Pace CCFB using actual wire overhead and is_bandwidth_limited status

This CL improves the RFC 8888 Congestion Control Feedback (CCFB) pacing logic to accurately calculate pacing delays and respect bandwidth limits:

1. **Wire Overhead Inclusion**: Includes the IP/UDP/SRTP wire overhead (defaulting to 42 bytes) in the CCFB pacing debt calculation. This prevents the generator from underestimating the feedback transmission rate and exceeding configured bandwidth limits on high packet rate connections.
2. **Bandwidth-Limited Pacing**: Only applies the feedback fraction limit (e.g., 5% of send BWE, if configured via the `feedback_fraction` option in the `WebRTC-RFC8888CongestionControlFeedback` field trial) when the link is actually marked as bandwidth-limited (`is_bandwidth_limited = true`). When not bandwidth-limited, the default pacing rate (500 kbps) applies.
3. **API Deprecation**: Deprecates the old `OnSendBandwidthEstimateChanged(DataRate)` signature in favor of a new one carrying `is_bandwidth_limited` and `transport_overhead` parameters.

These changes ensure CCFB complies with the RFC 8888 recommendation to not consume excessive feedback bandwidth (restricting it to a safe fraction of BWE when congested), while keeping feedback responsive and pacing correctly.

Bug: webrtc:519655843
Change-Id: I7b1b40f1f711df76b7beb4718b1566dd9d48aef5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/478442
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47951}
20 files changed
tree: d38d293f4ac9d441801d186aa0623caabd0ceb70
  1. .agents/
  2. agents/
  3. api/
  4. audio/
  5. build_overrides/
  6. call/
  7. common_audio/
  8. common_video/
  9. data/
  10. docs/
  11. examples/
  12. experiments/
  13. g3doc/
  14. infra/
  15. logging/
  16. media/
  17. modules/
  18. net/
  19. p2p/
  20. pc/
  21. resources/
  22. rtc_base/
  23. rtc_tools/
  24. rust/
  25. sdk/
  26. stats/
  27. system_wrappers/
  28. test/
  29. tools_webrtc/
  30. video/
  31. .clang-format
  32. .clang-tidy
  33. .git-blame-ignore-revs
  34. .gitignore
  35. .gn
  36. .mailmap
  37. .rustfmt.toml
  38. .style.mdformat
  39. .style.yapf
  40. .vpython3
  41. .yapfignore
  42. AUTHORS
  43. BUILD.gn
  44. CODE_OF_CONDUCT.md
  45. codereview.settings
  46. DEPS
  47. DIR_METADATA
  48. ENG_REVIEW_OWNERS
  49. GEMINI.md
  50. LICENSE
  51. license_template.txt
  52. native-api.md
  53. OWNERS
  54. OWNERS_INFRA
  55. PATENTS
  56. PRESUBMIT.py
  57. presubmit_test.py
  58. presubmit_test_mocks.py
  59. pylintrc
  60. pylintrc_old_style
  61. README.chromium
  62. README.md
  63. unsafe_buffers_paths.txt
  64. WATCHLISTS
  65. webrtc.gni
  66. webrtc_lib_link_test.cc
  67. 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