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gerrit-public.fairphone.software / platform / external / webrtc / 27c293665e3a2b0e2de12236529773a7b9057f6e / video / rtp_streams_synchronizer.cc
  1. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 9 months ago
  2. c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 5 years ago
  3. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  4. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  5. 8fdcac3 Remove clang:find_bad_constructs suppression from call:call. by Mirko Bonadei · 6 years ago
  6. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  7. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  8. 558cabf Refactor RtpToNtpEstimator and MovingMedianFilter by Ilya Nikolaevskiy · 7 years ago
  9. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  10. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/rtp_streams_synchronizer.cc]
  11. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  12. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  13. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  14. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 7 years ago
  15. fe50b4d Make class of static functions in rtp_to_ntp.h: - UpdateRtcpList - RtpToNtp by asapersson · 8 years ago
  16. b7e7b49 Use NtpTime in RtcpMeasurement instead of uint sec/uint frac. by asapersson · 8 years ago
  17. de9e5ff Add stats for frequency offset when converting RTP timestamp to NTP time. by asapersson · 8 years ago
  18. b0c1b4e Do not update stream synchronization if no new video packet has been received since last update (e.g. video muted). by asapersson · 8 years ago
  19. 4cd2790 Move RTP for synchroninzation and rename classes, files and variables. by mflodman · 8 years ago[Renamed (65%) from webrtc/video/vie_sync_module.cc]
  20. d28db7f Delete all use of tick_util.h. by Niels Möller · 8 years ago
  21. 0b25072 Use vcm::VideoReceiver on the receive side. by Peter Boström · 8 years ago
  22. 74f6e9e Replace NULL with nullptr in webrtc/video. by Peter Boström · 8 years ago
  23. f8cdd18 Add histogram stats for AV sync stream offset: "WebRTC.Video.AVSyncOffsetInMs" by asapersson · 8 years ago
  24. a26ac92 Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ ) by pbos · 8 years ago
  25. da33a8a Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ ) by torbjorng · 8 years ago
  26. f14c47a Remove ignored return code from modules. by Peter Boström · 8 years ago
  27. 1794b26 Extract ViESyncModule outside ViEChannel. by Peter Boström · 8 years ago
  28. 97888bd Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video. by Tommi · 9 years ago
  29. 7623ce4 Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ ) by Peter Boström · 9 years ago[Renamed (97%) from webrtc/video_engine/vie_sync_module.cc]
  30. 8237abf Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago[Renamed (97%) from webrtc/video/vie_sync_module.cc]
  31. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago[Renamed (97%) from webrtc/video_engine/vie_sync_module.cc]
  32. 2557b86 modules/video_coding refactorings by Henrik Kjellander · 9 years ago
  33. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  34. 74f0f35 Delete a chain of methods in ViE, VoE and ACM by henrik.lundin · 9 years ago
  35. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  36. 415d2cd Use webrtc/base/logging.h for video. by Peter Boström · 9 years ago
  37. e4f9650 Remove system_wrappers/interface/trace_event.h by tommi · 9 years ago
  38. 8fc7fa7 Base A/V synchronization on sync_labels. by pbos · 9 years ago
  39. 36a1438 Remove ViEFrameProviderBase. by Peter Boström · 9 years ago
  40. 0b1534c Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. by pkasting@chromium.org · 10 years ago
  41. 4e2806d Remove WEBRTC_TRACE uses in video_engine/ by pbos@webrtc.org · 10 years ago
  42. 66773a0 Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. by wu@webrtc.org · 10 years ago
  43. cd70119 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  44. 48df381 Fix for making sure that the packet in order checks are done prior to updating the last received packet state. by stefan@webrtc.org · 11 years ago
  45. 822fbd8 Update talk to 50918584. by wu@webrtc.org · 11 years ago
  46. aa4d96a Revert r4301 by tnakamura@webrtc.org · 11 years ago
  47. 66b2e5c Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  48. 7262ad1 Fix AV sync issue by hclam@chromium.org · 11 years ago
  49. 9b23ecb Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
  50. e46c8d3 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 11 years ago
  51. f5d4cb1 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  52. 1de0135 Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
  53. 806dc3b More trace events by hclam@chromium.org · 11 years ago
  54. b238d12 WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
  55. 79b0289 Adds event traces and counters for WebRTC receive side. by edjee@google.com · 11 years ago
  56. efe4edb Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 11 years ago
  57. ef9f76a Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
  58. 8d18526 Fixes an incorrect if statement in vie_sync_module.cc. by stefan@webrtc.org · 12 years ago
  59. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/video_engine/vie_sync_module.cc]
  60. 7c3523c Change audio/video sync to be based on mapping RTP timestamps to NTP. by stefan@webrtc.org · 12 years ago
  61. 6f8db36 Reorganize voice_engine/. by andrew@webrtc.org · 12 years ago
  62. ab2610f Removed the last lint warnings in video_engine. by mflodman@webrtc.org · 12 years ago
  63. 5f28498 First step in refactoring audio/video synchronization. Adds unittests. by stefan@webrtc.org · 12 years ago
  64. 139c467 Fixed a/v sync issue. by mflodman@webrtc.org · 12 years ago
  65. 2853dde Refactor the internal API to the rtp/rtcp module. by pwestin@webrtc.org · 12 years ago
  66. 3c383ab Revert 2211 - Refactor the internal API to the rtp/rtcp module. by turaj@webrtc.org · 12 years ago
  67. 0774838 Refactor the internal API to the rtp/rtcp module. by pwestin@webrtc.org · 12 years ago
  68. 39e9659 Correct wrong usage of WebRtc_Word8 in video enigne by leozwang@webrtc.org · 12 years ago
  69. d32c447 Changed constructor used for CriticalSectionScoped in ViE. by mflodman@webrtc.org · 13 years ago
  70. d2ee5d9 Changed sync bug introduced in refactoring. by mflodman@webrtc.org · 13 years ago
  71. 511f82e Refactored ViESyncModule. by mflodman@webrtc.org · 13 years ago
  72. 94ea32e Move video_engine/source* to video_engine/. No code changes except paths in gyp-files. by mflodman@webrtc.org · 13 years ago[Renamed from src/video_engine/main/source/vie_sync_module.cc]
  73. 470e71d by niklase@google.com · 13 years ago