Prevent delayed FrameDecoded from advancing RTP cleanup epoch

FrameDecoded() unwraps a stored uint16_t RTP sequence number relative
to the current receive state. If the frame is decoded only after the
receive sequence advances by more than half the 16-bit range, the old
value can unwrap into the next epoch. PacketBuffer::ClearTo() then
advances the cleanup boundary into that future epoch, and subsequent
packets from the current epoch are dropped.

Use PeekUnwrap() so delayed decode feedback cannot modify the receive
unwrapper. If the resulting cleanup boundary is ahead of
newest_media_seq_num_, remove only that frame's history entries and
skip both PacketBuffer and RtpFrameReferenceFinder cleanup.

Track the newest-media watermark on the H.264/H.265 receive path as
well, so the same validation applies on every codec path.

Add focused RtpVideoStreamReceiver2 coverage and an end-to-end VP9
flexible-mode regression test.

R=asapersson@webrtc.org

Bug: webrtc:556950523
Change-Id: I673ccb0397cb9df891cc4882503c0df5c926d632
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/499540
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48529}
4 files changed
tree: b3087055c42ecb4ee8aa1561bcaf4d8cc3fff34b
  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