Video timing simulator: Add RTX support.

This change adds RTX reception support to the video timing simulator.
This is done by introducing `Receiver`, which demuxes the stream of RTP
packets (video + RTX), so that video packets are passed through but RTX
packets are decapsulated. The resulting recovered stream is then sent to
the existing `Assembler` and then the pipeline continues as usual from
there.

Other related and required changes:
- Update `RtpPacketSimulator` to handle logged packets with RTX OSN.
- Update the stream factories to feed simulated packets to the `Receiver`, instead of the `Assembler`.
- Enable NACKing in `Assembler` by creating a `NackRequester` and corresponding `NackPeriodicProcessor`. (Without this, the jitter buffer adaptation for frames with recovered packets will be incorrectly done.)
- Add simulation-level smoke tests for a lossy RtcEventLog scenario.

Bug: b/423646186
Change-Id: I2bdccca07107234c148300629dd45e831ee61158
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/432961
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46755}
22 files changed
tree: f57b9515abd1a7154b65c2a30e860a4af37898af
  1. agents/
  2. api/
  3. audio/
  4. build_overrides/
  5. call/
  6. common_audio/
  7. common_video/
  8. data/
  9. docs/
  10. examples/
  11. experiments/
  12. g3doc/
  13. infra/
  14. logging/
  15. media/
  16. modules/
  17. net/
  18. p2p/
  19. pc/
  20. resources/
  21. rtc_base/
  22. rtc_tools/
  23. sdk/
  24. stats/
  25. system_wrappers/
  26. test/
  27. tools_webrtc/
  28. video/
  29. .clang-format
  30. .clang-tidy
  31. .git-blame-ignore-revs
  32. .gitignore
  33. .gn
  34. .mailmap
  35. .rustfmt.toml
  36. .style.yapf
  37. .vpython3
  38. AUTHORS
  39. BUILD.gn
  40. CODE_OF_CONDUCT.md
  41. codereview.settings
  42. DEPS
  43. DIR_METADATA
  44. ENG_REVIEW_OWNERS
  45. GEMINI.md
  46. LICENSE
  47. license_template.txt
  48. native-api.md
  49. OWNERS
  50. OWNERS_INFRA
  51. PATENTS
  52. PRESUBMIT.py
  53. presubmit_test.py
  54. presubmit_test_mocks.py
  55. pylintrc
  56. pylintrc_old_style
  57. README.chromium
  58. README.md
  59. unsafe_buffers_paths.txt
  60. WATCHLISTS
  61. webrtc.gni
  62. webrtc_lib_link_test.cc
  63. 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