shared_screencast_stream: Allow overwriting next shared frame

This makes the implementation in line with the existing X11
implementation:

https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc;l=240-243

The issue I am observing on slightly slower machines with 4k monitor
is that the frames tend to go back in time. I believe this happens
when the shared frame queue is full and has its frame shared. When
that happens, we still end up calling MoveToNextFrame and doing so
we will wrap around the queue and if the capturer captures a frame
again, it sees an older frame. This is causing screen glitches.

This CL normalizes the implementation with X11 (which is known to
work fine) and moves to next frame and always uses it. This helps
to keep the current_frame_ in sync for the caller / capturer and
the capturer will then always see the video moving forward.

On the same machine, these screencasts were taken:
Without this fix: https://youtu.be/7Toi8dL5eYw
With this fix: https://youtu.be/LOE8Si5iOuQ

Bug: chromium:1291247
Change-Id: I51d3d700d3417d31371b12a94f445fc7b530cf73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278700
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38342}
1 file changed
tree: d8928af1576bce1f28ae1f0ae6668970a9cdcee2
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. infra/
  12. logging/
  13. media/
  14. modules/
  15. net/
  16. p2p/
  17. pc/
  18. resources/
  19. rtc_base/
  20. rtc_tools/
  21. sdk/
  22. stats/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .mailmap
  32. .style.yapf
  33. .vpython
  34. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. g3doc.lua
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. README.chromium
  53. README.md
  54. WATCHLISTS
  55. webrtc.gni
  56. webrtc_lib_link_test.cc
  57. 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