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webrtc / src / 48ae01c5b0896ae407ac15795b41c8d82e8eae16
commit48ae01c5b0896ae407ac15795b41c8d82e8eae16[log] [tgz]
authorHenrik Boström <hbos@webrtc.org>Thu Apr 08 07:30:05 2021
committerCommit Bot <commit-bot@chromium.org>Thu Apr 15 11:35:30 2021
tree59b63b54e8f76297cf98386aff6cc4bb2723bfec
parentba8780fb851626daa89cf7c025460dd11fc78ed4 [diff]
[Merge-91] Remove RTCRemoteInboundRtpStreamStats duplicate members.

The RTCReceivedRtpStreamStats hierarchy, which inherit from
RTCRtpStreamStats, already contain members ssrc, kind, codec_id and
transport_id so there's no need to list them inside
RTCRemoteInboundrtpStreamStats.

This CL removes duplicates so that we don't DCHECK-crash on Android,
and adds a unit test ensuring we never accidentally list the same
member twice.

(cherry picked from commit 943ad970f4087bac5c503fe93fb99d953d475652)

Bug: webrtc:12658
Change-Id: I27925eadddc6224bf6d6a91784ed7cafd7a4cfb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214343
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#33649}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215233
Cr-Commit-Position: refs/branch-heads/4472@{#4}
Cr-Branched-From: 3e0c60ba4ef28a9f26fe991e5eec3150402c7dd3-refs/heads/master@{#33644}
2 files changed
tree: 59b63b54e8f76297cf98386aff6cc4bb2723bfec
  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. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. style-guide/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .vpython
  32. abseil-in-webrtc.md
  33. AUTHORS
  34. BUILD.gn
  35. CODE_OF_CONDUCT.md
  36. codereview.settings
  37. DEPS
  38. DIR_METADATA
  39. ENG_REVIEW_OWNERS
  40. g3doc.lua
  41. LICENSE
  42. license_template.txt
  43. native-api.md
  44. OWNERS
  45. PATENTS
  46. PRESUBMIT.py
  47. presubmit_test.py
  48. presubmit_test_mocks.py
  49. pylintrc
  50. README.chromium
  51. README.md
  52. style-guide.md
  53. WATCHLISTS
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. 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