Google Git
Sign in
webrtc / src / / 04158be200d821bc05c0980163987ccc4f934dc2 / call
  1. 41466b7 Revert "Extracts ssrc based feedback tracking from feedback adapter." by Sebastian Jansson · 5 years ago
  2. 977b265 Reduce some logging at INFO level by moving log statements by Harald Alvestrand · 5 years ago
  3. 08c46ad Extracts ssrc based feedback tracking from feedback adapter. by Sebastian Jansson · 5 years ago
  4. b04b2a1 Initial version of ResourceAdaptationProcessor and friends. by Henrik Boström · 5 years ago
  5. f18f920 Revert "Moves TransportFeedbackAdapter to TaskQueue." by Mirko Bonadei · 5 years ago
  6. ef3998f Add directive to make webrtc metrics optional. by Ying Wang · 5 years ago
  7. 62d01cd Moves TransportFeedbackAdapter to TaskQueue. by Sebastian Jansson · 5 years ago
  8. 33f9d2b Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class by Artem Titov · 5 years ago
  9. cec2433 Exposing more features in the network emulation manager API. by Sebastian Jansson · 5 years ago
  10. 17f82cf Verifies trials are populated when creating a Call. by Erik Språng · 5 years ago
  11. e19a375 Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets by Per Kjellander · 5 years ago
  12. 486cc55 TimeController: Rename Sleep to AdvanceTime. by Markus Handell · 5 years ago
  13. 269ac81 VideoReceiveStream: Enable encoded frame sink. by Markus Handell · 5 years ago
  14. 014dd3c Trials should always be populated in call config. by Erik Språng · 5 years ago
  15. 4314a49 Implements a task-queue based PacedSender, wires it up for field trials by Erik Språng · 5 years ago
  16. b529b7a Add string<->VideoCodecType conversion for all codec types. by Danil Chapovalov · 5 years ago
  17. dc36829 Add VideoCodecType::kVideoCodecAV1 value by Danil Chapovalov · 5 years ago
  18. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  19. 00cc836 Makes all of RtpVideoSenderTest use simulated time by Erik Språng · 5 years ago
  20. 00376e1 Add totalInterFrameDelay to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  21. 662678d Adds injectable trials from peerconnection down to transport controller. by Erik Språng · 5 years ago
  22. bae1275 Using unit types in TransportFeedbackAdapter. by Sebastian Jansson · 5 years ago
  23. 26452ff Cleanup of TransportFeedbackAdapter. by Sebastian Jansson · 5 years ago
  24. c3d1f9b Enable injection of a custom NetEqFactory into PeerConnectionFactory. by Ivo Creusen · 5 years ago
  25. 577c580 Do not stop SingleThreadedTaskQueueForTestingTest near the end of the tests by Danil Chapovalov · 5 years ago
  26. f298855 Cleanup of feedback observer interface by Sebastian Jansson · 5 years ago
  27. 05c4792 Removes OnPacketAdded callback from feedback adapter. by Sebastian Jansson · 5 years ago
  28. a922904 Calls OnPacketsAcknowledged on RtpRtcp instead of RTPSender directly. by Erik Språng · 5 years ago
  29. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  30. d15a028 Hide deprecated SingleThreadedTaskQueueForTest behind an accessor by Danil Chapovalov · 5 years ago
  31. 85a1000 Use deprecated SingleThreadedTaskQueueForTesting as regular task queue by Danil Chapovalov · 5 years ago
  32. e34fb87 Clarify NetworkControl interface: result of each function must be used by Danil Chapovalov · 5 years ago
  33. 9f5ae7b Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting by Danil Chapovalov · 5 years ago
  34. 82a3f0a Replace SingleThreadedTaskQueueForTesting::SendTask usage with ::webrtc::SendTask by Danil Chapovalov · 5 years ago
  35. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 5 years ago
  36. 04671b0 Delete unused method PacedSender::QueueSizePackets by Niels Möller · 5 years ago
  37. 7ea9b80 Set StreamDataCountersCallback on construction of RTP modules by Erik Språng · 5 years ago
  38. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
  39. 82ed2e8 Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender by Sebastian Jansson · 5 years ago
  40. 6841d25 Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 5 years ago
  41. f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 5 years ago
  42. e8a6bc3 Revert "Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"" by Erik Språng · 5 years ago
  43. c934821 Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 5 years ago
  44. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  45. 4ed0b52 Revert "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 5 years ago
  46. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  47. 17608dc RtpRtcp modules and below: Make media, RTX and FEC SSRCs const by Erik Språng · 5 years ago
  48. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  49. 51bf200 Reduce number of RTPVideoSender::SendVideo parameters by Danil Chapovalov · 5 years ago
  50. 28214cd Fix handling of large packets in RtxReceiveStream by Niels Möller · 5 years ago
  51. dc34a25 Adds RTPSenderVideo::Config struct with red/ulpfec config by Erik Språng · 5 years ago
  52. b9bfe65 Delete VCMEncodedFrame::VerifyAndAllocate by Niels Möller · 5 years ago
  53. 6cf554e Reduces locking in RtpSenderVideo. by Erik Språng · 5 years ago
  54. 44db436 Propagate task queue to create test::DirectTransport by TaskQueueBase interface by Danil Chapovalov · 5 years ago
  55. 538ca57 Converts const methods in BitrateAllocator to non-member functions. by Sebastian Jansson · 6 years ago
  56. f34116e Replacing bandwidth adaptation trial with stable target in Opus encoder. by Sebastian Jansson · 6 years ago
  57. 40de3cc Propagating TargetRate struct to BitrateAllocator. by Sebastian Jansson · 6 years ago
  58. 0a5ed89 Adds remote estimates to rtc event log. by Sebastian Jansson · 6 years ago
  59. 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 6 years ago
  60. 53227cc Remove webrtc::MinPositive from api/. by Mirko Bonadei · 6 years ago
  61. 738bfa7 Remove api/bitrate_constraints.h. by Mirko Bonadei · 6 years ago
  62. 4d461ba Reusing MediaStreamAllocationConfig struct in ObserverConfig. by Sebastian Jansson · 6 years ago
  63. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 6 years ago
  64. 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 6 years ago
  65. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 6 years ago
  66. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 6 years ago
  67. fcfeefe Move rtc_error.{h,cc} to its own build target. by Mirko Bonadei · 6 years ago
  68. 0987273 Add option to enable retransmission for all temporal layers in the constructor for rtp_sender_video. by Andrei Dumitru · 6 years ago
  69. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 6 years ago
  70. 6516f76 Deprecate SingleThreadedTaskQueueForTesting class. by Yves Gerey · 6 years ago
  71. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 6 years ago
  72. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 6 years ago
  73. 55dd72c Remove lock for process thread pointer from PacedSender. by Tommi · 6 years ago
  74. 25eb47c Make the RtpHeaderParserImpl available to tests and tools only. by Tommi · 6 years ago
  75. b4a6128 Delete unneeded dependencies on libjingle_peerconnection_api by Niels Möller · 6 years ago
  76. 6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 6 years ago
  77. fac7e31 Removes TransportSequenceNumberAllocator by Erik Språng · 6 years ago
  78. 0c141c5 Fix frames dropped statistics by Johannes Kron · 6 years ago
  79. d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 6 years ago
  80. 4e615d5 Wire the stable target bitrate from GoogCC to the BitrateAllocator by Florent Castelli · 6 years ago
  81. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 6 years ago
  82. 54d5d2c Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc by Erik Språng · 6 years ago
  83. 6b117a5 Make the callbacks to PollStats for RampUp* tests more regular. by Tommi · 6 years ago
  84. 93f5189 Remove some usage of RtpRtcp::SetSSRC() by Erik Språng · 6 years ago
  85. 6950b30 Fix thread naming in Call constructor by Erik Språng · 6 years ago
  86. c14b233 Disable the most flaky tests on iOS. by Sami Kalliomäki · 6 years ago
  87. eea605d Make fake network degradation work also for sent audio by Erik Språng · 6 years ago
  88. 48b48e5 Enable thread check in Call::GetStats(). by Tommi · 6 years ago
  89. 78a7138 Remove MediaTransport from Call. by Tommi · 6 years ago
  90. 891d393 Call Call::GetStats() from the correct thread in ProbingEndToEndTest. by Tommi · 6 years ago
  91. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 6 years ago
  92. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 6 years ago
  93. c24a5b1 Fix CallPerfTests to call Call::GetStats() from the right thread. by Tommi · 6 years ago
  94. c648819 DegradedCall: fake network using TaskQueue instead of ProcessThread by Erik Språng · 6 years ago
  95. 5e005f4 Fix RampUp tests to call Call::GetStats() from the right thread - and remove the need for a dedicated polling thread. by Tommi · 6 years ago
  96. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 6 years ago
  97. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 6 years ago
  98. 425d6aa Add RtpPacketPacer interface for pacer control by Erik Språng · 6 years ago
  99. 463d44a Don't crash when simulcast layer count is different from RID count by Florent Castelli · 6 years ago
  100. aa59eca Move RtpPacketSender and merge it with RtpPacketPacer. by Erik Språng · 6 years ago