screencast_portal: Allow hooks to adapt behavior for remote desktop

Change adds callbacks to the class so that the remote desktop portal can
still make use of this class for selecting sources but can provide its
own implementation on what to do after the sources are selected.
Furthermore, few getters are exposed in the class interface so as to
allow the remote desktop portal class to leverage them when sending the
captured pipewire frames onto the capture stream's consumer. Setters are
added for session, pipewire stream node id and few interfaces are made
public since remote desktop portal relies on them (e.g.
`SelectSources`).

The reason behind the change is that remote desktop portal depends on
screen cast portal for selecting sources. Also the setup to select
devices to control remotely as well as source selection should be
handled as part of the same session (and session should be
instantiated only once).

Currently, starting the screencast portal calls into a callback chain
that not only selects the sources but also starts the session but with
this change a consumer, such as remote desktop portal, can hook into
this callback chain by overriding the callbacks and provide a custom
callback chain from there onwards, if need be.

Bug: chromium:1291247
Change-Id: I983aff062ec2ddf52fdef5545fc58fede416e6ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249862
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#36285}
3 files changed
tree: bd00008ff0b9a4c82f5d25952fbc65dbb7db2ed5
  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