Refactor RTCStatsCollector to move stats production to signaling thread

This is CL 1 of 2 that moves population of stats objects from the
networking thread over to the signaling thread.

This CL refactors RTCStatsCollector to reduce the workload on the
network thread. Previously, the network thread was responsible for
gathering raw data and converting it to RTCStats objects. This change
moves the conversion logic (production of RTCStats objects) to the
signaling thread. The network thread now only gathers the necessary raw
data structs and passes them to the signaling thread.

Specific changes:
- ProducePartialResultsOnNetworkThread now gathers data into CollectionContext and posts a task to signaling thread.
- Produce*Stats_n methods are renamed to Produce*Stats_s and moved to run on the signaling thread.
- ProduceRTPStreamStats_n is moved to ProduceRTPStreamStats_s on the signaling thread.
- MergeNetworkReport_s is updated to MergeNetworkFinalResults_s.
- RTCStatsCollector::CollectionContext is updated to hold the intermediate raw data instead of a partial RTCStatsReport.

Bug: webrtc:42222804
Change-Id: I20332f63e0cef0ffdf0df17d0e4c468af90cb8d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/445000
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46806}
5 files changed
tree: 63858d9dd7ee0de717f7441c8521d0c6b93f65e3
  1. agents/
  2. api/
  3. audio/
  4. build_overrides/
  5. call/
  6. common_audio/
  7. common_video/
  8. data/
  9. docs/
  10. examples/
  11. experiments/
  12. g3doc/
  13. infra/
  14. logging/
  15. media/
  16. modules/
  17. net/
  18. p2p/
  19. pc/
  20. resources/
  21. rtc_base/
  22. rtc_tools/
  23. sdk/
  24. stats/
  25. system_wrappers/
  26. test/
  27. tools_webrtc/
  28. video/
  29. .clang-format
  30. .clang-tidy
  31. .git-blame-ignore-revs
  32. .gitignore
  33. .gn
  34. .mailmap
  35. .rustfmt.toml
  36. .style.yapf
  37. .vpython3
  38. AUTHORS
  39. BUILD.gn
  40. CODE_OF_CONDUCT.md
  41. codereview.settings
  42. DEPS
  43. DIR_METADATA
  44. ENG_REVIEW_OWNERS
  45. GEMINI.md
  46. LICENSE
  47. license_template.txt
  48. native-api.md
  49. OWNERS
  50. OWNERS_INFRA
  51. PATENTS
  52. PRESUBMIT.py
  53. presubmit_test.py
  54. presubmit_test_mocks.py
  55. pylintrc
  56. pylintrc_old_style
  57. README.chromium
  58. README.md
  59. unsafe_buffers_paths.txt
  60. WATCHLISTS
  61. webrtc.gni
  62. webrtc_lib_link_test.cc
  63. 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