1. dcc1534 Delete rtc::TaskQueue by Danil Chapovalov · 11 months ago
  2. bbff58d Introduce "well-known" SdpVideoFormat codecs by Philipp Hancke · 11 months ago
  3. 2825f0a Update WebRTC code version (2024-02-28T04:11:45). by webrtc-version-updater · 11 months ago
  4. 55d33f1 Roll chromium_revision aa1c478fe0..b2ff29df8d (1266091:1266241) by chromium-webrtc-autoroll · 11 months ago
  5. c27339f Roll chromium_revision 511e819ce4..aa1c478fe0 (1265924:1266091) by chromium-webrtc-autoroll · 11 months ago
  6. c3d937b In RtpFrameReferenceFinder discard frames with too large spatial id by Danil Chapovalov · 11 months ago
  7. 8cd50cb Delete PacketTransportInternal::SignalReadPacket by Per K · 11 months ago
  8. a8c4727 Roll chromium_revision a44e4a5e32..511e819ce4 (1265726:1265924) by chromium-webrtc-autoroll · 11 months ago
  9. 334e913 Video capture PipeWire: add support for DMABuf buffer type by Jan Grulich · 12 months ago
  10. 9e0bf9b Propagate rtc::ReceivedPacket further in RtpTransport by Per K · 11 months ago
  11. f7682f0 Delete RTCWrappedNativeVideoDecoder by Danil Chapovalov · 11 months ago
  12. f4aadf3 Change RtpTransport and DsctTransport to receives packets through ReceivedPacketCallback by Per K · 11 months ago
  13. 524a06b Change BuiltInNetworkBehaviorConfig.loss_percent to double by Florent Castelli · 11 months ago
  14. e39f6fd Roll chromium_revision af00cd208a..a44e4a5e32 (1264869:1265726) by chromium-webrtc-autoroll · 11 months ago
  15. 2514dd7 Increase WebRTC default receive buffer size to 1MB. by Erik Språng · 11 months ago
  16. 7a00882 Disable checks for googletest for WebRTC by Christoffer Dewerin · 11 months ago
  17. 51532fd Test handling of rejected m-lines without transport description by Philipp Hancke · 11 months ago
  18. 058bfe3 PipeWire capturer: set capturer as failed when session is closed by Jan Grulich · 11 months ago
  19. 8e137d0 Replace use of SignalReadPacket in DtlsTransport by Per K · 11 months ago
  20. 09e81cc Update WebRTC code version (2024-02-27T04:11:00). by webrtc-version-updater · 11 months ago
  21. c32a509 Export h.265 bitstream parser APIs. by Qiu Jianlin · 11 months ago
  22. 91ebd5f Add missing absl::optional includes by Danil Chapovalov · 11 months ago
  23. 7098d11 Revert "p2p: separate ICE tie breaker and foundation seed" by Christoffer Dewerin · 11 months ago
  24. 1461363 Make PeerConnectionInteface methods pure virtual. by Per K · 11 months ago
  25. 5261619 Remove rtc::TaskQueue in AudioDeviceBuffer by Danil Chapovalov · 1 year ago
  26. 3f7566a Cleanup rtc::TaskQueue in AsyncAudioProcessing by Danil Chapovalov · 1 year ago
  27. 97df932 Remove multiplex codec. by Markus Handell · 11 months ago
  28. db2f52b Reland "Make setCodecPreferences only look at receive codecs" by Philipp Hancke · 11 months ago
  29. d99499a p2p: separate ICE tie breaker and foundation seed by Philipp Hancke · 11 months ago
  30. 7c5f9cf Add nonstandard x-google-per-layer-pli fmtp for enabling per-layer keyFrames in response to PLIs by Philipp Hancke · 11 months ago
  31. 3fff83d Update WebRTC code version (2024-02-26T04:03:26). by webrtc-version-updater · 11 months ago
  32. 7391ecf Add directory for ChromiumOS specific tools by Li-Yu Yu · 11 months ago
  33. c7a4b2a Change internal candidate type to enum by Tommi · 11 months ago
  34. a021d99 Move implementation of PacketTransportInternal::RegisterReceivedPacketCallback by Per K · 11 months ago
  35. d0491a3 Update WebRTC code version (2024-02-25T04:13:05). by webrtc-version-updater · 11 months ago
  36. 4438dd3 Roll chromium_revision 1f0d2a10bd..af00cd208a (1264465:1264869) by chromium-webrtc-autoroll · 11 months ago
  37. 3fe69c5 Update MockPeerConnectionInterface and fake with missing methods. by Per K · 11 months ago
  38. 3f0d399 Update WebRTC code version (2024-02-24T04:01:40). by webrtc-version-updater · 11 months ago
  39. 8adb080 Roll chromium_revision a4279f2842..1f0d2a10bd (1259805:1264465) by Mirko Bonadei · 11 months ago
  40. ff0c960 Introduce PacketTransportInternal::NotifyPacketReceived by Per K · 11 months ago
  41. 845d6be Fix handling of rejected m-lines without transport description by Philipp Hancke · 11 months ago
  42. efbfc40 Demote RTC_CHECK for sctp_mid() to RTC_LOG(LS_ERROR) if unavailable by Tommi · 11 months ago
  43. 4f63ea4 Deprecate VP8Decoder::Create by Danil Chapovalov · 11 months ago
  44. bf20cf8 Implement Create instead of CreateVideoDecoder in remaining test VideoDecoderFactories by Danil Chapovalov · 11 months ago
  45. de3b1cd Revert "Make PeerConnectionInteface methods pure virtual." by Mirko Bonadei · 11 months ago
  46. 2fafcec Remove unused AsyncPacketSocket::NotifyPacketReceived by Per K · 11 months ago
  47. 2679468 Update WebRTC code version (2024-02-23T04:03:45). by webrtc-version-updater · 11 months ago
  48. 4a97488 Rename AudioLevel to AudioLevelExtension in rtp_header_extensions.h by Joachim Reiersen · 11 months ago
  49. b2f827c Remove extra trait to read only mandatory part of the dependency descriptor by Danil Chapovalov · 11 months ago
  50. d440358 Dont create RTX receive stream before media SSRC is known by Per K · 11 months ago
  51. 179444c Pass webrtc::Environment through InternalDecoderFactory::Create by Danil Chapovalov · 11 months ago
  52. 9cd5c3f Pass webrtc::Environment through VideoDecoderFactoryTemplate::Create by Danil Chapovalov · 11 months ago
  53. bff6858 Make PeerConnectionInteface methods pure virtual. by Per K · 11 months ago
  54. e166dec Update WebRTC code version (2024-02-22T04:03:12). by webrtc-version-updater · 11 months ago
  55. efea7bb Ignore WebRTC-LibvpxVp9Encoder-SvcFrameDropConfig in VP9 fuzzer by Sergey Silkin · 11 months ago
  56. fa01e3f Delete deprecated variant of the VideoDecoderSoftwareFallbackWrapper without Environment by Danil Chapovalov · 11 months ago
  57. dc6a001 Update and open `WebRtcAudioUtils` by Zoé Lepaul · 11 months ago
  58. 56d3cf0 Make VideoDecoderFactory::CreateVideoDecoder private by Danil Chapovalov · 11 months ago
  59. 74a4038 Limit max frame size in DAV1D decoder by Sergey Silkin · 11 months ago
  60. 88a8e44 Remove nonexempt field trials from POLICY_EXEMPT_FIELD_TRIALS by Emil Lundmark · 11 months ago
  61. f2431a9 Move ComputeFoundation to Candidate. by Tommi · 11 months ago
  62. 6bc92ce Update WebRTC code version (2024-02-21T04:01:38). by webrtc-version-updater · 11 months ago
  63. 9384bb2 Document how codec comparisons happen by Philipp Hancke · 11 months ago
  64. bc9af41 Sync definitions of IsSameCodecSpecific by Philipp Hancke · 11 months ago
  65. 2a3db31 Disable Android specific threading settings in libvpx VP8 encoder by Sergey Silkin · 11 months ago
  66. 0e9b8fe Compare codec number of channels and clockrate in MatchesRtpCodec for RTX too by Philipp Hancke · 11 months ago
  67. 41c44cd Add some comments for H265 RTP depacketizer. by Jianjun Zhu · 11 months ago
  68. 36d5eec Propagate webrtc::Environment through objc VideoDecoderFactory by Danil Chapovalov · 11 months ago
  69. 8c38eed Revert "Add google-java-format to DEPS." by Mirko Bonadei · 11 months ago
  70. 5437df3 Update WebRTC code version (2024-02-20T04:03:59). by webrtc-version-updater · 11 months ago
  71. dba3fd6 Correctly mark video frame type for FU packets. by Jianjun Zhu · 12 months ago
  72. 0355f45 Use Environment propagated through android sdk by Danil Chapovalov · 11 months ago
  73. bde80e3 Deprecate Candidate::set_id(), offer generate_id() instead by Tommi · 11 months ago
  74. a75459d Add google-java-format to DEPS. by Mirko Bonadei · 11 months ago
  75. d99da2c Allow to use propagated field trials in VideoDecoderSoftwareFallbackWrapper by Danil Chapovalov · 11 months ago
  76. 2bfb5db dcsctp: Update zero checksum option to v-06 draft by Victor Boivie · 12 months ago
  77. c49da7a Update WebRTC code version (2024-02-18T04:06:34). by webrtc-version-updater · 11 months ago
  78. 0ba663c Change a few uses of Candidate::type() to Candidate::type_name() by Tommi · 11 months ago
  79. 600503a Update WebRTC code version (2024-02-17T04:11:12). by webrtc-version-updater · 11 months ago
  80. 052bc3a Field trial to control SVC frame dropping mode in libvpx VP9 encoder by Sergey Silkin · 11 months ago
  81. 54d9cd0 Update iOS dimension to have more machines available. by Jeremy Leconte · 11 months ago
  82. 8bfc3e9 Fix variant name for iOS simulator 17.4. by Jeremy Leconte · 11 months ago
  83. 85b405b Switch all Linux tasks from Focal to Jammy (except *san). by Mirko Bonadei · 11 months ago
  84. 1b52d56 Fix generate_buildbot_json and switch to ios_runtime_cache_17_4. by Mirko Bonadei · 11 months ago
  85. 6596134 Update WebRTC code version (2024-02-16T04:14:44). by webrtc-version-updater · 11 months ago
  86. 62cbdce Allow getDisplayMedia capture HDR monitor. by Sunggook Chue · 11 months ago
  87. 7e0bd7a Reland "Add HEVC support for h264_packet_buffer." by Jianjun Zhu · 11 months ago
  88. 4636419 Propagate webrtc::Environment through MultiplexDecoderAdapter by Danil Chapovalov · 12 months ago
  89. ce1271a Do not guard AV1 SVC tests on VP9 define by Philipp Hancke · 11 months ago
  90. 2eee89e Cleanup webrtc::Environment propagation through java wrappers by Danil Chapovalov · 12 months ago
  91. 45242ad Add field trial property alloc_current_bwe_limit by Per K · 11 months ago
  92. 6a82366 Reject SDP with duplicate msid lines by Harald Alvestrand · 11 months ago
  93. 611f21d Revert "Add HEVC support for h264_packet_buffer." by Mirko Bonadei · 11 months ago
  94. b158537 Allow to propagate field trials into Vp8 Decoder by Danil Chapovalov · 12 months ago
  95. f7b22c6 Add Candidate::type_name() by Tommi · 11 months ago
  96. a265544 Add HEVC support for h264_packet_buffer. by Jianjun Zhu · 12 months ago
  97. 4efc830 Provide test output path with `OutputPathWithRandomDirectory` 1/n by Dor Hen · 11 months ago
  98. 3e9e4e7 Update WebRTC code version (2024-02-15T04:07:08). by webrtc-version-updater · 11 months ago
  99. 414c942 Reland "Extends WebRTC logs for software encoder fallback" by henrika · 11 months ago
  100. 23c32da Revert "Extends WebRTC logs for software encoder fallback" by Mirko Bonadei · 11 months ago