Avoid unnecessary copying in media engine and base

Code in media/engine and media/base copies codec lists, codecs and other
objects that it only reads, on paths such as codec negotiation,
SetParameters and GetStats, often once per element inside find_if
lambdas. Some by-value parameters and locals are also copied at their
last use instead of being moved.

- Use const references for read-only values: VideoCodecSettings vectors,
  the elements visited by find_if lambdas, StreamParams and the audio
  track id. Compare codec names as absl::string_view instead of
  constructing std::string temporaries.
- Move by-value parameters and locals at their last use, such as
  send_codec in the RTP parameter checks and the frame encryptor,
  decryptor and transformer scoped_refptrs. Use emplace instead of
  insert for decoder_map to avoid constructing a temporary pair.

Bug: webrtc:374845009
Change-Id: I254fb9bd79340929a233544a4d626e636bc44530
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/498960
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48811}
5 files changed
tree: d65278ccaae1924af54ac6fbacb54316112afc7d
  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