Google Git
Sign in
webrtc / src.git / 8401f56a5429401de9c9e0b8255315514d769a8c
  1. 8401f56 Add fieldtrials WebRTC-QCM-Static-{AV1, VP8, VP9} by Johannes Kron · 6 months ago
  2. 3d60f25 Fix gtest/gmock includes in apply-include-cleaner script. by Jeremy Leconte · 6 months ago
  3. 3881cb6 PipeWire camera: make member variable with the PipeWire status updated by Jan Grulich · 6 months ago
  4. 863c2c9 Roll chromium_revision b975bdde27..cae6b92cf5 (1348475:1349874) by Björn Terelius · 6 months ago
  5. 6e072e6 Rename is_key_frame to communicate_upper_bits in FrameInstrumentation*Data by Fanny Linderborg · 6 months ago
  6. 843a317 Fix requested_resolution orientation assumption in OnSinkWants(). by Henrik Boström · 6 months ago
  7. d34f3b8 Remove more self assignment in if-clause by Bjorn Terelius · 6 months ago
  8. 93c9aa1 Apply include-cleaner to call/ by Harald Alvestrand · 6 months ago
  9. 5eb8588 Move FrameInstrumentation*Data structs to common_video by Fanny Linderborg · 6 months ago
  10. a82eb4e Remove self assignment in if-clause by Björn Terelius · 6 months ago
  11. 55ed950 Propagate corruption score to VideoReceiverInfo. by Emil Vardar · 6 months ago
  12. 99874e7 Update WebRTC code version (2024-09-03T04:04:21). by webrtc-version-updater · 6 months ago
  13. 77eba46 Adding ChannelStatistics Logs by Daniel · 6 months ago
  14. 86251a0 rewrite SSLInfoCallback logging by Philipp Hancke · 6 months ago
  15. 04ab497 Review abseil-in-webrtc for freshness by Danil Chapovalov · 6 months ago
  16. 86ac1df Fix libsrtp openssl build by Philipp Hancke · 6 months ago
  17. 9212f09 Update Abseil instructions for absl::optional by Florent Castelli · 6 months ago
  18. 8037fc6 Migrate absl::optional to std::optional by Florent Castelli · 6 months ago
  19. 787b907 Update freshness of the h-cc-pairs section of the style guide by Danil Chapovalov · 6 months ago
  20. 4e41db2 Propagate Environment to RtpRtcp module in FlexfecReceiver by Danil Chapovalov · 6 months ago
  21. 164b3b3 Introduce ModuleRtpRtcpImpl factory that accepts Environment by Danil Chapovalov · 6 months ago
  22. cb00e16 Revert "Enable 'iwyu_verifier' bot." by Jeremy Leconte · 6 months ago
  23. af7155e Propagate Environment to video RtpRtcp modules by Danil Chapovalov · 6 months ago
  24. 5a92ddb Updates review date in ADM g3doc. by henrika · 6 months ago
  25. 24366b0 Propagate Environment to audio RtpRtcp modules by Danil Chapovalov · 6 months ago
  26. 0b4b5b0 Use AV1E_SET_AUTO_TILES by Sergey Silkin · 6 months ago
  27. a4cf34d Enable 'iwyu_verifier' bot. by Jeremy Leconte · 6 months ago
  28. dd86c95 Update WebRTC code version (2024-09-02T04:06:36). by webrtc-version-updater · 6 months ago
  29. 177788f Update WebRTC code version (2024-09-01T04:05:33). by webrtc-version-updater · 7 months ago
  30. 91eacf3 Update WebRTC code version (2024-08-31T04:05:52). by webrtc-version-updater · 7 months ago
  31. 738abe0 Upgrade ios version used for perf tests. by Jeremy Leconte · 7 months ago
  32. c4d7493 Add some flags to 'apply-include-cleaner'. by Jeremy Leconte · 7 months ago
  33. d385af5 Introduce ModuleRtpRtcpImpl2 constructor that accepts Environment by Danil Chapovalov · 7 months ago
  34. 058972f Make LAYER_DROP and max_consec_drop=2 to be default settings by Sergey Silkin · 7 months ago
  35. b5f4006 Inject field trials in NetEqTest instead of setting global. by Jakob Ivarsson · 7 months ago
  36. 8d478dd Roll chromium_revision 10ff7fa1e3..b975bdde27 by Jeremy Leconte · 7 months ago
  37. b4c1f2f6 Remove DegradedCall - To be submitted after 2024-07-01 by Per K · 7 months ago
  38. a49abbb Extend testing of prAnswer by Jonas Oreland · 7 months ago
  39. 2c637aa Register filter loop parameters' start position in VP9 frame header. by Emil Vardar · 7 months ago
  40. 427b712 Update WebRTC code version (2024-08-30T04:02:43). by webrtc-version-updater · 7 months ago
  41. e2fee23 Propagate Environment into RtpVideoStreamReceiver2 by Danil Chapovalov · 7 months ago
  42. 2f91bdc Declare corruption detection URI in RtpExtension by Fanny Linderborg · 7 months ago
  43. 058c005 Remove implicit `this` captures by Devon Loehr · 7 months ago
  44. 6ea1c96 Fix license metadata for spl_sqrt_floor, portaudio, sigslot by Andrew Grieve · 7 months ago
  45. a9ececd Only mute microphone while audio_unit is started. by Abby Yeh · 7 months ago
  46. 61a5214 In objc software video encoder wrappers expose functions to list supported scalability modes. by Danil Chapovalov · 7 months ago
  47. 41fffaa Fix requested_resolution bug where we get stuck with old restrictions. by Henrik Boström · 7 months ago
  48. 04cc4ce Deprecate NetEq::GetDecoderFormat and remove implementation. by Jakob Ivarsson · 7 months ago
  49. a99bf7f Delete deprecated AudioDecoderOpus::MakeAudioDecoder by Danil Chapovalov · 7 months ago
  50. f2487c0 [audio] Adjust the order of some definitions in audio_processing by Ho Cheung · 7 months ago
  51. 45af5a8 Update WebRTC code version (2024-08-29T04:04:15). by webrtc-version-updater · 7 months ago
  52. 2de37ef Roll chromium_revision c3a359139e..10ff7fa1e3 (1348059:1348232) by chromium-webrtc-autoroll · 7 months ago
  53. 2e10688 Roll chromium_revision ab7255fe8a..c3a359139e (1347197:1348059) by chromium-webrtc-autoroll · 7 months ago
  54. 44df591 Use NetEq::GetCurrentDecoderFormat in AcmReceiver. by Jakob Ivarsson · 7 months ago
  55. 4c862e7 Implement Create instead of MakeAudioDecoder in AudioDecoderFactory template by Danil Chapovalov · 7 months ago
  56. 32dd2ed Improve NetEq simulation frame size estimation. by Jakob Ivarsson · 7 months ago
  57. b6046ae Add NetEq API to get info about the current decoder. by Jakob Ivarsson · 7 months ago
  58. c22a1ae Fix linux_more_configs mb config. by Jeremy Leconte · 7 months ago
  59. 572280f Remove redundant mapping. by Emil Vardar · 7 months ago
  60. 54559d3 Fix formatting for corruption detection header explainer. by Erik Språng · 7 months ago
  61. b60f0ff Dont signal ReadyToSend in RtpTransport::SendPacket by Per K · 7 months ago
  62. 3f1e51d Aggregate and log corruption score. by Emil Vardar · 7 months ago
  63. 0a8204b Set libsrtp_build_boringssl to false in 'no_build_ssl'. by Jeremy Leconte · 7 months ago
  64. 6db0db5 Ensure TCPPort is notified of sent packets after reconnect by Per K · 7 months ago
  65. 6bed21c Extend objc RTCVideoCodecInfo to include scalability modes by Danil Chapovalov · 7 months ago
  66. 67ed656 Roll chromium_revision 30454db4a5..ab7255fe8a by Jeremy Leconte · 7 months ago
  67. c1a0d23 Update explainer text for corruption detection header extension. by Erik Språng · 7 months ago
  68. fd6f4b4 Add the corruption detection extension to RTPExtensionType by Fanny Linderborg · 7 months ago
  69. ad17756 Re-enable ApiCallDurationTest by Christoffer Jansson · 7 months ago
  70. 90e0829 Add test for PR-Answer functionality by Harald Alvestrand · 7 months ago
  71. fd90f1a Add Security Critical field to README.chromium. by Mirko Bonadei · 7 months ago
  72. 06a49f0 build: add options to configure libsrtp for boringssl or other libraries by Philipp Hancke · 7 months ago
  73. a46f103 Re-enable iOS simulator from CQ and LKGR. by Jeremy Leconte · 7 months ago
  74. 1d6ad04 Update WebRTC code version (2024-08-27T04:03:09). by webrtc-version-updater · 7 months ago
  75. c6b556f Roll chromium_revision cb10943d61..30454db4a5 (1346705:1346833) by chromium-webrtc-autoroll · 7 months ago
  76. 84ce545 Reland "Add PT lookup function to JsepTransportController" by Harald Alvestrand · 7 months ago
  77. 37bd18f Roll chromium_revision ef49a3ba49..cb10943d61 (1344824:1346705) by Jeremy Leconte · 7 months ago
  78. c54c85f Attach Mid/Rid RTP header extension to pure padding packets by Danil Chapovalov · 7 months ago
  79. ab009c2 Refactor WebRTC self assignments in if clauses by Benjamin Williams · 7 months ago
  80. 9e86528 Reland "Add first iteration of PayloadTypePicker.SuggestPayloadType" by Harald Alvestrand · 7 months ago
  81. 0b91688 Mark EncodedImage::{Set, Is}AtTargetQuality() as deprecated by Johannes Kron · 7 months ago
  82. 5308652 Reland "Add recording of PT->Codec mappings on setting SDP for transport" by Harald Alvestrand · 7 months ago
  83. 7348f82 dcsctp: Re-add lost validating in test case by Victor Boivie · 7 months ago
  84. b4dc789 Fix incorrect target for hamcrest and aapt2 and add back icu4j by Christoffer Dewerin · 7 months ago
  85. fc9d0cf Remove deprecated DEPS by Christoffer Dewerin · 7 months ago
  86. 5b47a7a [rtc] Adjust the sequence of rtc::Network definition by Ho Cheung · 7 months ago
  87. 4f1dcd9 rename shadowing variable "offer" in unit test by Philipp Hancke · 7 months ago
  88. 08cdf77 Update WebRTC code version (2024-08-26T04:05:49). by webrtc-version-updater · 7 months ago
  89. d4e8e61 Update WebRTC code version (2024-08-25T04:07:14). by webrtc-version-updater · 7 months ago
  90. 5a6a8fe Update WebRTC code version (2024-08-24T04:06:47). by webrtc-version-updater · 7 months ago
  91. b923456 [jumbo] Add begin()/end() to EncodedImage[BufferInterface]. by Peter Kasting · 7 months ago
  92. 7e37e5f Use xcode 16 for iOS debug simulators + fix version by Christoffer Dewerin · 7 months ago
  93. 8771cf4 Allow gap on packet buffer fix with GFD by Fan Zhou · 7 months ago
  94. 6793f83 Revert "Add recording of PT->Codec mappings on setting SDP for transport" by Jonas Oreland · 7 months ago
  95. 43c0cf9 Support borrowing of underused audio bitrate. by Dan Tan · 7 months ago
  96. 2e376cd Revert "Add first iteration of PayloadTypePicker.SuggestPayloadType" by Jonas Oreland · 7 months ago
  97. 0e3a326 Revert "Add PT lookup function to JsepTransportController" by Jonas Oreland · 7 months ago
  98. a691309 Update WebRTC code version (2024-08-23T04:07:24). by webrtc-version-updater · 7 months ago
  99. b31ade3 stun/turn: suppress icecandidateerror for incompatible address family by Philipp Hancke · 7 months ago
  100. d178532 Add PT lookup function to JsepTransportController by Harald Alvestrand · 7 months ago