all cases by Harald Alvestrand · 1 year, 10 months ago
  100. 428836d tools: fix small typo in python script by Alfred E. Heggestad · 1 year, 10 months ago