Google Git
Sign in
webrtc / src / c366d51836a9e533fb261320e577b32af269dbbd
commitc366d51836a9e533fb261320e577b32af269dbbd[log] [tgz]
authorAlessio Bazzica <alessiob@webrtc.org>Mon Mar 22 14:36:53 2021
committerCommit Bot <commit-bot@chromium.org>Mon Mar 22 18:47:33 2021
tree566ec8fc220f0d16cd977e41e7840fa6a4e3c1d6
parentc303f82f4d1f770824f5704b5a7da6d231b3f7bd [diff]
Fix unit for inbound RTP stat `lastPacketReceivedTimestamp` (s -> ms)

Both inbound RTP stats `estimatedPlayoutTimestamp` and
`lastPacketReceivedTimestamp` are surfaced to JS land as
`DOMHighResTimeStamp` - i.e., time values in milliseconds.
This CL fixes `lastPacketReceivedTimestamp` which is incorrectly
surfaced as time value in seconds.

Bug: webrtc:12605
Change-Id: I290103071cca3331d2a3066b6b6b9fcb4f4fd0af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212742
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33530}
2 files changed
tree: 566ec8fc220f0d16cd977e41e7840fa6a4e3c1d6
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. style-guide/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. abseil-in-webrtc.md
  32. AUTHORS
  33. BUILD.gn
  34. CODE_OF_CONDUCT.md
  35. codereview.settings
  36. DEPS
  37. DIR_METADATA
  38. ENG_REVIEW_OWNERS
  39. LICENSE
  40. license_template.txt
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. style-guide.md
  51. WATCHLISTS
  52. webrtc.gni
  53. webrtc_lib_link_test.cc
  54. whitespace.txt
README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info