Reland "Cache RtpSender parameters to reduce worker thread blocking"

This is a reland of commit 969e534de205b20ecdc2a73939d2219c8317323f

Original change's description:
> Cache RtpSender parameters to reduce worker thread blocking
>
> This avoids some blocking calls when collecting stats and also during
> negotiation, at most 8 (total blocking reduced by 8 in one line) for
> `RenegotiateManyVideoTransceiversAndWatchAudioDelay`.
>
> This change introduces a caching mechanism for internal fetching of
> RtpSender parameters to reduce the frequency of blocking calls to the
> worker thread. Previously, retrieving parameters always required a
> thread hop.
>
> Key changes include:
>
> Adding cached_parameters_ to RtpSenderBase to store the latest known
> state.
>
> Updating GetParametersInternal to return the cached value when available
> and requested.
>
> While running the `RenegotiateManyVideoTransceiversAndWatchAudioDelay`
> test, which does not measure stats collection, blocking call count is
> reduced to varying degrees, highest being 8 fewer blocking calls for a
> single method call.
>
> Bug: webrtc:42222804
> Change-Id: Ife4100e1d6e72e2287ae37ee1fc042a0cf44ebbc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/442443
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46699}

Bug: webrtc:42222804
Change-Id: Ifb5ce0e3086dc278e171ad15cb07df00b5923b18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/442922
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46714}
12 files changed
tree: ba84ee9b1ba6cb2f584707cc585ccee03088dc77
  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. WATCHLISTS
  60. webrtc.gni
  61. webrtc_lib_link_test.cc
  62. 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