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webrtc / src / / bd5874accf0faef28439ad32c1414bda8c9c25dc
  1. bd5874a Remove inter-arrival delay mode from DelayManager. by Jakob Ivarsson · 5 years ago
  2. 57218b4 Delete RtpDepacketizer::Create factory by Danil Chapovalov · 5 years ago
  3. 0b3a6e3 Make RTCAudioSession accessible to Swift. by Joe Chen · 5 years ago
  4. 0c20213 Roll chromium_revision 170b5c3c75..086dd4c442 (729099:729202) by chromium-webrtc-autoroll · 5 years ago
  5. 43e62fc Fix Heap-use-after-free. by Markus Handell · 5 years ago
  6. 2e6ca11 Roll chromium_revision 630d591cce..170b5c3c75 (728985:729099) by chromium-webrtc-autoroll · 5 years ago
  7. 1c1b99e Revert "Delete RtpPayloadParams::SetDependenciesVp8Deprecated as unused" by Qingsi Wang · 5 years ago
  8. 014c02f Roll chromium_revision c2f0727826..630d591cce (728843:728985) by chromium-webrtc-autoroll · 5 years ago
  9. 4f40fa5 Implement RTCOutboundRtpStreamStats::remoteId. by Henrik Boström · 5 years ago
  10. 266021d Add support for DegradationPreference in Android SDK by Florent Castelli · 5 years ago
  11. 75b5897 Allow nil degradationPreference in RTCRtpParameters. by Mirta Dvornicic · 5 years ago
  12. 0aa7e37 Add include of <cstdlib> by Niels Möller · 5 years ago
  13. 382cc6d Add incomplete ResourceAdaptationModuleInterface. by Henrik Boström · 5 years ago
  14. 1b4e4bf Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing. by Danil Chapovalov · 5 years ago
  15. ec7b36c Added exception handling to EncodedImage's release callback. by Raman Budny · 5 years ago
  16. 4cdd7fb Add degradationPreference to RTCRtpParameters in ObjC SDK. by Mirta Dvornicic · 5 years ago
  17. b08882b Refactor out VideoStreamEncoder's overuse logic to separate module. by Henrik Boström · 5 years ago
  18. 29e14e6 Delete compatibility defines for unused error codes by Niels Möller · 5 years ago
  19. 499de2d Fix tiny race condition when Vp9NonFlexMode_* tests were failing. by Yves Gerey · 5 years ago
  20. 27f4d32 Add VideoRtpDepacketizerGeneric by Danil Chapovalov · 5 years ago
  21. dc7fe40 Delete RtpPayloadParams::SetDependenciesVp8Deprecated as unused by Danil Chapovalov · 5 years ago
  22. 4442871 Adds srte to api/test/OWNERS. by Sebastian Jansson · 5 years ago
  23. 28b2184 Roll chromium_revision 2f5cb46774..c2f0727826 (728648:728843) by chromium-webrtc-autoroll · 5 years ago
  24. dc80017 Mark TCP connections that fail initialization as failed. by Harald Alvestrand · 5 years ago
  25. 1e63b9b Roll chromium_revision e0ea7be7d1..2f5cb46774 (728381:728648) by chromium-webrtc-autoroll · 5 years ago
  26. 434dfa7 [Android SDK] Add cmdline-tools cipd package to DEPS. by Yves Gerey · 5 years ago
  27. eb3beb8 Revert "Replace the ExperimentalAgc config with the new config format" by Yves Gerey · 5 years ago
  28. c33e491 Prevent 'use after free' by waiting for all queued tasks to be processed. by Yves Gerey · 5 years ago
  29. e8d54b9 Provide a default async resolver factory if none injected to PeerConnection. by Qingsi Wang · 5 years ago
  30. cfe75c1 Roll chromium_revision 236f912c9d..e0ea7be7d1 (728281:728381) by chromium-webrtc-autoroll · 5 years ago
  31. f3aa632 Replace the ExperimentalAgc config with the new config format by Per Åhgren · 5 years ago
  32. 1f74351 Roll chromium_revision 9427b39371..236f912c9d (728171:728281) by chromium-webrtc-autoroll · 5 years ago
  33. 2257c08 [Cleanup/Optim] Pass IPAddress by const reference. by Yves Gerey · 5 years ago
  34. d572748 Run delay tasks on time when using GlobalSimulatedTimeController. by philipel · 5 years ago
  35. 3828c30 Delete unused member BasicPortAllocator::allow_tcp_listen_ by Niels Möller · 5 years ago
  36. 12e319a Merge the preambles of the ProcessStream implementations by Sam Zackrisson · 5 years ago
  37. 0f14db2 Reduce for reallocations the pre-amplifier and high-pass filter by Per Åhgren · 5 years ago
  38. df344bb Roll chromium_revision d4992c6f92..9427b39371 (728071:728171) by chromium-webrtc-autoroll · 5 years ago
  39. 873610c Fix updating degradation preference in SetRtpParameters. by Mirta Dvornicic · 5 years ago
  40. 308bc64 Remove one acquisition of capture lock in APM AudioFrame API by Sam Zackrisson · 5 years ago
  41. 2bd85ab Avoid AGC2 runtime allocation and activate it on demand by Per Åhgren · 5 years ago
  42. 83ee982 Delete p2p/base/packet_transport_interface.h by Niels Möller · 5 years ago
  43. c073471 APM: Move the TransientSuppression activation to the apm_config by Per Åhgren · 5 years ago
  44. 045c36d Roll chromium_revision 81693dc9aa..d4992c6f92 (727940:728071) by chromium-webrtc-autoroll · 5 years ago
  45. 65bbcab [Android] Replace java_files with sources by Natalie Chouinard · 5 years ago
  46. 29fec66 AEC3: Remove metrics that are not used for analysis by Per Åhgren · 5 years ago
  47. d2fb5f5 Fixes WebRtcAudioTrack crash while stopping by Alex Narest · 5 years ago
  48. a688d11 Return unavailable rate rather than garbage value. by Yves Gerey · 5 years ago
  49. cf4c872 APM: Make the GetStatistics call independent of the locks in APM by Per Åhgren · 5 years ago
  50. a43777d Roll chromium_revision 2d48822491..81693dc9aa (727839:727940) by chromium-webrtc-autoroll · 5 years ago
  51. 1c34ca7 Roll chromium_revision f19d6cb823..2d48822491 (727734:727839) by chromium-webrtc-autoroll · 5 years ago
  52. 9d2c2db Roll chromium_revision 9986f2241a..f19d6cb823 (727633:727734) by chromium-webrtc-autoroll · 5 years ago
  53. 8ac7912 Roll chromium_revision d814fc7ea9..9986f2241a (727531:727633) by chromium-webrtc-autoroll · 5 years ago
  54. f2dc059 Roll chromium_revision 6f7e5e79ce..d814fc7ea9 (727038:727531) by chromium-webrtc-autoroll · 5 years ago
  55. 26762d0 Add video codec AV1 to the deprecated android decoder/encoder wrappers by Danil Chapovalov · 5 years ago
  56. 38a55a0 Roll chromium_revision 937a78378f..6f7e5e79ce (726883:727038) by chromium-webrtc-autoroll · 5 years ago
  57. a79fc59 Roll chromium_revision e4c6d7fe53..937a78378f (726742:726883) by Yves Gerey · 5 years ago
  58. b5159fe Revert "Reland "Reland "Distinguish between send and receive video codecs""" by Olga Sharonova · 5 years ago
  59. f6b875c Fixed crash on iOS13, methods beginGeneratingDeviceOrientationNotifications and endGeneratingDeviceOrientationNotifications. by Andrey Efremov · 5 years ago
  60. 4e64e60 Reland "Reland "Distinguish between send and receive video codecs"" by Johannes Kron · 5 years ago
  61. 23df143 Roll chromium_revision 26cf7e7d6c..e4c6d7fe53 (725941:726742) by Yves Gerey · 5 years ago
  62. 077ee35 Remove unused parameter in RtpFragmentize by Jiwon Kim · 5 years ago
  63. 41875aa add rotationOverride for RTCEAGLVideoView by CZ Theng · 5 years ago
  64. 2e8e1c6 Open up for do the noise suppressor analysis on the linear AEC output by Per Åhgren · 5 years ago
  65. 9136abb AEC3: Ensure that the data size in the reverb computer is not fixed by Per Åhgren · 5 years ago
  66. c8f3134 Parse max-fr and max-fs from SDP FMTP line by Johannes Kron · 5 years ago
  67. 5cad55b Signal requested resolution alignment requirements from sinks to sources. by Rasmus Brandt · 5 years ago
  68. c042425 Make the high-pass filter operate in full-band by Per Åhgren · 5 years ago
  69. 26335a9 Roll chromium_revision 98124fd660..26cf7e7d6c (725465:725941) by Yves Gerey · 5 years ago
  70. 7ab41e5 Fix typo in abseil-in-webrtc.md. by Rasmus Brandt · 5 years ago
  71. ae10029 Prevents probing while paused. by Erik Språng · 5 years ago
  72. 768c5f4 Roll chromium_revision faed30b47a..98124fd660 (724977:725465) by Yves Gerey · 5 years ago
  73. 6fd58b3 Add maxFramerate support to SimulcastEncoderAdapter by Florent Castelli · 5 years ago
  74. 9b540cb Correctly process disabled streams in FrameEncodeMetadataWriter by Ilya Nikolaevskiy · 5 years ago
  75. 00a1bcb Ensure that unset capture timestamp wouldn't cause incorrect SR rtp timestamps by Ilya Nikolaevskiy · 5 years ago
  76. f4cf4c7 Don't allow creation of sockets for wild card IPs in emulated networks. by Sebastian Jansson · 5 years ago
  77. 3a8df88 Add field trial to avoid extra backoffs in AIMD rate control. by Björn Terelius · 5 years ago
  78. 32fe4ef Move vp9 rtp depacketization to VideoRtpDepacketizerVp9 by Danil Chapovalov · 5 years ago
  79. 094396f Use a fake clock for rtc::Thread::PostDelayedTask test by Steve Anton · 5 years ago
  80. f9d92ed Revert "Reland "Distinguish between send and receive video codecs"" by Ilya Nikolaevskiy · 5 years ago
  81. 2697ac1 Stop an SCTP connection when the DTLS transport closes. by Harald Alvestrand · 5 years ago
  82. 8525a80 Add ability to resize buffers pool in decoder and use it in IVF generator by Artem Titov · 5 years ago
  83. 77eb338 Reland "Distinguish between send and receive video codecs" by Johannes Kron · 5 years ago
  84. 5331079 Protect against assigning current_offset_ negative value. by Raman Budny · 5 years ago
  85. cebdbf6 switch RtpVideoStreamReceiver to use VideoRtpDepacketizer interface by Danil Chapovalov · 5 years ago
  86. 0f6bf75 Make video engine tests aware of padding packets by Danil Chapovalov · 5 years ago
  87. 73eb784 Don't crash the test process when X11 isn't available. by Patrik Höglund · 5 years ago
  88. 774fb93 Roll chromium_revision cd7700164d..faed30b47a (724740:724977) by Yves Gerey · 5 years ago
  89. 17ea068 Integration test that verifies that data channels open. by Harald Alvestrand · 5 years ago
  90. 04158be Roll chromium_revision da78695105..cd7700164d (724157:724740) by Yves Gerey · 5 years ago
  91. 95059e0 Moved the legacy noise suppressor to a separate build target by Per Åhgren · 5 years ago
  92. eae6896 Move vp8 rtp depacketization to VideoRtpDepacketizerVp8 by Danil Chapovalov · 5 years ago
  93. 41466b7 Revert "Extracts ssrc based feedback tracking from feedback adapter." by Sebastian Jansson · 5 years ago
  94. 9d06bc2 Replace sequence checker with lock in IvfFrameGemerator. by Artem Titov · 5 years ago
  95. b57fe17 Migrate video tests and tool to VideoRtpDepacketizer interface by Danil Chapovalov · 5 years ago
  96. c9e532a Fix PacketBuffer::LastReceivedKeyframePacketMs by Danil Chapovalov · 5 years ago
  97. 5e9cac9 Don't try to resend packets that were removed out of order. by Sebastian Jansson · 5 years ago
  98. d77c829 Fix incorrect log message in FilterNetworks function. by Sungwoo · 5 years ago
  99. 9338bbc Replace assert() with RTC_DCHECK by Jerome Humbert · 5 years ago
  100. 0808a8c Explicitly set and use WEBRTC_USE_X11 instead of USE_X11. by Patrik Höglund · 5 years ago