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webrtc / src / / df810ab2adb57dfce650a399b3c42512ce650c6c
  1. df810ab Roll chromium_revision a7b5beaadc..1a5b138d55 (1185128:1185348) by chromium-webrtc-autoroll · 1 year, 7 months ago
  2. ad3f1bc Adopt EglThread in EglRenderer by Linus Nilsson · 1 year, 7 months ago
  3. 47f4e55 Log video encoder InitEncode error code by Philipp Hancke · 1 year, 7 months ago
  4. 972f283 Added HEVC parser for WebRTC by qwu16 · 1 year, 7 months ago
  5. b071871 Remove unused ALR experiment settings from VideoStreamEncoder. by philipel · 1 year, 7 months ago
  6. c146b5f Represent unset VideoPlayoutDelay with nullopt rather than special value by Danil Chapovalov · 1 year, 7 months ago
  7. 0250b69 Delete unused constructor of FakeNetworkPipe by Danil Chapovalov · 1 year, 7 months ago
  8. 44e54a7 Roll chromium_revision 4b024ef0b5..a7b5beaadc (1185015:1185128) by chromium-webrtc-autoroll · 1 year, 7 months ago
  9. 70483a5 Add unscaled getWidth/Height to TextureBuffer interface by Magnus Jedvert · 1 year, 7 months ago
  10. 31251fa Delete deprecated RTPSender::SetCsrcs by Danil Chapovalov · 1 year, 7 months ago
  11. 68dafcb Update WebRTC code version (2023-08-18T04:05:33). by webrtc-version-updater · 1 year, 7 months ago
  12. dce9913 Roll chromium_revision 80b8365885..4b024ef0b5 (1184627:1185015) by chromium-webrtc-autoroll · 1 year, 7 months ago
  13. e5af52e Make DesktopFrameCGImage::CreateFromCGImage() public by Johannes Kron · 1 year, 7 months ago
  14. a39d764 Roll chromium_revision 9f26c488cf..80b8365885 (1184507:1184627) by chromium-webrtc-autoroll · 1 year, 7 months ago
  15. 69e2637 Fix playout delay logging typo. by Qiu Jianlin · 1 year, 7 months ago
  16. 4f89506 Update WebRTC code version (2023-08-17T04:03:59). by webrtc-version-updater · 1 year, 7 months ago
  17. 6cb8b4f Roll chromium_revision c7e600fcc1..9f26c488cf (1184329:1184507) by chromium-webrtc-autoroll · 1 year, 7 months ago
  18. ee15bea Roll chromium_revision b88d5ed17e..c7e600fcc1 (1183080:1184329) by chromium-webrtc-autoroll · 1 year, 7 months ago
  19. 1e42d83 Make VCMReceiveCallback::FrameToRender pure virtual again by Philipp Hancke · 1 year, 7 months ago
  20. ff2462e Update WebRTC code version (2023-08-16T04:02:29). by webrtc-version-updater · 1 year, 7 months ago
  21. 392e471 Remove deprecated TransformableAudioFrameInterface::getHeader() method by Tony Herre · 1 year, 7 months ago
  22. d43af91 Remove internal overrides using old SendRtp and SendRtcp interfaces. by Harald Alvestrand · 1 year, 7 months ago
  23. 5165743 Reland "Fix definition of keyframes decoded statistics" by Philipp Hancke · 1 year, 7 months ago
  24. 48852c4 Drop email recipients from lkgr finder. by Mirko Bonadei · 1 year, 7 months ago
  25. c2da2e2 Update WebRTC code version (2023-08-15T04:12:47). by webrtc-version-updater · 1 year, 7 months ago
  26. 2dbf6e0 Revert "Fix definition of keyframes decoded statistics" by Mirko Bonadei · 1 year, 7 months ago
  27. b2be55d Roll chromium_revision ba4f77cde1..b88d5ed17e (1182677:1183080) by chromium-webrtc-autoroll · 1 year, 7 months ago
  28. ed2ce81 RtpSenderEgress: make potential crash more explicit. by Markus Handell · 1 year, 7 months ago
  29. 4c17e2d Refactor csrcs managment in RtpSender by Danil Chapovalov · 1 year, 7 months ago
  30. 6b51e72 fec: Skip traversal the list of recovered packets if possible by Andrey Volykhin · 1 year, 7 months ago
  31. 0e37f5e Fix definition of keyframes decoded statistics by Philipp Hancke · 1 year, 7 months ago
  32. 1b1f712 Update WebRTC code version (2023-08-12T04:04:01). by webrtc-version-updater · 1 year, 7 months ago
  33. 0bd0897 Roll chromium_revision d6e852857c..ba4f77cde1 (1182529:1182677) by chromium-webrtc-autoroll · 1 year, 7 months ago
  34. 0a740a9 Roll chromium_revision b5f5329314..d6e852857c (1182406:1182529) by chromium-webrtc-autoroll · 1 year, 7 months ago
  35. b987648 Exposed Adapter types for Android by MohamedRisaldarTA · 1 year, 7 months ago
  36. 721b908 Update WebRTC code version (2023-08-11T04:02:03). by webrtc-version-updater · 1 year, 7 months ago
  37. 497eacf Roll chromium_revision ac89fd7158..b5f5329314 (1182099:1182406) by chromium-webrtc-autoroll · 1 year, 7 months ago
  38. 24e704f Cleanup calculating time between RTCP reports by Danil Chapovalov · 1 year, 7 months ago
  39. 92ce470 Roll chromium_revision 80c35b98ae..ac89fd7158 (1180373:1182099) by chromium-webrtc-autoroll · 1 year, 7 months ago
  40. c252a40 Use layer/encode target resolution in DropDueToSize by Sergey Silkin · 1 year, 7 months ago
  41. 3b75c39 Revert "Cleanup calculating time between RTCP reports" by Danil Chapovalov · 1 year, 7 months ago
  42. da705ce Fix reclient download hook in DEPS. by Mirko Bonadei · 1 year, 7 months ago
  43. 490f0b8 Cleanup usage of csrcs in RtpSenderVideoFrameTransform by Danil Chapovalov · 1 year, 7 months ago
  44. 51f6a20 Roll chromium_revision 6b95b8aa08..80c35b98ae (1175249:1180373) by Björn Terelius · 1 year, 7 months ago
  45. d384456 Change FakeEncoder::kImplementationName from const char* to constexpr char* by Björn Terelius · 1 year, 8 months ago
  46. 86162d9 Implement setMetadata for receiver encoded video frames by Palak Agarwal · 1 year, 8 months ago
  47. 762f193 Cleanup calculating time between RTCP reports by Danil Chapovalov · 1 year, 8 months ago
  48. 240e783 Stop using invalid payload type 200 in audio/red unit test by Philipp Hancke · 1 year, 8 months ago
  49. 63dc56c Update WebRTC code version (2023-08-09T04:14:38). by webrtc-version-updater · 1 year, 8 months ago
  50. 8efd93d Encoder type agnostic resolution based fallback by Sergey Silkin · 1 year, 8 months ago
  51. 277fb3c Cleanup RtpSender unittest to use Timestamp instead of plain int by Danil Chapovalov · 1 year, 8 months ago
  52. 82e5f91 audio: fix handling of RED packets where the primary encoding is too large by Philipp Hancke · 1 year, 8 months ago
  53. 7bd90ba Remove templating from RtpTransceiver by Philipp Hancke · 1 year, 8 months ago
  54. c479e46 Update WebRTC code version (2023-08-08T04:02:52). by webrtc-version-updater · 1 year, 8 months ago
  55. 4f51b34 Add switch reason to verbose logging of SortAndSwitchConnection by Philipp Hancke · 1 year, 8 months ago
  56. c59be6d Roll chromium_revision 10080947c0..6b95b8aa08 (1174188:1175249) by Mirko Bonadei · 1 year, 8 months ago
  57. fc92d33 Add StunDictionary - patch 2 by Jonas Oreland · 1 year, 8 months ago
  58. 2bfa071 Add GN flag for an easier JNI generator upgrade. by Mirko Bonadei · 1 year, 8 months ago
  59. 34d82df Use ArrayView versions of SendRtp and SendRtcp by Harald Alvestrand · 1 year, 8 months ago
  60. 8dc5568 Reland "Skip NetworkTest if IPV4 is not available" by Christoffer Jansson · 1 year, 8 months ago
  61. c71bfcc Revert "Skip NetworkTest if IPV4 is not available" by Christoffer Jansson · 1 year, 8 months ago
  62. e2e0451 stats: implement fecSsrc on inbound-rtp by Philipp Hancke · 1 year, 8 months ago
  63. b17806a Add StunDictionary by Jonas Oreland · 1 year, 8 months ago
  64. b38d9d2 Add ArrayView versions of SendRtp and SendRtcp by Harald Alvestrand · 1 year, 8 months ago
  65. cad3aed Add setters to NetworkEmulationManager::SimulatedNetworkNode::Builder. by Mirko Bonadei · 1 year, 8 months ago
  66. 9b82b2f stats: implement RTX ssrc on inbound-rtp/outbound-rtp by Philipp Hancke · 1 year, 8 months ago
  67. 73ae5ca Skip NetworkTest if IPV4 is not available by Christoffer Jansson · 1 year, 8 months ago
  68. 9f3ea9d Increase concealement threshold for debug bots. by Henrik Boström · 1 year, 8 months ago
  69. 7f41b0b When simulating chains from VP9 codec specific info support first_active_layer > 0 by Danil Chapovalov · 1 year, 8 months ago
  70. fb5a4a3 build_ios_libs.py: Allow customizing the deployment target by Jordan Rose · 1 year, 8 months ago
  71. 33eff43 Update WebRTC code version (2023-08-03T04:03:48). by webrtc-version-updater · 1 year, 8 months ago
  72. ebb2bfd Delete deprecated member CodecSpecificInfoVP9::end_of_picture by Danil Chapovalov · 1 year, 8 months ago
  73. 50ca701 Mitigate scenario when VP9 HW encoder uses less spatial layers than configured by Danil Chapovalov · 1 year, 8 months ago
  74. 55b89a8 Rename cipher_suite to crypto_suite by Philipp Hancke · 1 year, 8 months ago
  75. 23fc2be Remove backwards compatibity #include by Mirko Bonadei · 1 year, 8 months ago
  76. 86a7969 Synchronize access to callbacks map by Sergey Silkin · 1 year, 8 months ago
  77. 9a09ed73 Cleanup RemoteBitrateEstimatorSingleStream by Danil Chapovalov · 1 year, 8 months ago
  78. fe2e7ea Reland "Add ARC enablement to WebRTC only if it isn't already present" by Christoffer Jansson · 1 year, 8 months ago
  79. a9d5141 Rename cricket::RtpParameters and derived classes by Philipp Hancke · 1 year, 8 months ago
  80. 787e326 Revert "Add ARC enablement to WebRTC only if it isn't already present" by Christoffer Jansson · 1 year, 8 months ago
  81. b396334 Update WebRTC code version (2023-08-01T04:05:22). by webrtc-version-updater · 1 year, 8 months ago
  82. 875cd32 Fix inconsistency with x-goog-max-bitrate and maxBitrate. by Henrik Boström · 1 year, 8 months ago
  83. 9613781 Add ARC enablement to WebRTC only if it isn't already present by Avi Drissman · 1 year, 8 months ago
  84. 4b87d7a Remove Codec template from RtpParameters and helper functions by Philipp Hancke · 1 year, 8 months ago
  85. 0d8b79e Reland "Extract HasIPv4Enabled/HasIPv6Enabled." by Mirko Bonadei · 1 year, 8 months ago
  86. d212551 Update RTC_LOG macro so it can be used when called from xyz::rtc namespaces by Joachim Reiersen · 1 year, 8 months ago
  87. 5eb5219 Correct typo from valee to value for color space definitions by Chunbo Hua · 1 year, 8 months ago
  88. bcb0b8e Update WebRTC code version (2023-07-29T04:02:10). by webrtc-version-updater · 1 year, 8 months ago
  89. a6c76d0 svc-av1: Fix to svc_e2e_tests by Marco Paniconi · 1 year, 8 months ago
  90. c787adc Remove decommisioned Pixel2 perf bots by Christoffer Jansson · 1 year, 8 months ago
  91. 4f4e989 In remote bitrate estimator pass packet using RtpPacketReceived class by Danil Chapovalov · 1 year, 8 months ago
  92. 6e937574 Update WebRTC code version (2023-07-28T04:04:17). by webrtc-version-updater · 1 year, 8 months ago
  93. 920abcc In RtpSenderVideo::UpdateConditionalRetransmit use typed time and framerate instead of plain ints by Danil Chapovalov · 1 year, 8 months ago
  94. b90cd91 Fix first encoding's maxBitrate being ignored when scalability is set. by Henrik Boström · 1 year, 8 months ago
  95. 6143ec9 [iOS testing] - Cut dependency from test module and app code. by Mirko Bonadei · 1 year, 8 months ago
  96. 3d7889a Revert "Extract HasIPv4Enabled/HasIPv6Enabled." by Mirko Bonadei · 1 year, 8 months ago
  97. db7a947 Update WebRTC code version (2023-07-27T04:02:37). by webrtc-version-updater · 1 year, 8 months ago
  98. 86cfe50 Extract HasIPv4Enabled/HasIPv6Enabled. by Mirko Bonadei · 1 year, 8 months ago
  99. 0145db4 Recreate the stream when switching from standard to legacy API. by Henrik Boström · 1 year, 8 months ago
  100. 9130431 Add possibility to set RTC_OBJC_TYPE_PREFIX from GN. by Mirko Bonadei · 1 year, 8 months ago