Fix video recovery freeze on RTP seq number wrap

After a prolonged decode stall on an SFU (where the sender does not
back off under loss), the 16-bit RTP sequence number can advance by
more than half its range. This breaks two independent state machines
in the video receive pipeline, each freezing video for 60-150s:

1. RtpFrameReferenceFinder wrapper: AheadOf<uint16_t>(cleared_to, seq)
   misjudges a fresh frame as "already cleared" and drops it before it
   reaches a per-codec finder. Fixed by requiring the 32-bit RTP
   timestamp (unambiguous over ~6.6 hours at 90 kHz) to also confirm
   the frame is older before dropping. This is wrap-safe for all frame
   types and backwards-compatible (callers that pass no timestamp keep
   the original sequence-number-only behavior).

2. RtpSeqNumOnlyRefFinder: stale GoP entries from before the wrap sit
   "ahead" of the new keyframe in 16-bit space, shadowing upper_bound()
   lookups so delta frames are dropped ("has no GoP"). Fixed by
   removing GoPs that appear ahead of a newly-arrived keyframe -- a
   keyframe is the newest frame received, so an "ahead" GoP can only be
   a stale leftover.

Both fixes address the same root cause (16-bit wrap ambiguity) at
different pipeline stages and are strictly less aggressive at dropping.

Tested: 3 new unit tests in rtp_frame_reference_finder_unittest.cc
(FrameNotDroppedWhenSeqWrapsButTimestampNewer,
ClearToSeqNumOnlyWithoutTimestamp,
DeltasNotDroppedByStaleGopAfterSeqNumJump); negative control confirms
the GoP test fails without the fix. Validated end-to-end on iOS against
a production SFU: 60-150s freeze plus 80s 1fps tail reduced to
immediate full-frame-rate recovery.

Bug: webrtc:516639936
Change-Id: I9790e3cf5cd6590b899de3e32d39f6d573ff205d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/477641
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48053}
7 files changed
tree: 9f3f314775446e5042c4a4b8667f0a3a202171f9
  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