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gerrit-public.fairphone.software / platform / external / webrtc / a13e7a1d4663e945195966f92298085700af46e4 / media
  1. 84bb634 Delete legacy cricket::RtpHeaderExtension struct as unused by Danil Chapovalov · 4 years ago
  2. c7c412a Check for null before accessing SctpTransport map. by Taylor Brandstetter · 4 years ago
  3. f7303e6 Migrate leftovers in media/ and modules/ to webrtc::Mutex. by Markus Handell · 4 years ago
  4. 1e257ca Migrate media/ to webrtc::Mutex. by Markus Handell · 4 years ago
  5. c212873 Relanding: Fix data channel message integrity violation by Taylor Brandstetter · 4 years ago
  6. f2a4ec1 sdp: parse and serialize non-key=value fmtp lines by Philipp Hancke · 4 years ago
  7. 51e08b8 Revert "Fix data channel message integrity violation" by Alessio Bazzica · 4 years ago
  8. 6cbff75 Fix data channel message integrity violation by Lennart Grahl · 4 years ago
  9. 3cb0985 Inclusive language in //media/engine. by Mirko Bonadei · 4 years ago
  10. 39adce1 Add RtpEncodingParameters.adaptive_ptime. by Jakob Ivarsson · 4 years ago
  11. edcd966 negotiate RED codec for audio by Philipp Hancke · 4 years ago
  12. 976faae Disable SCTP asconf and auth extensions. by Taylor Brandstetter · 4 years, 1 month ago
  13. a46fce2 Update rtp_utils to support two-byte RTP header extensions by Johannes Kron · 4 years, 1 month ago
  14. 2a70703 Delete MediaTransportInterface and DatagramTransportInterface by Niels Möller · 4 years, 1 month ago
  15. 9465978 Remove framemarking RTP extension. by philipel · 4 years, 1 month ago
  16. 7ff6355 Add decoder support for VP9 profile 1 I444 by Johannes Kron · 4 years, 1 month ago
  17. f4a9991 [Adaptation] Adding adaptation resources from Call. by Henrik Boström · 4 years, 1 month ago
  18. 2dcf348 Use absl_deps in order to preapre to the Abseil component build release. by Mirko Bonadei · 4 years, 1 month ago
  19. 301eb53 Prevent pointer from being sent in the clear over SCTP. by Taylor Brandstetter · 4 years, 1 month ago
  20. e917379 [Stats] Don't attempt to aggregate empty VideoSenderInfos. by Henrik Boström · 4 years, 1 month ago
  21. 3c9ae1f Adds fix for closing a prenegotiated DC without sending data. by Harald Alvestrand · 4 years, 1 month ago
  22. b59f337 Remove leftover SCTP "codec name" constants by Harald Alvestrand · 4 years, 1 month ago
  23. 461e387 use constants for CN and telephone-event codec names by Philipp Hancke · 4 years, 1 month ago
  24. f026592 Add HEVC codec name. by Andrey Logvin · 4 years, 2 months ago
  25. cdbebb0 Add INSTANTIATE_TEST_SUITE_P as needed. by Courtney Edwards · 4 years, 1 month ago
  26. f2c0f15 In media/ and modules/video_coding replace mock macros with unified MOCK_METHOD macro by Danil Chapovalov · 4 years, 2 months ago
  27. c5324fb VideoAdapter: remove lock recursions. by Markus Handell · 4 years, 2 months ago
  28. 3fa23ee VideoAdapter: add missing attribute thread annotations. by Markus Handell · 4 years, 2 months ago
  29. e7864f5 FakeNetworkInterface: remove lock recursions. by Markus Handell · 4 years, 2 months ago
  30. 772b149 MediaChannel: remove lock recursions. by Markus Handell · 4 years, 2 months ago
  31. 6efc14b VideoTrackSourceInterface: make some newly introduced methods pure virtual. by Markus Handell · 4 years, 2 months ago
  32. 09e9a83 Change the way that AecDumps are created in APM by Per Åhgren · 4 years, 2 months ago
  33. fc11519 Cleanup mocks in api/test by Danil Chapovalov · 4 years, 2 months ago
  34. b63331b Cleanup mocks for Video (en|de)coder factories by Danil Chapovalov · 4 years, 2 months ago
  35. a0ff50c Reland "Improve outbound-rtp statistics for simulcast" by Henrik Boström · 4 years, 2 months ago
  36. 9a925c9 Revert "Improve outbound-rtp statistics for simulcast" by Henrik Boström · 4 years, 2 months ago
  37. d9255b1 [getStats] Fix DCHECK crash in MergeInfoAboutOutboundRtpSubstreams(). by Henrik Boström · 4 years, 2 months ago
  38. cc73ed3 APM: Add build flag to allow building WebRTC without APM by Per Åhgren · 4 years, 2 months ago
  39. da6cda8 Improve outbound-rtp statistics for simulcast by Eldar Rello · 4 years, 3 months ago
  40. 6415dca Remove WebRTC-ExperimentalScreenshareSettings. by Mirko Bonadei · 4 years, 3 months ago
  41. 4a5bce9 Change to more idiomatic map erase. by Rasmus Brandt · 4 years, 3 months ago
  42. 2c99264 Remove std::vector<BandwidthEstimationInfo> template instantiation. by Mirko Bonadei · 4 years, 3 months ago
  43. adc4da3 [InsertableStreams] Fix video receiver simulcast. by Marina Ciocea · 4 years, 3 months ago
  44. 16d0d37 Apply performance-for-range-copy fixes. by Mirko Bonadei · 4 years, 3 months ago
  45. de6fa1e Reland "Let WebRtcVideoChannel::ResetUnsignaledRecvStream delete all default streams." by Rasmus Brandt · 4 years, 3 months ago
  46. c6b2f34 Revert "Let WebRtcVideoChannel::ResetUnsignaledRecvStream delete all default streams." by Rasmus Brandt · 4 years, 3 months ago
  47. 4553f45 Add AV1 to default video encoder factory by Danil Chapovalov · 4 years, 3 months ago
  48. d335426 Let WebRtcVideoChannel::ResetUnsignaledRecvStream delete all default streams. by Rasmus Brandt · 4 years, 3 months ago
  49. 3e9af7f Insert audio frame transformer between depacketizer and decoder. by Marina Ciocea · 4 years, 3 months ago
  50. 8cdd2c7 Regression test for SCTP transport. by Taylor Brandstetter · 4 years, 3 months ago
  51. e1aa22f [InsertableStreams] Set video frame transformer if RTP stream already started. by Guido Urdaneta · 4 years, 3 months ago
  52. d2aa8f9 Insert audio frame transformer between encoder and packetizer. by Marina Ciocea · 4 years, 3 months ago
  53. 3e98368 Reland "Distinguish between send and receive codecs" by Johannes Kron · 4 years, 3 months ago
  54. f45ca37 [Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats. by Henrik Boström · 4 years, 3 months ago
  55. 01611ed Fix potential divide by zero in SimulcastEncoderAdapter by Erik Språng · 4 years, 3 months ago
  56. dfeb0df RtpParameters: respect https://abseil.io/tips/1. by Markus Handell · 4 years, 4 months ago
  57. fc29b0a [Stats] Include RTX retransmissions in the VideoSenderInfo. by Henrik Boström · 4 years, 4 months ago
  58. 8bc8834 webrtc_video_engine_unittest.cc: fix using directives. by Markus Handell · 4 years, 4 months ago
  59. 0357b3e RtpTransceiverInterface: add header_extensions_to_offer() by Markus Handell · 4 years, 4 months ago
  60. e618cc9 Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API by Artem Titov · 4 years, 4 months ago
  61. c46385c Add Av1 Decoder wrapper behind a build flag by Danil Chapovalov · 4 years, 4 months ago
  62. b05ca4b Implement new specification for degradation preference by Florent Castelli · 4 years, 4 months ago
  63. 3f1aee3 Change network_priority from a double to an enum. by Taylor Brandstetter · 4 years, 4 months ago
  64. 24dbb21 Enable quality scaler for simulcast and SVC if only one stream is active by Ilya Nikolaevskiy · 4 years, 4 months ago
  65. 83762b2 Use NetworkToHost32 instead of HostToNetwork32 to translate PPID. by Taylor Brandstetter · 4 years, 4 months ago
  66. 412a31b Insert frame transformer between Depacketizer and Decoder. by Marina Ciocea · 4 years, 4 months ago
  67. e77912b Insert frame transformer between Encoded and Packetizer. by Marina Ciocea · 4 years, 4 months ago
  68. f0276c5 Lazy conversion to i420 buffer in SimulcastEncoderAdapter by Erik Språng · 4 years, 5 months ago
  69. 4a14f49 Remove wildcard ownership for build files. by Mirko Bonadei · 4 years, 5 months ago
  70. e52115a Remove inactive OWNERS. by Mirko Bonadei · 4 years, 5 months ago
  71. dea73ee Pass absolute capture time from WebRtcVoiceEngine to ACM. by Minyue Li · 4 years, 5 months ago
  72. cad3e0e Replace DataSize and DataRate factories with newer versions by Danil Chapovalov · 4 years, 5 months ago
  73. e457631 Save custom parameters provided by the SdpVideoFormat when requesting an encoder switch. by philipel · 4 years, 5 months ago
  74. 06901cf Use absl::c_any_of instead of a manual for-loop to finding an active encoding by Oskar Segersvärd · 4 years, 5 months ago
  75. dc81e11 Use webrtc::DataRate when referring to bitrates by Oskar Segersvärd · 4 years, 5 months ago
  76. e2b466e Stop advertising generic frame descriptor v1 by Danil Chapovalov · 4 years, 5 months ago
  77. 03d9096 Ensure that the first active layer isn't disabled by too low input resolution by Ilya Nikolaevskiy · 4 years, 5 months ago
  78. ccd49d9 Use I420Buffer::ScaleFrom to clean scaling in SimulcastEncoderAdapter by Evan Shrubsole · 4 years, 5 months ago
  79. c43fe2e Removing myself from OWNERS in webrtc. by Seth Hampson · 4 years, 5 months ago
  80. 0c626af Use newer version of TimeDelta and TimeStamp factories in webrtc by Danil Chapovalov · 4 years, 5 months ago
  81. 9b05803 Implement injectable EncoderSelectorInterface and wire it up in the VideoStreamEncoder. by philipel · 4 years, 5 months ago
  82. 8e8b36a Revert "Reland "Reland "Reland "Distinguish between send and receive codecs"""" by Johannes Kron · 4 years, 5 months ago
  83. cf2b382 Send bandwidth updates to all codecs, not just Opus by Karl Wiberg · 4 years, 5 months ago
  84. 72859e5 Make RtpEncodingParameters to not reverse active flags order by Ilya Nikolaevskiy · 4 years, 5 months ago
  85. 261f792 Allow software fallback on lowest simulcast stream for temporal support by Erik Språng · 4 years, 5 months ago
  86. 184ea66 Reland "Reland "Reland "Distinguish between send and receive codecs""" by Johannes Kron · 4 years, 5 months ago
  87. 99d6d81 Adding absolute capture timestamp to AudioTrackSinkInterface. by Minyue Li · 4 years, 5 months ago
  88. a104ceb Revert "Reland "Reland "Distinguish between send and receive codecs""" by Johannes Kron · 4 years, 5 months ago
  89. 9bac68c Reland "Reland "Distinguish between send and receive codecs"" by Johannes Kron · 4 years, 5 months ago
  90. 760fd52 Replace MockAudioDeviceModule mock refcounting with real refcounting by Steve Anton · 4 years, 5 months ago
  91. 00a3087 Revert "Reland "Distinguish between send and receive codecs"" by Johannes Kron · 4 years, 5 months ago
  92. 897776e Pass SDP video parameters to encoder. by Sergey Silkin · 4 years, 5 months ago
  93. 133bf2b Reland "Distinguish between send and receive codecs" by Johannes Kron · 4 years, 5 months ago
  94. 1acdc74 Split up EncoderStreamFactory::CreateEncoderStreams in two. by Rasmus Brandt · 4 years, 5 months ago
  95. 43bfe0b Enforce VideoEncoderConfig.num_temporal_layers >= 1. by Rasmus Brandt · 4 years, 5 months ago
  96. ccbe95f Reformat GN files. by Mirko Bonadei · 4 years, 5 months ago
  97. e57b266 Revert "Distinguish between send and receive codecs" by Steve Anton · 4 years, 6 months ago
  98. c0f25cf Distinguish between send and receive codecs by Johannes Kron · 4 years, 6 months ago
  99. b2b2031 Concatenate string literals at compile time. by Jonas Olsson · 4 years, 6 months ago
  100. 1546f99 Fixed timeout overflow in sctp reliability test. by Yura Yaroshevich · 4 years, 6 months ago