Don't redeliver received keyframes after seq num discontinuity

RtpVideoStreamReceiver2 ends a large sequence number discontinuity at a
keyframe with an RTP timestamp newer than the newest received media, but
took that timestamp from the packet with the newest sequence number. A
packet with a high sequence number and an old timestamp could thus make
an already received keyframe look new. A copy of that keyframe far
behind in sequence numbers then reset the receiver and was decoded a
second time under a new frame id. Track the newest media RTP timestamp
independently of the sequence number instead.

Decoding two frames with the same RTP timestamp before either is
rendered also hit a DCHECK in VideoQualityObserver::OnDecodedFrame. The
timestamp is set by the remote sender, so this is reachable without the
recovery logic too, e.g. with two keyframes that share a timestamp.
Remove the DCHECK; caching a blocky timestamp twice is harmless.

Bug: chromium:568946767
Change-Id: I905de47bd32806ad792e381b1139592fed9477f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/507641
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48804}
5 files changed
tree: 983e9cab19241fa1718291082e34b3361aa96a60
  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. .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. vpython.toml
  64. vpython.toml.uv.lock
  65. WATCHLISTS
  66. webrtc.gni
  67. webrtc_lib_link_test.cc
  68. 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