Unify worker and network threads in full stack tests

Unify worker and network threads by default in `webrtc_perf_tests` to eliminate the startup packet-loss race condition introduced in CL 472240.

By making worker and network threads refer to the same thread object, the receiver sink registration (`AddSink`) runs synchronously on the same thread. This avoids the race window where incoming packets fail demuxing on the network thread before the asynchronous `AddSink` task is processed.

Performance verification results (comparing separate vs unified threads):

- Pc_Simulcast_HD_High:
  * dropped_frames: 63 -> 34 (-46% improvement)
  * max_skipped: 0.048 -> 0.025 (-47% improvement)
  * actual_encode_bitrate: 2011.686 kbps -> 2071.463 kbps (+3% recovery)
  * retransmission_bitrate: 0.134 kbps -> 0.000 kbps (-100% fully eliminated)

- Pc_Screenshare_Slides_Simulcast_No_Conference_Mode:
  * dropped_frames: 2 -> 1 (-50% improvement)
  * max_skipped: 0.045 -> 0.022 (-50% improvement)
  * actual_encode_bitrate: 240.117 kbps -> 352.958 kbps (+47% recovery)
  * transmission_bitrate: 371.318 kbps -> 520.972 kbps (+40% recovery)

- Pc_Simulcast_Vp8_3sl_High:
  * dropped_frames: 39 -> 27 (-30% improvement)
  * max_skipped: 0.029 -> 0.020 (-31% improvement)

Bug: webrtc:514760674
Change-Id: Ie6ad4542ac30a365af464854df383f0f14a03ed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/474120
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47772}
1 file changed
tree: 1158017fe631d46e6ffe46af39d6d1514c9633b4
  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