Add screenshare CBR rate control tests for V2 Video Encoder API.

Implement screenshare test cases in video_encoder_rate_control_unittest:
- ScreenshareSlideChangesFullHd: Evaluates 1080p slide presentations with
  mostly static content punctuated by intermittent slide changes every 2s,
  verifying bounded scene change bursts and compact static encoding.
- ScreenshareScrollingHd: Evaluates 720p scrolling over high-resolution
  content (1850x1110) with alternating scrolling and paused intervals,
  verifying CBR sliding window and total bitrate deviation.
- ScreenshareZeroHzHd: Evaluates Zero-Hz mode where a static frame is
  encoded at 1 Hz repeat rate while each frame's encoded CBR duration
  remains 1/target_fps, and subsequent resumption of 30 fps motion,
  verifying no oversized bursts on idle frames and quick convergence
  upon resuming motion.

Bug: webrtc:496266459
Change-Id: I0137fd256ab3daae29771fc508b4bd70a311acd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/501920
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48573}
4 files changed
tree: f4f1e6da8afa4f64866400c011e1a7d7ac1fc7e9
  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