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  1. b4c1f2f6 Remove DegradedCall - To be submitted after 2024-07-01 by Per K · 5 weeks ago
  2. a49abbb Extend testing of prAnswer by Jonas Oreland · 5 weeks ago
  3. 2c637aa Register filter loop parameters' start position in VP9 frame header. by Emil Vardar · 5 weeks ago
  4. 427b712 Update WebRTC code version (2024-08-30T04:02:43). by webrtc-version-updater · 5 weeks ago
  5. e2fee23 Propagate Environment into RtpVideoStreamReceiver2 by Danil Chapovalov · 5 weeks ago
  6. 2f91bdc Declare corruption detection URI in RtpExtension by Fanny Linderborg · 5 weeks ago
  7. 058c005 Remove implicit `this` captures by Devon Loehr · 3 months ago
  8. 6ea1c96 Fix license metadata for spl_sqrt_floor, portaudio, sigslot by Andrew Grieve · 5 weeks ago
  9. a9ececd Only mute microphone while audio_unit is started. by Abby Yeh · 5 weeks ago
  10. 61a5214 In objc software video encoder wrappers expose functions to list supported scalability modes. by Danil Chapovalov · 5 weeks ago
  11. 41fffaa Fix requested_resolution bug where we get stuck with old restrictions. by Henrik Boström · 6 weeks ago
  12. 04cc4ce Deprecate NetEq::GetDecoderFormat and remove implementation. by Jakob Ivarsson · 5 weeks ago
  13. a99bf7f Delete deprecated AudioDecoderOpus::MakeAudioDecoder by Danil Chapovalov · 5 weeks ago
  14. f2487c0 [audio] Adjust the order of some definitions in audio_processing by Ho Cheung · 5 weeks ago
  15. 45af5a8 Update WebRTC code version (2024-08-29T04:04:15). by webrtc-version-updater · 5 weeks ago
  16. 2de37ef Roll chromium_revision c3a359139e..10ff7fa1e3 (1348059:1348232) by chromium-webrtc-autoroll · 5 weeks ago
  17. 2e10688 Roll chromium_revision ab7255fe8a..c3a359139e (1347197:1348059) by chromium-webrtc-autoroll · 5 weeks ago
  18. 44df591 Use NetEq::GetCurrentDecoderFormat in AcmReceiver. by Jakob Ivarsson · 5 weeks ago
  19. 4c862e7 Implement Create instead of MakeAudioDecoder in AudioDecoderFactory template by Danil Chapovalov · 7 weeks ago
  20. 32dd2ed Improve NetEq simulation frame size estimation. by Jakob Ivarsson · 6 weeks ago
  21. b6046ae Add NetEq API to get info about the current decoder. by Jakob Ivarsson · 6 weeks ago
  22. c22a1ae Fix linux_more_configs mb config. by Jeremy Leconte · 5 weeks ago
  23. 572280f Remove redundant mapping. by Emil Vardar · 6 weeks ago
  24. 54559d3 Fix formatting for corruption detection header explainer. by Erik Språng · 6 weeks ago
  25. b60f0ff Dont signal ReadyToSend in RtpTransport::SendPacket by Per K · 6 weeks ago
  26. 3f1e51d Aggregate and log corruption score. by Emil Vardar · 6 weeks ago
  27. 0a8204b Set libsrtp_build_boringssl to false in 'no_build_ssl'. by Jeremy Leconte · 6 weeks ago
  28. 6db0db5 Ensure TCPPort is notified of sent packets after reconnect by Per K · 6 weeks ago
  29. 6bed21c Extend objc RTCVideoCodecInfo to include scalability modes by Danil Chapovalov · 6 weeks ago
  30. 67ed656 Roll chromium_revision 30454db4a5..ab7255fe8a by Jeremy Leconte · 6 weeks ago
  31. c1a0d23 Update explainer text for corruption detection header extension. by Erik Språng · 6 weeks ago
  32. fd6f4b4 Add the corruption detection extension to RTPExtensionType by Fanny Linderborg · 6 weeks ago
  33. ad17756 Re-enable ApiCallDurationTest by Christoffer Jansson · 1 year, 10 months ago
  34. 90e0829 Add test for PR-Answer functionality by Harald Alvestrand · 6 weeks ago
  35. fd90f1a Add Security Critical field to README.chromium. by Mirko Bonadei · 6 weeks ago
  36. 06a49f0 build: add options to configure libsrtp for boringssl or other libraries by Philipp Hancke · 6 weeks ago
  37. a46f103 Re-enable iOS simulator from CQ and LKGR. by Jeremy Leconte · 6 weeks ago
  38. 1d6ad04 Update WebRTC code version (2024-08-27T04:03:09). by webrtc-version-updater · 6 weeks ago
  39. c6b556f Roll chromium_revision cb10943d61..30454db4a5 (1346705:1346833) by chromium-webrtc-autoroll · 6 weeks ago
  40. 84ce545 Reland "Add PT lookup function to JsepTransportController" by Harald Alvestrand · 6 weeks ago
  41. 37bd18f Roll chromium_revision ef49a3ba49..cb10943d61 (1344824:1346705) by Jeremy Leconte · 6 weeks ago
  42. c54c85f Attach Mid/Rid RTP header extension to pure padding packets by Danil Chapovalov · 6 weeks ago
  43. ab009c2 Refactor WebRTC self assignments in if clauses by Benjamin Williams · 6 weeks ago
  44. 9e86528 Reland "Add first iteration of PayloadTypePicker.SuggestPayloadType" by Harald Alvestrand · 6 weeks ago
  45. 0b91688 Mark EncodedImage::{Set, Is}AtTargetQuality() as deprecated by Johannes Kron · 7 weeks ago
  46. 5308652 Reland "Add recording of PT->Codec mappings on setting SDP for transport" by Harald Alvestrand · 6 weeks ago
  47. 7348f82 dcsctp: Re-add lost validating in test case by Victor Boivie · 6 weeks ago
  48. b4dc789 Fix incorrect target for hamcrest and aapt2 and add back icu4j by Christoffer Dewerin · 6 weeks ago
  49. fc9d0cf Remove deprecated DEPS by Christoffer Dewerin · 6 weeks ago
  50. 5b47a7a [rtc] Adjust the sequence of rtc::Network definition by Ho Cheung · 6 weeks ago
  51. 4f1dcd9 rename shadowing variable "offer" in unit test by Philipp Hancke · 6 weeks ago
  52. 08cdf77 Update WebRTC code version (2024-08-26T04:05:49). by webrtc-version-updater · 6 weeks ago
  53. d4e8e61 Update WebRTC code version (2024-08-25T04:07:14). by webrtc-version-updater · 6 weeks ago
  54. 5a6a8fe Update WebRTC code version (2024-08-24T04:06:47). by webrtc-version-updater · 6 weeks ago
  55. b923456 [jumbo] Add begin()/end() to EncodedImage[BufferInterface]. by Peter Kasting · 8 weeks ago
  56. 7e37e5f Use xcode 16 for iOS debug simulators + fix version by Christoffer Dewerin · 6 weeks ago
  57. 8771cf4 Allow gap on packet buffer fix with GFD by Fan Zhou · 6 weeks ago
  58. 6793f83 Revert "Add recording of PT->Codec mappings on setting SDP for transport" by Jonas Oreland · 6 weeks ago
  59. 43c0cf9 Support borrowing of underused audio bitrate. by Dan Tan · 6 weeks ago
  60. 2e376cd Revert "Add first iteration of PayloadTypePicker.SuggestPayloadType" by Jonas Oreland · 6 weeks ago
  61. 0e3a326 Revert "Add PT lookup function to JsepTransportController" by Jonas Oreland · 6 weeks ago
  62. a691309 Update WebRTC code version (2024-08-23T04:07:24). by webrtc-version-updater · 6 weeks ago
  63. b31ade3 stun/turn: suppress icecandidateerror for incompatible address family by Philipp Hancke · 8 weeks ago
  64. d178532 Add PT lookup function to JsepTransportController by Harald Alvestrand · 6 weeks ago
  65. e2869de Add first iteration of PayloadTypePicker.SuggestPayloadType by Harald Alvestrand · 6 weeks ago
  66. abb6388 remove deprecated <codecvt> by Helmut Januschka · 6 weeks ago
  67. c03edf6 Add missing includes and remove unused includes by Fanny Linderborg · 6 weeks ago
  68. 5e70fd3 fix of a compilation error in Visual Studio 2022 due to a warning C4244. by Denis Genestier · 6 weeks ago
  69. bbd1467 Update WebRTC code version (2024-08-22T04:09:17). by webrtc-version-updater · 6 weeks ago
  70. 69f7916 Roll chromium_revision bb795520d5..ef49a3ba49 (1344631:1344824) by chromium-webrtc-autoroll · 6 weeks ago
  71. da72666 Support standard simulcast with `requested_resolution`. by Henrik Boström · 6 weeks ago
  72. 90bd500 Roll chromium_revision 3484724f00..bb795520d5 (1344182:1344631) by chromium-webrtc-autoroll · 6 weeks ago
  73. 1f26102 Adjust fuzzers group to respect build variables by Danil Chapovalov · 6 weeks ago
  74. 1571723 Add recording of PT->Codec mappings on setting SDP for transport by Harald Alvestrand · 7 weeks ago
  75. fea60ef Fixed issue with missing network interfaces on iOS by Corby Hoback · 7 weeks ago
  76. 2efd4fd Update WebRTC code version (2024-08-21T04:04:12). by webrtc-version-updater · 6 weeks ago
  77. c1a7827 Roll chromium_revision f031fbef87..3484724f00 (1343352:1344182) by chromium-webrtc-autoroll · 7 weeks ago
  78. f009e38 Fix AudioSendStream reconfigure - stop processing during unconfigured state by Guy Hershenbaum · 7 weeks ago
  79. f2d3136 Reland "Include fuzzers to build by default" by Danil Chapovalov · 7 weeks ago
  80. a6fad74 Add missing optional deps by Christoffer Dewerin · 7 weeks ago
  81. c478ff6 Sync bot configs with Chromium. by Jeremy Leconte · 7 weeks ago
  82. f654998 Add opus_audio_decoder_factory to the static library. by Patrick Reynolds · 7 weeks ago
  83. 2cfedb2 Remove vestiges of GTURN by Philipp Hancke · 8 weeks ago
  84. 37c406a Clean up 32 byte mid support from the demuxer by Philipp Hancke · 7 weeks ago
  85. b1ffa9b Export SdpTypeFromString by Harald Alvestrand · 7 weeks ago
  86. a6186b2 Add helper that generate filter data given a captured and an encoded frame by Fanny Linderborg · 7 weeks ago
  87. 52e4624 Apply include-cleaner to api/voip by Dor Hen · 7 weeks ago
  88. f4dd393 Initial implementation of PayloadTypePicker by Harald Alvestrand · 7 weeks ago
  89. ecbba45 Roll chromium_revision af7f3f9345..f031fbef87 (1339036:1343352) by Mirko Bonadei · 7 weeks ago
  90. 06391de Revert "Include fuzzers to build by default" by Ilya Nikolaevskiy · 7 weeks ago
  91. d7d940e Apply include-cleaner to api/video_codecs by Dor Hen · 7 weeks ago
  92. 9fb83a1 Apply include-cleaner to api/video by Dor Hen · 7 weeks ago
  93. 668e905 Add support for determining which pixels to sample by Fanny Linderborg · 7 weeks ago
  94. c146587 Update WebRTC code version (2024-08-19T04:06:43). by webrtc-version-updater · 7 weeks ago
  95. 13b327b srtp: demonstrate wraparound with loss decryption failure by Philipp Hancke · 7 weeks ago
  96. adb224b dcsctp: Simplify congestion control algorithm by Victor Boivie · 7 weeks ago
  97. de972c1 Apply include-cleaner to api/units by Dor Hen · 7 weeks ago
  98. a09878c Update WebRTC code version (2024-08-17T04:08:01). by webrtc-version-updater · 7 weeks ago
  99. e777c65 Include fuzzers to build by default by Danil Chapovalov · 7 weeks ago
  100. 24823c5 Add AudioDecoderOpus::MakeAudioDecoder overload taking Environment by Danil Chapovalov · 7 weeks ago