Add move operations to EncoderSettings

EncoderSettings declares a copy constructor and a copy assignment
operator, so it has no implicit move operations, and std::move on an
EncoderSettings copies it, including its VideoEncoderConfig and
EncoderInfo. For example, SetEncoderSettings() in
VideoStreamEncoderResourceManager and OnEncoderSettingsUpdated() in
BitrateConstraint take the settings by value and move them into a
member, which made a copy.

Add a defaulted move constructor and move assignment operator. Like the
existing copy operations, they are defined out of line.

Also avoid the two copies made in
VideoStreamEncoder::OnEncoderSettingsChanged():
VideoStreamInputStateProvider only reads the settings, so take them by
const reference in its OnEncoderSettingsChanged(). Call that before
SetEncoderSettings(), which neither reads from nor writes to the input
state provider, so that the settings can be moved into
SetEncoderSettings(). Read the screenshare and zero-hertz flags from
`encoder_config_`, which the settings' config is a copy of.

An encoder settings change now copies EncoderSettings once, into
BitrateConstraint, instead of five times.

Bug: webrtc:374845009
Change-Id: I06419eda299836ae282ef88b95a1a1c6c8a4bb1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/507860
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48821}
5 files changed
tree: 99d7de2df47440ce22540cbf7f1c55e524b7d4f0
  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. .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. vpython.toml
  64. vpython.toml.uv.lock
  65. WATCHLISTS
  66. webrtc.gni
  67. webrtc_lib_link_test.cc
  68. 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