SCReAMv2: Replace EWMA loss rate filter with asymmetric step filter

Replaces the complex EWMA loss rate filter in SCReAMv2 with an intuitive
and highly robust Asymmetric Step Filter (leaky bucket counter) to estimate
the short-term congestion level of the connection.

The new filter tracks short-term congestion on RTT boundaries and is normalized
between 0.0 (loss-free) and 1.0 (congested):
- Increment step (loss RTT): 1.0 / rtts_with_loss_before_backoff (default 3).
  This ensures exactly 3 consecutive loss RTTs are required to trigger backoff from a clean state.
- Decrement step (lossless RTT): 1.0 / lossless_rtts_before_clear (default 2).
  This ensures the filter is fully cleared back to 0.0 after exactly 2 consecutive lossless RTTs.

The expected drift under 1% uniform random loss is strongly negative (-0.5),
guaranteeing spurious wireless link losses are filtered out, while a hardcoded
congested() helper check protects against floating-point rounding errors across
various tuned parameter values.

Additional Cleanups:
- Renamed LossRateEstimator class and files to LossEstimator.
- Renamed loss_event_rate() getter and variables to congestion_level() and congestion_level_.
- Simplified scream_v2.cc window growth by checking congestion_level() == 0.0 directly,
  removing the redundant loss_event_rate_threshold_increase and loss_event_rate_threshold_discard
  parameters.
- Updated all unittests and verified that all integration tests pass.

Bug: webrtc:447037083
Change-Id: Iaf1ac08a5230f6e72af3cb0ed84423dfb6997e39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/475580
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47839}
15 files changed
tree: b366841cf47463a2ac253cd5de39d18022a59f80
  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.yapf
  39. .vpython3
  40. .yapfignore
  41. AUTHORS
  42. BUILD.gn
  43. CODE_OF_CONDUCT.md
  44. codereview.settings
  45. DEPS
  46. DIR_METADATA
  47. ENG_REVIEW_OWNERS
  48. GEMINI.md
  49. LICENSE
  50. license_template.txt
  51. native-api.md
  52. OWNERS
  53. OWNERS_INFRA
  54. PATENTS
  55. PRESUBMIT.py
  56. presubmit_test.py
  57. presubmit_test_mocks.py
  58. pylintrc
  59. pylintrc_old_style
  60. README.chromium
  61. README.md
  62. unsafe_buffers_paths.txt
  63. WATCHLISTS
  64. webrtc.gni
  65. webrtc_lib_link_test.cc
  66. 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