Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / fcf79cca7bb92d9ec9b8d2f8146d3c076a6ad365 / call / video_receive_stream.h
  1. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 9 months ago
  2. 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 4 years, 10 months ago
  3. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 4 years, 10 months ago
  4. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 4 years, 10 months ago
  5. 0c141c5 Fix frames dropped statistics by Johannes Kron · 4 years, 11 months ago
  6. d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 4 years, 11 months ago
  7. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  8. 6737841 Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Guido Urdaneta · 5 years ago
  9. fadb181 Negotiate use of RTCP loss notification feedback (LNTF) by Elad Alon · 5 years ago
  10. fe68daa Add option to configure raw RTP packetization per payload type. by Mirta Dvornicic · 5 years ago
  11. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  12. a556448 Don't recreate the VideoReceiveStream on SetFrameDecryptor in the MediaEngine. by Benjamin Wright · 5 years ago
  13. dd41da6 Delete unused methods from VCMReceiveStatisticsCallback by Niels Möller · 5 years ago
  14. 493a650 Propagate base minimum delay from video jitter buffer to webrtc/api. by Ruslan Burakov · 5 years ago
  15. 0237106 Expose video freeze metrics in GetStats. by Sergey Silkin · 5 years ago
  16. 1e27fec Negate flag name for prerender smoothing and update comments. by Rasmus Brandt · 5 years ago
  17. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 5 years ago
  18. 4687915 Enable use of MediaTransportInterface for video streams. by Niels Möller · 6 years ago
  19. f331de6 Remove unused VideoReceiveStream::Config::AddRtxBinding. by Rasmus Brandt · 6 years ago
  20. 514f084 New statistic added to VideoReceiveStream to determine latency to first decode. by Benjamin Wright · 6 years ago
  21. 53382cb Move RtcpStatistics from common_types.h to a new header file by Niels Möller · 6 years ago
  22. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  23. 192eeec Enable End-to-End Encrypted Video Frames. by Benjamin Wright · 6 years ago
  24. 88be972 Delete post_encode_callback by Niels Möller · 6 years ago
  25. 5ca2912 Delete VideoReceiveStream::EnableEncodedFrameRecording by Niels Möller · 6 years ago
  26. cbcbc22 Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Niels Möller · 6 years ago
  27. 377b26e Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Sebastian Jansson · 6 years ago
  28. efb94d5 Revert "Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."" by Oleh Prypin · 6 years ago
  29. 7961dc2 Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." by Niels Moller · 6 years ago
  30. 529d0d9 Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config. by Niels Möller · 6 years ago
  31. 49ac595 Add GetSources to VideoRtpReceiver by Jonas Oreland · 6 years ago
  32. cb7e1d2 Use SdpVideoFormat in VideoReceiveStream::Decoder by Niels Möller · 6 years ago
  33. f88a22c Delete pre_decode_callback. by Niels Möller · 6 years ago
  34. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  35. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  36. c6ce9c5 New file api/video/BUILD.gn by Niels Möller · 6 years ago
  37. be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
  38. 3e11343 Fix circular dependencies in webrtc_common. by Patrik Höglund · 7 years ago
  39. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 7 years ago
  40. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  41. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  42. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/video_receive_stream.h]
  43. ca5706d Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) by nisse · 7 years ago
  44. 8e7eee0 Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) by nisse · 7 years ago
  45. 35713ea Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) by nisse · 7 years ago
  46. 2e1b40b Implement googContentType GetStats metric reported on receive side. by ilnik · 7 years ago
  47. 3c39c01 Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) by nisse · 7 years ago
  48. 75204c5 Change reporting of timing frames conditions in GetStats on receive side by ilnik · 7 years ago
  49. 5c0f6c6 Use RtxReceiveStream. by nisse · 7 years ago
  50. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  51. 26e3abb Reverse |rtx_payload_types| map, and rename. by nisse · 7 years ago
  52. 23bdb67 New accessor function VideoReceiveStream::Config::Rtp::AddRtxBinding by Niels Möller · 7 years ago
  53. a79cc28 Report max interframe delay over window insdead of interframe delay sum by ilnik · 7 years ago
  54. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago